diff options
Diffstat (limited to 'sound')
262 files changed, 20305 insertions, 2841 deletions
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 38590b322c54..fbd5dad0c484 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> #include <sound/core.h> #include <sound/pcm.h> @@ -43,7 +44,11 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_reset, }; -static unsigned long pxa2xx_ac97_pcm_out_req = 12; +static struct pxad_param pxa2xx_ac97_pcm_out_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 12, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -51,7 +56,11 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .filter_data = &pxa2xx_ac97_pcm_out_req, }; -static unsigned long pxa2xx_ac97_pcm_in_req = 11; +static struct pxad_param pxa2xx_ac97_pcm_in_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 11, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 01f8fdc42b1b..e9b98af6b52c 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/dma-mapping.h> #include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> #include <sound/core.h> #include <sound/pcm.h> @@ -15,8 +16,6 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include <mach/dma.h> - #include "pxa2xx-pcm.h" static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { @@ -31,7 +30,7 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { .period_bytes_min = 32, .period_bytes_max = 8192 - 32, .periods_min = 1, - .periods_max = PAGE_SIZE/sizeof(pxa_dma_desc), + .periods_max = 256, .buffer_bytes_max = 128 * 1024, .fifo_size = 32, }; @@ -39,65 +38,29 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd = runtime->private_data; - size_t totsize = params_buffer_bytes(params); - size_t period = params_period_bytes(params); - pxa_dma_desc *dma_desc; - dma_addr_t dma_buff_phys, next_desc_phys; - u32 dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG; + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_dmaengine_dai_dma_data *dma_params; + struct dma_slave_config config; + int ret; - /* temporary transition hack */ - switch (rtd->params->addr_width) { - case DMA_SLAVE_BUSWIDTH_1_BYTE: - dcmd |= DCMD_WIDTH1; - break; - case DMA_SLAVE_BUSWIDTH_2_BYTES: - dcmd |= DCMD_WIDTH2; - break; - case DMA_SLAVE_BUSWIDTH_4_BYTES: - dcmd |= DCMD_WIDTH4; - break; - default: - /* can't happen */ - break; - } + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dma_params) + return 0; - switch (rtd->params->maxburst) { - case 8: - dcmd |= DCMD_BURST8; - break; - case 16: - dcmd |= DCMD_BURST16; - break; - case 32: - dcmd |= DCMD_BURST32; - break; - } + ret = snd_hwparams_to_dma_slave_config(substream, params, &config); + if (ret) + return ret; - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = totsize; + snd_dmaengine_pcm_set_config_from_dai_data(substream, + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream), + &config); - dma_desc = rtd->dma_desc_array; - next_desc_phys = rtd->dma_desc_array_phys; - dma_buff_phys = runtime->dma_addr; - do { - next_desc_phys += sizeof(pxa_dma_desc); - dma_desc->ddadr = next_desc_phys; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dma_desc->dsadr = dma_buff_phys; - dma_desc->dtadr = rtd->params->addr; - } else { - dma_desc->dsadr = rtd->params->addr; - dma_desc->dtadr = dma_buff_phys; - } - if (period > totsize) - period = totsize; - dma_desc->dcmd = dcmd | period | DCMD_ENDIRQEN; - dma_desc++; - dma_buff_phys += period; - } while (totsize -= period); - dma_desc[-1].ddadr = rtd->dma_desc_array_phys; + ret = dmaengine_slave_config(chan, &config); + if (ret) + return ret; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); return 0; } @@ -105,13 +68,6 @@ EXPORT_SYMBOL(__pxa2xx_pcm_hw_params); int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { - struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - - if (rtd && rtd->params && rtd->params->filter_data) { - unsigned long req = *(unsigned long *) rtd->params->filter_data; - DRCMR(req) = 0; - } - snd_pcm_set_runtime_buffer(substream, NULL); return 0; } @@ -119,100 +75,36 @@ EXPORT_SYMBOL(__pxa2xx_pcm_hw_free); int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys; - DCSR(prtd->dma_ch) = DCSR_RUN; - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - DCSR(prtd->dma_ch) &= ~DCSR_RUN; - break; - - case SNDRV_PCM_TRIGGER_RESUME: - DCSR(prtd->dma_ch) |= DCSR_RUN; - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys; - DCSR(prtd->dma_ch) |= DCSR_RUN; - break; - - default: - ret = -EINVAL; - } - - return ret; + return snd_dmaengine_pcm_trigger(substream, cmd); } EXPORT_SYMBOL(pxa2xx_pcm_trigger); snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *prtd = runtime->private_data; - - dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch); - snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr); - - if (x == runtime->buffer_size) - x = 0; - return x; + return snd_dmaengine_pcm_pointer(substream); } EXPORT_SYMBOL(pxa2xx_pcm_pointer); int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) { - struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; - unsigned long req; - - if (!prtd || !prtd->params) - return 0; - - if (prtd->dma_ch == -1) - return -EINVAL; - - DCSR(prtd->dma_ch) &= ~DCSR_RUN; - DCSR(prtd->dma_ch) = 0; - DCMD(prtd->dma_ch) = 0; - req = *(unsigned long *) prtd->params->filter_data; - DRCMR(req) = prtd->dma_ch | DRCMR_MAPVLD; - return 0; } EXPORT_SYMBOL(__pxa2xx_pcm_prepare); -void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) -{ - struct snd_pcm_substream *substream = dev_id; - int dcsr; - - dcsr = DCSR(dma_ch); - DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN; - - if (dcsr & DCSR_ENDINTR) { - snd_pcm_period_elapsed(substream); - } else { - printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n", - dma_ch, dcsr); - snd_pcm_stop_xrun(substream); - } -} -EXPORT_SYMBOL(pxa2xx_pcm_dma_irq); - int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd; + struct snd_dmaengine_dai_dma_data *dma_params; int ret; runtime->hw = pxa2xx_pcm_hardware; + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dma_params) + return 0; + /* * For mysterious reasons (and despite what the manual says) * playback samples are lost if the DMA count is not a multiple @@ -221,48 +113,27 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); if (ret) - goto out; + return ret; ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); if (ret) - goto out; + return ret; ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) - goto out; - - ret = -ENOMEM; - rtd = kzalloc(sizeof(*rtd), GFP_KERNEL); - if (!rtd) - goto out; - rtd->dma_desc_array = - dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE, - &rtd->dma_desc_array_phys, GFP_KERNEL); - if (!rtd->dma_desc_array) - goto err1; + return ret; - rtd->dma_ch = -1; - runtime->private_data = rtd; - return 0; - - err1: - kfree(rtd); - out: - return ret; + return snd_dmaengine_pcm_open_request_chan(substream, + pxad_filter_fn, + dma_params->filter_data); } EXPORT_SYMBOL(__pxa2xx_pcm_open); int __pxa2xx_pcm_close(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd = runtime->private_data; - - dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE, - rtd->dma_desc_array, rtd->dma_desc_array_phys); - kfree(rtd); - return 0; + return snd_dmaengine_pcm_close_release_chan(substream); } EXPORT_SYMBOL(__pxa2xx_pcm_close); diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index 83be8e3f095e..83fcfac97739 100644 --- a/sound/arm/pxa2xx-pcm.c +++ b/sound/arm/pxa2xx-pcm.c @@ -46,17 +46,13 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? client->playback_params : client->capture_params; - ret = pxa_request_dma("dma", DMA_PRIO_LOW, - pxa2xx_pcm_dma_irq, substream); - if (ret < 0) - goto err2; - rtd->dma_ch = ret; ret = client->startup(substream); if (!ret) - goto out; + goto err2; + + return 0; - pxa_free_dma(rtd->dma_ch); err2: __pxa2xx_pcm_close(substream); out: @@ -66,9 +62,7 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) static int pxa2xx_pcm_close(struct snd_pcm_substream *substream) { struct pxa2xx_pcm_client *client = substream->private_data; - struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - pxa_free_dma(rtd->dma_ch); client->shutdown(substream); return __pxa2xx_pcm_close(substream); diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h index 00330985beec..8fa2b7c9e6b8 100644 --- a/sound/arm/pxa2xx-pcm.h +++ b/sound/arm/pxa2xx-pcm.h @@ -13,8 +13,6 @@ struct pxa2xx_runtime_data { int dma_ch; struct snd_dmaengine_dai_dma_data *params; - struct pxa_dma_desc *dma_desc_array; - dma_addr_t dma_desc_array_phys; }; struct pxa2xx_pcm_client { diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6c96feeaf01e..e3e949126a56 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -4,7 +4,7 @@ config SND_TIMER config SND_PCM tristate - select SND_TIMER + select SND_TIMER if SND_PCM_TIMER config SND_PCM_ELD bool @@ -93,6 +93,17 @@ config SND_PCM_OSS_PLUGINS support conversion of channels, formats and rates. It will behave like most of new OSS/Free drivers in 2.4/2.6 kernels. +config SND_PCM_TIMER + bool "PCM timer interface" if EXPERT + default y + help + If you disable this option, pcm timer will be inavailable, so + those stubs used pcm timer (e.g. dmix, dsnoop & co) may work + incorrectlly. + + For some embedded device, we may disable it to reduce memory + footprint, about 20KB on x86_64 platform. + config SND_SEQUENCER_OSS bool "OSS Sequencer API" depends on SND_SEQUENCER diff --git a/sound/core/Makefile b/sound/core/Makefile index 3354f91e003a..48ab4b8f8279 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -13,8 +13,9 @@ snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o -snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ +snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \ pcm_memory.o memalloc.o +snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index a99f7200ff3f..7a8c79dd9734 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1177,7 +1177,8 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, struct snd_mixer_oss *mixer = entry->private_data; char line[128], str[32], idxstr[16]; const char *cptr; - int ch, idx; + unsigned int idx; + int ch; struct snd_mixer_oss_assign_table *tbl; struct slot *slot; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 02bd96954dc4..308c9ecf73db 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1014,9 +1014,6 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) snd_free_pages((void*)runtime->control, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))); kfree(runtime->hw_constraints.rules); -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - kfree(runtime->hwptr_log); -#endif kfree(runtime); substream->runtime = NULL; put_pid(substream->pid); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7d45645f10ba..6b5a811e01a5 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -801,7 +801,7 @@ void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, * negative error code. */ int snd_interval_ratnum(struct snd_interval *i, - unsigned int rats_count, struct snd_ratnum *rats, + unsigned int rats_count, const struct snd_ratnum *rats, unsigned int *nump, unsigned int *denp) { unsigned int best_num, best_den; @@ -920,7 +920,8 @@ EXPORT_SYMBOL(snd_interval_ratnum); * negative error code. */ static int snd_interval_ratden(struct snd_interval *i, - unsigned int rats_count, struct snd_ratden *rats, + unsigned int rats_count, + const struct snd_ratden *rats, unsigned int *nump, unsigned int *denp) { unsigned int best_num, best_diff, best_den; @@ -1339,7 +1340,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_ranges); static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hw_constraint_ratnums *r = rule->private; + const struct snd_pcm_hw_constraint_ratnums *r = rule->private; unsigned int num = 0, den = 0; int err; err = snd_interval_ratnum(hw_param_interval(params, rule->var), @@ -1363,10 +1364,10 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratnums *r) + const struct snd_pcm_hw_constraint_ratnums *r) { return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_ratnums, r, + snd_pcm_hw_rule_ratnums, (void *)r, var, -1); } @@ -1375,7 +1376,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_ratnums); static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hw_constraint_ratdens *r = rule->private; + const struct snd_pcm_hw_constraint_ratdens *r = rule->private; unsigned int num = 0, den = 0; int err = snd_interval_ratden(hw_param_interval(params, rule->var), r->nrats, r->rats, &num, &den); @@ -1398,10 +1399,10 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratdens *r) + const struct snd_pcm_hw_constraint_ratdens *r) { return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_ratdens, r, + snd_pcm_hw_rule_ratdens, (void *)r, var, -1); } @@ -1875,20 +1876,17 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) return; runtime = substream->runtime; - if (runtime->transfer_ack_begin) - runtime->transfer_ack_begin(substream); - snd_pcm_stream_lock_irqsave(substream, flags); if (!snd_pcm_running(substream) || snd_pcm_update_hw_ptr0(substream, 1) < 0) goto _end; +#ifdef CONFIG_SND_PCM_TIMER if (substream->timer_running) snd_timer_interrupt(substream->timer, 1); +#endif _end: snd_pcm_stream_unlock_irqrestore(substream, flags); - if (runtime->transfer_ack_end) - runtime->transfer_ack_end(substream); kill_fasync(&runtime->fasync, SIGIO, POLL_IN); } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 75888dd38a7f..a8b27cdc2844 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -486,6 +486,16 @@ static void snd_pcm_set_state(struct snd_pcm_substream *substream, int state) snd_pcm_stream_unlock_irq(substream); } +static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream, + int event) +{ +#ifdef CONFIG_SND_PCM_TIMER + if (substream->timer) + snd_timer_notify(substream->timer, event, + &substream->runtime->trigger_tstamp); +#endif +} + static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -650,7 +660,8 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, } snd_pcm_stream_unlock_irq(substream); - if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) + if (params->tstamp_mode < 0 || + params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) return -EINVAL; if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12) && params->tstamp_type > SNDRV_PCM_TSTAMP_TYPE_LAST) @@ -1042,9 +1053,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTART, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } static struct action_ops snd_pcm_action_start = { @@ -1092,9 +1101,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) if (runtime->status->state != state) { snd_pcm_trigger_tstamp(substream); runtime->status->state = state; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTOP); } wake_up(&runtime->sleep); wake_up(&runtime->tsleep); @@ -1208,18 +1215,12 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push) snd_pcm_trigger_tstamp(substream); if (push) { runtime->status->state = SNDRV_PCM_STATE_PAUSED; - if (substream->timer) - snd_timer_notify(substream->timer, - SNDRV_TIMER_EVENT_MPAUSE, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MPAUSE); wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } else { runtime->status->state = SNDRV_PCM_STATE_RUNNING; - if (substream->timer) - snd_timer_notify(substream->timer, - SNDRV_TIMER_EVENT_MCONTINUE, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MCONTINUE); } } @@ -1267,9 +1268,7 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) snd_pcm_trigger_tstamp(substream); runtime->status->suspended_state = runtime->status->state; runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSUSPEND, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSUSPEND); wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } @@ -1373,9 +1372,7 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state) struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->status->state = runtime->status->suspended_state; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MRESUME); } static struct action_ops snd_pcm_action_resume = { @@ -2226,7 +2223,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_drop(substream); if (substream->hw_opened) { - if (substream->ops->hw_free != NULL) + if (substream->ops->hw_free && + substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) substream->ops->hw_free(substream); substream->ops->close(substream); substream->hw_opened = 0; diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c index ccd893566f1d..046cb586fb2f 100644 --- a/sound/core/seq/oss/seq_oss_readq.c +++ b/sound/core/seq/oss/seq_oss_readq.c @@ -91,8 +91,7 @@ snd_seq_oss_readq_clear(struct seq_oss_readq *q) q->head = q->tail = 0; } /* if someone sleeping, wake'em up */ - if (waitqueue_active(&q->midi_sleep)) - wake_up(&q->midi_sleep); + wake_up(&q->midi_sleep); q->input_time = (unsigned long)-1; } @@ -138,8 +137,7 @@ snd_seq_oss_readq_put_event(struct seq_oss_readq *q, union evrec *ev) q->qlen++; /* wake up sleeper */ - if (waitqueue_active(&q->midi_sleep)) - wake_up(&q->midi_sleep); + wake_up(&q->midi_sleep); spin_unlock_irqrestore(&q->lock, flags); diff --git a/sound/core/seq/oss/seq_oss_writeq.c b/sound/core/seq/oss/seq_oss_writeq.c index d50338bbc21f..1f6788a18444 100644 --- a/sound/core/seq/oss/seq_oss_writeq.c +++ b/sound/core/seq/oss/seq_oss_writeq.c @@ -138,9 +138,7 @@ snd_seq_oss_writeq_wakeup(struct seq_oss_writeq *q, abstime_t time) spin_lock_irqsave(&q->sync_lock, flags); q->sync_time = time; q->sync_event_put = 0; - if (waitqueue_active(&q->sync_sleep)) { - wake_up(&q->sync_sleep); - } + wake_up(&q->sync_sleep); spin_unlock_irqrestore(&q->sync_lock, flags); } diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 8850b7de1d38..bee0e5f1a116 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -120,4 +120,31 @@ config SND_BEBOB To compile this driver as a module, choose M here: the module will be called snd-bebob. +config SND_FIREWIRE_DIGI00X + tristate "Digidesign Digi 002/003 family support" + select SND_FIREWIRE_LIB + select SND_HWDEP + help + Say Y here to include support for Digidesign Digi 002/003 family. + * Digi 002 Console + * Digi 002 Rack + * Digi 003 Console + * Digi 003 Rack + * Digi 003 Rack+ + + To compile this driver as a module, choose M here: the module + will be called snd-firewire-digi00x. + +config SND_FIREWIRE_TASCAM + tristate "TASCAM FireWire series support" + select SND_FIREWIRE_LIB + select SND_HWDEP + help + Say Y here to include support for TASCAM. + * FW-1884 + * FW-1082 + + To compile this driver as a module, choose M here: the module + will be called snd-firewire-tascam. + endif # SND_FIREWIRE diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile index 8b37f084b2ab..f5fb62551c60 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -1,6 +1,5 @@ snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \ - fcp.o cmp.o amdtp.o -snd-oxfw-objs := oxfw.o + fcp.o cmp.o amdtp-stream.o amdtp-am824.o snd-isight-objs := isight.o snd-scs1x-objs := scs1x.o @@ -11,3 +10,5 @@ obj-$(CONFIG_SND_ISIGHT) += snd-isight.o obj-$(CONFIG_SND_SCS1X) += snd-scs1x.o obj-$(CONFIG_SND_FIREWORKS) += fireworks/ obj-$(CONFIG_SND_BEBOB) += bebob/ +obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += digi00x/ +obj-$(CONFIG_SND_FIREWIRE_TASCAM) += tascam/ diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c new file mode 100644 index 000000000000..bebddc60fde8 --- /dev/null +++ b/sound/firewire/amdtp-am824.c @@ -0,0 +1,465 @@ +/* + * AM824 format in Audio and Music Data Transmission Protocol (IEC 61883-6) + * + * Copyright (c) Clemens Ladisch <clemens@ladisch.de> + * Copyright (c) 2015 Takashi Sakamoto <o-takashi@sakamocchi.jp> + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include <linux/slab.h> + +#include "amdtp-am824.h" + +#define CIP_FMT_AM 0x10 + +/* "Clock-based rate control mode" is just supported. */ +#define AMDTP_FDF_AM824 0x00 + +/* + * Nominally 3125 bytes/second, but the MIDI port's clock might be + * 1% too slow, and the bus clock 100 ppm too fast. + */ +#define MIDI_BYTES_PER_SECOND 3093 + +/* + * Several devices look only at the first eight data blocks. + * In any case, this is more than enough for the MIDI data rate. + */ +#define MAX_MIDI_RX_BLOCKS 8 + +struct amdtp_am824 { + struct snd_rawmidi_substream *midi[AM824_MAX_CHANNELS_FOR_MIDI * 8]; + int midi_fifo_limit; + int midi_fifo_used[AM824_MAX_CHANNELS_FOR_MIDI * 8]; + unsigned int pcm_channels; + unsigned int midi_ports; + + u8 pcm_positions[AM824_MAX_CHANNELS_FOR_PCM]; + u8 midi_position; + + void (*transfer_samples)(struct amdtp_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames); + + unsigned int frame_multiplier; +}; + +/** + * amdtp_am824_set_parameters - set stream parameters + * @s: the AMDTP stream to configure + * @rate: the sample rate + * @pcm_channels: the number of PCM samples in each data block, to be encoded + * as AM824 multi-bit linear audio + * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels) + * @double_pcm_frames: one data block transfers two PCM frames + * + * The parameters must be set before the stream is started, and must not be + * changed while the stream is running. + */ +int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate, + unsigned int pcm_channels, + unsigned int midi_ports, + bool double_pcm_frames) +{ + struct amdtp_am824 *p = s->protocol; + unsigned int midi_channels; + unsigned int i; + int err; + + if (amdtp_stream_running(s)) + return -EINVAL; + + if (pcm_channels > AM824_MAX_CHANNELS_FOR_PCM) + return -EINVAL; + + midi_channels = DIV_ROUND_UP(midi_ports, 8); + if (midi_channels > AM824_MAX_CHANNELS_FOR_MIDI) + return -EINVAL; + + if (WARN_ON(amdtp_stream_running(s)) || + WARN_ON(pcm_channels > AM824_MAX_CHANNELS_FOR_PCM) || + WARN_ON(midi_channels > AM824_MAX_CHANNELS_FOR_MIDI)) + return -EINVAL; + + err = amdtp_stream_set_parameters(s, rate, + pcm_channels + midi_channels); + if (err < 0) + return err; + + s->fdf = AMDTP_FDF_AM824 | s->sfc; + + p->pcm_channels = pcm_channels; + p->midi_ports = midi_ports; + + /* + * In IEC 61883-6, one data block represents one event. In ALSA, one + * event equals to one PCM frame. But Dice has a quirk at higher + * sampling rate to transfer two PCM frames in one data block. + */ + if (double_pcm_frames) + p->frame_multiplier = 2; + else + p->frame_multiplier = 1; + + /* init the position map for PCM and MIDI channels */ + for (i = 0; i < pcm_channels; i++) + p->pcm_positions[i] = i; + p->midi_position = p->pcm_channels; + + /* + * We do not know the actual MIDI FIFO size of most devices. Just + * assume two bytes, i.e., one byte can be received over the bus while + * the previous one is transmitted over MIDI. + * (The value here is adjusted for midi_ratelimit_per_packet().) + */ + p->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1; + + return 0; +} +EXPORT_SYMBOL_GPL(amdtp_am824_set_parameters); + +/** + * amdtp_am824_set_pcm_position - set an index of data channel for a channel + * of PCM frame + * @s: the AMDTP stream + * @index: the index of data channel in an data block + * @position: the channel of PCM frame + */ +void amdtp_am824_set_pcm_position(struct amdtp_stream *s, unsigned int index, + unsigned int position) +{ + struct amdtp_am824 *p = s->protocol; + + if (index < p->pcm_channels) + p->pcm_positions[index] = position; +} +EXPORT_SYMBOL_GPL(amdtp_am824_set_pcm_position); + +/** + * amdtp_am824_set_midi_position - set a index of data channel for MIDI + * conformant data channel + * @s: the AMDTP stream + * @position: the index of data channel in an data block + */ +void amdtp_am824_set_midi_position(struct amdtp_stream *s, + unsigned int position) +{ + struct amdtp_am824 *p = s->protocol; + + p->midi_position = position; +} +EXPORT_SYMBOL_GPL(amdtp_am824_set_midi_position); + +static void write_pcm_s32(struct amdtp_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames) +{ + struct amdtp_am824 *p = s->protocol; + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int channels, remaining_frames, i, c; + const u32 *src; + + channels = p->pcm_channels; + src = (void *)runtime->dma_area + + frames_to_bytes(runtime, s->pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; + + for (i = 0; i < frames; ++i) { + for (c = 0; c < channels; ++c) { + buffer[p->pcm_positions[c]] = + cpu_to_be32((*src >> 8) | 0x40000000); + src++; + } + buffer += s->data_block_quadlets; + if (--remaining_frames == 0) + src = (void *)runtime->dma_area; + } +} + +static void write_pcm_s16(struct amdtp_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames) +{ + struct amdtp_am824 *p = s->protocol; + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int channels, remaining_frames, i, c; + const u16 *src; + + channels = p->pcm_channels; + src = (void *)runtime->dma_area + + frames_to_bytes(runtime, s->pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; + + for (i = 0; i < frames; ++i) { + for (c = 0; c < channels; ++c) { + buffer[p->pcm_positions[c]] = + cpu_to_be32((*src << 8) | 0x42000000); + src++; + } + buffer += s->data_block_quadlets; + if (--remaining_frames == 0) + src = (void *)runtime->dma_area; + } +} + +static void read_pcm_s32(struct amdtp_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames) +{ + struct amdtp_am824 *p = s->protocol; + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int channels, remaining_frames, i, c; + u32 *dst; + + channels = p->pcm_channels; + dst = (void *)runtime->dma_area + + frames_to_bytes(runtime, s->pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; + + for (i = 0; i < frames; ++i) { + for (c = 0; c < channels; ++c) { + *dst = be32_to_cpu(buffer[p->pcm_positions[c]]) << 8; + dst++; + } + buffer += s->data_block_quadlets; + if (--remaining_frames == 0) + dst = (void *)runtime->dma_area; + } +} + +static void write_pcm_silence(struct amdtp_stream *s, + __be32 *buffer, unsigned int frames) +{ + struct amdtp_am824 *p = s->protocol; + unsigned int i, c, channels = p->pcm_channels; + + for (i = 0; i < frames; ++i) { + for (c = 0; c < channels; ++c) + buffer[p->pcm_positions[c]] = cpu_to_be32(0x40000000); + buffer += s->data_block_quadlets; + } +} + +/** + * amdtp_am824_set_pcm_format - set the PCM format + * @s: the AMDTP stream to configure + * @format: the format of the ALSA PCM device + * + * The sample format must be set after the other parameters (rate/PCM channels/ + * MIDI) and before the stream is started, and must not be changed while the + * stream is running. + */ +void amdtp_am824_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format) +{ + struct amdtp_am824 *p = s->protocol; + + if (WARN_ON(amdtp_stream_pcm_running(s))) + return; + + switch (format) { + default: + WARN_ON(1); + /* fall through */ + case SNDRV_PCM_FORMAT_S16: + if (s->direction == AMDTP_OUT_STREAM) { + p->transfer_samples = write_pcm_s16; + break; + } + WARN_ON(1); + /* fall through */ + case SNDRV_PCM_FORMAT_S32: + if (s->direction == AMDTP_OUT_STREAM) + p->transfer_samples = write_pcm_s32; + else + p->transfer_samples = read_pcm_s32; + break; + } +} +EXPORT_SYMBOL_GPL(amdtp_am824_set_pcm_format); + +/** + * amdtp_am824_add_pcm_hw_constraints - add hw constraints for PCM substream + * @s: the AMDTP stream for AM824 data block, must be initialized. + * @runtime: the PCM substream runtime + * + */ +int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s, + struct snd_pcm_runtime *runtime) +{ + int err; + + err = amdtp_stream_add_pcm_hw_constraints(s, runtime); + if (err < 0) + return err; + + /* AM824 in IEC 61883-6 can deliver 24bit data. */ + return snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); +} +EXPORT_SYMBOL_GPL(amdtp_am824_add_pcm_hw_constraints); + +/** + * amdtp_am824_midi_trigger - start/stop playback/capture with a MIDI device + * @s: the AMDTP stream + * @port: index of MIDI port + * @midi: the MIDI device to be started, or %NULL to stop the current device + * + * Call this function on a running isochronous stream to enable the actual + * transmission of MIDI data. This function should be called from the MIDI + * device's .trigger callback. + */ +void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port, + struct snd_rawmidi_substream *midi) +{ + struct amdtp_am824 *p = s->protocol; + + if (port < p->midi_ports) + ACCESS_ONCE(p->midi[port]) = midi; +} +EXPORT_SYMBOL_GPL(amdtp_am824_midi_trigger); + +/* + * To avoid sending MIDI bytes at too high a rate, assume that the receiving + * device has a FIFO, and track how much it is filled. This values increases + * by one whenever we send one byte in a packet, but the FIFO empties at + * a constant rate independent of our packet rate. One packet has syt_interval + * samples, so the number of bytes that empty out of the FIFO, per packet(!), + * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate. To avoid storing + * fractional values, the values in midi_fifo_used[] are measured in bytes + * multiplied by the sample rate. + */ +static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port) +{ + struct amdtp_am824 *p = s->protocol; + int used; + + used = p->midi_fifo_used[port]; + if (used == 0) /* common shortcut */ + return true; + + used -= MIDI_BYTES_PER_SECOND * s->syt_interval; + used = max(used, 0); + p->midi_fifo_used[port] = used; + + return used < p->midi_fifo_limit; +} + +static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port) +{ + struct amdtp_am824 *p = s->protocol; + + p->midi_fifo_used[port] += amdtp_rate_table[s->sfc]; +} + +static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer, + unsigned int frames) +{ + struct amdtp_am824 *p = s->protocol; + unsigned int f, port; + u8 *b; + + for (f = 0; f < frames; f++) { + b = (u8 *)&buffer[p->midi_position]; + + port = (s->data_block_counter + f) % 8; + if (f < MAX_MIDI_RX_BLOCKS && + midi_ratelimit_per_packet(s, port) && + p->midi[port] != NULL && + snd_rawmidi_transmit(p->midi[port], &b[1], 1) == 1) { + midi_rate_use_one_byte(s, port); + b[0] = 0x81; + } else { + b[0] = 0x80; + b[1] = 0; + } + b[2] = 0; + b[3] = 0; + + buffer += s->data_block_quadlets; + } +} + +static void read_midi_messages(struct amdtp_stream *s, + __be32 *buffer, unsigned int frames) +{ + struct amdtp_am824 *p = s->protocol; + unsigned int f, port; + int len; + u8 *b; + + for (f = 0; f < frames; f++) { + port = (s->data_block_counter + f) % 8; + b = (u8 *)&buffer[p->midi_position]; + + len = b[0] - 0x80; + if ((1 <= len) && (len <= 3) && (p->midi[port])) + snd_rawmidi_receive(p->midi[port], b + 1, len); + + buffer += s->data_block_quadlets; + } +} + +static unsigned int process_rx_data_blocks(struct amdtp_stream *s, __be32 *buffer, + unsigned int data_blocks, unsigned int *syt) +{ + struct amdtp_am824 *p = s->protocol; + struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm); + unsigned int pcm_frames; + + if (pcm) { + p->transfer_samples(s, pcm, buffer, data_blocks); + pcm_frames = data_blocks * p->frame_multiplier; + } else { + write_pcm_silence(s, buffer, data_blocks); + pcm_frames = 0; + } + + if (p->midi_ports) + write_midi_messages(s, buffer, data_blocks); + + return pcm_frames; +} + +static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffer, + unsigned int data_blocks, unsigned int *syt) +{ + struct amdtp_am824 *p = s->protocol; + struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm); + unsigned int pcm_frames; + + if (pcm) { + p->transfer_samples(s, pcm, buffer, data_blocks); + pcm_frames = data_blocks * p->frame_multiplier; + } else { + pcm_frames = 0; + } + + if (p->midi_ports) + read_midi_messages(s, buffer, data_blocks); + + return pcm_frames; +} + +/** + * amdtp_am824_init - initialize an AMDTP stream structure to handle AM824 + * data block + * @s: the AMDTP stream to initialize + * @unit: the target of the stream + * @dir: the direction of stream + * @flags: the packet transmission method to use + */ +int amdtp_am824_init(struct amdtp_stream *s, struct fw_unit *unit, + enum amdtp_stream_direction dir, enum cip_flags flags) +{ + amdtp_stream_process_data_blocks_t process_data_blocks; + + if (dir == AMDTP_IN_STREAM) + process_data_blocks = process_tx_data_blocks; + else + process_data_blocks = process_rx_data_blocks; + + return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM, + process_data_blocks, + sizeof(struct amdtp_am824)); +} +EXPORT_SYMBOL_GPL(amdtp_am824_init); diff --git a/sound/firewire/amdtp-am824.h b/sound/firewire/amdtp-am824.h new file mode 100644 index 000000000000..73b07b3109db --- /dev/null +++ b/sound/firewire/amdtp-am824.h @@ -0,0 +1,52 @@ +#ifndef SOUND_FIREWIRE_AMDTP_AM824_H_INCLUDED +#define SOUND_FIREWIRE_AMDTP_AM824_H_INCLUDED + +#include <sound/pcm.h> +#include <sound/rawmidi.h> + +#include "amdtp-stream.h" + +#define AM824_IN_PCM_FORMAT_BITS SNDRV_PCM_FMTBIT_S32 + +#define AM824_OUT_PCM_FORMAT_BITS (SNDRV_PCM_FMTBIT_S16 | \ + SNDRV_PCM_FMTBIT_S32) + +/* + * This module supports maximum 64 PCM channels for one PCM stream + * This is for our convenience. + */ +#define AM824_MAX_CHANNELS_FOR_PCM 64 + +/* + * AMDTP packet can include channels for MIDI conformant data. + * Each MIDI conformant data channel includes 8 MPX-MIDI data stream. + * Each MPX-MIDI data stream includes one data stream from/to MIDI ports. + * + * This module supports maximum 1 MIDI conformant data channels. + * Then this AMDTP packets can transfer maximum 8 MIDI data streams. + */ +#define AM824_MAX_CHANNELS_FOR_MIDI 1 + +int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate, + unsigned int pcm_channels, + unsigned int midi_ports, + bool double_pcm_frames); + +void amdtp_am824_set_pcm_position(struct amdtp_stream *s, unsigned int index, + unsigned int position); + +void amdtp_am824_set_midi_position(struct amdtp_stream *s, + unsigned int position); + +int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s, + struct snd_pcm_runtime *runtime); + +void amdtp_am824_set_pcm_format(struct amdtp_stream *s, + snd_pcm_format_t format); + +void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port, + struct snd_rawmidi_substream *midi); + +int amdtp_am824_init(struct amdtp_stream *s, struct fw_unit *unit, + enum amdtp_stream_direction dir, enum cip_flags flags); +#endif diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp-stream.c index 2a153d260836..ed2902609a4c 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp-stream.c @@ -11,28 +11,14 @@ #include <linux/firewire.h> #include <linux/module.h> #include <linux/slab.h> -#include <linux/sched.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -#include <sound/rawmidi.h> -#include "amdtp.h" +#include "amdtp-stream.h" #define TICKS_PER_CYCLE 3072 #define CYCLES_PER_SECOND 8000 #define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND) -/* - * Nominally 3125 bytes/second, but the MIDI port's clock might be - * 1% too slow, and the bus clock 100 ppm too fast. - */ -#define MIDI_BYTES_PER_SECOND 3093 - -/* - * Several devices look only at the first eight data blocks. - * In any case, this is more than enough for the MIDI data rate. - */ -#define MAX_MIDI_RX_BLOCKS 8 - #define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */ /* isochronous header parameters */ @@ -55,12 +41,8 @@ #define CIP_SYT_MASK 0x0000ffff #define CIP_SYT_NO_INFO 0xffff -/* - * Audio and Music transfer protocol specific parameters - * only "Clock-based rate control mode" is supported - */ -#define CIP_FMT_AM (0x10 << CIP_FMT_SHIFT) -#define AMDTP_FDF_AM824 (0 << (CIP_FDF_SHIFT + 3)) +/* Audio and Music transfer protocol specific parameters */ +#define CIP_FMT_AM 0x10 #define AMDTP_FDF_NO_DATA 0xff /* TODO: make these configurable */ @@ -78,10 +60,23 @@ static void pcm_period_tasklet(unsigned long data); * @unit: the target of the stream * @dir: the direction of stream * @flags: the packet transmission method to use + * @fmt: the value of fmt field in CIP header + * @process_data_blocks: callback handler to process data blocks + * @protocol_size: the size to allocate newly for protocol */ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, - enum amdtp_stream_direction dir, enum cip_flags flags) + enum amdtp_stream_direction dir, enum cip_flags flags, + unsigned int fmt, + amdtp_stream_process_data_blocks_t process_data_blocks, + unsigned int protocol_size) { + if (process_data_blocks == NULL) + return -EINVAL; + + s->protocol = kzalloc(protocol_size, GFP_KERNEL); + if (!s->protocol) + return -ENOMEM; + s->unit = unit; s->direction = dir; s->flags = flags; @@ -94,6 +89,9 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->callbacked = false; s->sync_slave = NULL; + s->fmt = fmt; + s->process_data_blocks = process_data_blocks; + return 0; } EXPORT_SYMBOL(amdtp_stream_init); @@ -105,6 +103,7 @@ EXPORT_SYMBOL(amdtp_stream_init); void amdtp_stream_destroy(struct amdtp_stream *s) { WARN_ON(amdtp_stream_running(s)); + kfree(s->protocol); mutex_destroy(&s->mutex); } EXPORT_SYMBOL(amdtp_stream_destroy); @@ -141,11 +140,6 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, { int err; - /* AM824 in IEC 61883-6 can deliver 24bit data */ - err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - if (err < 0) - goto end; - /* * Currently firewire-lib processes 16 packets in one software * interrupt callback. This equals to 2msec but actually the @@ -190,39 +184,25 @@ EXPORT_SYMBOL(amdtp_stream_add_pcm_hw_constraints); * amdtp_stream_set_parameters - set stream parameters * @s: the AMDTP stream to configure * @rate: the sample rate - * @pcm_channels: the number of PCM samples in each data block, to be encoded - * as AM824 multi-bit linear audio - * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels) + * @data_block_quadlets: the size of a data block in quadlet unit * * The parameters must be set before the stream is started, and must not be * changed while the stream is running. */ -void amdtp_stream_set_parameters(struct amdtp_stream *s, - unsigned int rate, - unsigned int pcm_channels, - unsigned int midi_ports) +int amdtp_stream_set_parameters(struct amdtp_stream *s, unsigned int rate, + unsigned int data_block_quadlets) { - unsigned int i, sfc, midi_channels; + unsigned int sfc; - midi_channels = DIV_ROUND_UP(midi_ports, 8); - - if (WARN_ON(amdtp_stream_running(s)) | - WARN_ON(pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM) | - WARN_ON(midi_channels > AMDTP_MAX_CHANNELS_FOR_MIDI)) - return; - - for (sfc = 0; sfc < ARRAY_SIZE(amdtp_rate_table); ++sfc) + for (sfc = 0; sfc < ARRAY_SIZE(amdtp_rate_table); ++sfc) { if (amdtp_rate_table[sfc] == rate) - goto sfc_found; - WARN_ON(1); - return; + break; + } + if (sfc == ARRAY_SIZE(amdtp_rate_table)) + return -EINVAL; -sfc_found: - s->pcm_channels = pcm_channels; s->sfc = sfc; - s->data_block_quadlets = s->pcm_channels + midi_channels; - s->midi_ports = midi_ports; - + s->data_block_quadlets = data_block_quadlets; s->syt_interval = amdtp_syt_intervals[sfc]; /* default buffering in the device */ @@ -231,18 +211,7 @@ sfc_found: /* additional buffering needed to adjust for no-data packets */ s->transfer_delay += TICKS_PER_SECOND * s->syt_interval / rate; - /* init the position map for PCM and MIDI channels */ - for (i = 0; i < pcm_channels; i++) - s->pcm_positions[i] = i; - s->midi_position = s->pcm_channels; - - /* - * We do not know the actual MIDI FIFO size of most devices. Just - * assume two bytes, i.e., one byte can be received over the bus while - * the previous one is transmitted over MIDI. - * (The value here is adjusted for midi_ratelimit_per_packet().) - */ - s->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1; + return 0; } EXPORT_SYMBOL(amdtp_stream_set_parameters); @@ -264,52 +233,6 @@ unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s) } EXPORT_SYMBOL(amdtp_stream_get_max_payload); -static void write_pcm_s16(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames); -static void write_pcm_s32(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames); -static void read_pcm_s32(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames); - -/** - * amdtp_stream_set_pcm_format - set the PCM format - * @s: the AMDTP stream to configure - * @format: the format of the ALSA PCM device - * - * The sample format must be set after the other parameters (rate/PCM channels/ - * MIDI) and before the stream is started, and must not be changed while the - * stream is running. - */ -void amdtp_stream_set_pcm_format(struct amdtp_stream *s, - snd_pcm_format_t format) -{ - if (WARN_ON(amdtp_stream_pcm_running(s))) - return; - - switch (format) { - default: - WARN_ON(1); - /* fall through */ - case SNDRV_PCM_FORMAT_S16: - if (s->direction == AMDTP_OUT_STREAM) { - s->transfer_samples = write_pcm_s16; - break; - } - WARN_ON(1); - /* fall through */ - case SNDRV_PCM_FORMAT_S32: - if (s->direction == AMDTP_OUT_STREAM) - s->transfer_samples = write_pcm_s32; - else - s->transfer_samples = read_pcm_s32; - break; - } -} -EXPORT_SYMBOL(amdtp_stream_set_pcm_format); - /** * amdtp_stream_pcm_prepare - prepare PCM device for running * @s: the AMDTP stream @@ -412,182 +335,12 @@ static unsigned int calculate_syt(struct amdtp_stream *s, } } -static void write_pcm_s32(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames) -{ - struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; - const u32 *src; - - channels = s->pcm_channels; - src = (void *)runtime->dma_area + - frames_to_bytes(runtime, s->pcm_buffer_pointer); - remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; - - for (i = 0; i < frames; ++i) { - for (c = 0; c < channels; ++c) { - buffer[s->pcm_positions[c]] = - cpu_to_be32((*src >> 8) | 0x40000000); - src++; - } - buffer += s->data_block_quadlets; - if (--remaining_frames == 0) - src = (void *)runtime->dma_area; - } -} - -static void write_pcm_s16(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames) -{ - struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; - const u16 *src; - - channels = s->pcm_channels; - src = (void *)runtime->dma_area + - frames_to_bytes(runtime, s->pcm_buffer_pointer); - remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; - - for (i = 0; i < frames; ++i) { - for (c = 0; c < channels; ++c) { - buffer[s->pcm_positions[c]] = - cpu_to_be32((*src << 8) | 0x42000000); - src++; - } - buffer += s->data_block_quadlets; - if (--remaining_frames == 0) - src = (void *)runtime->dma_area; - } -} - -static void read_pcm_s32(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames) -{ - struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; - u32 *dst; - - channels = s->pcm_channels; - dst = (void *)runtime->dma_area + - frames_to_bytes(runtime, s->pcm_buffer_pointer); - remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; - - for (i = 0; i < frames; ++i) { - for (c = 0; c < channels; ++c) { - *dst = be32_to_cpu(buffer[s->pcm_positions[c]]) << 8; - dst++; - } - buffer += s->data_block_quadlets; - if (--remaining_frames == 0) - dst = (void *)runtime->dma_area; - } -} - -static void write_pcm_silence(struct amdtp_stream *s, - __be32 *buffer, unsigned int frames) -{ - unsigned int i, c; - - for (i = 0; i < frames; ++i) { - for (c = 0; c < s->pcm_channels; ++c) - buffer[s->pcm_positions[c]] = cpu_to_be32(0x40000000); - buffer += s->data_block_quadlets; - } -} - -/* - * To avoid sending MIDI bytes at too high a rate, assume that the receiving - * device has a FIFO, and track how much it is filled. This values increases - * by one whenever we send one byte in a packet, but the FIFO empties at - * a constant rate independent of our packet rate. One packet has syt_interval - * samples, so the number of bytes that empty out of the FIFO, per packet(!), - * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate. To avoid storing - * fractional values, the values in midi_fifo_used[] are measured in bytes - * multiplied by the sample rate. - */ -static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port) -{ - int used; - - used = s->midi_fifo_used[port]; - if (used == 0) /* common shortcut */ - return true; - - used -= MIDI_BYTES_PER_SECOND * s->syt_interval; - used = max(used, 0); - s->midi_fifo_used[port] = used; - - return used < s->midi_fifo_limit; -} - -static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port) -{ - s->midi_fifo_used[port] += amdtp_rate_table[s->sfc]; -} - -static void write_midi_messages(struct amdtp_stream *s, - __be32 *buffer, unsigned int frames) -{ - unsigned int f, port; - u8 *b; - - for (f = 0; f < frames; f++) { - b = (u8 *)&buffer[s->midi_position]; - - port = (s->data_block_counter + f) % 8; - if (f < MAX_MIDI_RX_BLOCKS && - midi_ratelimit_per_packet(s, port) && - s->midi[port] != NULL && - snd_rawmidi_transmit(s->midi[port], &b[1], 1) == 1) { - midi_rate_use_one_byte(s, port); - b[0] = 0x81; - } else { - b[0] = 0x80; - b[1] = 0; - } - b[2] = 0; - b[3] = 0; - - buffer += s->data_block_quadlets; - } -} - -static void read_midi_messages(struct amdtp_stream *s, - __be32 *buffer, unsigned int frames) -{ - unsigned int f, port; - int len; - u8 *b; - - for (f = 0; f < frames; f++) { - port = (s->data_block_counter + f) % 8; - b = (u8 *)&buffer[s->midi_position]; - - len = b[0] - 0x80; - if ((1 <= len) && (len <= 3) && (s->midi[port])) - snd_rawmidi_receive(s->midi[port], b + 1, len); - - buffer += s->data_block_quadlets; - } -} - static void update_pcm_pointers(struct amdtp_stream *s, struct snd_pcm_substream *pcm, unsigned int frames) { unsigned int ptr; - /* - * In IEC 61883-6, one data block represents one event. In ALSA, one - * event equals to one PCM frame. But Dice has a quirk to transfer - * two PCM frames in one data block. - */ - if (s->double_pcm_frames) - frames *= 2; - ptr = s->pcm_buffer_pointer + frames; if (ptr >= pcm->runtime->buffer_size) ptr -= pcm->runtime->buffer_size; @@ -656,23 +409,19 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, { __be32 *buffer; unsigned int payload_length; + unsigned int pcm_frames; struct snd_pcm_substream *pcm; buffer = s->buffer.packets[s->packet_index].buffer; + pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt); + buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) | (s->data_block_quadlets << CIP_DBS_SHIFT) | s->data_block_counter); - buffer[1] = cpu_to_be32(CIP_EOH | CIP_FMT_AM | AMDTP_FDF_AM824 | - (s->sfc << CIP_FDF_SHIFT) | syt); - buffer += 2; - - pcm = ACCESS_ONCE(s->pcm); - if (pcm) - s->transfer_samples(s, pcm, buffer, data_blocks); - else - write_pcm_silence(s, buffer, data_blocks); - if (s->midi_ports) - write_midi_messages(s, buffer, data_blocks); + buffer[1] = cpu_to_be32(CIP_EOH | + ((s->fmt << CIP_FMT_SHIFT) & CIP_FMT_MASK) | + ((s->fdf << CIP_FDF_SHIFT) & CIP_FDF_MASK) | + (syt & CIP_SYT_MASK)); s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff; @@ -680,8 +429,9 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, if (queue_out_packet(s, payload_length, false) < 0) return -EIO; - if (pcm) - update_pcm_pointers(s, pcm, data_blocks); + pcm = ACCESS_ONCE(s->pcm); + if (pcm && pcm_frames > 0) + update_pcm_pointers(s, pcm, pcm_frames); /* No need to return the number of handled data blocks. */ return 0; @@ -689,11 +439,13 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, static int handle_in_packet(struct amdtp_stream *s, unsigned int payload_quadlets, __be32 *buffer, - unsigned int *data_blocks) + unsigned int *data_blocks, unsigned int syt) { u32 cip_header[2]; + unsigned int fmt, fdf; unsigned int data_block_quadlets, data_block_counter, dbc_interval; - struct snd_pcm_substream *pcm = NULL; + struct snd_pcm_substream *pcm; + unsigned int pcm_frames; bool lost; cip_header[0] = be32_to_cpu(buffer[0]); @@ -704,19 +456,30 @@ static int handle_in_packet(struct amdtp_stream *s, * For convenience, also check FMT field is AM824 or not. */ if (((cip_header[0] & CIP_EOH_MASK) == CIP_EOH) || - ((cip_header[1] & CIP_EOH_MASK) != CIP_EOH) || - ((cip_header[1] & CIP_FMT_MASK) != CIP_FMT_AM)) { + ((cip_header[1] & CIP_EOH_MASK) != CIP_EOH)) { dev_info_ratelimited(&s->unit->device, "Invalid CIP header for AMDTP: %08X:%08X\n", cip_header[0], cip_header[1]); *data_blocks = 0; + pcm_frames = 0; + goto end; + } + + /* Check valid protocol or not. */ + fmt = (cip_header[1] & CIP_FMT_MASK) >> CIP_FMT_SHIFT; + if (fmt != s->fmt) { + dev_info_ratelimited(&s->unit->device, + "Detect unexpected protocol: %08x %08x\n", + cip_header[0], cip_header[1]); + *data_blocks = 0; + pcm_frames = 0; goto end; } /* Calculate data blocks */ + fdf = (cip_header[1] & CIP_FDF_MASK) >> CIP_FDF_SHIFT; if (payload_quadlets < 3 || - ((cip_header[1] & CIP_FDF_MASK) == - (AMDTP_FDF_NO_DATA << CIP_FDF_SHIFT))) { + (fmt == CIP_FMT_AM && fdf == AMDTP_FDF_NO_DATA)) { *data_blocks = 0; } else { data_block_quadlets = @@ -763,16 +526,7 @@ static int handle_in_packet(struct amdtp_stream *s, return -EIO; } - if (*data_blocks > 0) { - buffer += 2; - - pcm = ACCESS_ONCE(s->pcm); - if (pcm) - s->transfer_samples(s, pcm, buffer, *data_blocks); - - if (s->midi_ports) - read_midi_messages(s, buffer, *data_blocks); - } + pcm_frames = s->process_data_blocks(s, buffer + 2, *data_blocks, &syt); if (s->flags & CIP_DBC_IS_END_EVENT) s->data_block_counter = data_block_counter; @@ -783,8 +537,9 @@ end: if (queue_in_packet(s) < 0) return -EIO; - if (pcm) - update_pcm_pointers(s, pcm, *data_blocks); + pcm = ACCESS_ONCE(s->pcm); + if (pcm && pcm_frames > 0) + update_pcm_pointers(s, pcm, pcm_frames); return 0; } @@ -854,15 +609,15 @@ static void in_stream_callback(struct fw_iso_context *context, u32 cycle, break; } + syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK; if (handle_in_packet(s, payload_quadlets, buffer, - &data_blocks) < 0) { + &data_blocks, syt) < 0) { s->packet_index = -1; break; } /* Process sync slave stream */ if (s->sync_slave && s->sync_slave->callbacked) { - syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK; if (handle_out_packet(s->sync_slave, data_blocks, syt) < 0) { s->packet_index = -1; diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp-stream.h index b2cf9e75693b..8775704a3665 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp-stream.h @@ -4,6 +4,7 @@ #include <linux/err.h> #include <linux/interrupt.h> #include <linux/mutex.h> +#include <linux/sched.h> #include <sound/asound.h> #include "packets-buffer.h" @@ -80,100 +81,78 @@ enum cip_sfc { CIP_SFC_COUNT }; -#define AMDTP_IN_PCM_FORMAT_BITS SNDRV_PCM_FMTBIT_S32 - -#define AMDTP_OUT_PCM_FORMAT_BITS (SNDRV_PCM_FMTBIT_S16 | \ - SNDRV_PCM_FMTBIT_S32) - - -/* - * This module supports maximum 64 PCM channels for one PCM stream - * This is for our convenience. - */ -#define AMDTP_MAX_CHANNELS_FOR_PCM 64 - -/* - * AMDTP packet can include channels for MIDI conformant data. - * Each MIDI conformant data channel includes 8 MPX-MIDI data stream. - * Each MPX-MIDI data stream includes one data stream from/to MIDI ports. - * - * This module supports maximum 1 MIDI conformant data channels. - * Then this AMDTP packets can transfer maximum 8 MIDI data streams. - */ -#define AMDTP_MAX_CHANNELS_FOR_MIDI 1 - struct fw_unit; struct fw_iso_context; struct snd_pcm_substream; struct snd_pcm_runtime; -struct snd_rawmidi_substream; enum amdtp_stream_direction { AMDTP_OUT_STREAM = 0, AMDTP_IN_STREAM }; +struct amdtp_stream; +typedef unsigned int (*amdtp_stream_process_data_blocks_t)( + struct amdtp_stream *s, + __be32 *buffer, + unsigned int data_blocks, + unsigned int *syt); struct amdtp_stream { struct fw_unit *unit; enum cip_flags flags; enum amdtp_stream_direction direction; - struct fw_iso_context *context; struct mutex mutex; - enum cip_sfc sfc; - unsigned int data_block_quadlets; - unsigned int pcm_channels; - unsigned int midi_ports; - void (*transfer_samples)(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames); - u8 pcm_positions[AMDTP_MAX_CHANNELS_FOR_PCM]; - u8 midi_position; - - unsigned int syt_interval; - unsigned int transfer_delay; - unsigned int source_node_id_field; + /* For packet processing. */ + struct fw_iso_context *context; struct iso_packets_buffer buffer; - - struct snd_pcm_substream *pcm; - struct tasklet_struct period_tasklet; - int packet_index; + + /* For CIP headers. */ + unsigned int source_node_id_field; + unsigned int data_block_quadlets; unsigned int data_block_counter; + unsigned int fmt; + unsigned int fdf; + /* quirk: fixed interval of dbc between previos/current packets. */ + unsigned int tx_dbc_interval; + /* quirk: indicate the value of dbc field in a first packet. */ + unsigned int tx_first_dbc; + /* Internal flags. */ + enum cip_sfc sfc; + unsigned int syt_interval; + unsigned int transfer_delay; unsigned int data_block_state; - unsigned int last_syt_offset; unsigned int syt_offset_state; + /* For a PCM substream processing. */ + struct snd_pcm_substream *pcm; + struct tasklet_struct period_tasklet; unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; bool pointer_flush; - bool double_pcm_frames; - - struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8]; - int midi_fifo_limit; - int midi_fifo_used[AMDTP_MAX_CHANNELS_FOR_MIDI * 8]; - - /* quirk: fixed interval of dbc between previos/current packets. */ - unsigned int tx_dbc_interval; - /* quirk: indicate the value of dbc field in a first packet. */ - unsigned int tx_first_dbc; + /* To wait for first packet. */ bool callbacked; wait_queue_head_t callback_wait; struct amdtp_stream *sync_slave; + + /* For backends to process data blocks. */ + void *protocol; + amdtp_stream_process_data_blocks_t process_data_blocks; }; int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, - enum amdtp_stream_direction dir, - enum cip_flags flags); + enum amdtp_stream_direction dir, enum cip_flags flags, + unsigned int fmt, + amdtp_stream_process_data_blocks_t process_data_blocks, + unsigned int protocol_size); void amdtp_stream_destroy(struct amdtp_stream *s); -void amdtp_stream_set_parameters(struct amdtp_stream *s, - unsigned int rate, - unsigned int pcm_channels, - unsigned int midi_ports); +int amdtp_stream_set_parameters(struct amdtp_stream *s, unsigned int rate, + unsigned int data_block_quadlets); unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s); int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed); @@ -182,8 +161,7 @@ void amdtp_stream_stop(struct amdtp_stream *s); int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, struct snd_pcm_runtime *runtime); -void amdtp_stream_set_pcm_format(struct amdtp_stream *s, - snd_pcm_format_t format); + void amdtp_stream_pcm_prepare(struct amdtp_stream *s); unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s); void amdtp_stream_pcm_abort(struct amdtp_stream *s); @@ -240,24 +218,6 @@ static inline void amdtp_stream_pcm_trigger(struct amdtp_stream *s, ACCESS_ONCE(s->pcm) = pcm; } -/** - * amdtp_stream_midi_trigger - start/stop playback/capture with a MIDI device - * @s: the AMDTP stream - * @port: index of MIDI port - * @midi: the MIDI device to be started, or %NULL to stop the current device - * - * Call this function on a running isochronous stream to enable the actual - * transmission of MIDI data. This function should be called from the MIDI - * device's .trigger callback. - */ -static inline void amdtp_stream_midi_trigger(struct amdtp_stream *s, - unsigned int port, - struct snd_rawmidi_substream *midi) -{ - if (port < s->midi_ports) - ACCESS_ONCE(s->midi[port]) = midi; -} - static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc) { return sfc & 1; diff --git a/sound/firewire/bebob/Makefile b/sound/firewire/bebob/Makefile index 6cf470c80d1f..af7ed6643266 100644 --- a/sound/firewire/bebob/Makefile +++ b/sound/firewire/bebob/Makefile @@ -1,4 +1,4 @@ snd-bebob-objs := bebob_command.o bebob_stream.o bebob_proc.o bebob_midi.o \ bebob_pcm.o bebob_hwdep.o bebob_terratec.o bebob_yamaha.o \ bebob_focusrite.o bebob_maudio.o bebob.o -obj-m += snd-bebob.o +obj-$(CONFIG_SND_BEBOB) += snd-bebob.o diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 27a04ac8ffee..091290d1f3ea 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -41,7 +41,8 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define VEN_EDIROL 0x000040ab #define VEN_PRESONUS 0x00000a92 #define VEN_BRIDGECO 0x000007f5 -#define VEN_MACKIE 0x0000000f +#define VEN_MACKIE1 0x0000000f +#define VEN_MACKIE2 0x00000ff2 #define VEN_STANTON 0x00001260 #define VEN_TASCAM 0x0000022e #define VEN_BEHRINGER 0x00001564 @@ -334,7 +335,7 @@ static void bebob_remove(struct fw_unit *unit) snd_card_free_when_closed(bebob->card); } -static struct snd_bebob_rate_spec normal_rate_spec = { +static const struct snd_bebob_rate_spec normal_rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate }; @@ -360,9 +361,9 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* BridgeCo, Audio5 */ SND_BEBOB_DEV_ENTRY(VEN_BRIDGECO, 0x00010049, &spec_normal), /* Mackie, Onyx 1220/1620/1640 (Firewire I/O Card) */ - SND_BEBOB_DEV_ENTRY(VEN_MACKIE, 0x00010065, &spec_normal), + SND_BEBOB_DEV_ENTRY(VEN_MACKIE2, 0x00010065, &spec_normal), /* Mackie, d.2 (Firewire Option) */ - SND_BEBOB_DEV_ENTRY(VEN_MACKIE, 0x00010067, &spec_normal), + SND_BEBOB_DEV_ENTRY(VEN_MACKIE1, 0x00010067, &spec_normal), /* Stanton, ScratchAmp */ SND_BEBOB_DEV_ENTRY(VEN_STANTON, 0x00000001, &spec_normal), /* Tascam, IF-FW DM */ diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index d23caca7f369..4d8fcc78e747 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -31,7 +31,7 @@ #include "../fcp.h" #include "../packets-buffer.h" #include "../iso-resources.h" -#include "../amdtp.h" +#include "../amdtp-am824.h" #include "../cmp.h" /* basic register addresses on DM1000/DM1100/DM1500 */ @@ -70,9 +70,9 @@ struct snd_bebob_meter_spec { int (*get)(struct snd_bebob *bebob, u32 *target, unsigned int size); }; struct snd_bebob_spec { - struct snd_bebob_clock_spec *clock; - struct snd_bebob_rate_spec *rate; - struct snd_bebob_meter_spec *meter; + const struct snd_bebob_clock_spec *clock; + const struct snd_bebob_rate_spec *rate; + const struct snd_bebob_meter_spec *meter; }; struct snd_bebob { @@ -235,19 +235,19 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob); int snd_bebob_create_hwdep_device(struct snd_bebob *bebob); /* model specific operations */ -extern struct snd_bebob_spec phase88_rack_spec; -extern struct snd_bebob_spec phase24_series_spec; -extern struct snd_bebob_spec yamaha_go_spec; -extern struct snd_bebob_spec saffirepro_26_spec; -extern struct snd_bebob_spec saffirepro_10_spec; -extern struct snd_bebob_spec saffire_le_spec; -extern struct snd_bebob_spec saffire_spec; -extern struct snd_bebob_spec maudio_fw410_spec; -extern struct snd_bebob_spec maudio_audiophile_spec; -extern struct snd_bebob_spec maudio_solo_spec; -extern struct snd_bebob_spec maudio_ozonic_spec; -extern struct snd_bebob_spec maudio_nrv10_spec; -extern struct snd_bebob_spec maudio_special_spec; +extern const struct snd_bebob_spec phase88_rack_spec; +extern const struct snd_bebob_spec phase24_series_spec; +extern const struct snd_bebob_spec yamaha_go_spec; +extern const struct snd_bebob_spec saffirepro_26_spec; +extern const struct snd_bebob_spec saffirepro_10_spec; +extern const struct snd_bebob_spec saffire_le_spec; +extern const struct snd_bebob_spec saffire_spec; +extern const struct snd_bebob_spec maudio_fw410_spec; +extern const struct snd_bebob_spec maudio_audiophile_spec; +extern const struct snd_bebob_spec maudio_solo_spec; +extern const struct snd_bebob_spec maudio_ozonic_spec; +extern const struct snd_bebob_spec maudio_nrv10_spec; +extern const struct snd_bebob_spec maudio_special_spec; int snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814); int snd_bebob_maudio_load_firmware(struct fw_unit *unit); diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c index a1a39494ea6c..f11090057949 100644 --- a/sound/firewire/bebob/bebob_focusrite.c +++ b/sound/firewire/bebob/bebob_focusrite.c @@ -200,7 +200,7 @@ end: return err; } -struct snd_bebob_spec saffire_le_spec; +const struct snd_bebob_spec saffire_le_spec; static enum snd_bebob_clock_type saffire_both_clk_src_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, SND_BEBOB_CLOCK_TYPE_EXTERNAL, @@ -229,7 +229,7 @@ static const char *const saffire_meter_labels[] = { static int saffire_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size) { - struct snd_bebob_meter_spec *spec = bebob->spec->meter; + const struct snd_bebob_meter_spec *spec = bebob->spec->meter; unsigned int channels; u64 offset; int err; @@ -260,60 +260,60 @@ saffire_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size) return err; } -static struct snd_bebob_rate_spec saffirepro_both_rate_spec = { +static const struct snd_bebob_rate_spec saffirepro_both_rate_spec = { .get = &saffirepro_both_clk_freq_get, .set = &saffirepro_both_clk_freq_set, }; /* Saffire Pro 26 I/O */ -static struct snd_bebob_clock_spec saffirepro_26_clk_spec = { +static const struct snd_bebob_clock_spec saffirepro_26_clk_spec = { .num = ARRAY_SIZE(saffirepro_26_clk_src_types), .types = saffirepro_26_clk_src_types, .get = &saffirepro_both_clk_src_get, }; -struct snd_bebob_spec saffirepro_26_spec = { +const struct snd_bebob_spec saffirepro_26_spec = { .clock = &saffirepro_26_clk_spec, .rate = &saffirepro_both_rate_spec, .meter = NULL }; /* Saffire Pro 10 I/O */ -static struct snd_bebob_clock_spec saffirepro_10_clk_spec = { +static const struct snd_bebob_clock_spec saffirepro_10_clk_spec = { .num = ARRAY_SIZE(saffirepro_10_clk_src_types), .types = saffirepro_10_clk_src_types, .get = &saffirepro_both_clk_src_get, }; -struct snd_bebob_spec saffirepro_10_spec = { +const struct snd_bebob_spec saffirepro_10_spec = { .clock = &saffirepro_10_clk_spec, .rate = &saffirepro_both_rate_spec, .meter = NULL }; -static struct snd_bebob_rate_spec saffire_both_rate_spec = { +static const struct snd_bebob_rate_spec saffire_both_rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate, }; -static struct snd_bebob_clock_spec saffire_both_clk_spec = { +static const struct snd_bebob_clock_spec saffire_both_clk_spec = { .num = ARRAY_SIZE(saffire_both_clk_src_types), .types = saffire_both_clk_src_types, .get = &saffire_both_clk_src_get, }; /* Saffire LE */ -static struct snd_bebob_meter_spec saffire_le_meter_spec = { +static const struct snd_bebob_meter_spec saffire_le_meter_spec = { .num = ARRAY_SIZE(saffire_le_meter_labels), .labels = saffire_le_meter_labels, .get = &saffire_meter_get, }; -struct snd_bebob_spec saffire_le_spec = { +const struct snd_bebob_spec saffire_le_spec = { .clock = &saffire_both_clk_spec, .rate = &saffire_both_rate_spec, .meter = &saffire_le_meter_spec }; /* Saffire */ -static struct snd_bebob_meter_spec saffire_meter_spec = { +static const struct snd_bebob_meter_spec saffire_meter_spec = { .num = ARRAY_SIZE(saffire_meter_labels), .labels = saffire_meter_labels, .get = &saffire_meter_get, }; -struct snd_bebob_spec saffire_spec = { +const struct snd_bebob_spec saffire_spec = { .clock = &saffire_both_clk_spec, .rate = &saffire_both_rate_spec, .meter = &saffire_meter_spec diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index 057495d54ab0..07e5abdbceb5 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -628,7 +628,7 @@ static const char *const special_meter_labels[] = { static int special_meter_get(struct snd_bebob *bebob, u32 *target, unsigned int size) { - u16 *buf; + __be16 *buf; unsigned int i, c, channels; int err; @@ -687,7 +687,7 @@ static const char *const nrv10_meter_labels[] = { static int normal_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size) { - struct snd_bebob_meter_spec *spec = bebob->spec->meter; + const struct snd_bebob_meter_spec *spec = bebob->spec->meter; unsigned int c, channels; int err; @@ -712,85 +712,85 @@ end: } /* for special customized devices */ -static struct snd_bebob_rate_spec special_rate_spec = { +static const struct snd_bebob_rate_spec special_rate_spec = { .get = &special_get_rate, .set = &special_set_rate, }; -static struct snd_bebob_clock_spec special_clk_spec = { +static const struct snd_bebob_clock_spec special_clk_spec = { .num = ARRAY_SIZE(special_clk_types), .types = special_clk_types, .get = &special_clk_get, }; -static struct snd_bebob_meter_spec special_meter_spec = { +static const struct snd_bebob_meter_spec special_meter_spec = { .num = ARRAY_SIZE(special_meter_labels), .labels = special_meter_labels, .get = &special_meter_get }; -struct snd_bebob_spec maudio_special_spec = { +const struct snd_bebob_spec maudio_special_spec = { .clock = &special_clk_spec, .rate = &special_rate_spec, .meter = &special_meter_spec }; /* Firewire 410 specification */ -static struct snd_bebob_rate_spec usual_rate_spec = { +static const struct snd_bebob_rate_spec usual_rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate, }; -static struct snd_bebob_meter_spec fw410_meter_spec = { +static const struct snd_bebob_meter_spec fw410_meter_spec = { .num = ARRAY_SIZE(fw410_meter_labels), .labels = fw410_meter_labels, .get = &normal_meter_get }; -struct snd_bebob_spec maudio_fw410_spec = { +const struct snd_bebob_spec maudio_fw410_spec = { .clock = NULL, .rate = &usual_rate_spec, .meter = &fw410_meter_spec }; /* Firewire Audiophile specification */ -static struct snd_bebob_meter_spec audiophile_meter_spec = { +static const struct snd_bebob_meter_spec audiophile_meter_spec = { .num = ARRAY_SIZE(audiophile_meter_labels), .labels = audiophile_meter_labels, .get = &normal_meter_get }; -struct snd_bebob_spec maudio_audiophile_spec = { +const struct snd_bebob_spec maudio_audiophile_spec = { .clock = NULL, .rate = &usual_rate_spec, .meter = &audiophile_meter_spec }; /* Firewire Solo specification */ -static struct snd_bebob_meter_spec solo_meter_spec = { +static const struct snd_bebob_meter_spec solo_meter_spec = { .num = ARRAY_SIZE(solo_meter_labels), .labels = solo_meter_labels, .get = &normal_meter_get }; -struct snd_bebob_spec maudio_solo_spec = { +const struct snd_bebob_spec maudio_solo_spec = { .clock = NULL, .rate = &usual_rate_spec, .meter = &solo_meter_spec }; /* Ozonic specification */ -static struct snd_bebob_meter_spec ozonic_meter_spec = { +static const struct snd_bebob_meter_spec ozonic_meter_spec = { .num = ARRAY_SIZE(ozonic_meter_labels), .labels = ozonic_meter_labels, .get = &normal_meter_get }; -struct snd_bebob_spec maudio_ozonic_spec = { +const struct snd_bebob_spec maudio_ozonic_spec = { .clock = NULL, .rate = &usual_rate_spec, .meter = &ozonic_meter_spec }; /* NRV10 specification */ -static struct snd_bebob_meter_spec nrv10_meter_spec = { +static const struct snd_bebob_meter_spec nrv10_meter_spec = { .num = ARRAY_SIZE(nrv10_meter_labels), .labels = nrv10_meter_labels, .get = &normal_meter_get }; -struct snd_bebob_spec maudio_nrv10_spec = { +const struct snd_bebob_spec maudio_nrv10_spec = { .clock = NULL, .rate = &usual_rate_spec, .meter = &nrv10_meter_spec diff --git a/sound/firewire/bebob/bebob_midi.c b/sound/firewire/bebob/bebob_midi.c index 5681143925cd..90d95be499b0 100644 --- a/sound/firewire/bebob/bebob_midi.c +++ b/sound/firewire/bebob/bebob_midi.c @@ -72,11 +72,11 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up) spin_lock_irqsave(&bebob->lock, flags); if (up) - amdtp_stream_midi_trigger(&bebob->tx_stream, - substrm->number, substrm); + amdtp_am824_midi_trigger(&bebob->tx_stream, + substrm->number, substrm); else - amdtp_stream_midi_trigger(&bebob->tx_stream, - substrm->number, NULL); + amdtp_am824_midi_trigger(&bebob->tx_stream, + substrm->number, NULL); spin_unlock_irqrestore(&bebob->lock, flags); } @@ -89,11 +89,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up) spin_lock_irqsave(&bebob->lock, flags); if (up) - amdtp_stream_midi_trigger(&bebob->rx_stream, - substrm->number, substrm); + amdtp_am824_midi_trigger(&bebob->rx_stream, + substrm->number, substrm); else - amdtp_stream_midi_trigger(&bebob->rx_stream, - substrm->number, NULL); + amdtp_am824_midi_trigger(&bebob->rx_stream, + substrm->number, NULL); spin_unlock_irqrestore(&bebob->lock, flags); } diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index c0f018a61fdc..ef224d6f5c24 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -122,11 +122,11 @@ pcm_init_hw_params(struct snd_bebob *bebob, SNDRV_PCM_INFO_MMAP_VALID; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS; + runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS; s = &bebob->tx_stream; formations = bebob->tx_stream_formations; } else { - runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS; + runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS; s = &bebob->rx_stream; formations = bebob->rx_stream_formations; } @@ -146,7 +146,7 @@ pcm_init_hw_params(struct snd_bebob *bebob, if (err < 0) goto end; - err = amdtp_stream_add_pcm_hw_constraints(s, runtime); + err = amdtp_am824_add_pcm_hw_constraints(s, runtime); end: return err; } @@ -155,7 +155,7 @@ static int pcm_open(struct snd_pcm_substream *substream) { struct snd_bebob *bebob = substream->private_data; - struct snd_bebob_rate_spec *spec = bebob->spec->rate; + const struct snd_bebob_rate_spec *spec = bebob->spec->rate; unsigned int sampling_rate; enum snd_bebob_clock_type src; int err; @@ -220,8 +220,8 @@ pcm_capture_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) atomic_inc(&bebob->substreams_counter); - amdtp_stream_set_pcm_format(&bebob->tx_stream, - params_format(hw_params)); + + amdtp_am824_set_pcm_format(&bebob->tx_stream, params_format(hw_params)); return 0; } @@ -239,8 +239,8 @@ pcm_playback_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) atomic_inc(&bebob->substreams_counter); - amdtp_stream_set_pcm_format(&bebob->rx_stream, - params_format(hw_params)); + + amdtp_am824_set_pcm_format(&bebob->rx_stream, params_format(hw_params)); return 0; } diff --git a/sound/firewire/bebob/bebob_proc.c b/sound/firewire/bebob/bebob_proc.c index 301cc6a93945..ec24f96794f5 100644 --- a/sound/firewire/bebob/bebob_proc.c +++ b/sound/firewire/bebob/bebob_proc.c @@ -73,7 +73,7 @@ proc_read_meters(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_bebob *bebob = entry->private_data; - struct snd_bebob_meter_spec *spec = bebob->spec->meter; + const struct snd_bebob_meter_spec *spec = bebob->spec->meter; u32 *buf; unsigned int i, c, channels, size; @@ -138,8 +138,8 @@ proc_read_clock(struct snd_info_entry *entry, "SYT-Match", }; struct snd_bebob *bebob = entry->private_data; - struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; - struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock; + const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; + const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock; enum snd_bebob_clock_type src; unsigned int rate; diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 5be5242e1ed8..926e5dcbb66a 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -119,7 +119,7 @@ end: int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob, enum snd_bebob_clock_type *src) { - struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock; + const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock; u8 addr[AVC_BRIDGECO_ADDR_BYTES], input[7]; unsigned int id; enum avc_bridgeco_plug_type type; @@ -338,7 +338,7 @@ map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) err = -ENOSYS; goto end; } - s->midi_position = stm_pos; + amdtp_am824_set_midi_position(s, stm_pos); midi = stm_pos; break; /* for PCM data channel */ @@ -354,11 +354,12 @@ map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) case 0x09: /* Digital */ default: location = pcm + sec_loc; - if (location >= AMDTP_MAX_CHANNELS_FOR_PCM) { + if (location >= AM824_MAX_CHANNELS_FOR_PCM) { err = -ENOSYS; goto end; } - s->pcm_positions[location] = stm_pos; + amdtp_am824_set_pcm_position(s, location, + stm_pos); break; } } @@ -427,12 +428,19 @@ make_both_connections(struct snd_bebob *bebob, unsigned int rate) index = get_formation_index(rate); pcm_channels = bebob->tx_stream_formations[index].pcm; midi_channels = bebob->tx_stream_formations[index].midi; - amdtp_stream_set_parameters(&bebob->tx_stream, - rate, pcm_channels, midi_channels * 8); + err = amdtp_am824_set_parameters(&bebob->tx_stream, rate, + pcm_channels, midi_channels * 8, + false); + if (err < 0) + goto end; + pcm_channels = bebob->rx_stream_formations[index].pcm; midi_channels = bebob->rx_stream_formations[index].midi; - amdtp_stream_set_parameters(&bebob->rx_stream, - rate, pcm_channels, midi_channels * 8); + err = amdtp_am824_set_parameters(&bebob->rx_stream, rate, + pcm_channels, midi_channels * 8, + false); + if (err < 0) + goto end; /* establish connections for both streams */ err = cmp_connection_establish(&bebob->out_conn, @@ -530,8 +538,8 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) if (err < 0) goto end; - err = amdtp_stream_init(&bebob->tx_stream, bebob->unit, - AMDTP_IN_STREAM, CIP_BLOCKING); + err = amdtp_am824_init(&bebob->tx_stream, bebob->unit, + AMDTP_IN_STREAM, CIP_BLOCKING); if (err < 0) { amdtp_stream_destroy(&bebob->tx_stream); destroy_both_connections(bebob); @@ -559,8 +567,8 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) if (bebob->maudio_special_quirk) bebob->tx_stream.flags |= CIP_EMPTY_HAS_WRONG_DBC; - err = amdtp_stream_init(&bebob->rx_stream, bebob->unit, - AMDTP_OUT_STREAM, CIP_BLOCKING); + err = amdtp_am824_init(&bebob->rx_stream, bebob->unit, + AMDTP_OUT_STREAM, CIP_BLOCKING); if (err < 0) { amdtp_stream_destroy(&bebob->tx_stream); amdtp_stream_destroy(&bebob->rx_stream); @@ -572,7 +580,7 @@ end: int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) { - struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; + const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; struct amdtp_stream *master, *slave; enum cip_flags sync_mode; unsigned int curr_rate; @@ -864,8 +872,8 @@ parse_stream_formation(u8 *buf, unsigned int len, } } - if (formation[i].pcm > AMDTP_MAX_CHANNELS_FOR_PCM || - formation[i].midi > AMDTP_MAX_CHANNELS_FOR_MIDI) + if (formation[i].pcm > AM824_MAX_CHANNELS_FOR_PCM || + formation[i].midi > AM824_MAX_CHANNELS_FOR_MIDI) return -ENOSYS; return 0; @@ -959,7 +967,7 @@ end: int snd_bebob_stream_discover(struct snd_bebob *bebob) { - struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock; + const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock; u8 plugs[AVC_PLUG_INFO_BUF_BYTES], addr[AVC_BRIDGECO_ADDR_BYTES]; enum avc_bridgeco_plug_type type; unsigned int i; diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c index 9242e33d2cf1..c38358b82ada 100644 --- a/sound/firewire/bebob/bebob_terratec.c +++ b/sound/firewire/bebob/bebob_terratec.c @@ -55,30 +55,30 @@ phase24_series_clk_src_get(struct snd_bebob *bebob, unsigned int *id) return 0; } -static struct snd_bebob_rate_spec phase_series_rate_spec = { +static const struct snd_bebob_rate_spec phase_series_rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate, }; /* PHASE 88 Rack FW */ -static struct snd_bebob_clock_spec phase88_rack_clk = { +static const struct snd_bebob_clock_spec phase88_rack_clk = { .num = ARRAY_SIZE(phase88_rack_clk_src_types), .types = phase88_rack_clk_src_types, .get = &phase88_rack_clk_src_get, }; -struct snd_bebob_spec phase88_rack_spec = { +const struct snd_bebob_spec phase88_rack_spec = { .clock = &phase88_rack_clk, .rate = &phase_series_rate_spec, .meter = NULL }; /* 'PHASE 24 FW' and 'PHASE X24 FW' */ -static struct snd_bebob_clock_spec phase24_series_clk = { +static const struct snd_bebob_clock_spec phase24_series_clk = { .num = ARRAY_SIZE(phase24_series_clk_src_types), .types = phase24_series_clk_src_types, .get = &phase24_series_clk_src_get, }; -struct snd_bebob_spec phase24_series_spec = { +const struct snd_bebob_spec phase24_series_spec = { .clock = &phase24_series_clk, .rate = &phase_series_rate_spec, .meter = NULL diff --git a/sound/firewire/bebob/bebob_yamaha.c b/sound/firewire/bebob/bebob_yamaha.c index 58101702410b..90d4404f77ce 100644 --- a/sound/firewire/bebob/bebob_yamaha.c +++ b/sound/firewire/bebob/bebob_yamaha.c @@ -46,16 +46,16 @@ clk_src_get(struct snd_bebob *bebob, unsigned int *id) return 0; } -static struct snd_bebob_clock_spec clock_spec = { +static const struct snd_bebob_clock_spec clock_spec = { .num = ARRAY_SIZE(clk_src_types), .types = clk_src_types, .get = &clk_src_get, }; -static struct snd_bebob_rate_spec rate_spec = { +static const struct snd_bebob_rate_spec rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate, }; -struct snd_bebob_spec yamaha_go_spec = { +const struct snd_bebob_spec yamaha_go_spec = { .clock = &clock_spec, .rate = &rate_spec, .meter = NULL diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 9ef228ef7baf..55b4be9b0034 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,3 +1,3 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o -obj-m += snd-dice.o +obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c index fe43ce791f84..151b09f240f2 100644 --- a/sound/firewire/dice/dice-midi.c +++ b/sound/firewire/dice/dice-midi.c @@ -52,10 +52,10 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up) spin_lock_irqsave(&dice->lock, flags); if (up) - amdtp_stream_midi_trigger(&dice->tx_stream, + amdtp_am824_midi_trigger(&dice->tx_stream, substrm->number, substrm); else - amdtp_stream_midi_trigger(&dice->tx_stream, + amdtp_am824_midi_trigger(&dice->tx_stream, substrm->number, NULL); spin_unlock_irqrestore(&dice->lock, flags); @@ -69,11 +69,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up) spin_lock_irqsave(&dice->lock, flags); if (up) - amdtp_stream_midi_trigger(&dice->rx_stream, - substrm->number, substrm); + amdtp_am824_midi_trigger(&dice->rx_stream, + substrm->number, substrm); else - amdtp_stream_midi_trigger(&dice->rx_stream, - substrm->number, NULL); + amdtp_am824_midi_trigger(&dice->rx_stream, + substrm->number, NULL); spin_unlock_irqrestore(&dice->lock, flags); } diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 4e67b1da0fe6..9b3431999fc8 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -133,11 +133,11 @@ static int init_hw_info(struct snd_dice *dice, SNDRV_PCM_INFO_BLOCK_TRANSFER; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - hw->formats = AMDTP_IN_PCM_FORMAT_BITS; + hw->formats = AM824_IN_PCM_FORMAT_BITS; stream = &dice->tx_stream; pcm_channels = dice->tx_channels; } else { - hw->formats = AMDTP_OUT_PCM_FORMAT_BITS; + hw->formats = AM824_OUT_PCM_FORMAT_BITS; stream = &dice->rx_stream; pcm_channels = dice->rx_channels; } @@ -156,7 +156,7 @@ static int init_hw_info(struct snd_dice *dice, if (err < 0) goto end; - err = amdtp_stream_add_pcm_hw_constraints(stream, runtime); + err = amdtp_am824_add_pcm_hw_constraints(stream, runtime); end: return err; } @@ -243,8 +243,7 @@ static int capture_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&dice->mutex); } - amdtp_stream_set_pcm_format(&dice->tx_stream, - params_format(hw_params)); + amdtp_am824_set_pcm_format(&dice->tx_stream, params_format(hw_params)); return 0; } @@ -265,8 +264,7 @@ static int playback_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&dice->mutex); } - amdtp_stream_set_pcm_format(&dice->rx_stream, - params_format(hw_params)); + amdtp_am824_set_pcm_format(&dice->rx_stream, params_format(hw_params)); return 0; } diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 07dbd01d7a6b..a6a39f7ef58d 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -44,16 +44,16 @@ int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate, static void release_resources(struct snd_dice *dice, struct fw_iso_resources *resources) { - unsigned int channel; + __be32 channel; /* Reset channel number */ channel = cpu_to_be32((u32)-1); if (resources == &dice->tx_resources) snd_dice_transaction_write_tx(dice, TX_ISOCHRONOUS, - &channel, 4); + &channel, sizeof(channel)); else snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, - &channel, 4); + &channel, sizeof(channel)); fw_iso_resources_free(resources); } @@ -62,7 +62,7 @@ static int keep_resources(struct snd_dice *dice, struct fw_iso_resources *resources, unsigned int max_payload_bytes) { - unsigned int channel; + __be32 channel; int err; err = fw_iso_resources_allocate(resources, max_payload_bytes, @@ -74,10 +74,10 @@ static int keep_resources(struct snd_dice *dice, channel = cpu_to_be32(resources->channel); if (resources == &dice->tx_resources) err = snd_dice_transaction_write_tx(dice, TX_ISOCHRONOUS, - &channel, 4); + &channel, sizeof(channel)); else err = snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, - &channel, 4); + &channel, sizeof(channel)); if (err < 0) release_resources(dice, resources); end: @@ -100,6 +100,7 @@ static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream, { struct fw_iso_resources *resources; unsigned int i, mode, pcm_chs, midi_ports; + bool double_pcm_frames; int err; err = snd_dice_stream_get_rate_mode(dice, rate, &mode); @@ -125,21 +126,24 @@ static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream, * For this quirk, blocking mode is required and PCM buffer size should * be aligned to SYT_INTERVAL. */ - if (mode > 1) { + double_pcm_frames = mode > 1; + if (double_pcm_frames) { rate /= 2; pcm_chs *= 2; - stream->double_pcm_frames = true; - } else { - stream->double_pcm_frames = false; } - amdtp_stream_set_parameters(stream, rate, pcm_chs, midi_ports); - if (mode > 1) { + err = amdtp_am824_set_parameters(stream, rate, pcm_chs, midi_ports, + double_pcm_frames); + if (err < 0) + goto end; + + if (double_pcm_frames) { pcm_chs /= 2; for (i = 0; i < pcm_chs; i++) { - stream->pcm_positions[i] = i * 2; - stream->pcm_positions[i + pcm_chs] = i * 2 + 1; + amdtp_am824_set_pcm_position(stream, i, i * 2); + amdtp_am824_set_pcm_position(stream, i + pcm_chs, + i * 2 + 1); } } @@ -302,7 +306,7 @@ static int init_stream(struct snd_dice *dice, struct amdtp_stream *stream) goto end; resources->channels_mask = 0x00000000ffffffffuLL; - err = amdtp_stream_init(stream, dice->unit, dir, CIP_BLOCKING); + err = amdtp_am824_init(stream, dice->unit, dir, CIP_BLOCKING); if (err < 0) { amdtp_stream_destroy(stream); fw_iso_resources_destroy(resources); diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 70a111d7f428..5d99436dfcae 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -29,7 +29,8 @@ static int dice_interface_check(struct fw_unit *unit) struct fw_csr_iterator it; int key, val, vendor = -1, model = -1, err; unsigned int category, i; - __be32 *pointers, value; + __be32 *pointers; + u32 value; __be32 version; pointers = kmalloc_array(ARRAY_SIZE(min_values), sizeof(__be32), diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index ecf5dc862235..101550ac1a24 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -34,7 +34,7 @@ #include <sound/pcm_params.h> #include <sound/rawmidi.h> -#include "../amdtp.h" +#include "../amdtp-am824.h" #include "../iso-resources.h" #include "../lib.h" #include "dice-interface.h" diff --git a/sound/firewire/digi00x/Makefile b/sound/firewire/digi00x/Makefile new file mode 100644 index 000000000000..1123e68c8b28 --- /dev/null +++ b/sound/firewire/digi00x/Makefile @@ -0,0 +1,4 @@ +snd-firewire-digi00x-objs := amdtp-dot.o digi00x-stream.o digi00x-proc.o \ + digi00x-pcm.o digi00x-hwdep.o \ + digi00x-transaction.o digi00x-midi.o digi00x.o +obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += snd-firewire-digi00x.o diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c new file mode 100644 index 000000000000..b02a5e8cad44 --- /dev/null +++ b/sound/firewire/digi00x/amdtp-dot.c @@ -0,0 +1,442 @@ +/* + * amdtp-dot.c - a part of driver for Digidesign Digi 002/003 family + * + * Copyright (c) 2014-2015 Takashi Sakamoto + * Copyright (C) 2012 Robin Gareus <robin@gareus.org> + * Copyright (C) 2012 Damien Zammit <damien@zamaudio.com> + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include <sound/pcm.h> +#include "digi00x.h" + +#define CIP_FMT_AM 0x10 + +/* 'Clock-based rate control mode' is just supported. */ +#define AMDTP_FDF_AM824 0x00 + +/* + * Nominally 3125 bytes/second, but the MIDI port's clock might be + * 1% too slow, and the bus clock 100 ppm too fast. + */ +#define MIDI_BYTES_PER_SECOND 3093 + +/* + * Several devices look only at the first eight data blocks. + * In any case, this is more than enough for the MIDI data rate. + */ +#define MAX_MIDI_RX_BLOCKS 8 + +/* + * The double-oh-three algorithm was discovered by Robin Gareus and Damien + * Zammit in 2012, with reverse-engineering for Digi 003 Rack. + */ +struct dot_state { + u8 carry; + u8 idx; + unsigned int off; +}; + +struct amdtp_dot { + unsigned int pcm_channels; + struct dot_state state; + + unsigned int midi_ports; + /* 2 = MAX(DOT_MIDI_IN_PORTS, DOT_MIDI_OUT_PORTS) */ + struct snd_rawmidi_substream *midi[2]; + int midi_fifo_used[2]; + int midi_fifo_limit; + + void (*transfer_samples)(struct amdtp_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames); +}; + +/* + * double-oh-three look up table + * + * @param idx index byte (audio-sample data) 0x00..0xff + * @param off channel offset shift + * @return salt to XOR with given data + */ +#define BYTE_PER_SAMPLE (4) +#define MAGIC_DOT_BYTE (2) +#define MAGIC_BYTE_OFF(x) (((x) * BYTE_PER_SAMPLE) + MAGIC_DOT_BYTE) +static const u8 dot_scrt(const u8 idx, const unsigned int off) +{ + /* + * the length of the added pattern only depends on the lower nibble + * of the last non-zero data + */ + static const u8 len[16] = {0, 1, 3, 5, 7, 9, 11, 13, 14, + 12, 10, 8, 6, 4, 2, 0}; + + /* + * the lower nibble of the salt. Interleaved sequence. + * this is walked backwards according to len[] + */ + static const u8 nib[15] = {0x8, 0x7, 0x9, 0x6, 0xa, 0x5, 0xb, 0x4, + 0xc, 0x3, 0xd, 0x2, 0xe, 0x1, 0xf}; + + /* circular list for the salt's hi nibble. */ + static const u8 hir[15] = {0x0, 0x6, 0xf, 0x8, 0x7, 0x5, 0x3, 0x4, + 0xc, 0xd, 0xe, 0x1, 0x2, 0xb, 0xa}; + + /* + * start offset for upper nibble mapping. + * note: 9 is /special/. In the case where the high nibble == 0x9, + * hir[] is not used and - coincidentally - the salt's hi nibble is + * 0x09 regardless of the offset. + */ + static const u8 hio[16] = {0, 11, 12, 6, 7, 5, 1, 4, + 3, 0x00, 14, 13, 8, 9, 10, 2}; + + const u8 ln = idx & 0xf; + const u8 hn = (idx >> 4) & 0xf; + const u8 hr = (hn == 0x9) ? 0x9 : hir[(hio[hn] + off) % 15]; + + if (len[ln] < off) + return 0x00; + + return ((nib[14 + off - len[ln]]) | (hr << 4)); +} + +static void dot_encode_step(struct dot_state *state, __be32 *const buffer) +{ + u8 * const data = (u8 *) buffer; + + if (data[MAGIC_DOT_BYTE] != 0x00) { + state->off = 0; + state->idx = data[MAGIC_DOT_BYTE] ^ state->carry; + } + data[MAGIC_DOT_BYTE] ^= state->carry; + state->carry = dot_scrt(state->idx, ++(state->off)); +} + +int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate, + unsigned int pcm_channels) +{ + struct amdtp_dot *p = s->protocol; + int err; + + if (amdtp_stream_running(s)) + return -EBUSY; + + /* + * A first data channel is for MIDI conformant data channel, the rest is + * Multi Bit Linear Audio data channel. + */ + err = amdtp_stream_set_parameters(s, rate, pcm_channels + 1); + if (err < 0) + return err; + + s->fdf = AMDTP_FDF_AM824 | s->sfc; + + p->pcm_channels = pcm_channels; + + if (s->direction == AMDTP_IN_STREAM) + p->midi_ports = DOT_MIDI_IN_PORTS; + else + p->midi_ports = DOT_MIDI_OUT_PORTS; + + /* + * We do not know the actual MIDI FIFO size of most devices. Just + * assume two bytes, i.e., one byte can be received over the bus while + * the previous one is transmitted over MIDI. + * (The value here is adjusted for midi_ratelimit_per_packet().) + */ + p->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1; + + return 0; +} + +static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames) +{ + struct amdtp_dot *p = s->protocol; + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int channels, remaining_frames, i, c; + const u32 *src; + + channels = p->pcm_channels; + src = (void *)runtime->dma_area + + frames_to_bytes(runtime, s->pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; + + buffer++; + for (i = 0; i < frames; ++i) { + for (c = 0; c < channels; ++c) { + buffer[c] = cpu_to_be32((*src >> 8) | 0x40000000); + dot_encode_step(&p->state, &buffer[c]); + src++; + } + buffer += s->data_block_quadlets; + if (--remaining_frames == 0) + src = (void *)runtime->dma_area; + } +} + +static void write_pcm_s16(struct amdtp_stream *s, struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames) +{ + struct amdtp_dot *p = s->protocol; + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int channels, remaining_frames, i, c; + const u16 *src; + + channels = p->pcm_channels; + src = (void *)runtime->dma_area + + frames_to_bytes(runtime, s->pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; + + buffer++; + for (i = 0; i < frames; ++i) { + for (c = 0; c < channels; ++c) { + buffer[c] = cpu_to_be32((*src << 8) | 0x40000000); + dot_encode_step(&p->state, &buffer[c]); + src++; + } + buffer += s->data_block_quadlets; + if (--remaining_frames == 0) + src = (void *)runtime->dma_area; + } +} + +static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames) +{ + struct amdtp_dot *p = s->protocol; + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int channels, remaining_frames, i, c; + u32 *dst; + + channels = p->pcm_channels; + dst = (void *)runtime->dma_area + + frames_to_bytes(runtime, s->pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; + + buffer++; + for (i = 0; i < frames; ++i) { + for (c = 0; c < channels; ++c) { + *dst = be32_to_cpu(buffer[c]) << 8; + dst++; + } + buffer += s->data_block_quadlets; + if (--remaining_frames == 0) + dst = (void *)runtime->dma_area; + } +} + +static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer, + unsigned int data_blocks) +{ + struct amdtp_dot *p = s->protocol; + unsigned int channels, i, c; + + channels = p->pcm_channels; + + buffer++; + for (i = 0; i < data_blocks; ++i) { + for (c = 0; c < channels; ++c) + buffer[c] = cpu_to_be32(0x40000000); + buffer += s->data_block_quadlets; + } +} + +static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port) +{ + struct amdtp_dot *p = s->protocol; + int used; + + used = p->midi_fifo_used[port]; + if (used == 0) + return true; + + used -= MIDI_BYTES_PER_SECOND * s->syt_interval; + used = max(used, 0); + p->midi_fifo_used[port] = used; + + return used < p->midi_fifo_limit; +} + +static inline void midi_use_bytes(struct amdtp_stream *s, + unsigned int port, unsigned int count) +{ + struct amdtp_dot *p = s->protocol; + + p->midi_fifo_used[port] += amdtp_rate_table[s->sfc] * count; +} + +static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer, + unsigned int data_blocks) +{ + struct amdtp_dot *p = s->protocol; + unsigned int f, port; + int len; + u8 *b; + + for (f = 0; f < data_blocks; f++) { + port = (s->data_block_counter + f) % 8; + b = (u8 *)&buffer[0]; + + len = 0; + if (port < p->midi_ports && + midi_ratelimit_per_packet(s, port) && + p->midi[port] != NULL) + len = snd_rawmidi_transmit(p->midi[port], b + 1, 2); + + if (len > 0) { + b[3] = (0x10 << port) | len; + midi_use_bytes(s, port, len); + } else { + b[1] = 0; + b[2] = 0; + b[3] = 0; + } + b[0] = 0x80; + + buffer += s->data_block_quadlets; + } +} + +static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer, + unsigned int data_blocks) +{ + struct amdtp_dot *p = s->protocol; + unsigned int f, port, len; + u8 *b; + + for (f = 0; f < data_blocks; f++) { + b = (u8 *)&buffer[0]; + port = b[3] >> 4; + len = b[3] & 0x0f; + + if (port < p->midi_ports && p->midi[port] && len > 0) + snd_rawmidi_receive(p->midi[port], b + 1, len); + + buffer += s->data_block_quadlets; + } +} + +int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s, + struct snd_pcm_runtime *runtime) +{ + int err; + + /* This protocol delivers 24 bit data in 32bit data channel. */ + err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + if (err < 0) + return err; + + return amdtp_stream_add_pcm_hw_constraints(s, runtime); +} + +void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format) +{ + struct amdtp_dot *p = s->protocol; + + if (WARN_ON(amdtp_stream_pcm_running(s))) + return; + + switch (format) { + default: + WARN_ON(1); + /* fall through */ + case SNDRV_PCM_FORMAT_S16: + if (s->direction == AMDTP_OUT_STREAM) { + p->transfer_samples = write_pcm_s16; + break; + } + WARN_ON(1); + /* fall through */ + case SNDRV_PCM_FORMAT_S32: + if (s->direction == AMDTP_OUT_STREAM) + p->transfer_samples = write_pcm_s32; + else + p->transfer_samples = read_pcm_s32; + break; + } +} + +void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port, + struct snd_rawmidi_substream *midi) +{ + struct amdtp_dot *p = s->protocol; + + if (port < p->midi_ports) + ACCESS_ONCE(p->midi[port]) = midi; +} + +static unsigned int process_tx_data_blocks(struct amdtp_stream *s, + __be32 *buffer, + unsigned int data_blocks, + unsigned int *syt) +{ + struct amdtp_dot *p = (struct amdtp_dot *)s->protocol; + struct snd_pcm_substream *pcm; + unsigned int pcm_frames; + + pcm = ACCESS_ONCE(s->pcm); + if (pcm) { + p->transfer_samples(s, pcm, buffer, data_blocks); + pcm_frames = data_blocks; + } else { + pcm_frames = 0; + } + + read_midi_messages(s, buffer, data_blocks); + + return pcm_frames; +} + +static unsigned int process_rx_data_blocks(struct amdtp_stream *s, + __be32 *buffer, + unsigned int data_blocks, + unsigned int *syt) +{ + struct amdtp_dot *p = (struct amdtp_dot *)s->protocol; + struct snd_pcm_substream *pcm; + unsigned int pcm_frames; + + pcm = ACCESS_ONCE(s->pcm); + if (pcm) { + p->transfer_samples(s, pcm, buffer, data_blocks); + pcm_frames = data_blocks; + } else { + write_pcm_silence(s, buffer, data_blocks); + pcm_frames = 0; + } + + write_midi_messages(s, buffer, data_blocks); + + return pcm_frames; +} + +int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit, + enum amdtp_stream_direction dir) +{ + amdtp_stream_process_data_blocks_t process_data_blocks; + enum cip_flags flags; + + /* Use different mode between incoming/outgoing. */ + if (dir == AMDTP_IN_STREAM) { + flags = CIP_NONBLOCKING | CIP_SKIP_INIT_DBC_CHECK; + process_data_blocks = process_tx_data_blocks; + } else { + flags = CIP_BLOCKING; + process_data_blocks = process_rx_data_blocks; + } + + return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM, + process_data_blocks, sizeof(struct amdtp_dot)); +} + +void amdtp_dot_reset(struct amdtp_stream *s) +{ + struct amdtp_dot *p = s->protocol; + + p->state.carry = 0x00; + p->state.idx = 0x00; + p->state.off = 0; +} diff --git a/sound/firewire/digi00x/digi00x-hwdep.c b/sound/firewire/digi00x/digi00x-hwdep.c new file mode 100644 index 000000000000..f188e4758fd2 --- /dev/null +++ b/sound/firewire/digi00x/digi00x-hwdep.c @@ -0,0 +1,200 @@ +/* + * digi00x-hwdep.c - a part of driver for Digidesign Digi 002/003 family + * + * Copyright (c) 2014-2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +/* + * This codes give three functionality. + * + * 1.get firewire node information + * 2.get notification about starting/stopping stream + * 3.lock/unlock stream + * 4.get asynchronous messaging + */ + +#include "digi00x.h" + +static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, + loff_t *offset) +{ + struct snd_dg00x *dg00x = hwdep->private_data; + DEFINE_WAIT(wait); + union snd_firewire_event event; + + spin_lock_irq(&dg00x->lock); + + while (!dg00x->dev_lock_changed && dg00x->msg == 0) { + prepare_to_wait(&dg00x->hwdep_wait, &wait, TASK_INTERRUPTIBLE); + spin_unlock_irq(&dg00x->lock); + schedule(); + finish_wait(&dg00x->hwdep_wait, &wait); + if (signal_pending(current)) + return -ERESTARTSYS; + spin_lock_irq(&dg00x->lock); + } + + memset(&event, 0, sizeof(event)); + if (dg00x->dev_lock_changed) { + event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; + event.lock_status.status = (dg00x->dev_lock_count > 0); + dg00x->dev_lock_changed = false; + + count = min_t(long, count, sizeof(event.lock_status)); + } else { + event.digi00x_message.type = + SNDRV_FIREWIRE_EVENT_DIGI00X_MESSAGE; + event.digi00x_message.message = dg00x->msg; + dg00x->msg = 0; + + count = min_t(long, count, sizeof(event.digi00x_message)); + } + + spin_unlock_irq(&dg00x->lock); + + if (copy_to_user(buf, &event, count)) + return -EFAULT; + + return count; +} + +static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file, + poll_table *wait) +{ + struct snd_dg00x *dg00x = hwdep->private_data; + unsigned int events; + + poll_wait(file, &dg00x->hwdep_wait, wait); + + spin_lock_irq(&dg00x->lock); + if (dg00x->dev_lock_changed || dg00x->msg) + events = POLLIN | POLLRDNORM; + else + events = 0; + spin_unlock_irq(&dg00x->lock); + + return events; +} + +static int hwdep_get_info(struct snd_dg00x *dg00x, void __user *arg) +{ + struct fw_device *dev = fw_parent_device(dg00x->unit); + struct snd_firewire_get_info info; + + memset(&info, 0, sizeof(info)); + info.type = SNDRV_FIREWIRE_TYPE_DIGI00X; + info.card = dev->card->index; + *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); + *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); + strlcpy(info.device_name, dev_name(&dev->device), + sizeof(info.device_name)); + + if (copy_to_user(arg, &info, sizeof(info))) + return -EFAULT; + + return 0; +} + +static int hwdep_lock(struct snd_dg00x *dg00x) +{ + int err; + + spin_lock_irq(&dg00x->lock); + + if (dg00x->dev_lock_count == 0) { + dg00x->dev_lock_count = -1; + err = 0; + } else { + err = -EBUSY; + } + + spin_unlock_irq(&dg00x->lock); + + return err; +} + +static int hwdep_unlock(struct snd_dg00x *dg00x) +{ + int err; + + spin_lock_irq(&dg00x->lock); + + if (dg00x->dev_lock_count == -1) { + dg00x->dev_lock_count = 0; + err = 0; + } else { + err = -EBADFD; + } + + spin_unlock_irq(&dg00x->lock); + + return err; +} + +static int hwdep_release(struct snd_hwdep *hwdep, struct file *file) +{ + struct snd_dg00x *dg00x = hwdep->private_data; + + spin_lock_irq(&dg00x->lock); + if (dg00x->dev_lock_count == -1) + dg00x->dev_lock_count = 0; + spin_unlock_irq(&dg00x->lock); + + return 0; +} + +static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + struct snd_dg00x *dg00x = hwdep->private_data; + + switch (cmd) { + case SNDRV_FIREWIRE_IOCTL_GET_INFO: + return hwdep_get_info(dg00x, (void __user *)arg); + case SNDRV_FIREWIRE_IOCTL_LOCK: + return hwdep_lock(dg00x); + case SNDRV_FIREWIRE_IOCTL_UNLOCK: + return hwdep_unlock(dg00x); + default: + return -ENOIOCTLCMD; + } +} + +#ifdef CONFIG_COMPAT +static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + return hwdep_ioctl(hwdep, file, cmd, + (unsigned long)compat_ptr(arg)); +} +#else +#define hwdep_compat_ioctl NULL +#endif + +static const struct snd_hwdep_ops hwdep_ops = { + .read = hwdep_read, + .release = hwdep_release, + .poll = hwdep_poll, + .ioctl = hwdep_ioctl, + .ioctl_compat = hwdep_compat_ioctl, +}; + +int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x) +{ + struct snd_hwdep *hwdep; + int err; + + err = snd_hwdep_new(dg00x->card, "Digi00x", 0, &hwdep); + if (err < 0) + return err; + + strcpy(hwdep->name, "Digi00x"); + hwdep->iface = SNDRV_HWDEP_IFACE_FW_DIGI00X; + hwdep->ops = hwdep_ops; + hwdep->private_data = dg00x; + hwdep->exclusive = true; + + return err; +} diff --git a/sound/firewire/digi00x/digi00x-midi.c b/sound/firewire/digi00x/digi00x-midi.c new file mode 100644 index 000000000000..1a72a382b384 --- /dev/null +++ b/sound/firewire/digi00x/digi00x-midi.c @@ -0,0 +1,223 @@ +/* + * digi00x-midi.h - a part of driver for Digidesign Digi 002/003 family + * + * Copyright (c) 2014-2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "digi00x.h" + +static int midi_phys_open(struct snd_rawmidi_substream *substream) +{ + struct snd_dg00x *dg00x = substream->rmidi->private_data; + int err; + + err = snd_dg00x_stream_lock_try(dg00x); + if (err < 0) + return err; + + mutex_lock(&dg00x->mutex); + dg00x->substreams_counter++; + err = snd_dg00x_stream_start_duplex(dg00x, 0); + mutex_unlock(&dg00x->mutex); + if (err < 0) + snd_dg00x_stream_lock_release(dg00x); + + return err; +} + +static int midi_phys_close(struct snd_rawmidi_substream *substream) +{ + struct snd_dg00x *dg00x = substream->rmidi->private_data; + + mutex_lock(&dg00x->mutex); + dg00x->substreams_counter--; + snd_dg00x_stream_stop_duplex(dg00x); + mutex_unlock(&dg00x->mutex); + + snd_dg00x_stream_lock_release(dg00x); + return 0; +} + +static void midi_phys_capture_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct snd_dg00x *dg00x = substream->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&dg00x->lock, flags); + + if (up) + amdtp_dot_midi_trigger(&dg00x->tx_stream, substream->number, + substream); + else + amdtp_dot_midi_trigger(&dg00x->tx_stream, substream->number, + NULL); + + spin_unlock_irqrestore(&dg00x->lock, flags); +} + +static void midi_phys_playback_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct snd_dg00x *dg00x = substream->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&dg00x->lock, flags); + + if (up) + amdtp_dot_midi_trigger(&dg00x->rx_stream, substream->number, + substream); + else + amdtp_dot_midi_trigger(&dg00x->rx_stream, substream->number, + NULL); + + spin_unlock_irqrestore(&dg00x->lock, flags); +} + +static struct snd_rawmidi_ops midi_phys_capture_ops = { + .open = midi_phys_open, + .close = midi_phys_close, + .trigger = midi_phys_capture_trigger, +}; + +static struct snd_rawmidi_ops midi_phys_playback_ops = { + .open = midi_phys_open, + .close = midi_phys_close, + .trigger = midi_phys_playback_trigger, +}; + +static int midi_ctl_open(struct snd_rawmidi_substream *substream) +{ + /* Do nothing. */ + return 0; +} + +static int midi_ctl_capture_close(struct snd_rawmidi_substream *substream) +{ + /* Do nothing. */ + return 0; +} + +static int midi_ctl_playback_close(struct snd_rawmidi_substream *substream) +{ + struct snd_dg00x *dg00x = substream->rmidi->private_data; + + snd_fw_async_midi_port_finish(&dg00x->out_control); + + return 0; +} + +static void midi_ctl_capture_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct snd_dg00x *dg00x = substream->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&dg00x->lock, flags); + + if (up) + dg00x->in_control = substream; + else + dg00x->in_control = NULL; + + spin_unlock_irqrestore(&dg00x->lock, flags); +} + +static void midi_ctl_playback_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct snd_dg00x *dg00x = substream->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&dg00x->lock, flags); + + if (up) + snd_fw_async_midi_port_run(&dg00x->out_control, substream); + + spin_unlock_irqrestore(&dg00x->lock, flags); +} + +static struct snd_rawmidi_ops midi_ctl_capture_ops = { + .open = midi_ctl_open, + .close = midi_ctl_capture_close, + .trigger = midi_ctl_capture_trigger, +}; + +static struct snd_rawmidi_ops midi_ctl_playback_ops = { + .open = midi_ctl_open, + .close = midi_ctl_playback_close, + .trigger = midi_ctl_playback_trigger, +}; + +static void set_midi_substream_names(struct snd_dg00x *dg00x, + struct snd_rawmidi_str *str, + bool is_ctl) +{ + struct snd_rawmidi_substream *subs; + + list_for_each_entry(subs, &str->substreams, list) { + if (!is_ctl) + snprintf(subs->name, sizeof(subs->name), + "%s MIDI %d", + dg00x->card->shortname, subs->number + 1); + else + /* This port is for asynchronous transaction. */ + snprintf(subs->name, sizeof(subs->name), + "%s control", + dg00x->card->shortname); + } +} + +int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x) +{ + struct snd_rawmidi *rmidi[2]; + struct snd_rawmidi_str *str; + unsigned int i; + int err; + + /* Add physical midi ports. */ + err = snd_rawmidi_new(dg00x->card, dg00x->card->driver, 0, + DOT_MIDI_OUT_PORTS, DOT_MIDI_IN_PORTS, &rmidi[0]); + if (err < 0) + return err; + + snprintf(rmidi[0]->name, sizeof(rmidi[0]->name), + "%s MIDI", dg00x->card->shortname); + + snd_rawmidi_set_ops(rmidi[0], SNDRV_RAWMIDI_STREAM_INPUT, + &midi_phys_capture_ops); + snd_rawmidi_set_ops(rmidi[0], SNDRV_RAWMIDI_STREAM_OUTPUT, + &midi_phys_playback_ops); + + /* Add a pair of control midi ports. */ + err = snd_rawmidi_new(dg00x->card, dg00x->card->driver, 1, + 1, 1, &rmidi[1]); + if (err < 0) + return err; + + snprintf(rmidi[1]->name, sizeof(rmidi[1]->name), + "%s control", dg00x->card->shortname); + + snd_rawmidi_set_ops(rmidi[1], SNDRV_RAWMIDI_STREAM_INPUT, + &midi_ctl_capture_ops); + snd_rawmidi_set_ops(rmidi[1], SNDRV_RAWMIDI_STREAM_OUTPUT, + &midi_ctl_playback_ops); + + for (i = 0; i < ARRAY_SIZE(rmidi); i++) { + rmidi[i]->private_data = dg00x; + + rmidi[i]->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + str = &rmidi[i]->streams[SNDRV_RAWMIDI_STREAM_INPUT]; + set_midi_substream_names(dg00x, str, i); + + rmidi[i]->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + str = &rmidi[i]->streams[SNDRV_RAWMIDI_STREAM_OUTPUT]; + set_midi_substream_names(dg00x, str, i); + + rmidi[i]->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX; + } + + return 0; +} diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c new file mode 100644 index 000000000000..cac28f70aef7 --- /dev/null +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -0,0 +1,373 @@ +/* + * digi00x-pcm.c - a part of driver for Digidesign Digi 002/003 family + * + * Copyright (c) 2014-2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "digi00x.h" + +static int hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + const struct snd_interval *c = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval t = { + .min = UINT_MAX, .max = 0, .integer = 1, + }; + unsigned int i; + + for (i = 0; i < SND_DG00X_RATE_COUNT; i++) { + if (!snd_interval_test(c, + snd_dg00x_stream_pcm_channels[i])) + continue; + + t.min = min(t.min, snd_dg00x_stream_rates[i]); + t.max = max(t.max, snd_dg00x_stream_rates[i]); + } + + return snd_interval_refine(r, &t); +} + +static int hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *c = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + const struct snd_interval *r = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval t = { + .min = UINT_MAX, .max = 0, .integer = 1, + }; + unsigned int i; + + for (i = 0; i < SND_DG00X_RATE_COUNT; i++) { + if (!snd_interval_test(r, snd_dg00x_stream_rates[i])) + continue; + + t.min = min(t.min, snd_dg00x_stream_pcm_channels[i]); + t.max = max(t.max, snd_dg00x_stream_pcm_channels[i]); + } + + return snd_interval_refine(c, &t); +} + +static int pcm_init_hw_params(struct snd_dg00x *dg00x, + struct snd_pcm_substream *substream) +{ + static const struct snd_pcm_hardware hardware = { + .info = SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_JOINT_DUPLEX | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .channels_min = 10, + .channels_max = 18, + .period_bytes_min = 4 * 18, + .period_bytes_max = 4 * 18 * 2048, + .buffer_bytes_max = 4 * 18 * 2048 * 2, + .periods_min = 2, + .periods_max = UINT_MAX, + }; + struct amdtp_stream *s; + int err; + + substream->runtime->hw = hardware; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; + s = &dg00x->tx_stream; + } else { + substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16 | + SNDRV_PCM_FMTBIT_S32; + s = &dg00x->rx_stream; + } + + err = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + + err = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + hw_rule_rate, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + return err; + + return amdtp_dot_add_pcm_hw_constraints(s, substream->runtime); +} + +static int pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_dg00x *dg00x = substream->private_data; + enum snd_dg00x_clock clock; + bool detect; + unsigned int rate; + int err; + + err = snd_dg00x_stream_lock_try(dg00x); + if (err < 0) + goto end; + + err = pcm_init_hw_params(dg00x, substream); + if (err < 0) + goto err_locked; + + /* Check current clock source. */ + err = snd_dg00x_stream_get_clock(dg00x, &clock); + if (err < 0) + goto err_locked; + if (clock != SND_DG00X_CLOCK_INTERNAL) { + err = snd_dg00x_stream_check_external_clock(dg00x, &detect); + if (err < 0) + goto err_locked; + if (!detect) { + err = -EBUSY; + goto err_locked; + } + } + + if ((clock != SND_DG00X_CLOCK_INTERNAL) || + amdtp_stream_pcm_running(&dg00x->rx_stream) || + amdtp_stream_pcm_running(&dg00x->tx_stream)) { + err = snd_dg00x_stream_get_external_rate(dg00x, &rate); + if (err < 0) + goto err_locked; + substream->runtime->hw.rate_min = rate; + substream->runtime->hw.rate_max = rate; + } + + snd_pcm_set_sync(substream); +end: + return err; +err_locked: + snd_dg00x_stream_lock_release(dg00x); + return err; +} + +static int pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_dg00x *dg00x = substream->private_data; + + snd_dg00x_stream_lock_release(dg00x); + + return 0; +} + +static int pcm_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_dg00x *dg00x = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + + if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + mutex_lock(&dg00x->mutex); + dg00x->substreams_counter++; + mutex_unlock(&dg00x->mutex); + } + + amdtp_dot_set_pcm_format(&dg00x->tx_stream, params_format(hw_params)); + + return 0; +} + +static int pcm_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_dg00x *dg00x = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + + if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + mutex_lock(&dg00x->mutex); + dg00x->substreams_counter++; + mutex_unlock(&dg00x->mutex); + } + + amdtp_dot_set_pcm_format(&dg00x->rx_stream, params_format(hw_params)); + + return 0; +} + +static int pcm_capture_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_dg00x *dg00x = substream->private_data; + + mutex_lock(&dg00x->mutex); + + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + dg00x->substreams_counter--; + + snd_dg00x_stream_stop_duplex(dg00x); + + mutex_unlock(&dg00x->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int pcm_playback_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_dg00x *dg00x = substream->private_data; + + mutex_lock(&dg00x->mutex); + + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + dg00x->substreams_counter--; + + snd_dg00x_stream_stop_duplex(dg00x); + + mutex_unlock(&dg00x->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_dg00x *dg00x = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + mutex_lock(&dg00x->mutex); + + err = snd_dg00x_stream_start_duplex(dg00x, runtime->rate); + if (err >= 0) + amdtp_stream_pcm_prepare(&dg00x->tx_stream); + + mutex_unlock(&dg00x->mutex); + + return err; +} + +static int pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_dg00x *dg00x = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + mutex_lock(&dg00x->mutex); + + err = snd_dg00x_stream_start_duplex(dg00x, runtime->rate); + if (err >= 0) { + amdtp_stream_pcm_prepare(&dg00x->rx_stream); + amdtp_dot_reset(&dg00x->rx_stream); + } + + mutex_unlock(&dg00x->mutex); + + return err; +} + +static int pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_dg00x *dg00x = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + amdtp_stream_pcm_trigger(&dg00x->tx_stream, substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + amdtp_stream_pcm_trigger(&dg00x->tx_stream, NULL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_dg00x *dg00x = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + amdtp_stream_pcm_trigger(&dg00x->rx_stream, substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + amdtp_stream_pcm_trigger(&dg00x->rx_stream, NULL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm) +{ + struct snd_dg00x *dg00x = sbstrm->private_data; + + return amdtp_stream_pcm_pointer(&dg00x->tx_stream); +} + +static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm) +{ + struct snd_dg00x *dg00x = sbstrm->private_data; + + return amdtp_stream_pcm_pointer(&dg00x->rx_stream); +} + +static struct snd_pcm_ops pcm_capture_ops = { + .open = pcm_open, + .close = pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pcm_capture_hw_params, + .hw_free = pcm_capture_hw_free, + .prepare = pcm_capture_prepare, + .trigger = pcm_capture_trigger, + .pointer = pcm_capture_pointer, + .page = snd_pcm_lib_get_vmalloc_page, +}; + +static struct snd_pcm_ops pcm_playback_ops = { + .open = pcm_open, + .close = pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pcm_playback_hw_params, + .hw_free = pcm_playback_hw_free, + .prepare = pcm_playback_prepare, + .trigger = pcm_playback_trigger, + .pointer = pcm_playback_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, +}; + +int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x) +{ + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(dg00x->card, dg00x->card->driver, 0, 1, 1, &pcm); + if (err < 0) + return err; + + pcm->private_data = dg00x; + snprintf(pcm->name, sizeof(pcm->name), + "%s PCM", dg00x->card->shortname); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops); + + return 0; +} diff --git a/sound/firewire/digi00x/digi00x-proc.c b/sound/firewire/digi00x/digi00x-proc.c new file mode 100644 index 000000000000..a1d601f31165 --- /dev/null +++ b/sound/firewire/digi00x/digi00x-proc.c @@ -0,0 +1,99 @@ +/* + * digi00x-proc.c - a part of driver for Digidesign Digi 002/003 family + * + * Copyright (c) 2014-2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "digi00x.h" + +static int get_optical_iface_mode(struct snd_dg00x *dg00x, + enum snd_dg00x_optical_mode *mode) +{ + __be32 data; + int err; + + err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_OPT_IFACE_MODE, + &data, sizeof(data), 0); + if (err >= 0) + *mode = be32_to_cpu(data) & 0x01; + + return err; +} + +static void proc_read_clock(struct snd_info_entry *entry, + struct snd_info_buffer *buf) +{ + static const char *const source_name[] = { + [SND_DG00X_CLOCK_INTERNAL] = "internal", + [SND_DG00X_CLOCK_SPDIF] = "s/pdif", + [SND_DG00X_CLOCK_ADAT] = "adat", + [SND_DG00X_CLOCK_WORD] = "word clock", + }; + static const char *const optical_name[] = { + [SND_DG00X_OPT_IFACE_MODE_ADAT] = "adat", + [SND_DG00X_OPT_IFACE_MODE_SPDIF] = "s/pdif", + }; + struct snd_dg00x *dg00x = entry->private_data; + enum snd_dg00x_optical_mode mode; + unsigned int rate; + enum snd_dg00x_clock clock; + bool detect; + + if (get_optical_iface_mode(dg00x, &mode) < 0) + return; + if (snd_dg00x_stream_get_local_rate(dg00x, &rate) < 0) + return; + if (snd_dg00x_stream_get_clock(dg00x, &clock) < 0) + return; + + snd_iprintf(buf, "Optical mode: %s\n", optical_name[mode]); + snd_iprintf(buf, "Sampling Rate: %d\n", rate); + snd_iprintf(buf, "Clock Source: %s\n", source_name[clock]); + + if (clock == SND_DG00X_CLOCK_INTERNAL) + return; + + if (snd_dg00x_stream_check_external_clock(dg00x, &detect) < 0) + return; + snd_iprintf(buf, "External source: %s\n", detect ? "detected" : "not"); + if (!detect) + return; + + if (snd_dg00x_stream_get_external_rate(dg00x, &rate) >= 0) + snd_iprintf(buf, "External sampling rate: %d\n", rate); +} + +void snd_dg00x_proc_init(struct snd_dg00x *dg00x) +{ + struct snd_info_entry *root, *entry; + + /* + * All nodes are automatically removed at snd_card_disconnect(), + * by following to link list. + */ + root = snd_info_create_card_entry(dg00x->card, "firewire", + dg00x->card->proc_root); + if (root == NULL) + return; + + root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + if (snd_info_register(root) < 0) { + snd_info_free_entry(root); + return; + } + + entry = snd_info_create_card_entry(dg00x->card, "clock", root); + if (entry == NULL) { + snd_info_free_entry(root); + return; + } + + snd_info_set_text_ops(entry, dg00x, proc_read_clock); + if (snd_info_register(entry) < 0) { + snd_info_free_entry(entry); + snd_info_free_entry(root); + } +} diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c new file mode 100644 index 000000000000..4d3b4ebbdd49 --- /dev/null +++ b/sound/firewire/digi00x/digi00x-stream.c @@ -0,0 +1,422 @@ +/* + * digi00x-stream.c - a part of driver for Digidesign Digi 002/003 family + * + * Copyright (c) 2014-2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "digi00x.h" + +#define CALLBACK_TIMEOUT 500 + +const unsigned int snd_dg00x_stream_rates[SND_DG00X_RATE_COUNT] = { + [SND_DG00X_RATE_44100] = 44100, + [SND_DG00X_RATE_48000] = 48000, + [SND_DG00X_RATE_88200] = 88200, + [SND_DG00X_RATE_96000] = 96000, +}; + +/* Multi Bit Linear Audio data channels for each sampling transfer frequency. */ +const unsigned int +snd_dg00x_stream_pcm_channels[SND_DG00X_RATE_COUNT] = { + /* Analog/ADAT/SPDIF */ + [SND_DG00X_RATE_44100] = (8 + 8 + 2), + [SND_DG00X_RATE_48000] = (8 + 8 + 2), + /* Analog/SPDIF */ + [SND_DG00X_RATE_88200] = (8 + 2), + [SND_DG00X_RATE_96000] = (8 + 2), +}; + +int snd_dg00x_stream_get_local_rate(struct snd_dg00x *dg00x, unsigned int *rate) +{ + u32 data; + __be32 reg; + int err; + + err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_LOCAL_RATE, + ®, sizeof(reg), 0); + if (err < 0) + return err; + + data = be32_to_cpu(reg) & 0x0f; + if (data < ARRAY_SIZE(snd_dg00x_stream_rates)) + *rate = snd_dg00x_stream_rates[data]; + else + err = -EIO; + + return err; +} + +int snd_dg00x_stream_set_local_rate(struct snd_dg00x *dg00x, unsigned int rate) +{ + __be32 reg; + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(snd_dg00x_stream_rates); i++) { + if (rate == snd_dg00x_stream_rates[i]) + break; + } + if (i == ARRAY_SIZE(snd_dg00x_stream_rates)) + return -EINVAL; + + reg = cpu_to_be32(i); + return snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_LOCAL_RATE, + ®, sizeof(reg), 0); +} + +int snd_dg00x_stream_get_clock(struct snd_dg00x *dg00x, + enum snd_dg00x_clock *clock) +{ + __be32 reg; + int err; + + err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_CLOCK_SOURCE, + ®, sizeof(reg), 0); + if (err < 0) + return err; + + *clock = be32_to_cpu(reg) & 0x0f; + if (*clock >= SND_DG00X_CLOCK_COUNT) + err = -EIO; + + return err; +} + +int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x, bool *detect) +{ + __be32 reg; + int err; + + err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_DETECT_EXTERNAL, + ®, sizeof(reg), 0); + if (err >= 0) + *detect = be32_to_cpu(reg) > 0; + + return err; +} + +int snd_dg00x_stream_get_external_rate(struct snd_dg00x *dg00x, + unsigned int *rate) +{ + u32 data; + __be32 reg; + int err; + + err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_EXTERNAL_RATE, + ®, sizeof(reg), 0); + if (err < 0) + return err; + + data = be32_to_cpu(reg) & 0x0f; + if (data < ARRAY_SIZE(snd_dg00x_stream_rates)) + *rate = snd_dg00x_stream_rates[data]; + /* This means desync. */ + else + err = -EBUSY; + + return err; +} + +static void finish_session(struct snd_dg00x *dg00x) +{ + __be32 data = cpu_to_be32(0x00000003); + + snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_SET, + &data, sizeof(data), 0); +} + +static int begin_session(struct snd_dg00x *dg00x) +{ + __be32 data; + u32 curr; + int err; + + err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_STATE, + &data, sizeof(data), 0); + if (err < 0) + goto error; + curr = be32_to_cpu(data); + + if (curr == 0) + curr = 2; + + curr--; + while (curr > 0) { + data = cpu_to_be32(curr); + err = snd_fw_transaction(dg00x->unit, + TCODE_WRITE_QUADLET_REQUEST, + DG00X_ADDR_BASE + + DG00X_OFFSET_STREAMING_SET, + &data, sizeof(data), 0); + if (err < 0) + goto error; + + msleep(20); + curr--; + } + + return 0; +error: + finish_session(dg00x); + return err; +} + +static void release_resources(struct snd_dg00x *dg00x) +{ + __be32 data = 0; + + /* Unregister isochronous channels for both direction. */ + snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS, + &data, sizeof(data), 0); + + /* Release isochronous resources. */ + fw_iso_resources_free(&dg00x->tx_resources); + fw_iso_resources_free(&dg00x->rx_resources); +} + +static int keep_resources(struct snd_dg00x *dg00x, unsigned int rate) +{ + unsigned int i; + __be32 data; + int err; + + /* Check sampling rate. */ + for (i = 0; i < SND_DG00X_RATE_COUNT; i++) { + if (snd_dg00x_stream_rates[i] == rate) + break; + } + if (i == SND_DG00X_RATE_COUNT) + return -EINVAL; + + /* Keep resources for out-stream. */ + err = amdtp_dot_set_parameters(&dg00x->rx_stream, rate, + snd_dg00x_stream_pcm_channels[i]); + if (err < 0) + return err; + err = fw_iso_resources_allocate(&dg00x->rx_resources, + amdtp_stream_get_max_payload(&dg00x->rx_stream), + fw_parent_device(dg00x->unit)->max_speed); + if (err < 0) + return err; + + /* Keep resources for in-stream. */ + err = amdtp_dot_set_parameters(&dg00x->tx_stream, rate, + snd_dg00x_stream_pcm_channels[i]); + if (err < 0) + return err; + err = fw_iso_resources_allocate(&dg00x->tx_resources, + amdtp_stream_get_max_payload(&dg00x->tx_stream), + fw_parent_device(dg00x->unit)->max_speed); + if (err < 0) + goto error; + + /* Register isochronous channels for both direction. */ + data = cpu_to_be32((dg00x->tx_resources.channel << 16) | + dg00x->rx_resources.channel); + err = snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS, + &data, sizeof(data), 0); + if (err < 0) + goto error; + + return 0; +error: + release_resources(dg00x); + return err; +} + +int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x) +{ + int err; + + /* For out-stream. */ + err = fw_iso_resources_init(&dg00x->rx_resources, dg00x->unit); + if (err < 0) + goto error; + err = amdtp_dot_init(&dg00x->rx_stream, dg00x->unit, AMDTP_OUT_STREAM); + if (err < 0) + goto error; + + /* For in-stream. */ + err = fw_iso_resources_init(&dg00x->tx_resources, dg00x->unit); + if (err < 0) + goto error; + err = amdtp_dot_init(&dg00x->tx_stream, dg00x->unit, AMDTP_IN_STREAM); + if (err < 0) + goto error; + + return 0; +error: + snd_dg00x_stream_destroy_duplex(dg00x); + return err; +} + +/* + * This function should be called before starting streams or after stopping + * streams. + */ +void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x) +{ + amdtp_stream_destroy(&dg00x->rx_stream); + fw_iso_resources_destroy(&dg00x->rx_resources); + + amdtp_stream_destroy(&dg00x->tx_stream); + fw_iso_resources_destroy(&dg00x->tx_resources); +} + +int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate) +{ + unsigned int curr_rate; + int err = 0; + + if (dg00x->substreams_counter == 0) + goto end; + + /* Check current sampling rate. */ + err = snd_dg00x_stream_get_local_rate(dg00x, &curr_rate); + if (err < 0) + goto error; + if (rate == 0) + rate = curr_rate; + if (curr_rate != rate || + amdtp_streaming_error(&dg00x->tx_stream) || + amdtp_streaming_error(&dg00x->rx_stream)) { + finish_session(dg00x); + + amdtp_stream_stop(&dg00x->tx_stream); + amdtp_stream_stop(&dg00x->rx_stream); + release_resources(dg00x); + } + + /* + * No packets are transmitted without receiving packets, reagardless of + * which source of clock is used. + */ + if (!amdtp_stream_running(&dg00x->rx_stream)) { + err = snd_dg00x_stream_set_local_rate(dg00x, rate); + if (err < 0) + goto error; + + err = keep_resources(dg00x, rate); + if (err < 0) + goto error; + + err = begin_session(dg00x); + if (err < 0) + goto error; + + err = amdtp_stream_start(&dg00x->rx_stream, + dg00x->rx_resources.channel, + fw_parent_device(dg00x->unit)->max_speed); + if (err < 0) + goto error; + + if (!amdtp_stream_wait_callback(&dg00x->rx_stream, + CALLBACK_TIMEOUT)) { + err = -ETIMEDOUT; + goto error; + } + } + + /* + * The value of SYT field in transmitted packets is always 0x0000. Thus, + * duplex streams with timestamp synchronization cannot be built. + */ + if (!amdtp_stream_running(&dg00x->tx_stream)) { + err = amdtp_stream_start(&dg00x->tx_stream, + dg00x->tx_resources.channel, + fw_parent_device(dg00x->unit)->max_speed); + if (err < 0) + goto error; + + if (!amdtp_stream_wait_callback(&dg00x->tx_stream, + CALLBACK_TIMEOUT)) { + err = -ETIMEDOUT; + goto error; + } + } +end: + return err; +error: + finish_session(dg00x); + + amdtp_stream_stop(&dg00x->tx_stream); + amdtp_stream_stop(&dg00x->rx_stream); + release_resources(dg00x); + + return err; +} + +void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x) +{ + if (dg00x->substreams_counter > 0) + return; + + amdtp_stream_stop(&dg00x->tx_stream); + amdtp_stream_stop(&dg00x->rx_stream); + finish_session(dg00x); + release_resources(dg00x); + + /* + * Just after finishing the session, the device may lost transmitting + * functionality for a short time. + */ + msleep(50); +} + +void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x) +{ + fw_iso_resources_update(&dg00x->tx_resources); + fw_iso_resources_update(&dg00x->rx_resources); + + amdtp_stream_update(&dg00x->tx_stream); + amdtp_stream_update(&dg00x->rx_stream); +} + +void snd_dg00x_stream_lock_changed(struct snd_dg00x *dg00x) +{ + dg00x->dev_lock_changed = true; + wake_up(&dg00x->hwdep_wait); +} + +int snd_dg00x_stream_lock_try(struct snd_dg00x *dg00x) +{ + int err; + + spin_lock_irq(&dg00x->lock); + + /* user land lock this */ + if (dg00x->dev_lock_count < 0) { + err = -EBUSY; + goto end; + } + + /* this is the first time */ + if (dg00x->dev_lock_count++ == 0) + snd_dg00x_stream_lock_changed(dg00x); + err = 0; +end: + spin_unlock_irq(&dg00x->lock); + return err; +} + +void snd_dg00x_stream_lock_release(struct snd_dg00x *dg00x) +{ + spin_lock_irq(&dg00x->lock); + + if (WARN_ON(dg00x->dev_lock_count <= 0)) + goto end; + if (--dg00x->dev_lock_count == 0) + snd_dg00x_stream_lock_changed(dg00x); +end: + spin_unlock_irq(&dg00x->lock); +} diff --git a/sound/firewire/digi00x/digi00x-transaction.c b/sound/firewire/digi00x/digi00x-transaction.c new file mode 100644 index 000000000000..554324d8c602 --- /dev/null +++ b/sound/firewire/digi00x/digi00x-transaction.c @@ -0,0 +1,137 @@ +/* + * digi00x-transaction.c - a part of driver for Digidesign Digi 002/003 family + * + * Copyright (c) 2014-2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include <sound/asound.h> +#include "digi00x.h" + +static int fill_midi_message(struct snd_rawmidi_substream *substream, u8 *buf) +{ + int bytes; + + buf[0] = 0x80; + bytes = snd_rawmidi_transmit_peek(substream, buf + 1, 2); + if (bytes >= 0) + buf[3] = 0xc0 | bytes; + + return bytes; +} + +static void handle_midi_control(struct snd_dg00x *dg00x, __be32 *buf, + unsigned int length) +{ + struct snd_rawmidi_substream *substream; + unsigned int i; + unsigned int len; + u8 *b; + + substream = ACCESS_ONCE(dg00x->in_control); + if (substream == NULL) + return; + + length /= 4; + + for (i = 0; i < length; i++) { + b = (u8 *)&buf[i]; + len = b[3] & 0xf; + if (len > 0) + snd_rawmidi_receive(dg00x->in_control, b + 1, len); + } +} + +static void handle_unknown_message(struct snd_dg00x *dg00x, + unsigned long long offset, __be32 *buf) +{ + unsigned long flags; + + spin_lock_irqsave(&dg00x->lock, flags); + dg00x->msg = be32_to_cpu(*buf); + spin_unlock_irqrestore(&dg00x->lock, flags); + + wake_up(&dg00x->hwdep_wait); +} + +static void handle_message(struct fw_card *card, struct fw_request *request, + int tcode, int destination, int source, + int generation, unsigned long long offset, + void *data, size_t length, void *callback_data) +{ + struct snd_dg00x *dg00x = callback_data; + __be32 *buf = (__be32 *)data; + + if (offset == dg00x->async_handler.offset) + handle_unknown_message(dg00x, offset, buf); + else if (offset == dg00x->async_handler.offset + 4) + handle_midi_control(dg00x, buf, length); + + fw_send_response(card, request, RCODE_COMPLETE); +} + +int snd_dg00x_transaction_reregister(struct snd_dg00x *dg00x) +{ + struct fw_device *device = fw_parent_device(dg00x->unit); + __be32 data[2]; + int err; + + /* Unknown. 4bytes. */ + data[0] = cpu_to_be32((device->card->node_id << 16) | + (dg00x->async_handler.offset >> 32)); + data[1] = cpu_to_be32(dg00x->async_handler.offset); + err = snd_fw_transaction(dg00x->unit, TCODE_WRITE_BLOCK_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_MESSAGE_ADDR, + &data, sizeof(data), 0); + if (err < 0) + return err; + + /* Asynchronous transactions for MIDI control message. */ + data[0] = cpu_to_be32((device->card->node_id << 16) | + (dg00x->async_handler.offset >> 32)); + data[1] = cpu_to_be32(dg00x->async_handler.offset + 4); + return snd_fw_transaction(dg00x->unit, TCODE_WRITE_BLOCK_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_MIDI_CTL_ADDR, + &data, sizeof(data), 0); +} + +int snd_dg00x_transaction_register(struct snd_dg00x *dg00x) +{ + static const struct fw_address_region resp_register_region = { + .start = 0xffffe0000000ull, + .end = 0xffffe000ffffull, + }; + int err; + + dg00x->async_handler.length = 12; + dg00x->async_handler.address_callback = handle_message; + dg00x->async_handler.callback_data = dg00x; + + err = fw_core_add_address_handler(&dg00x->async_handler, + &resp_register_region); + if (err < 0) + return err; + + err = snd_dg00x_transaction_reregister(dg00x); + if (err < 0) + goto error; + + err = snd_fw_async_midi_port_init(&dg00x->out_control, dg00x->unit, + DG00X_ADDR_BASE + DG00X_OFFSET_MMC, + 4, fill_midi_message); + if (err < 0) + goto error; + + return err; +error: + fw_core_remove_address_handler(&dg00x->async_handler); + dg00x->async_handler.address_callback = NULL; + return err; +} + +void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x) +{ + snd_fw_async_midi_port_destroy(&dg00x->out_control); + fw_core_remove_address_handler(&dg00x->async_handler); +} diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c new file mode 100644 index 000000000000..1f33b7a1fca4 --- /dev/null +++ b/sound/firewire/digi00x/digi00x.c @@ -0,0 +1,170 @@ +/* + * digi00x.c - a part of driver for Digidesign Digi 002/003 family + * + * Copyright (c) 2014-2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "digi00x.h" + +MODULE_DESCRIPTION("Digidesign Digi 002/003 family Driver"); +MODULE_AUTHOR("Takashi Sakamoto <o-takashi@sakamocchi.jp>"); +MODULE_LICENSE("GPL v2"); + +#define VENDOR_DIGIDESIGN 0x00a07e +#define MODEL_DIGI00X 0x000002 + +static int name_card(struct snd_dg00x *dg00x) +{ + struct fw_device *fw_dev = fw_parent_device(dg00x->unit); + char name[32] = {0}; + char *model; + int err; + + err = fw_csr_string(dg00x->unit->directory, CSR_MODEL, name, + sizeof(name)); + if (err < 0) + return err; + + model = skip_spaces(name); + + strcpy(dg00x->card->driver, "Digi00x"); + strcpy(dg00x->card->shortname, model); + strcpy(dg00x->card->mixername, model); + snprintf(dg00x->card->longname, sizeof(dg00x->card->longname), + "Digidesign %s, GUID %08x%08x at %s, S%d", model, + fw_dev->config_rom[3], fw_dev->config_rom[4], + dev_name(&dg00x->unit->device), 100 << fw_dev->max_speed); + + return 0; +} + +static void dg00x_card_free(struct snd_card *card) +{ + struct snd_dg00x *dg00x = card->private_data; + + snd_dg00x_stream_destroy_duplex(dg00x); + snd_dg00x_transaction_unregister(dg00x); + + fw_unit_put(dg00x->unit); + + mutex_destroy(&dg00x->mutex); +} + +static int snd_dg00x_probe(struct fw_unit *unit, + const struct ieee1394_device_id *entry) +{ + struct snd_card *card; + struct snd_dg00x *dg00x; + int err; + + /* create card */ + err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, + sizeof(struct snd_dg00x), &card); + if (err < 0) + return err; + card->private_free = dg00x_card_free; + + /* initialize myself */ + dg00x = card->private_data; + dg00x->card = card; + dg00x->unit = fw_unit_get(unit); + + mutex_init(&dg00x->mutex); + spin_lock_init(&dg00x->lock); + init_waitqueue_head(&dg00x->hwdep_wait); + + err = name_card(dg00x); + if (err < 0) + goto error; + + err = snd_dg00x_stream_init_duplex(dg00x); + if (err < 0) + goto error; + + snd_dg00x_proc_init(dg00x); + + err = snd_dg00x_create_pcm_devices(dg00x); + if (err < 0) + goto error; + + err = snd_dg00x_create_midi_devices(dg00x); + if (err < 0) + goto error; + + err = snd_dg00x_create_hwdep_device(dg00x); + if (err < 0) + goto error; + + err = snd_dg00x_transaction_register(dg00x); + if (err < 0) + goto error; + + err = snd_card_register(card); + if (err < 0) + goto error; + + dev_set_drvdata(&unit->device, dg00x); + + return err; +error: + snd_card_free(card); + return err; +} + +static void snd_dg00x_update(struct fw_unit *unit) +{ + struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device); + + snd_dg00x_transaction_reregister(dg00x); + + mutex_lock(&dg00x->mutex); + snd_dg00x_stream_update_duplex(dg00x); + mutex_unlock(&dg00x->mutex); +} + +static void snd_dg00x_remove(struct fw_unit *unit) +{ + struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device); + + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(dg00x->card); +} + +static const struct ieee1394_device_id snd_dg00x_id_table[] = { + /* Both of 002/003 use the same ID. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = VENDOR_DIGIDESIGN, + .model_id = MODEL_DIGI00X, + }, + {} +}; +MODULE_DEVICE_TABLE(ieee1394, snd_dg00x_id_table); + +static struct fw_driver dg00x_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "snd-firewire-digi00x", + .bus = &fw_bus_type, + }, + .probe = snd_dg00x_probe, + .update = snd_dg00x_update, + .remove = snd_dg00x_remove, + .id_table = snd_dg00x_id_table, +}; + +static int __init snd_dg00x_init(void) +{ + return driver_register(&dg00x_driver.driver); +} + +static void __exit snd_dg00x_exit(void) +{ + driver_unregister(&dg00x_driver.driver); +} + +module_init(snd_dg00x_init); +module_exit(snd_dg00x_exit); diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h new file mode 100644 index 000000000000..907e73993677 --- /dev/null +++ b/sound/firewire/digi00x/digi00x.h @@ -0,0 +1,157 @@ +/* + * digi00x.h - a part of driver for Digidesign Digi 002/003 family + * + * Copyright (c) 2014-2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#ifndef SOUND_DIGI00X_H_INCLUDED +#define SOUND_DIGI00X_H_INCLUDED + +#include <linux/compat.h> +#include <linux/device.h> +#include <linux/firewire.h> +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/delay.h> +#include <linux/slab.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/info.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/firewire.h> +#include <sound/hwdep.h> +#include <sound/rawmidi.h> + +#include "../lib.h" +#include "../iso-resources.h" +#include "../amdtp-stream.h" + +struct snd_dg00x { + struct snd_card *card; + struct fw_unit *unit; + + struct mutex mutex; + spinlock_t lock; + + struct amdtp_stream tx_stream; + struct fw_iso_resources tx_resources; + + struct amdtp_stream rx_stream; + struct fw_iso_resources rx_resources; + + unsigned int substreams_counter; + + /* for uapi */ + int dev_lock_count; + bool dev_lock_changed; + wait_queue_head_t hwdep_wait; + + /* For asynchronous messages. */ + struct fw_address_handler async_handler; + u32 msg; + + /* For asynchronous MIDI controls. */ + struct snd_rawmidi_substream *in_control; + struct snd_fw_async_midi_port out_control; +}; + +#define DG00X_ADDR_BASE 0xffffe0000000ull + +#define DG00X_OFFSET_STREAMING_STATE 0x0000 +#define DG00X_OFFSET_STREAMING_SET 0x0004 +#define DG00X_OFFSET_MIDI_CTL_ADDR 0x0008 +/* For LSB of the address 0x000c */ +/* unknown 0x0010 */ +#define DG00X_OFFSET_MESSAGE_ADDR 0x0014 +/* For LSB of the address 0x0018 */ +/* unknown 0x001c */ +/* unknown 0x0020 */ +/* not used 0x0024--0x00ff */ +#define DG00X_OFFSET_ISOC_CHANNELS 0x0100 +/* unknown 0x0104 */ +/* unknown 0x0108 */ +/* unknown 0x010c */ +#define DG00X_OFFSET_LOCAL_RATE 0x0110 +#define DG00X_OFFSET_EXTERNAL_RATE 0x0114 +#define DG00X_OFFSET_CLOCK_SOURCE 0x0118 +#define DG00X_OFFSET_OPT_IFACE_MODE 0x011c +/* unknown 0x0120 */ +/* Mixer control on/off 0x0124 */ +/* unknown 0x0128 */ +#define DG00X_OFFSET_DETECT_EXTERNAL 0x012c +/* unknown 0x0138 */ +#define DG00X_OFFSET_MMC 0x0400 + +enum snd_dg00x_rate { + SND_DG00X_RATE_44100 = 0, + SND_DG00X_RATE_48000, + SND_DG00X_RATE_88200, + SND_DG00X_RATE_96000, + SND_DG00X_RATE_COUNT, +}; + +enum snd_dg00x_clock { + SND_DG00X_CLOCK_INTERNAL = 0, + SND_DG00X_CLOCK_SPDIF, + SND_DG00X_CLOCK_ADAT, + SND_DG00X_CLOCK_WORD, + SND_DG00X_CLOCK_COUNT, +}; + +enum snd_dg00x_optical_mode { + SND_DG00X_OPT_IFACE_MODE_ADAT = 0, + SND_DG00X_OPT_IFACE_MODE_SPDIF, + SND_DG00X_OPT_IFACE_MODE_COUNT, +}; + +#define DOT_MIDI_IN_PORTS 1 +#define DOT_MIDI_OUT_PORTS 2 + +int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit, + enum amdtp_stream_direction dir); +int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate, + unsigned int pcm_channels); +void amdtp_dot_reset(struct amdtp_stream *s); +int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s, + struct snd_pcm_runtime *runtime); +void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format); +void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port, + struct snd_rawmidi_substream *midi); + +int snd_dg00x_transaction_register(struct snd_dg00x *dg00x); +int snd_dg00x_transaction_reregister(struct snd_dg00x *dg00x); +void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x); + +extern const unsigned int snd_dg00x_stream_rates[SND_DG00X_RATE_COUNT]; +extern const unsigned int snd_dg00x_stream_pcm_channels[SND_DG00X_RATE_COUNT]; +int snd_dg00x_stream_get_external_rate(struct snd_dg00x *dg00x, + unsigned int *rate); +int snd_dg00x_stream_get_local_rate(struct snd_dg00x *dg00x, + unsigned int *rate); +int snd_dg00x_stream_set_local_rate(struct snd_dg00x *dg00x, unsigned int rate); +int snd_dg00x_stream_get_clock(struct snd_dg00x *dg00x, + enum snd_dg00x_clock *clock); +int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x, + bool *detect); +int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x); +int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate); +void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x); +void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x); +void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x); + +void snd_dg00x_stream_lock_changed(struct snd_dg00x *dg00x); +int snd_dg00x_stream_lock_try(struct snd_dg00x *dg00x); +void snd_dg00x_stream_lock_release(struct snd_dg00x *dg00x); + +void snd_dg00x_proc_init(struct snd_dg00x *dg00x); + +int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x); + +int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x); + +int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x); +#endif diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c index 0619597e3a3f..cce19768f43d 100644 --- a/sound/firewire/fcp.c +++ b/sound/firewire/fcp.c @@ -17,7 +17,7 @@ #include <linux/delay.h> #include "fcp.h" #include "lib.h" -#include "amdtp.h" +#include "amdtp-stream.h" #define CTS_AVC 0x00 diff --git a/sound/firewire/fireworks/Makefile b/sound/firewire/fireworks/Makefile index 0c7440826db8..15ef7f75a8ef 100644 --- a/sound/firewire/fireworks/Makefile +++ b/sound/firewire/fireworks/Makefile @@ -1,4 +1,4 @@ snd-fireworks-objs := fireworks_transaction.o fireworks_command.o \ fireworks_stream.o fireworks_proc.o fireworks_midi.o \ fireworks_pcm.o fireworks_hwdep.o fireworks.o -obj-m += snd-fireworks.o +obj-$(CONFIG_SND_FIREWORKS) += snd-fireworks.o diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index c94a432f7cc6..d5b19bc11e59 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -138,12 +138,12 @@ get_hardware_info(struct snd_efw *efw) efw->midi_out_ports = hwinfo->midi_out_ports; efw->midi_in_ports = hwinfo->midi_in_ports; - if (hwinfo->amdtp_tx_pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM || - hwinfo->amdtp_tx_pcm_channels_2x > AMDTP_MAX_CHANNELS_FOR_PCM || - hwinfo->amdtp_tx_pcm_channels_4x > AMDTP_MAX_CHANNELS_FOR_PCM || - hwinfo->amdtp_rx_pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM || - hwinfo->amdtp_rx_pcm_channels_2x > AMDTP_MAX_CHANNELS_FOR_PCM || - hwinfo->amdtp_rx_pcm_channels_4x > AMDTP_MAX_CHANNELS_FOR_PCM) { + if (hwinfo->amdtp_tx_pcm_channels > AM824_MAX_CHANNELS_FOR_PCM || + hwinfo->amdtp_tx_pcm_channels_2x > AM824_MAX_CHANNELS_FOR_PCM || + hwinfo->amdtp_tx_pcm_channels_4x > AM824_MAX_CHANNELS_FOR_PCM || + hwinfo->amdtp_rx_pcm_channels > AM824_MAX_CHANNELS_FOR_PCM || + hwinfo->amdtp_rx_pcm_channels_2x > AM824_MAX_CHANNELS_FOR_PCM || + hwinfo->amdtp_rx_pcm_channels_4x > AM824_MAX_CHANNELS_FOR_PCM) { err = -ENOSYS; goto end; } diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 084d414b228c..c7cb7deafe48 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -29,7 +29,7 @@ #include "../packets-buffer.h" #include "../iso-resources.h" -#include "../amdtp.h" +#include "../amdtp-am824.h" #include "../cmp.h" #include "../lib.h" diff --git a/sound/firewire/fireworks/fireworks_command.c b/sound/firewire/fireworks/fireworks_command.c index 166f80584c2a..94bab0476a65 100644 --- a/sound/firewire/fireworks/fireworks_command.c +++ b/sound/firewire/fireworks/fireworks_command.c @@ -257,7 +257,7 @@ int snd_efw_command_get_phys_meters(struct snd_efw *efw, struct snd_efw_phys_meters *meters, unsigned int len) { - __be32 *buf = (__be32 *)meters; + u32 *buf = (u32 *)meters; unsigned int i; int err; diff --git a/sound/firewire/fireworks/fireworks_midi.c b/sound/firewire/fireworks/fireworks_midi.c index cf9c65260439..fba01bbba456 100644 --- a/sound/firewire/fireworks/fireworks_midi.c +++ b/sound/firewire/fireworks/fireworks_midi.c @@ -73,10 +73,10 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up) spin_lock_irqsave(&efw->lock, flags); if (up) - amdtp_stream_midi_trigger(&efw->tx_stream, + amdtp_am824_midi_trigger(&efw->tx_stream, substrm->number, substrm); else - amdtp_stream_midi_trigger(&efw->tx_stream, + amdtp_am824_midi_trigger(&efw->tx_stream, substrm->number, NULL); spin_unlock_irqrestore(&efw->lock, flags); @@ -90,11 +90,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up) spin_lock_irqsave(&efw->lock, flags); if (up) - amdtp_stream_midi_trigger(&efw->rx_stream, - substrm->number, substrm); + amdtp_am824_midi_trigger(&efw->rx_stream, + substrm->number, substrm); else - amdtp_stream_midi_trigger(&efw->rx_stream, - substrm->number, NULL); + amdtp_am824_midi_trigger(&efw->rx_stream, + substrm->number, NULL); spin_unlock_irqrestore(&efw->lock, flags); } diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index c30b2ffa8dfb..d27135bac513 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -159,11 +159,11 @@ pcm_init_hw_params(struct snd_efw *efw, SNDRV_PCM_INFO_MMAP_VALID; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS; + runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS; s = &efw->tx_stream; pcm_channels = efw->pcm_capture_channels; } else { - runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS; + runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS; s = &efw->rx_stream; pcm_channels = efw->pcm_playback_channels; } @@ -187,7 +187,7 @@ pcm_init_hw_params(struct snd_efw *efw, if (err < 0) goto end; - err = amdtp_stream_add_pcm_hw_constraints(s, runtime); + err = amdtp_am824_add_pcm_hw_constraints(s, runtime); end: return err; } @@ -253,7 +253,8 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) atomic_inc(&efw->capture_substreams); - amdtp_stream_set_pcm_format(&efw->tx_stream, params_format(hw_params)); + + amdtp_am824_set_pcm_format(&efw->tx_stream, params_format(hw_params)); return 0; } @@ -270,7 +271,8 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) atomic_inc(&efw->playback_substreams); - amdtp_stream_set_pcm_format(&efw->rx_stream, params_format(hw_params)); + + amdtp_am824_set_pcm_format(&efw->rx_stream, params_format(hw_params)); return 0; } diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index 7e353f1f7bff..759f6e3ed44a 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -31,7 +31,7 @@ init_stream(struct snd_efw *efw, struct amdtp_stream *stream) if (err < 0) goto end; - err = amdtp_stream_init(stream, efw->unit, s_dir, CIP_BLOCKING); + err = amdtp_am824_init(stream, efw->unit, s_dir, CIP_BLOCKING); if (err < 0) { amdtp_stream_destroy(stream); cmp_connection_destroy(conn); @@ -73,8 +73,10 @@ start_stream(struct snd_efw *efw, struct amdtp_stream *stream, midi_ports = efw->midi_in_ports; } - amdtp_stream_set_parameters(stream, sampling_rate, - pcm_channels, midi_ports); + err = amdtp_am824_set_parameters(stream, sampling_rate, + pcm_channels, midi_ports, false); + if (err < 0) + goto end; /* establish connection via CMP */ err = cmp_connection_establish(conn, diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c index 7409edba9f06..f80aafa44c89 100644 --- a/sound/firewire/lib.c +++ b/sound/firewire/lib.c @@ -9,6 +9,7 @@ #include <linux/device.h> #include <linux/firewire.h> #include <linux/module.h> +#include <linux/slab.h> #include "lib.h" #define ERROR_RETRY_DELAY_MS 20 @@ -66,6 +67,147 @@ int snd_fw_transaction(struct fw_unit *unit, int tcode, } EXPORT_SYMBOL(snd_fw_transaction); +static void async_midi_port_callback(struct fw_card *card, int rcode, + void *data, size_t length, + void *callback_data) +{ + struct snd_fw_async_midi_port *port = callback_data; + struct snd_rawmidi_substream *substream = ACCESS_ONCE(port->substream); + + /* This port is closed. */ + if (substream == NULL) + return; + + if (rcode == RCODE_COMPLETE) + snd_rawmidi_transmit_ack(substream, port->consume_bytes); + else if (!rcode_is_permanent_error(rcode)) + /* To start next transaction immediately for recovery. */ + port->next_ktime = ktime_set(0, 0); + else + /* Don't continue processing. */ + port->error = true; + + port->idling = true; + + if (!snd_rawmidi_transmit_empty(substream)) + schedule_work(&port->work); +} + +static void midi_port_work(struct work_struct *work) +{ + struct snd_fw_async_midi_port *port = + container_of(work, struct snd_fw_async_midi_port, work); + struct snd_rawmidi_substream *substream = ACCESS_ONCE(port->substream); + int generation; + int type; + + /* Under transacting or error state. */ + if (!port->idling || port->error) + return; + + /* Nothing to do. */ + if (substream == NULL || snd_rawmidi_transmit_empty(substream)) + return; + + /* Do it in next chance. */ + if (ktime_after(port->next_ktime, ktime_get())) { + schedule_work(&port->work); + return; + } + + /* + * Fill the buffer. The callee must use snd_rawmidi_transmit_peek(). + * Later, snd_rawmidi_transmit_ack() is called. + */ + memset(port->buf, 0, port->len); + port->consume_bytes = port->fill(substream, port->buf); + if (port->consume_bytes <= 0) { + /* Do it in next chance, immediately. */ + if (port->consume_bytes == 0) { + port->next_ktime = ktime_set(0, 0); + schedule_work(&port->work); + } else { + /* Fatal error. */ + port->error = true; + } + return; + } + + /* Calculate type of transaction. */ + if (port->len == 4) + type = TCODE_WRITE_QUADLET_REQUEST; + else + type = TCODE_WRITE_BLOCK_REQUEST; + + /* Set interval to next transaction. */ + port->next_ktime = ktime_add_ns(ktime_get(), + port->consume_bytes * 8 * NSEC_PER_SEC / 31250); + + /* Start this transaction. */ + port->idling = false; + + /* + * In Linux FireWire core, when generation is updated with memory + * barrier, node id has already been updated. In this module, After + * this smp_rmb(), load/store instructions to memory are completed. + * Thus, both of generation and node id are available with recent + * values. This is a light-serialization solution to handle bus reset + * events on IEEE 1394 bus. + */ + generation = port->parent->generation; + smp_rmb(); + + fw_send_request(port->parent->card, &port->transaction, type, + port->parent->node_id, generation, + port->parent->max_speed, port->addr, + port->buf, port->len, async_midi_port_callback, + port); +} + +/** + * snd_fw_async_midi_port_init - initialize asynchronous MIDI port structure + * @port: the asynchronous MIDI port to initialize + * @unit: the target of the asynchronous transaction + * @addr: the address to which transactions are transferred + * @len: the length of transaction + * @fill: the callback function to fill given buffer, and returns the + * number of consumed bytes for MIDI message. + * + */ +int snd_fw_async_midi_port_init(struct snd_fw_async_midi_port *port, + struct fw_unit *unit, u64 addr, unsigned int len, + snd_fw_async_midi_port_fill fill) +{ + port->len = DIV_ROUND_UP(len, 4) * 4; + port->buf = kzalloc(port->len, GFP_KERNEL); + if (port->buf == NULL) + return -ENOMEM; + + port->parent = fw_parent_device(unit); + port->addr = addr; + port->fill = fill; + port->idling = true; + port->next_ktime = ktime_set(0, 0); + port->error = false; + + INIT_WORK(&port->work, midi_port_work); + + return 0; +} +EXPORT_SYMBOL(snd_fw_async_midi_port_init); + +/** + * snd_fw_async_midi_port_destroy - free asynchronous MIDI port structure + * @port: the asynchronous MIDI port structure + */ +void snd_fw_async_midi_port_destroy(struct snd_fw_async_midi_port *port) +{ + snd_fw_async_midi_port_finish(port); + cancel_work_sync(&port->work); + kfree(port->buf); +} +EXPORT_SYMBOL(snd_fw_async_midi_port_destroy); + MODULE_DESCRIPTION("FireWire audio helper functions"); MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); MODULE_LICENSE("GPL v2"); diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h index 02cfabc9c3c4..f3f6f84c48d6 100644 --- a/sound/firewire/lib.h +++ b/sound/firewire/lib.h @@ -3,6 +3,8 @@ #include <linux/firewire-constants.h> #include <linux/types.h> +#include <linux/sched.h> +#include <sound/rawmidi.h> struct fw_unit; @@ -20,4 +22,58 @@ static inline bool rcode_is_permanent_error(int rcode) return rcode == RCODE_TYPE_ERROR || rcode == RCODE_ADDRESS_ERROR; } +struct snd_fw_async_midi_port; +typedef int (*snd_fw_async_midi_port_fill)( + struct snd_rawmidi_substream *substream, + u8 *buf); + +struct snd_fw_async_midi_port { + struct fw_device *parent; + struct work_struct work; + bool idling; + ktime_t next_ktime; + bool error; + + u64 addr; + struct fw_transaction transaction; + + u8 *buf; + unsigned int len; + + struct snd_rawmidi_substream *substream; + snd_fw_async_midi_port_fill fill; + unsigned int consume_bytes; +}; + +int snd_fw_async_midi_port_init(struct snd_fw_async_midi_port *port, + struct fw_unit *unit, u64 addr, unsigned int len, + snd_fw_async_midi_port_fill fill); +void snd_fw_async_midi_port_destroy(struct snd_fw_async_midi_port *port); + +/** + * snd_fw_async_midi_port_run - run transactions for the async MIDI port + * @port: the asynchronous MIDI port + * @substream: the MIDI substream + */ +static inline void +snd_fw_async_midi_port_run(struct snd_fw_async_midi_port *port, + struct snd_rawmidi_substream *substream) +{ + if (!port->error) { + port->substream = substream; + schedule_work(&port->work); + } +} + +/** + * snd_fw_async_midi_port_finish - finish the asynchronous MIDI port + * @port: the asynchronous MIDI port + */ +static inline void +snd_fw_async_midi_port_finish(struct snd_fw_async_midi_port *port) +{ + port->substream = NULL; + port->error = false; +} + #endif diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile index a926850864f6..06ff50f4e6c0 100644 --- a/sound/firewire/oxfw/Makefile +++ b/sound/firewire/oxfw/Makefile @@ -1,3 +1,3 @@ snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-control.o oxfw-pcm.o \ oxfw-proc.o oxfw-midi.o oxfw-hwdep.o oxfw.o -obj-m += snd-oxfw.o +obj-$(CONFIG_SND_OXFW) += snd-oxfw.o diff --git a/sound/firewire/oxfw/oxfw-midi.c b/sound/firewire/oxfw/oxfw-midi.c index 540a30338516..8665e1043d41 100644 --- a/sound/firewire/oxfw/oxfw-midi.c +++ b/sound/firewire/oxfw/oxfw-midi.c @@ -90,11 +90,11 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up) spin_lock_irqsave(&oxfw->lock, flags); if (up) - amdtp_stream_midi_trigger(&oxfw->tx_stream, - substrm->number, substrm); + amdtp_am824_midi_trigger(&oxfw->tx_stream, + substrm->number, substrm); else - amdtp_stream_midi_trigger(&oxfw->tx_stream, - substrm->number, NULL); + amdtp_am824_midi_trigger(&oxfw->tx_stream, + substrm->number, NULL); spin_unlock_irqrestore(&oxfw->lock, flags); } @@ -107,11 +107,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up) spin_lock_irqsave(&oxfw->lock, flags); if (up) - amdtp_stream_midi_trigger(&oxfw->rx_stream, - substrm->number, substrm); + amdtp_am824_midi_trigger(&oxfw->rx_stream, + substrm->number, substrm); else - amdtp_stream_midi_trigger(&oxfw->rx_stream, - substrm->number, NULL); + amdtp_am824_midi_trigger(&oxfw->rx_stream, + substrm->number, NULL); spin_unlock_irqrestore(&oxfw->lock, flags); } @@ -142,29 +142,11 @@ static void set_midi_substream_names(struct snd_oxfw *oxfw, int snd_oxfw_create_midi(struct snd_oxfw *oxfw) { - struct snd_oxfw_stream_formation formation; struct snd_rawmidi *rmidi; struct snd_rawmidi_str *str; - u8 *format; - int i, err; - - /* If its stream has MIDI conformant data channel, add one MIDI port */ - for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { - format = oxfw->tx_stream_formats[i]; - if (format != NULL) { - err = snd_oxfw_stream_parse_format(format, &formation); - if (err >= 0 && formation.midi > 0) - oxfw->midi_input_ports = 1; - } - - format = oxfw->rx_stream_formats[i]; - if (format != NULL) { - err = snd_oxfw_stream_parse_format(format, &formation); - if (err >= 0 && formation.midi > 0) - oxfw->midi_output_ports = 1; - } - } - if ((oxfw->midi_input_ports == 0) && (oxfw->midi_output_ports == 0)) + int err; + + if (oxfw->midi_input_ports == 0 && oxfw->midi_output_ports == 0) return 0; /* create midi ports */ diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 9c73930d0278..8d233417695d 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -134,11 +134,11 @@ static int init_hw_params(struct snd_oxfw *oxfw, SNDRV_PCM_INFO_MMAP_VALID; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS; + runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS; stream = &oxfw->tx_stream; formats = oxfw->tx_stream_formats; } else { - runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS; + runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS; stream = &oxfw->rx_stream; formats = oxfw->rx_stream_formats; } @@ -158,7 +158,7 @@ static int init_hw_params(struct snd_oxfw *oxfw, if (err < 0) goto end; - err = amdtp_stream_add_pcm_hw_constraints(stream, runtime); + err = amdtp_am824_add_pcm_hw_constraints(stream, runtime); end: return err; } @@ -244,7 +244,7 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&oxfw->mutex); } - amdtp_stream_set_pcm_format(&oxfw->tx_stream, params_format(hw_params)); + amdtp_am824_set_pcm_format(&oxfw->tx_stream, params_format(hw_params)); return 0; } @@ -265,7 +265,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&oxfw->mutex); } - amdtp_stream_set_pcm_format(&oxfw->rx_stream, params_format(hw_params)); + amdtp_am824_set_pcm_format(&oxfw->rx_stream, params_format(hw_params)); return 0; } diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 77ad5b98e806..7cb5743c073b 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -148,14 +148,17 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream, } pcm_channels = formation.pcm; - midi_ports = DIV_ROUND_UP(formation.midi, 8); + midi_ports = formation.midi * 8; /* The stream should have one pcm channels at least */ if (pcm_channels == 0) { err = -EINVAL; goto end; } - amdtp_stream_set_parameters(stream, rate, pcm_channels, midi_ports); + err = amdtp_am824_set_parameters(stream, rate, pcm_channels, midi_ports, + false); + if (err < 0) + goto end; err = cmp_connection_establish(conn, amdtp_stream_get_max_payload(stream)); @@ -225,7 +228,7 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw, if (err < 0) goto end; - err = amdtp_stream_init(stream, oxfw->unit, s_dir, CIP_NONBLOCKING); + err = amdtp_am824_init(stream, oxfw->unit, s_dir, CIP_NONBLOCKING); if (err < 0) { amdtp_stream_destroy(stream); cmp_connection_destroy(conn); @@ -238,9 +241,12 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw, * packets. As a result, next isochronous packet includes more data * blocks than IEC 61883-6 defines. */ - if (stream == &oxfw->tx_stream) + if (stream == &oxfw->tx_stream) { oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK | CIP_JUMBO_PAYLOAD; + if (oxfw->wrong_dbs) + oxfw->tx_stream.flags |= CIP_WRONG_DBS; + } end: return err; } @@ -480,8 +486,8 @@ int snd_oxfw_stream_parse_format(u8 *format, } } - if (formation->pcm > AMDTP_MAX_CHANNELS_FOR_PCM || - formation->midi > AMDTP_MAX_CHANNELS_FOR_MIDI) + if (formation->pcm > AM824_MAX_CHANNELS_FOR_PCM || + formation->midi > AM824_MAX_CHANNELS_FOR_MIDI) return -ENOSYS; return 0; @@ -623,6 +629,9 @@ end: int snd_oxfw_stream_discover(struct snd_oxfw *oxfw) { u8 plugs[AVC_PLUG_INFO_BUF_BYTES]; + struct snd_oxfw_stream_formation formation; + u8 *format; + unsigned int i; int err; /* the number of plugs for isoc in/out, ext in/out */ @@ -642,12 +651,42 @@ int snd_oxfw_stream_discover(struct snd_oxfw *oxfw) err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_OUT, 0); if (err < 0) goto end; + + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { + format = oxfw->tx_stream_formats[i]; + if (format == NULL) + continue; + err = snd_oxfw_stream_parse_format(format, &formation); + if (err < 0) + continue; + + /* Add one MIDI port. */ + if (formation.midi > 0) + oxfw->midi_input_ports = 1; + } + oxfw->has_output = true; } /* use iPCR[0] if exists */ - if (plugs[0] > 0) + if (plugs[0] > 0) { err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_IN, 0); + if (err < 0) + goto end; + + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { + format = oxfw->rx_stream_formats[i]; + if (format == NULL) + continue; + err = snd_oxfw_stream_parse_format(format, &formation); + if (err < 0) + continue; + + /* Add one MIDI port. */ + if (formation.midi > 0) + oxfw->midi_output_ports = 1; + } + } end: return err; } diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 8c6ce019f437..588b93f20c2e 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -18,6 +18,9 @@ #define VENDOR_GRIFFIN 0x001292 #define VENDOR_BEHRINGER 0x001564 #define VENDOR_LACIE 0x00d04b +#define VENDOR_TASCAM 0x00022e + +#define MODEL_SATELLITE 0x00200f #define SPECIFIER_1394TA 0x00a02d #define VERSION_AVC 0x010001 @@ -129,6 +132,40 @@ static void oxfw_card_free(struct snd_card *card) mutex_destroy(&oxfw->mutex); } +static void detect_quirks(struct snd_oxfw *oxfw) +{ + struct fw_device *fw_dev = fw_parent_device(oxfw->unit); + struct fw_csr_iterator it; + int key, val; + int vendor, model; + + /* Seek from Root Directory of Config ROM. */ + vendor = model = 0; + fw_csr_iterator_init(&it, fw_dev->config_rom + 5); + while (fw_csr_iterator_next(&it, &key, &val)) { + if (key == CSR_VENDOR) + vendor = val; + else if (key == CSR_MODEL) + model = val; + } + + /* + * Mackie Onyx Satellite with base station has a quirk to report a wrong + * value in 'dbs' field of CIP header against its format information. + */ + if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE) + oxfw->wrong_dbs = true; + + /* + * TASCAM FireOne has physical control and requires a pair of additional + * MIDI ports. + */ + if (vendor == VENDOR_TASCAM) { + oxfw->midi_input_ports++; + oxfw->midi_output_ports++; + } +} + static int oxfw_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) { @@ -157,6 +194,8 @@ static int oxfw_probe(struct fw_unit *unit, if (err < 0) goto error; + detect_quirks(oxfw); + err = name_card(oxfw); if (err < 0) goto error; @@ -294,6 +333,13 @@ static const struct ieee1394_device_id oxfw_id_table[] = { .specifier_id = SPECIFIER_1394TA, .version = VERSION_AVC, }, + /* TASCAM, FireOne */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = VENDOR_TASCAM, + .model_id = 0x800007, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table); diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index cace5ad4fe76..8392c424ad1d 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -28,7 +28,7 @@ #include "../fcp.h" #include "../packets-buffer.h" #include "../iso-resources.h" -#include "../amdtp.h" +#include "../amdtp-am824.h" #include "../cmp.h" struct device_info { @@ -49,6 +49,7 @@ struct snd_oxfw { struct mutex mutex; spinlock_t lock; + bool wrong_dbs; bool has_output; u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; u8 *rx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; diff --git a/sound/firewire/tascam/Makefile b/sound/firewire/tascam/Makefile new file mode 100644 index 000000000000..0fc955d5bd15 --- /dev/null +++ b/sound/firewire/tascam/Makefile @@ -0,0 +1,4 @@ +snd-firewire-tascam-objs := tascam-proc.o amdtp-tascam.o tascam-stream.o \ + tascam-pcm.o tascam-hwdep.o tascam-transaction.o \ + tascam-midi.o tascam.o +obj-$(CONFIG_SND_FIREWIRE_TASCAM) += snd-firewire-tascam.o diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c new file mode 100644 index 000000000000..9dd0fccd5ccc --- /dev/null +++ b/sound/firewire/tascam/amdtp-tascam.c @@ -0,0 +1,243 @@ +/* + * amdtp-tascam.c - a part of driver for TASCAM FireWire series + * + * Copyright (c) 2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include <sound/pcm.h> +#include "tascam.h" + +#define AMDTP_FMT_TSCM_TX 0x1e +#define AMDTP_FMT_TSCM_RX 0x3e + +struct amdtp_tscm { + unsigned int pcm_channels; + + void (*transfer_samples)(struct amdtp_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames); +}; + +int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate) +{ + struct amdtp_tscm *p = s->protocol; + unsigned int data_channels; + + if (amdtp_stream_running(s)) + return -EBUSY; + + data_channels = p->pcm_channels; + + /* Packets in in-stream have extra 2 data channels. */ + if (s->direction == AMDTP_IN_STREAM) + data_channels += 2; + + return amdtp_stream_set_parameters(s, rate, data_channels); +} + +static void write_pcm_s32(struct amdtp_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames) +{ + struct amdtp_tscm *p = s->protocol; + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int channels, remaining_frames, i, c; + const u32 *src; + + channels = p->pcm_channels; + src = (void *)runtime->dma_area + + frames_to_bytes(runtime, s->pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; + + for (i = 0; i < frames; ++i) { + for (c = 0; c < channels; ++c) { + buffer[c] = cpu_to_be32(*src); + src++; + } + buffer += s->data_block_quadlets; + if (--remaining_frames == 0) + src = (void *)runtime->dma_area; + } +} + +static void write_pcm_s16(struct amdtp_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames) +{ + struct amdtp_tscm *p = s->protocol; + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int channels, remaining_frames, i, c; + const u16 *src; + + channels = p->pcm_channels; + src = (void *)runtime->dma_area + + frames_to_bytes(runtime, s->pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; + + for (i = 0; i < frames; ++i) { + for (c = 0; c < channels; ++c) { + buffer[c] = cpu_to_be32(*src << 16); + src++; + } + buffer += s->data_block_quadlets; + if (--remaining_frames == 0) + src = (void *)runtime->dma_area; + } +} + +static void read_pcm_s32(struct amdtp_stream *s, + struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames) +{ + struct amdtp_tscm *p = s->protocol; + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int channels, remaining_frames, i, c; + u32 *dst; + + channels = p->pcm_channels; + dst = (void *)runtime->dma_area + + frames_to_bytes(runtime, s->pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; + + /* The first data channel is for event counter. */ + buffer += 1; + + for (i = 0; i < frames; ++i) { + for (c = 0; c < channels; ++c) { + *dst = be32_to_cpu(buffer[c]); + dst++; + } + buffer += s->data_block_quadlets; + if (--remaining_frames == 0) + dst = (void *)runtime->dma_area; + } +} + +static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer, + unsigned int data_blocks) +{ + struct amdtp_tscm *p = s->protocol; + unsigned int channels, i, c; + + channels = p->pcm_channels; + + for (i = 0; i < data_blocks; ++i) { + for (c = 0; c < channels; ++c) + buffer[c] = 0x00000000; + buffer += s->data_block_quadlets; + } +} + +int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s, + struct snd_pcm_runtime *runtime) +{ + int err; + + /* + * Our implementation allows this protocol to deliver 24 bit sample in + * 32bit data channel. + */ + err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + if (err < 0) + return err; + + return amdtp_stream_add_pcm_hw_constraints(s, runtime); +} + +void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format) +{ + struct amdtp_tscm *p = s->protocol; + + if (WARN_ON(amdtp_stream_pcm_running(s))) + return; + + switch (format) { + default: + WARN_ON(1); + /* fall through */ + case SNDRV_PCM_FORMAT_S16: + if (s->direction == AMDTP_OUT_STREAM) { + p->transfer_samples = write_pcm_s16; + break; + } + WARN_ON(1); + /* fall through */ + case SNDRV_PCM_FORMAT_S32: + if (s->direction == AMDTP_OUT_STREAM) + p->transfer_samples = write_pcm_s32; + else + p->transfer_samples = read_pcm_s32; + break; + } +} + +static unsigned int process_tx_data_blocks(struct amdtp_stream *s, + __be32 *buffer, + unsigned int data_blocks, + unsigned int *syt) +{ + struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol; + struct snd_pcm_substream *pcm; + + pcm = ACCESS_ONCE(s->pcm); + if (data_blocks > 0 && pcm) + p->transfer_samples(s, pcm, buffer, data_blocks); + + /* A place holder for control messages. */ + + return data_blocks; +} + +static unsigned int process_rx_data_blocks(struct amdtp_stream *s, + __be32 *buffer, + unsigned int data_blocks, + unsigned int *syt) +{ + struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol; + struct snd_pcm_substream *pcm; + + /* This field is not used. */ + *syt = 0x0000; + + pcm = ACCESS_ONCE(s->pcm); + if (pcm) + p->transfer_samples(s, pcm, buffer, data_blocks); + else + write_pcm_silence(s, buffer, data_blocks); + + return data_blocks; +} + +int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit, + enum amdtp_stream_direction dir, unsigned int pcm_channels) +{ + amdtp_stream_process_data_blocks_t process_data_blocks; + struct amdtp_tscm *p; + unsigned int fmt; + int err; + + if (dir == AMDTP_IN_STREAM) { + fmt = AMDTP_FMT_TSCM_TX; + process_data_blocks = process_tx_data_blocks; + } else { + fmt = AMDTP_FMT_TSCM_RX; + process_data_blocks = process_rx_data_blocks; + } + + err = amdtp_stream_init(s, unit, dir, + CIP_NONBLOCKING | CIP_SKIP_DBC_ZERO_CHECK, fmt, + process_data_blocks, sizeof(struct amdtp_tscm)); + if (err < 0) + return 0; + + /* Use fixed value for FDF field. */ + s->fdf = 0x00; + + /* This protocol uses fixed number of data channels for PCM samples. */ + p = s->protocol; + p->pcm_channels = pcm_channels; + + return 0; +} diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c new file mode 100644 index 000000000000..131267c3a042 --- /dev/null +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -0,0 +1,201 @@ +/* + * tascam-hwdep.c - a part of driver for TASCAM FireWire series + * + * Copyright (c) 2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +/* + * This codes give three functionality. + * + * 1.get firewire node information + * 2.get notification about starting/stopping stream + * 3.lock/unlock stream + */ + +#include "tascam.h" + +static long hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, + long count) +{ + union snd_firewire_event event; + + memset(&event, 0, sizeof(event)); + + event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; + event.lock_status.status = (tscm->dev_lock_count > 0); + tscm->dev_lock_changed = false; + + count = min_t(long, count, sizeof(event.lock_status)); + + if (copy_to_user(buf, &event, count)) + return -EFAULT; + + return count; +} + +static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, + loff_t *offset) +{ + struct snd_tscm *tscm = hwdep->private_data; + DEFINE_WAIT(wait); + union snd_firewire_event event; + + spin_lock_irq(&tscm->lock); + + while (!tscm->dev_lock_changed) { + prepare_to_wait(&tscm->hwdep_wait, &wait, TASK_INTERRUPTIBLE); + spin_unlock_irq(&tscm->lock); + schedule(); + finish_wait(&tscm->hwdep_wait, &wait); + if (signal_pending(current)) + return -ERESTARTSYS; + spin_lock_irq(&tscm->lock); + } + + memset(&event, 0, sizeof(event)); + count = hwdep_read_locked(tscm, buf, count); + spin_unlock_irq(&tscm->lock); + + return count; +} + +static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file, + poll_table *wait) +{ + struct snd_tscm *tscm = hwdep->private_data; + unsigned int events; + + poll_wait(file, &tscm->hwdep_wait, wait); + + spin_lock_irq(&tscm->lock); + if (tscm->dev_lock_changed) + events = POLLIN | POLLRDNORM; + else + events = 0; + spin_unlock_irq(&tscm->lock); + + return events; +} + +static int hwdep_get_info(struct snd_tscm *tscm, void __user *arg) +{ + struct fw_device *dev = fw_parent_device(tscm->unit); + struct snd_firewire_get_info info; + + memset(&info, 0, sizeof(info)); + info.type = SNDRV_FIREWIRE_TYPE_TASCAM; + info.card = dev->card->index; + *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); + *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); + strlcpy(info.device_name, dev_name(&dev->device), + sizeof(info.device_name)); + + if (copy_to_user(arg, &info, sizeof(info))) + return -EFAULT; + + return 0; +} + +static int hwdep_lock(struct snd_tscm *tscm) +{ + int err; + + spin_lock_irq(&tscm->lock); + + if (tscm->dev_lock_count == 0) { + tscm->dev_lock_count = -1; + err = 0; + } else { + err = -EBUSY; + } + + spin_unlock_irq(&tscm->lock); + + return err; +} + +static int hwdep_unlock(struct snd_tscm *tscm) +{ + int err; + + spin_lock_irq(&tscm->lock); + + if (tscm->dev_lock_count == -1) { + tscm->dev_lock_count = 0; + err = 0; + } else { + err = -EBADFD; + } + + spin_unlock_irq(&tscm->lock); + + return err; +} + +static int hwdep_release(struct snd_hwdep *hwdep, struct file *file) +{ + struct snd_tscm *tscm = hwdep->private_data; + + spin_lock_irq(&tscm->lock); + if (tscm->dev_lock_count == -1) + tscm->dev_lock_count = 0; + spin_unlock_irq(&tscm->lock); + + return 0; +} + +static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + struct snd_tscm *tscm = hwdep->private_data; + + switch (cmd) { + case SNDRV_FIREWIRE_IOCTL_GET_INFO: + return hwdep_get_info(tscm, (void __user *)arg); + case SNDRV_FIREWIRE_IOCTL_LOCK: + return hwdep_lock(tscm); + case SNDRV_FIREWIRE_IOCTL_UNLOCK: + return hwdep_unlock(tscm); + default: + return -ENOIOCTLCMD; + } +} + +#ifdef CONFIG_COMPAT +static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + return hwdep_ioctl(hwdep, file, cmd, + (unsigned long)compat_ptr(arg)); +} +#else +#define hwdep_compat_ioctl NULL +#endif + +static const struct snd_hwdep_ops hwdep_ops = { + .read = hwdep_read, + .release = hwdep_release, + .poll = hwdep_poll, + .ioctl = hwdep_ioctl, + .ioctl_compat = hwdep_compat_ioctl, +}; + +int snd_tscm_create_hwdep_device(struct snd_tscm *tscm) +{ + struct snd_hwdep *hwdep; + int err; + + err = snd_hwdep_new(tscm->card, "Tascam", 0, &hwdep); + if (err < 0) + return err; + + strcpy(hwdep->name, "Tascam"); + hwdep->iface = SNDRV_HWDEP_IFACE_FW_TASCAM; + hwdep->ops = hwdep_ops; + hwdep->private_data = tscm; + hwdep->exclusive = true; + + return err; +} diff --git a/sound/firewire/tascam/tascam-midi.c b/sound/firewire/tascam/tascam-midi.c new file mode 100644 index 000000000000..41f842079d9d --- /dev/null +++ b/sound/firewire/tascam/tascam-midi.c @@ -0,0 +1,135 @@ +/* + * tascam-midi.c - a part of driver for TASCAM FireWire series + * + * Copyright (c) 2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "tascam.h" + +static int midi_capture_open(struct snd_rawmidi_substream *substream) +{ + /* Do nothing. */ + return 0; +} + +static int midi_playback_open(struct snd_rawmidi_substream *substream) +{ + struct snd_tscm *tscm = substream->rmidi->private_data; + + /* Initialize internal status. */ + tscm->running_status[substream->number] = 0; + tscm->on_sysex[substream->number] = 0; + return 0; +} + +static int midi_capture_close(struct snd_rawmidi_substream *substream) +{ + /* Do nothing. */ + return 0; +} + +static int midi_playback_close(struct snd_rawmidi_substream *substream) +{ + struct snd_tscm *tscm = substream->rmidi->private_data; + + snd_fw_async_midi_port_finish(&tscm->out_ports[substream->number]); + + return 0; +} + +static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up) +{ + struct snd_tscm *tscm = substrm->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&tscm->lock, flags); + + if (up) + tscm->tx_midi_substreams[substrm->number] = substrm; + else + tscm->tx_midi_substreams[substrm->number] = NULL; + + spin_unlock_irqrestore(&tscm->lock, flags); +} + +static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up) +{ + struct snd_tscm *tscm = substrm->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&tscm->lock, flags); + + if (up) + snd_fw_async_midi_port_run(&tscm->out_ports[substrm->number], + substrm); + + spin_unlock_irqrestore(&tscm->lock, flags); +} + +static struct snd_rawmidi_ops midi_capture_ops = { + .open = midi_capture_open, + .close = midi_capture_close, + .trigger = midi_capture_trigger, +}; + +static struct snd_rawmidi_ops midi_playback_ops = { + .open = midi_playback_open, + .close = midi_playback_close, + .trigger = midi_playback_trigger, +}; + +int snd_tscm_create_midi_devices(struct snd_tscm *tscm) +{ + struct snd_rawmidi *rmidi; + struct snd_rawmidi_str *stream; + struct snd_rawmidi_substream *subs; + int err; + + err = snd_rawmidi_new(tscm->card, tscm->card->driver, 0, + tscm->spec->midi_playback_ports, + tscm->spec->midi_capture_ports, + &rmidi); + if (err < 0) + return err; + + snprintf(rmidi->name, sizeof(rmidi->name), + "%s MIDI", tscm->card->shortname); + rmidi->private_data = tscm; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &midi_capture_ops); + stream = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT]; + + /* Set port names for MIDI input. */ + list_for_each_entry(subs, &stream->substreams, list) { + /* TODO: support virtual MIDI ports. */ + if (subs->number < tscm->spec->midi_capture_ports) { + /* Hardware MIDI ports. */ + snprintf(subs->name, sizeof(subs->name), + "%s MIDI %d", + tscm->card->shortname, subs->number + 1); + } + } + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &midi_playback_ops); + stream = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT]; + + /* Set port names for MIDI ourput. */ + list_for_each_entry(subs, &stream->substreams, list) { + if (subs->number < tscm->spec->midi_playback_ports) { + /* Hardware MIDI ports only. */ + snprintf(subs->name, sizeof(subs->name), + "%s MIDI %d", + tscm->card->shortname, subs->number + 1); + } + } + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX; + + return 0; +} diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c new file mode 100644 index 000000000000..380d3db969a5 --- /dev/null +++ b/sound/firewire/tascam/tascam-pcm.c @@ -0,0 +1,312 @@ +/* + * tascam-pcm.c - a part of driver for TASCAM FireWire series + * + * Copyright (c) 2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "tascam.h" + +static void set_buffer_params(struct snd_pcm_hardware *hw) +{ + hw->period_bytes_min = 4 * hw->channels_min; + hw->period_bytes_max = hw->period_bytes_min * 2048; + hw->buffer_bytes_max = hw->period_bytes_max * 2; + + hw->periods_min = 2; + hw->periods_max = UINT_MAX; +} + +static int pcm_init_hw_params(struct snd_tscm *tscm, + struct snd_pcm_substream *substream) +{ + static const struct snd_pcm_hardware hardware = { + .info = SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_JOINT_DUPLEX | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .channels_min = 10, + .channels_max = 18, + }; + struct snd_pcm_runtime *runtime = substream->runtime; + struct amdtp_stream *stream; + unsigned int pcm_channels; + + runtime->hw = hardware; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; + stream = &tscm->tx_stream; + pcm_channels = tscm->spec->pcm_capture_analog_channels; + } else { + runtime->hw.formats = + SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32; + stream = &tscm->rx_stream; + pcm_channels = tscm->spec->pcm_playback_analog_channels; + } + + if (tscm->spec->has_adat) + pcm_channels += 8; + if (tscm->spec->has_spdif) + pcm_channels += 2; + runtime->hw.channels_min = runtime->hw.channels_max = pcm_channels; + + set_buffer_params(&runtime->hw); + + return amdtp_tscm_add_pcm_hw_constraints(stream, runtime); +} + +static int pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_tscm *tscm = substream->private_data; + enum snd_tscm_clock clock; + unsigned int rate; + int err; + + err = snd_tscm_stream_lock_try(tscm); + if (err < 0) + goto end; + + err = pcm_init_hw_params(tscm, substream); + if (err < 0) + goto err_locked; + + err = snd_tscm_stream_get_clock(tscm, &clock); + if (clock != SND_TSCM_CLOCK_INTERNAL || + amdtp_stream_pcm_running(&tscm->rx_stream) || + amdtp_stream_pcm_running(&tscm->tx_stream)) { + err = snd_tscm_stream_get_rate(tscm, &rate); + if (err < 0) + goto err_locked; + substream->runtime->hw.rate_min = rate; + substream->runtime->hw.rate_max = rate; + } + + snd_pcm_set_sync(substream); +end: + return err; +err_locked: + snd_tscm_stream_lock_release(tscm); + return err; +} + +static int pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_tscm *tscm = substream->private_data; + + snd_tscm_stream_lock_release(tscm); + + return 0; +} + +static int pcm_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_tscm *tscm = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + + if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + mutex_lock(&tscm->mutex); + tscm->substreams_counter++; + mutex_unlock(&tscm->mutex); + } + + amdtp_tscm_set_pcm_format(&tscm->tx_stream, params_format(hw_params)); + + return 0; +} + +static int pcm_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_tscm *tscm = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + + if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + mutex_lock(&tscm->mutex); + tscm->substreams_counter++; + mutex_unlock(&tscm->mutex); + } + + amdtp_tscm_set_pcm_format(&tscm->rx_stream, params_format(hw_params)); + + return 0; +} + +static int pcm_capture_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_tscm *tscm = substream->private_data; + + mutex_lock(&tscm->mutex); + + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + tscm->substreams_counter--; + + snd_tscm_stream_stop_duplex(tscm); + + mutex_unlock(&tscm->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int pcm_playback_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_tscm *tscm = substream->private_data; + + mutex_lock(&tscm->mutex); + + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + tscm->substreams_counter--; + + snd_tscm_stream_stop_duplex(tscm); + + mutex_unlock(&tscm->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_tscm *tscm = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + mutex_lock(&tscm->mutex); + + err = snd_tscm_stream_start_duplex(tscm, runtime->rate); + if (err >= 0) + amdtp_stream_pcm_prepare(&tscm->tx_stream); + + mutex_unlock(&tscm->mutex); + + return err; +} + +static int pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_tscm *tscm = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + mutex_lock(&tscm->mutex); + + err = snd_tscm_stream_start_duplex(tscm, runtime->rate); + if (err >= 0) + amdtp_stream_pcm_prepare(&tscm->rx_stream); + + mutex_unlock(&tscm->mutex); + + return err; +} + +static int pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_tscm *tscm = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + amdtp_stream_pcm_trigger(&tscm->tx_stream, substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + amdtp_stream_pcm_trigger(&tscm->tx_stream, NULL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_tscm *tscm = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + amdtp_stream_pcm_trigger(&tscm->rx_stream, substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + amdtp_stream_pcm_trigger(&tscm->rx_stream, NULL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm) +{ + struct snd_tscm *tscm = sbstrm->private_data; + + return amdtp_stream_pcm_pointer(&tscm->tx_stream); +} + +static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm) +{ + struct snd_tscm *tscm = sbstrm->private_data; + + return amdtp_stream_pcm_pointer(&tscm->rx_stream); +} + +static struct snd_pcm_ops pcm_capture_ops = { + .open = pcm_open, + .close = pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pcm_capture_hw_params, + .hw_free = pcm_capture_hw_free, + .prepare = pcm_capture_prepare, + .trigger = pcm_capture_trigger, + .pointer = pcm_capture_pointer, + .page = snd_pcm_lib_get_vmalloc_page, +}; + +static struct snd_pcm_ops pcm_playback_ops = { + .open = pcm_open, + .close = pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pcm_playback_hw_params, + .hw_free = pcm_playback_hw_free, + .prepare = pcm_playback_prepare, + .trigger = pcm_playback_trigger, + .pointer = pcm_playback_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, +}; + +int snd_tscm_create_pcm_devices(struct snd_tscm *tscm) +{ + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(tscm->card, tscm->card->driver, 0, 1, 1, &pcm); + if (err < 0) + return err; + + pcm->private_data = tscm; + snprintf(pcm->name, sizeof(pcm->name), + "%s PCM", tscm->card->shortname); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops); + + return 0; +} diff --git a/sound/firewire/tascam/tascam-proc.c b/sound/firewire/tascam/tascam-proc.c new file mode 100644 index 000000000000..bfd4a4c06914 --- /dev/null +++ b/sound/firewire/tascam/tascam-proc.c @@ -0,0 +1,88 @@ +/* + * tascam-proc.h - a part of driver for TASCAM FireWire series + * + * Copyright (c) 2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "./tascam.h" + +static void proc_read_firmware(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_tscm *tscm = entry->private_data; + __be32 data; + unsigned int reg, fpga, arm, hw; + int err; + + err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_REGISTER, + &data, sizeof(data), 0); + if (err < 0) + return; + reg = be32_to_cpu(data); + + err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_FPGA, + &data, sizeof(data), 0); + if (err < 0) + return; + fpga = be32_to_cpu(data); + + err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_ARM, + &data, sizeof(data), 0); + if (err < 0) + return; + arm = be32_to_cpu(data); + + err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_HW, + &data, sizeof(data), 0); + if (err < 0) + return; + hw = be32_to_cpu(data); + + snd_iprintf(buffer, "Register: %d (0x%08x)\n", reg & 0xffff, reg); + snd_iprintf(buffer, "FPGA: %d (0x%08x)\n", fpga & 0xffff, fpga); + snd_iprintf(buffer, "ARM: %d (0x%08x)\n", arm & 0xffff, arm); + snd_iprintf(buffer, "Hardware: %d (0x%08x)\n", hw >> 16, hw); +} + +static void add_node(struct snd_tscm *tscm, struct snd_info_entry *root, + const char *name, + void (*op)(struct snd_info_entry *e, + struct snd_info_buffer *b)) +{ + struct snd_info_entry *entry; + + entry = snd_info_create_card_entry(tscm->card, name, root); + if (entry == NULL) + return; + + snd_info_set_text_ops(entry, tscm, op); + if (snd_info_register(entry) < 0) + snd_info_free_entry(entry); +} + +void snd_tscm_proc_init(struct snd_tscm *tscm) +{ + struct snd_info_entry *root; + + /* + * All nodes are automatically removed at snd_card_disconnect(), + * by following to link list. + */ + root = snd_info_create_card_entry(tscm->card, "firewire", + tscm->card->proc_root); + if (root == NULL) + return; + root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + if (snd_info_register(root) < 0) { + snd_info_free_entry(root); + return; + } + + add_node(tscm, root, "firmware", proc_read_firmware); +} diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c new file mode 100644 index 000000000000..0e6dd5c61f53 --- /dev/null +++ b/sound/firewire/tascam/tascam-stream.c @@ -0,0 +1,496 @@ +/* + * tascam-stream.c - a part of driver for TASCAM FireWire series + * + * Copyright (c) 2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include <linux/delay.h> +#include "tascam.h" + +#define CALLBACK_TIMEOUT 500 + +static int get_clock(struct snd_tscm *tscm, u32 *data) +{ + __be32 reg; + int err; + + err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS, + ®, sizeof(reg), 0); + if (err >= 0) + *data = be32_to_cpu(reg); + + return err; +} + +static int set_clock(struct snd_tscm *tscm, unsigned int rate, + enum snd_tscm_clock clock) +{ + u32 data; + __be32 reg; + int err; + + err = get_clock(tscm, &data); + if (err < 0) + return err; + data &= 0x0000ffff; + + if (rate > 0) { + data &= 0x000000ff; + /* Base rate. */ + if ((rate % 44100) == 0) { + data |= 0x00000100; + /* Multiplier. */ + if (rate / 44100 == 2) + data |= 0x00008000; + } else if ((rate % 48000) == 0) { + data |= 0x00000200; + /* Multiplier. */ + if (rate / 48000 == 2) + data |= 0x00008000; + } else { + return -EAGAIN; + } + } + + if (clock != INT_MAX) { + data &= 0x0000ff00; + data |= clock + 1; + } + + reg = cpu_to_be32(data); + + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS, + ®, sizeof(reg), 0); + if (err < 0) + return err; + + if (data & 0x00008000) + reg = cpu_to_be32(0x0000001a); + else + reg = cpu_to_be32(0x0000000d); + + return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_MULTIPLEX_MODE, + ®, sizeof(reg), 0); +} + +int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate) +{ + u32 data = 0x0; + unsigned int trials = 0; + int err; + + while (data == 0x0 || trials++ < 5) { + err = get_clock(tscm, &data); + if (err < 0) + return err; + + data = (data & 0xff000000) >> 24; + } + + /* Check base rate. */ + if ((data & 0x0f) == 0x01) + *rate = 44100; + else if ((data & 0x0f) == 0x02) + *rate = 48000; + else + return -EAGAIN; + + /* Check multiplier. */ + if ((data & 0xf0) == 0x80) + *rate *= 2; + else if ((data & 0xf0) != 0x00) + return -EAGAIN; + + return err; +} + +int snd_tscm_stream_get_clock(struct snd_tscm *tscm, enum snd_tscm_clock *clock) +{ + u32 data; + int err; + + err = get_clock(tscm, &data); + if (err < 0) + return err; + + *clock = ((data & 0x00ff0000) >> 16) - 1; + if (*clock < 0 || *clock > SND_TSCM_CLOCK_ADAT) + return -EIO; + + return 0; +} + +static int enable_data_channels(struct snd_tscm *tscm) +{ + __be32 reg; + u32 data; + unsigned int i; + int err; + + data = 0; + for (i = 0; i < tscm->spec->pcm_capture_analog_channels; ++i) + data |= BIT(i); + if (tscm->spec->has_adat) + data |= 0x0000ff00; + if (tscm->spec->has_spdif) + data |= 0x00030000; + + reg = cpu_to_be32(data); + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_TX_PCM_CHANNELS, + ®, sizeof(reg), 0); + if (err < 0) + return err; + + data = 0; + for (i = 0; i < tscm->spec->pcm_playback_analog_channels; ++i) + data |= BIT(i); + if (tscm->spec->has_adat) + data |= 0x0000ff00; + if (tscm->spec->has_spdif) + data |= 0x00030000; + + reg = cpu_to_be32(data); + return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_RX_PCM_CHANNELS, + ®, sizeof(reg), 0); +} + +static int set_stream_formats(struct snd_tscm *tscm, unsigned int rate) +{ + __be32 reg; + int err; + + /* Set an option for unknown purpose. */ + reg = cpu_to_be32(0x00200000); + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_SET_OPTION, + ®, sizeof(reg), 0); + if (err < 0) + return err; + + err = enable_data_channels(tscm); + if (err < 0) + return err; + + return set_clock(tscm, rate, INT_MAX); +} + +static void finish_session(struct snd_tscm *tscm) +{ + __be32 reg; + + reg = 0; + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_START_STREAMING, + ®, sizeof(reg), 0); + + reg = 0; + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_ON, + ®, sizeof(reg), 0); + +} + +static int begin_session(struct snd_tscm *tscm) +{ + __be32 reg; + int err; + + reg = cpu_to_be32(0x00000001); + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_START_STREAMING, + ®, sizeof(reg), 0); + if (err < 0) + return err; + + reg = cpu_to_be32(0x00000001); + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_ON, + ®, sizeof(reg), 0); + if (err < 0) + return err; + + /* Set an option for unknown purpose. */ + reg = cpu_to_be32(0x00002000); + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_SET_OPTION, + ®, sizeof(reg), 0); + if (err < 0) + return err; + + /* Start multiplexing PCM samples on packets. */ + reg = cpu_to_be32(0x00000001); + return snd_fw_transaction(tscm->unit, + TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_ON, + ®, sizeof(reg), 0); +} + +static void release_resources(struct snd_tscm *tscm) +{ + __be32 reg; + + /* Unregister channels. */ + reg = cpu_to_be32(0x00000000); + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_CH, + ®, sizeof(reg), 0); + reg = cpu_to_be32(0x00000000); + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_UNKNOWN, + ®, sizeof(reg), 0); + reg = cpu_to_be32(0x00000000); + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_CH, + ®, sizeof(reg), 0); + + /* Release isochronous resources. */ + fw_iso_resources_free(&tscm->tx_resources); + fw_iso_resources_free(&tscm->rx_resources); +} + +static int keep_resources(struct snd_tscm *tscm, unsigned int rate) +{ + __be32 reg; + int err; + + /* Keep resources for in-stream. */ + err = amdtp_tscm_set_parameters(&tscm->tx_stream, rate); + if (err < 0) + return err; + err = fw_iso_resources_allocate(&tscm->tx_resources, + amdtp_stream_get_max_payload(&tscm->tx_stream), + fw_parent_device(tscm->unit)->max_speed); + if (err < 0) + goto error; + + /* Keep resources for out-stream. */ + err = amdtp_tscm_set_parameters(&tscm->rx_stream, rate); + if (err < 0) + return err; + err = fw_iso_resources_allocate(&tscm->rx_resources, + amdtp_stream_get_max_payload(&tscm->rx_stream), + fw_parent_device(tscm->unit)->max_speed); + if (err < 0) + return err; + + /* Register the isochronous channel for transmitting stream. */ + reg = cpu_to_be32(tscm->tx_resources.channel); + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_CH, + ®, sizeof(reg), 0); + if (err < 0) + goto error; + + /* Unknown */ + reg = cpu_to_be32(0x00000002); + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_UNKNOWN, + ®, sizeof(reg), 0); + if (err < 0) + goto error; + + /* Register the isochronous channel for receiving stream. */ + reg = cpu_to_be32(tscm->rx_resources.channel); + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_CH, + ®, sizeof(reg), 0); + if (err < 0) + goto error; + + return 0; +error: + release_resources(tscm); + return err; +} + +int snd_tscm_stream_init_duplex(struct snd_tscm *tscm) +{ + unsigned int pcm_channels; + int err; + + /* For out-stream. */ + err = fw_iso_resources_init(&tscm->rx_resources, tscm->unit); + if (err < 0) + return err; + pcm_channels = tscm->spec->pcm_playback_analog_channels; + if (tscm->spec->has_adat) + pcm_channels += 8; + if (tscm->spec->has_spdif) + pcm_channels += 2; + err = amdtp_tscm_init(&tscm->rx_stream, tscm->unit, AMDTP_OUT_STREAM, + pcm_channels); + if (err < 0) + return err; + + /* For in-stream. */ + err = fw_iso_resources_init(&tscm->tx_resources, tscm->unit); + if (err < 0) + return err; + pcm_channels = tscm->spec->pcm_capture_analog_channels; + if (tscm->spec->has_adat) + pcm_channels += 8; + if (tscm->spec->has_spdif) + pcm_channels += 2; + err = amdtp_tscm_init(&tscm->tx_stream, tscm->unit, AMDTP_IN_STREAM, + pcm_channels); + if (err < 0) + amdtp_stream_destroy(&tscm->rx_stream); + + return 0; +} + +/* At bus reset, streaming is stopped and some registers are clear. */ +void snd_tscm_stream_update_duplex(struct snd_tscm *tscm) +{ + amdtp_stream_pcm_abort(&tscm->tx_stream); + amdtp_stream_stop(&tscm->tx_stream); + + amdtp_stream_pcm_abort(&tscm->rx_stream); + amdtp_stream_stop(&tscm->rx_stream); +} + +/* + * This function should be called before starting streams or after stopping + * streams. + */ +void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm) +{ + amdtp_stream_destroy(&tscm->rx_stream); + amdtp_stream_destroy(&tscm->tx_stream); + + fw_iso_resources_destroy(&tscm->rx_resources); + fw_iso_resources_destroy(&tscm->tx_resources); +} + +int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) +{ + unsigned int curr_rate; + int err; + + if (tscm->substreams_counter == 0) + return 0; + + err = snd_tscm_stream_get_rate(tscm, &curr_rate); + if (err < 0) + return err; + if (curr_rate != rate || + amdtp_streaming_error(&tscm->tx_stream) || + amdtp_streaming_error(&tscm->rx_stream)) { + finish_session(tscm); + + amdtp_stream_stop(&tscm->tx_stream); + amdtp_stream_stop(&tscm->rx_stream); + + release_resources(tscm); + } + + if (!amdtp_stream_running(&tscm->tx_stream)) { + amdtp_stream_set_sync(CIP_SYNC_TO_DEVICE, + &tscm->tx_stream, &tscm->rx_stream); + err = keep_resources(tscm, rate); + if (err < 0) + goto error; + + err = set_stream_formats(tscm, rate); + if (err < 0) + goto error; + + err = begin_session(tscm); + if (err < 0) + goto error; + + err = amdtp_stream_start(&tscm->tx_stream, + tscm->tx_resources.channel, + fw_parent_device(tscm->unit)->max_speed); + if (err < 0) + goto error; + + if (!amdtp_stream_wait_callback(&tscm->tx_stream, + CALLBACK_TIMEOUT)) { + err = -ETIMEDOUT; + goto error; + } + } + + if (!amdtp_stream_running(&tscm->rx_stream)) { + err = amdtp_stream_start(&tscm->rx_stream, + tscm->rx_resources.channel, + fw_parent_device(tscm->unit)->max_speed); + if (err < 0) + goto error; + + if (!amdtp_stream_wait_callback(&tscm->rx_stream, + CALLBACK_TIMEOUT)) { + err = -ETIMEDOUT; + goto error; + } + } + + return 0; +error: + amdtp_stream_stop(&tscm->tx_stream); + amdtp_stream_stop(&tscm->rx_stream); + + finish_session(tscm); + release_resources(tscm); + + return err; +} + +void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm) +{ + if (tscm->substreams_counter > 0) + return; + + amdtp_stream_stop(&tscm->tx_stream); + amdtp_stream_stop(&tscm->rx_stream); + + finish_session(tscm); + release_resources(tscm); +} + +void snd_tscm_stream_lock_changed(struct snd_tscm *tscm) +{ + tscm->dev_lock_changed = true; + wake_up(&tscm->hwdep_wait); +} + +int snd_tscm_stream_lock_try(struct snd_tscm *tscm) +{ + int err; + + spin_lock_irq(&tscm->lock); + + /* user land lock this */ + if (tscm->dev_lock_count < 0) { + err = -EBUSY; + goto end; + } + + /* this is the first time */ + if (tscm->dev_lock_count++ == 0) + snd_tscm_stream_lock_changed(tscm); + err = 0; +end: + spin_unlock_irq(&tscm->lock); + return err; +} + +void snd_tscm_stream_lock_release(struct snd_tscm *tscm) +{ + spin_lock_irq(&tscm->lock); + + if (WARN_ON(tscm->dev_lock_count <= 0)) + goto end; + if (--tscm->dev_lock_count == 0) + snd_tscm_stream_lock_changed(tscm); +end: + spin_unlock_irq(&tscm->lock); +} diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c new file mode 100644 index 000000000000..904ce0329fa1 --- /dev/null +++ b/sound/firewire/tascam/tascam-transaction.c @@ -0,0 +1,302 @@ +/* + * tascam-transaction.c - a part of driver for TASCAM FireWire series + * + * Copyright (c) 2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "tascam.h" + +/* + * When return minus value, given argument is not MIDI status. + * When return 0, given argument is a beginning of system exclusive. + * When return the others, given argument is MIDI data. + */ +static inline int calculate_message_bytes(u8 status) +{ + switch (status) { + case 0xf6: /* Tune request. */ + case 0xf8: /* Timing clock. */ + case 0xfa: /* Start. */ + case 0xfb: /* Continue. */ + case 0xfc: /* Stop. */ + case 0xfe: /* Active sensing. */ + case 0xff: /* System reset. */ + return 1; + case 0xf1: /* MIDI time code quarter frame. */ + case 0xf3: /* Song select. */ + return 2; + case 0xf2: /* Song position pointer. */ + return 3; + case 0xf0: /* Exclusive. */ + return 0; + case 0xf7: /* End of exclusive. */ + break; + case 0xf4: /* Undefined. */ + case 0xf5: /* Undefined. */ + case 0xf9: /* Undefined. */ + case 0xfd: /* Undefined. */ + break; + default: + switch (status & 0xf0) { + case 0x80: /* Note on. */ + case 0x90: /* Note off. */ + case 0xa0: /* Polyphonic key pressure. */ + case 0xb0: /* Control change and Mode change. */ + case 0xe0: /* Pitch bend change. */ + return 3; + case 0xc0: /* Program change. */ + case 0xd0: /* Channel pressure. */ + return 2; + default: + break; + } + break; + } + + return -EINVAL; +} + +static int fill_message(struct snd_rawmidi_substream *substream, u8 *buf) +{ + struct snd_tscm *tscm = substream->rmidi->private_data; + unsigned int port = substream->number; + int i, len, consume; + u8 *label, *msg; + u8 status; + + /* The first byte is used for label, the rest for MIDI bytes. */ + label = buf; + msg = buf + 1; + + consume = snd_rawmidi_transmit_peek(substream, msg, 3); + if (consume == 0) + return 0; + + /* On exclusive message. */ + if (tscm->on_sysex[port]) { + /* Seek the end of exclusives. */ + for (i = 0; i < consume; ++i) { + if (msg[i] == 0xf7) { + tscm->on_sysex[port] = false; + break; + } + } + + /* At the end of exclusive message, use label 0x07. */ + if (!tscm->on_sysex[port]) { + consume = i + 1; + *label = (port << 4) | 0x07; + /* During exclusive message, use label 0x04. */ + } else if (consume == 3) { + *label = (port << 4) | 0x04; + /* We need to fill whole 3 bytes. Go to next change. */ + } else { + return 0; + } + + len = consume; + } else { + /* The beginning of exclusives. */ + if (msg[0] == 0xf0) { + /* Transfer it in next chance in another condition. */ + tscm->on_sysex[port] = true; + return 0; + } else { + /* On running-status. */ + if ((msg[0] & 0x80) != 0x80) + status = tscm->running_status[port]; + else + status = msg[0]; + + /* Calculate consume bytes. */ + len = calculate_message_bytes(status); + if (len <= 0) + return 0; + + /* On running-status. */ + if ((msg[0] & 0x80) != 0x80) { + /* Enough MIDI bytes were not retrieved. */ + if (consume < len - 1) + return 0; + consume = len - 1; + + msg[2] = msg[1]; + msg[1] = msg[0]; + msg[0] = tscm->running_status[port]; + } else { + /* Enough MIDI bytes were not retrieved. */ + if (consume < len) + return 0; + consume = len; + + tscm->running_status[port] = msg[0]; + } + } + + *label = (port << 4) | (msg[0] >> 4); + } + + if (len > 0 && len < 3) + memset(msg + len, 0, 3 - len); + + return consume; +} + +static void handle_midi_tx(struct fw_card *card, struct fw_request *request, + int tcode, int destination, int source, + int generation, unsigned long long offset, + void *data, size_t length, void *callback_data) +{ + struct snd_tscm *tscm = callback_data; + u32 *buf = (u32 *)data; + unsigned int messages; + unsigned int i; + unsigned int port; + struct snd_rawmidi_substream *substream; + u8 *b; + int bytes; + + if (offset != tscm->async_handler.offset) + goto end; + + messages = length / 8; + for (i = 0; i < messages; i++) { + b = (u8 *)(buf + i * 2); + + port = b[0] >> 4; + /* TODO: support virtual MIDI ports. */ + if (port >= tscm->spec->midi_capture_ports) + goto end; + + /* Assume the message length. */ + bytes = calculate_message_bytes(b[1]); + /* On MIDI data or exclusives. */ + if (bytes <= 0) { + /* Seek the end of exclusives. */ + for (bytes = 1; bytes < 4; bytes++) { + if (b[bytes] == 0xf7) + break; + } + if (bytes == 4) + bytes = 3; + } + + substream = ACCESS_ONCE(tscm->tx_midi_substreams[port]); + if (substream != NULL) + snd_rawmidi_receive(substream, b + 1, bytes); + } +end: + fw_send_response(card, request, RCODE_COMPLETE); +} + +int snd_tscm_transaction_register(struct snd_tscm *tscm) +{ + static const struct fw_address_region resp_register_region = { + .start = 0xffffe0000000ull, + .end = 0xffffe000ffffull, + }; + unsigned int i; + int err; + + /* + * Usually, two quadlets are transferred by one transaction. The first + * quadlet has MIDI messages, the rest includes timestamp. + * Sometimes, 8 set of the data is transferred by a block transaction. + */ + tscm->async_handler.length = 8 * 8; + tscm->async_handler.address_callback = handle_midi_tx; + tscm->async_handler.callback_data = tscm; + + err = fw_core_add_address_handler(&tscm->async_handler, + &resp_register_region); + if (err < 0) + return err; + + err = snd_tscm_transaction_reregister(tscm); + if (err < 0) + goto error; + + for (i = 0; i < TSCM_MIDI_OUT_PORT_MAX; i++) { + err = snd_fw_async_midi_port_init( + &tscm->out_ports[i], tscm->unit, + TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_RX_QUAD, + 4, fill_message); + if (err < 0) + goto error; + } + + return err; +error: + fw_core_remove_address_handler(&tscm->async_handler); + return err; +} + +/* At bus reset, these registers are cleared. */ +int snd_tscm_transaction_reregister(struct snd_tscm *tscm) +{ + struct fw_device *device = fw_parent_device(tscm->unit); + __be32 reg; + int err; + + /* Register messaging address. Block transaction is not allowed. */ + reg = cpu_to_be32((device->card->node_id << 16) | + (tscm->async_handler.offset >> 32)); + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_HI, + ®, sizeof(reg), 0); + if (err < 0) + return err; + + reg = cpu_to_be32(tscm->async_handler.offset); + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_LO, + ®, sizeof(reg), 0); + if (err < 0) + return err; + + /* Turn on messaging. */ + reg = cpu_to_be32(0x00000001); + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON, + ®, sizeof(reg), 0); + if (err < 0) + return err; + + /* Turn on FireWire LED. */ + reg = cpu_to_be32(0x0001008e); + return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_LED_POWER, + ®, sizeof(reg), 0); +} + +void snd_tscm_transaction_unregister(struct snd_tscm *tscm) +{ + __be32 reg; + unsigned int i; + + /* Turn off FireWire LED. */ + reg = cpu_to_be32(0x0000008e); + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_LED_POWER, + ®, sizeof(reg), 0); + + /* Turn off messaging. */ + reg = cpu_to_be32(0x00000000); + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON, + ®, sizeof(reg), 0); + + /* Unregister the address. */ + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_HI, + ®, sizeof(reg), 0); + snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_LO, + ®, sizeof(reg), 0); + + fw_core_remove_address_handler(&tscm->async_handler); + for (i = 0; i < TSCM_MIDI_OUT_PORT_MAX; i++) + snd_fw_async_midi_port_destroy(&tscm->out_ports[i]); +} diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c new file mode 100644 index 000000000000..ee0bc1839508 --- /dev/null +++ b/sound/firewire/tascam/tascam.c @@ -0,0 +1,209 @@ +/* + * tascam.c - a part of driver for TASCAM FireWire series + * + * Copyright (c) 2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "tascam.h" + +MODULE_DESCRIPTION("TASCAM FireWire series Driver"); +MODULE_AUTHOR("Takashi Sakamoto <o-takashi@sakamocchi.jp>"); +MODULE_LICENSE("GPL v2"); + +static struct snd_tscm_spec model_specs[] = { + { + .name = "FW-1884", + .has_adat = true, + .has_spdif = true, + .pcm_capture_analog_channels = 8, + .pcm_playback_analog_channels = 8, + .midi_capture_ports = 4, + .midi_playback_ports = 4, + .is_controller = true, + }, + { + .name = "FW-1082", + .has_adat = false, + .has_spdif = true, + .pcm_capture_analog_channels = 8, + .pcm_playback_analog_channels = 2, + .midi_capture_ports = 2, + .midi_playback_ports = 2, + .is_controller = true, + }, + /* FW-1804 may be supported. */ +}; + +static int identify_model(struct snd_tscm *tscm) +{ + struct fw_device *fw_dev = fw_parent_device(tscm->unit); + const u32 *config_rom = fw_dev->config_rom; + char model[9]; + unsigned int i; + u8 c; + + if (fw_dev->config_rom_length < 30) { + dev_err(&tscm->unit->device, + "Configuration ROM is too short.\n"); + return -ENODEV; + } + + /* Pick up model name from certain addresses. */ + for (i = 0; i < 8; i++) { + c = config_rom[28 + i / 4] >> (24 - 8 * (i % 4)); + if (c == '\0') + break; + model[i] = c; + } + model[i] = '\0'; + + for (i = 0; i < ARRAY_SIZE(model_specs); i++) { + if (strcmp(model, model_specs[i].name) == 0) { + tscm->spec = &model_specs[i]; + break; + } + } + if (tscm->spec == NULL) + return -ENODEV; + + strcpy(tscm->card->driver, "FW-TASCAM"); + strcpy(tscm->card->shortname, model); + strcpy(tscm->card->mixername, model); + snprintf(tscm->card->longname, sizeof(tscm->card->longname), + "TASCAM %s, GUID %08x%08x at %s, S%d", model, + fw_dev->config_rom[3], fw_dev->config_rom[4], + dev_name(&tscm->unit->device), 100 << fw_dev->max_speed); + + return 0; +} + +static void tscm_card_free(struct snd_card *card) +{ + struct snd_tscm *tscm = card->private_data; + + snd_tscm_transaction_unregister(tscm); + snd_tscm_stream_destroy_duplex(tscm); + + fw_unit_put(tscm->unit); + + mutex_destroy(&tscm->mutex); +} + +static int snd_tscm_probe(struct fw_unit *unit, + const struct ieee1394_device_id *entry) +{ + struct snd_card *card; + struct snd_tscm *tscm; + int err; + + /* create card */ + err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, + sizeof(struct snd_tscm), &card); + if (err < 0) + return err; + card->private_free = tscm_card_free; + + /* initialize myself */ + tscm = card->private_data; + tscm->card = card; + tscm->unit = fw_unit_get(unit); + + mutex_init(&tscm->mutex); + spin_lock_init(&tscm->lock); + init_waitqueue_head(&tscm->hwdep_wait); + + err = identify_model(tscm); + if (err < 0) + goto error; + + snd_tscm_proc_init(tscm); + + err = snd_tscm_stream_init_duplex(tscm); + if (err < 0) + goto error; + + err = snd_tscm_create_pcm_devices(tscm); + if (err < 0) + goto error; + + err = snd_tscm_transaction_register(tscm); + if (err < 0) + goto error; + + err = snd_tscm_create_midi_devices(tscm); + if (err < 0) + goto error; + + err = snd_tscm_create_hwdep_device(tscm); + if (err < 0) + goto error; + + err = snd_card_register(card); + if (err < 0) + goto error; + + dev_set_drvdata(&unit->device, tscm); + + return err; +error: + snd_card_free(card); + return err; +} + +static void snd_tscm_update(struct fw_unit *unit) +{ + struct snd_tscm *tscm = dev_get_drvdata(&unit->device); + + snd_tscm_transaction_reregister(tscm); + + mutex_lock(&tscm->mutex); + snd_tscm_stream_update_duplex(tscm); + mutex_unlock(&tscm->mutex); +} + +static void snd_tscm_remove(struct fw_unit *unit) +{ + struct snd_tscm *tscm = dev_get_drvdata(&unit->device); + + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(tscm->card); +} + +static const struct ieee1394_device_id snd_tscm_id_table[] = { + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID, + .vendor_id = 0x00022e, + .specifier_id = 0x00022e, + }, + /* FE-08 requires reverse-engineering because it just has faders. */ + {} +}; +MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table); + +static struct fw_driver tscm_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "snd-firewire-tascam", + .bus = &fw_bus_type, + }, + .probe = snd_tscm_probe, + .update = snd_tscm_update, + .remove = snd_tscm_remove, + .id_table = snd_tscm_id_table, +}; + +static int __init snd_tscm_init(void) +{ + return driver_register(&tscm_driver.driver); +} + +static void __exit snd_tscm_exit(void) +{ + driver_unregister(&tscm_driver.driver); +} + +module_init(snd_tscm_init); +module_exit(snd_tscm_exit); diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h new file mode 100644 index 000000000000..2d028d2bd3bd --- /dev/null +++ b/sound/firewire/tascam/tascam.h @@ -0,0 +1,147 @@ +/* + * tascam.h - a part of driver for TASCAM FireWire series + * + * Copyright (c) 2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#ifndef SOUND_TASCAM_H_INCLUDED +#define SOUND_TASCAM_H_INCLUDED + +#include <linux/device.h> +#include <linux/firewire.h> +#include <linux/firewire-constants.h> +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/mutex.h> +#include <linux/slab.h> +#include <linux/compat.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/info.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/firewire.h> +#include <sound/hwdep.h> +#include <sound/rawmidi.h> + +#include "../lib.h" +#include "../amdtp-stream.h" +#include "../iso-resources.h" + +struct snd_tscm_spec { + const char *const name; + bool has_adat; + bool has_spdif; + unsigned int pcm_capture_analog_channels; + unsigned int pcm_playback_analog_channels; + unsigned int midi_capture_ports; + unsigned int midi_playback_ports; + bool is_controller; +}; + +#define TSCM_MIDI_IN_PORT_MAX 4 +#define TSCM_MIDI_OUT_PORT_MAX 4 + +struct snd_tscm { + struct snd_card *card; + struct fw_unit *unit; + + struct mutex mutex; + spinlock_t lock; + + const struct snd_tscm_spec *spec; + + struct fw_iso_resources tx_resources; + struct fw_iso_resources rx_resources; + struct amdtp_stream tx_stream; + struct amdtp_stream rx_stream; + unsigned int substreams_counter; + + int dev_lock_count; + bool dev_lock_changed; + wait_queue_head_t hwdep_wait; + + /* For MIDI message incoming transactions. */ + struct fw_address_handler async_handler; + struct snd_rawmidi_substream *tx_midi_substreams[TSCM_MIDI_IN_PORT_MAX]; + + /* For MIDI message outgoing transactions. */ + struct snd_fw_async_midi_port out_ports[TSCM_MIDI_OUT_PORT_MAX]; + u8 running_status[TSCM_MIDI_OUT_PORT_MAX]; + bool on_sysex[TSCM_MIDI_OUT_PORT_MAX]; + + /* For control messages. */ + struct snd_firewire_tascam_status *status; +}; + +#define TSCM_ADDR_BASE 0xffff00000000ull + +#define TSCM_OFFSET_FIRMWARE_REGISTER 0x0000 +#define TSCM_OFFSET_FIRMWARE_FPGA 0x0004 +#define TSCM_OFFSET_FIRMWARE_ARM 0x0008 +#define TSCM_OFFSET_FIRMWARE_HW 0x000c + +#define TSCM_OFFSET_ISOC_TX_CH 0x0200 +#define TSCM_OFFSET_UNKNOWN 0x0204 +#define TSCM_OFFSET_START_STREAMING 0x0208 +#define TSCM_OFFSET_ISOC_RX_CH 0x020c +#define TSCM_OFFSET_ISOC_RX_ON 0x0210 /* Little conviction. */ +#define TSCM_OFFSET_TX_PCM_CHANNELS 0x0214 +#define TSCM_OFFSET_RX_PCM_CHANNELS 0x0218 +#define TSCM_OFFSET_MULTIPLEX_MODE 0x021c +#define TSCM_OFFSET_ISOC_TX_ON 0x0220 +/* Unknown 0x0224 */ +#define TSCM_OFFSET_CLOCK_STATUS 0x0228 +#define TSCM_OFFSET_SET_OPTION 0x022c + +#define TSCM_OFFSET_MIDI_TX_ON 0x0300 +#define TSCM_OFFSET_MIDI_TX_ADDR_HI 0x0304 +#define TSCM_OFFSET_MIDI_TX_ADDR_LO 0x0308 + +#define TSCM_OFFSET_LED_POWER 0x0404 + +#define TSCM_OFFSET_MIDI_RX_QUAD 0x4000 + +enum snd_tscm_clock { + SND_TSCM_CLOCK_INTERNAL = 0, + SND_TSCM_CLOCK_WORD = 1, + SND_TSCM_CLOCK_SPDIF = 2, + SND_TSCM_CLOCK_ADAT = 3, +}; + +int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit, + enum amdtp_stream_direction dir, unsigned int pcm_channels); +int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate); +int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s, + struct snd_pcm_runtime *runtime); +void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format); + +int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate); +int snd_tscm_stream_get_clock(struct snd_tscm *tscm, + enum snd_tscm_clock *clock); +int snd_tscm_stream_init_duplex(struct snd_tscm *tscm); +void snd_tscm_stream_update_duplex(struct snd_tscm *tscm); +void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm); +int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate); +void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm); + +void snd_tscm_stream_lock_changed(struct snd_tscm *tscm); +int snd_tscm_stream_lock_try(struct snd_tscm *tscm); +void snd_tscm_stream_lock_release(struct snd_tscm *tscm); + +int snd_tscm_transaction_register(struct snd_tscm *tscm); +int snd_tscm_transaction_reregister(struct snd_tscm *tscm); +void snd_tscm_transaction_unregister(struct snd_tscm *tscm); + +void snd_tscm_proc_init(struct snd_tscm *tscm); + +int snd_tscm_create_pcm_devices(struct snd_tscm *tscm); + +int snd_tscm_create_midi_devices(struct snd_tscm *tscm); + +int snd_tscm_create_hwdep_device(struct snd_tscm *tscm); + +#endif diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 33ba77dd32f2..cb89ec7c8147 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -227,7 +227,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_link_stream_setup); void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link, int stream) { - snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 0); + snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 1 << stream); } EXPORT_SYMBOL_GPL(snd_hdac_ext_link_set_stream_id); @@ -385,14 +385,13 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type) break; case HDAC_EXT_STREAM_TYPE_HOST: - if (stream->decoupled) { + if (stream->decoupled && !stream->link_locked) snd_hdac_ext_stream_decouple(ebus, stream, false); - snd_hdac_stream_release(&stream->hstream); - } + snd_hdac_stream_release(&stream->hstream); break; case HDAC_EXT_STREAM_TYPE_LINK: - if (stream->decoupled) + if (stream->decoupled && !stream->hstream.opened) snd_hdac_ext_stream_decouple(ebus, stream, false); spin_lock_irq(&bus->reg_lock); stream->link_locked = 0; diff --git a/sound/hda/hda_bus_type.c b/sound/hda/hda_bus_type.c index 89c2711baaaf..3060e2aee36f 100644 --- a/sound/hda/hda_bus_type.c +++ b/sound/hda/hda_bus_type.c @@ -4,6 +4,7 @@ #include <linux/init.h> #include <linux/device.h> #include <linux/module.h> +#include <linux/mod_devicetable.h> #include <linux/export.h> #include <sound/hdaudio.h> @@ -63,9 +64,21 @@ static int hda_bus_match(struct device *dev, struct device_driver *drv) return 1; } +static int hda_uevent(struct device *dev, struct kobj_uevent_env *env) +{ + char modalias[32]; + + snd_hdac_codec_modalias(dev_to_hdac_dev(dev), modalias, + sizeof(modalias)); + if (add_uevent_var(env, "MODALIAS=%s", modalias)) + return -ENOMEM; + return 0; +} + struct bus_type snd_hda_bus_type = { .name = "hdaudio", .match = hda_bus_match, + .uevent = hda_uevent, }; EXPORT_SYMBOL_GPL(snd_hda_bus_type); diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 27c447e4fe5c..0e81ea89a596 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -172,6 +172,15 @@ static void process_unsol_events(struct work_struct *work) } } +/** + * snd_hdac_bus_add_device - Add a codec to bus + * @bus: HDA core bus + * @codec: HDA core device to add + * + * Adds the given codec to the list in the bus. The caddr_tbl array + * and codec_powered bits are updated, as well. + * Returns zero if success, or a negative error code. + */ int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec) { if (bus->caddr_tbl[codec->addr]) { @@ -188,6 +197,11 @@ int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec) } EXPORT_SYMBOL_GPL(snd_hdac_bus_add_device); +/** + * snd_hdac_bus_remove_device - Remove a codec from bus + * @bus: HDA core bus + * @codec: HDA core device to remove + */ void snd_hdac_bus_remove_device(struct hdac_bus *bus, struct hdac_device *codec) { diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index db96042a497f..e361024eabb6 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -164,6 +164,43 @@ void snd_hdac_device_unregister(struct hdac_device *codec) EXPORT_SYMBOL_GPL(snd_hdac_device_unregister); /** + * snd_hdac_device_set_chip_name - set/update the codec name + * @codec: the HDAC device + * @name: name string to set + * + * Returns 0 if the name is set or updated, or a negative error code. + */ +int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name) +{ + char *newname; + + if (!name) + return 0; + newname = kstrdup(name, GFP_KERNEL); + if (!newname) + return -ENOMEM; + kfree(codec->chip_name); + codec->chip_name = newname; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name); + +/** + * snd_hdac_codec_modalias - give the module alias name + * @codec: HDAC device + * @buf: string buffer to store + * @size: string buffer size + * + * Returns the size of string, like snprintf(), or a negative error code. + */ +int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size) +{ + return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n", + codec->vendor_id, codec->revision_id, codec->type); +} +EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias); + +/** * snd_hdac_make_cmd - compose a 32bit command word to be sent to the * HD-audio controller * @codec: the codec object @@ -592,8 +629,10 @@ int snd_hdac_power_down_pm(struct hdac_device *codec) EXPORT_SYMBOL_GPL(snd_hdac_power_down_pm); #endif -/* - * Enable/disable the link power for a codec. +/** + * snd_hdac_link_power - Enable/disable the link power for a codec + * @codec: the codec object + * @bool: enable or disable the link power */ int snd_hdac_link_power(struct hdac_device *codec, bool enable) { @@ -952,3 +991,84 @@ bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid, return true; } EXPORT_SYMBOL_GPL(snd_hdac_is_supported_format); + +static unsigned int codec_read(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm); + unsigned int res; + + if (snd_hdac_exec_verb(hdac, cmd, flags, &res)) + return -1; + + return res; +} + +static int codec_write(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm); + + return snd_hdac_exec_verb(hdac, cmd, flags, NULL); +} + +/** + * snd_hdac_codec_read - send a command and get the response + * @hdac: the HDAC device + * @nid: NID to send the command + * @flags: optional bit flags + * @verb: the verb to send + * @parm: the parameter for the verb + * + * Send a single command and read the corresponding response. + * + * Returns the obtained response value, or -1 for an error. + */ +int snd_hdac_codec_read(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + return codec_read(hdac, nid, flags, verb, parm); +} +EXPORT_SYMBOL_GPL(snd_hdac_codec_read); + +/** + * snd_hdac_codec_write - send a single command without waiting for response + * @hdac: the HDAC device + * @nid: NID to send the command + * @flags: optional bit flags + * @verb: the verb to send + * @parm: the parameter for the verb + * + * Send a single command without waiting for response. + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + return codec_write(hdac, nid, flags, verb, parm); +} +EXPORT_SYMBOL_GPL(snd_hdac_codec_write); + +/** + * snd_hdac_check_power_state - check whether the actual power state matches + * with the target state + * + * @hdac: the HDAC device + * @nid: NID to send the command + * @target_state: target state to check for + * + * Return true if state matches, false if not + */ +bool snd_hdac_check_power_state(struct hdac_device *hdac, + hda_nid_t nid, unsigned int target_state) +{ + unsigned int state = codec_read(hdac, nid, 0, + AC_VERB_GET_POWER_STATE, 0); + + if (state & AC_PWRST_ERROR) + return true; + state = (state >> 4) & 0x0f; + return (state == target_state); +} +EXPORT_SYMBOL_GPL(snd_hdac_check_power_state); diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 55c3df4458f7..8fef1b8d1fd8 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -23,6 +23,19 @@ static struct i915_audio_component *hdac_acomp; +/** + * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup + * @bus: HDA core bus + * @enable: enable or disable the wakeup + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with i915 graphics. + * + * This function should be called during the chip reset, also called at + * resume for updating STATESTS register read. + * + * Returns zero for success or a negative error code. + */ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) { struct i915_audio_component *acomp = bus->audio_component; @@ -45,6 +58,19 @@ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) } EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup); +/** + * snd_hdac_display_power - Power up / down the power refcount + * @bus: HDA core bus + * @enable: power up or down + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with i915 graphics. + * + * This function manages a refcount and calls the i915 get_power() and + * put_power() ops accordingly, toggling the codec wakeup, too. + * + * Returns zero for success or a negative error code. + */ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) { struct i915_audio_component *acomp = bus->audio_component; @@ -71,6 +97,16 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) } EXPORT_SYMBOL_GPL(snd_hdac_display_power); +/** + * snd_hdac_get_display_clk - Get CDCLK in kHz + * @bus: HDA core bus + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with i915 graphics. + * + * This function queries CDCLK value in kHz from the graphics driver and + * returns the value. A negative code is returned in error. + */ int snd_hdac_get_display_clk(struct hdac_bus *bus) { struct i915_audio_component *acomp = bus->audio_component; @@ -134,6 +170,17 @@ static int hdac_component_master_match(struct device *dev, void *data) return !strcmp(dev->driver->name, "i915"); } +/** + * snd_hdac_i915_register_notifier - Register i915 audio component ops + * @aops: i915 audio component ops + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with i915 graphics. + * + * This function sets the given ops to be called by the i915 graphics driver. + * + * Returns zero for success or a negative error code. + */ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops) { if (WARN_ON(!hdac_acomp)) @@ -144,6 +191,18 @@ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops } EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); +/** + * snd_hdac_i915_init - Initialize i915 audio component + * @bus: HDA core bus + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with i915 graphics. + * + * This function initializes and sets up the audio component to communicate + * with i915 graphics driver. + * + * Returns zero for success or a negative error code. + */ int snd_hdac_i915_init(struct hdac_bus *bus) { struct component_match *match = NULL; @@ -187,6 +246,17 @@ out_err: } EXPORT_SYMBOL_GPL(snd_hdac_i915_init); +/** + * snd_hdac_i915_exit - Finalize i915 audio component + * @bus: HDA core bus + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with i915 graphics. + * + * This function releases the i915 audio component that has been used. + * + * Returns zero for success or a negative error code. + */ int snd_hdac_i915_exit(struct hdac_bus *bus) { struct device *dev = bus->dev; diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index b0ed870ffb88..eb8f7c30cb09 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -339,6 +339,12 @@ static const struct regmap_config hda_regmap_cfg = { .use_single_rw = true, }; +/** + * snd_hdac_regmap_init - Initialize regmap for HDA register accesses + * @codec: the codec object + * + * Returns zero for success or a negative error code. + */ int snd_hdac_regmap_init(struct hdac_device *codec) { struct regmap *regmap; @@ -352,6 +358,10 @@ int snd_hdac_regmap_init(struct hdac_device *codec) } EXPORT_SYMBOL_GPL(snd_hdac_regmap_init); +/** + * snd_hdac_regmap_init - Release the regmap from HDA codec + * @codec: the codec object + */ void snd_hdac_regmap_exit(struct hdac_device *codec) { if (codec->regmap) { diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 8981159813ef..38990a77d7b7 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -426,7 +426,8 @@ int snd_hdac_stream_setup_periods(struct hdac_stream *azx_dev) } EXPORT_SYMBOL_GPL(snd_hdac_stream_setup_periods); -/* snd_hdac_stream_set_params - set stream parameters +/** + * snd_hdac_stream_set_params - set stream parameters * @azx_dev: HD-audio core stream for which parameters are to be set * @format_val: format value parameter * diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index c71142dea98a..42d61bf41969 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -45,6 +45,13 @@ CODEC_ATTR(mfg); CODEC_ATTR_STR(vendor_name); CODEC_ATTR_STR(chip_name); +static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, + char *buf) +{ + return snd_hdac_codec_modalias(dev_to_hdac_dev(dev), buf, 256); +} +static DEVICE_ATTR_RO(modalias); + static struct attribute *hdac_dev_attrs[] = { &dev_attr_type.attr, &dev_attr_vendor_id.attr, @@ -54,6 +61,7 @@ static struct attribute *hdac_dev_attrs[] = { &dev_attr_mfg.attr, &dev_attr_vendor_name.attr, &dev_attr_chip_name.attr, + &dev_attr_modalias.attr, NULL }; diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 2a9f4a345171..2706f271a83b 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -1864,7 +1864,7 @@ int snd_cs46xx_pcm_iec958(struct snd_cs46xx *chip, int device) /* global setup */ pcm->info_flags = 0; strcpy(pcm->name, "CS46xx - IEC958"); - chip->pcm_rear = pcm; + chip->pcm_iec958 = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), 64*1024, 256*1024); @@ -2528,7 +2528,7 @@ int snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device) #ifdef CONFIG_SND_CS46XX_NEW_DSP if (chip->nr_ac97_codecs == 1) { unsigned int id2 = chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]->id & 0xffff; - if (id2 == 0x592b || id2 == 0x592d) { + if ((id2 & 0xfff0) == 0x5920) { /* CS4294 and CS4298 */ err = snd_ctl_add(card, snd_ctl_new1(&snd_cs46xx_front_dup_ctl, chip)); if (err < 0) return err; @@ -3780,6 +3780,11 @@ static int snd_cs46xx_suspend(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->in_suspend = 1; snd_pcm_suspend_all(chip->pcm); +#ifdef CONFIG_SND_CS46XX_NEW_DSP + snd_pcm_suspend_all(chip->pcm_rear); + snd_pcm_suspend_all(chip->pcm_center_lfe); + snd_pcm_suspend_all(chip->pcm_iec958); +#endif // chip->ac97_powerdown = snd_cs46xx_codec_read(chip, AC97_POWER_CONTROL); // chip->ac97_general_purpose = snd_cs46xx_codec_read(chip, BA0_AC97_GENERAL_PURPOSE); diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index d5ac25cc7fee..70671ad65d24 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -15,21 +15,22 @@ #include "hda_local.h" /* - * find a matching codec preset + * find a matching codec id */ static int hda_codec_match(struct hdac_device *dev, struct hdac_driver *drv) { struct hda_codec *codec = container_of(dev, struct hda_codec, core); struct hda_codec_driver *driver = container_of(drv, struct hda_codec_driver, core); - const struct hda_codec_preset *preset; + const struct hda_device_id *list; /* check probe_id instead of vendor_id if set */ u32 id = codec->probe_id ? codec->probe_id : codec->core.vendor_id; + u32 rev_id = codec->core.revision_id; - for (preset = driver->preset; preset->id; preset++) { - if (preset->id == id && - (!preset->rev || preset->rev == codec->core.revision_id)) { - codec->preset = preset; + for (list = driver->id; list->vendor_id; list++) { + if (list->vendor_id == id && + (!list->rev_id || list->rev_id == rev_id)) { + codec->preset = list; return 1; } } @@ -45,26 +46,45 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) codec->patch_ops.unsol_event(codec, ev); } -/* reset the codec name from the preset */ -static int codec_refresh_name(struct hda_codec *codec, const char *name) +/** + * snd_hda_codec_set_name - set the codec name + * @codec: the HDA codec + * @name: name string to set + */ +int snd_hda_codec_set_name(struct hda_codec *codec, const char *name) { - if (name) { - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup(name, GFP_KERNEL); + int err; + + if (!name) + return 0; + err = snd_hdac_device_set_chip_name(&codec->core, name); + if (err < 0) + return err; + + /* update the mixer name */ + if (!*codec->card->mixername || + codec->bus->mixer_assigned >= codec->core.addr) { + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), "%s %s", + codec->core.vendor_name, codec->core.chip_name); + codec->bus->mixer_assigned = codec->core.addr; } - return codec->core.chip_name ? 0 : -ENOMEM; + + return 0; } +EXPORT_SYMBOL_GPL(snd_hda_codec_set_name); static int hda_codec_driver_probe(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); struct module *owner = dev->driver->owner; + hda_codec_patch_t patch; int err; if (WARN_ON(!codec->preset)) return -EINVAL; - err = codec_refresh_name(codec, codec->preset->name); + err = snd_hda_codec_set_name(codec, codec->preset->name); if (err < 0) goto error; err = snd_hdac_regmap_init(&codec->core); @@ -76,9 +96,12 @@ static int hda_codec_driver_probe(struct device *dev) goto error; } - err = codec->preset->patch(codec); - if (err < 0) - goto error_module; + patch = (hda_codec_patch_t)codec->preset->driver_data; + if (patch) { + err = patch(codec); + if (err < 0) + goto error_module; + } err = snd_hda_codec_build_pcms(codec); if (err < 0) @@ -155,11 +178,10 @@ static inline bool codec_probed(struct hda_codec *codec) static void codec_bind_module(struct hda_codec *codec) { #ifdef MODULE - request_module("snd-hda-codec-id:%08x", codec->core.vendor_id); - if (codec_probed(codec)) - return; - request_module("snd-hda-codec-id:%04x*", - (codec->core.vendor_id >> 16) & 0xffff); + char modalias[32]; + + snd_hdac_codec_modalias(&codec->core, modalias, sizeof(modalias)); + request_module(modalias); if (codec_probed(codec)) return; #endif @@ -251,11 +273,6 @@ int snd_hda_codec_configure(struct hda_codec *codec) } } - /* audio codec should override the mixer name */ - if (codec->core.afg || !*codec->card->mixername) - snprintf(codec->card->mixername, - sizeof(codec->card->mixername), "%s %s", - codec->core.vendor_name, codec->core.chip_name); return 0; error: diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a249d5486889..83741887faa1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -91,50 +91,6 @@ static int codec_exec_verb(struct hdac_device *dev, unsigned int cmd, } /** - * snd_hda_codec_read - send a command and get the response - * @codec: the HDA codec - * @nid: NID to send the command - * @flags: optional bit flags - * @verb: the verb to send - * @parm: the parameter for the verb - * - * Send a single command and read the corresponding response. - * - * Returns the obtained response value, or -1 for an error. - */ -unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, - int flags, - unsigned int verb, unsigned int parm) -{ - unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm); - unsigned int res; - if (snd_hdac_exec_verb(&codec->core, cmd, flags, &res)) - return -1; - return res; -} -EXPORT_SYMBOL_GPL(snd_hda_codec_read); - -/** - * snd_hda_codec_write - send a single command without waiting for response - * @codec: the HDA codec - * @nid: NID to send the command - * @flags: optional bit flags - * @verb: the verb to send - * @parm: the parameter for the verb - * - * Send a single command without waiting for response. - * - * Returns 0 if successful, or a negative error code. - */ -int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, - unsigned int verb, unsigned int parm) -{ - unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm); - return snd_hdac_exec_verb(&codec->core, cmd, flags, NULL); -} -EXPORT_SYMBOL_GPL(snd_hda_codec_write); - -/** * snd_hda_sequence_write - sequence writes * @codec: the HDA codec * @seq: VERB array to send diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2970413f18a0..373fcad840ea 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -22,6 +22,7 @@ #define __SOUND_HDA_CODEC_H #include <linux/kref.h> +#include <linux/mod_devicetable.h> #include <sound/info.h> #include <sound/control.h> #include <sound/pcm.h> @@ -69,6 +70,7 @@ struct hda_bus { unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ int primary_dig_out_type; /* primary digital out PCM type */ + unsigned int mixer_assigned; /* codec addr for mixer name */ }; /* from hdac_bus to hda_bus */ @@ -80,19 +82,21 @@ struct hda_bus { * Known codecs have the patch to build and set up the controls/PCMs * better than the generic parser. */ -struct hda_codec_preset { - unsigned int id; - unsigned int rev; - const char *name; - int (*patch)(struct hda_codec *codec); -}; +typedef int (*hda_codec_patch_t)(struct hda_codec *); #define HDA_CODEC_ID_GENERIC_HDMI 0x00000101 #define HDA_CODEC_ID_GENERIC 0x00000201 +#define HDA_CODEC_REV_ENTRY(_vid, _rev, _name, _patch) \ + { .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \ + .api_version = HDA_DEV_LEGACY, \ + .driver_data = (unsigned long)(_patch) } +#define HDA_CODEC_ENTRY(_vid, _name, _patch) \ + HDA_CODEC_REV_ENTRY(_vid, 0, _name, _patch) + struct hda_codec_driver { struct hdac_driver core; - const struct hda_codec_preset *preset; + const struct hda_device_id *id; }; int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, @@ -183,7 +187,7 @@ struct hda_codec { u32 probe_id; /* overridden id for probing */ /* detected preset */ - const struct hda_codec_preset *preset; + const struct hda_device_id *preset; const char *modelname; /* model name for preset */ /* set by patch */ @@ -297,10 +301,6 @@ struct hda_codec { /* * constructors */ -int snd_hda_bus_new(struct snd_card *card, - const struct hdac_bus_ops *ops, - const struct hdac_io_ops *io_ops, - struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, unsigned int codec_addr, struct hda_codec **codecp); int snd_hda_codec_configure(struct hda_codec *codec); @@ -309,11 +309,21 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec); /* * low level functions */ -unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, +static inline unsigned int +snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int flags, - unsigned int verb, unsigned int parm); -int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, - unsigned int verb, unsigned int parm); + unsigned int verb, unsigned int parm) +{ + return snd_hdac_codec_read(&codec->core, nid, flags, verb, parm); +} + +static inline int +snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, + unsigned int verb, unsigned int parm) +{ + return snd_hdac_codec_write(&codec->core, nid, flags, verb, parm); +} + #define snd_hda_param_read(codec, nid, param) \ snd_hdac_read_parm(&(codec)->core, nid, param) #define snd_hda_get_sub_nodes(codec, nid, start_nid) \ @@ -453,6 +463,8 @@ void snd_hda_unlock_devices(struct hda_bus *bus); void snd_hda_bus_reset(struct hda_bus *bus); void snd_hda_bus_reset_codecs(struct hda_bus *bus); +int snd_hda_codec_set_name(struct hda_codec *codec, const char *name); + /* * power management */ diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 944455997fdc..d6b93a20361b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1045,6 +1045,7 @@ int azx_bus_init(struct azx *chip, const char *model, mutex_init(&bus->prepare_mutex); bus->pci = chip->pci; bus->modelname = model; + bus->mixer_assigned = -1; bus->core.snoop = azx_snoop(chip); if (chip->get_position[0] != azx_get_pos_lpib || chip->get_position[1] != azx_get_pos_lpib) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 24f91114a32c..c6e8a651cea1 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5877,13 +5877,14 @@ error: return err; } -static const struct hda_codec_preset snd_hda_preset_generic[] = { - { .id = HDA_CODEC_ID_GENERIC, .patch = snd_hda_parse_generic_codec }, +static const struct hda_device_id snd_hda_id_generic[] = { + HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC, "Generic", snd_hda_parse_generic_codec), {} /* terminator */ }; +MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_generic); static struct hda_codec_driver generic_driver = { - .preset = snd_hda_preset_generic, + .id = snd_hda_id_generic, }; module_hda_codec_driver(generic_driver); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 4a21c2199e02..d0e066e4c985 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -681,12 +681,7 @@ static inline bool snd_hda_check_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int target_state) { - unsigned int state = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_POWER_STATE, 0); - if (state & AC_PWRST_ERROR) - return true; - state = (state >> 4) & 0x0f; - return (state == target_state); + return snd_hdac_check_power_state(&codec->core, nid, target_state); } unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec, diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index a6e3d9b511ab..64e0d1d81ca5 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -595,8 +595,7 @@ static void parse_model_mode(char *buf, struct hda_bus *bus, static void parse_chip_name_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - kfree((*codecp)->core.chip_name); - (*codecp)->core.chip_name = kstrdup(buf, GFP_KERNEL); + snd_hda_codec_set_name(*codecp, buf); } #define DEFINE_PARSE_ID_MODE(name) \ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index c033a4ee6547..e0fb8c6d1bc2 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1165,32 +1165,31 @@ static int patch_ad1882(struct hda_codec *codec) /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_analog[] = { - { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884 }, - { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, - { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884 }, - { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, - { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884 }, - { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884 }, - { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, - { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, - { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1884 }, - { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, - { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 }, - { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 }, - { .id = 0x11d4882a, .name = "AD1882A", .patch = patch_ad1882 }, - { .id = 0x11d4989a, .name = "AD1989A", .patch = patch_ad1988 }, - { .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 }, +static const struct hda_device_id snd_hda_id_analog[] = { + HDA_CODEC_ENTRY(0x11d4184a, "AD1884A", patch_ad1884), + HDA_CODEC_ENTRY(0x11d41882, "AD1882", patch_ad1882), + HDA_CODEC_ENTRY(0x11d41883, "AD1883", patch_ad1884), + HDA_CODEC_ENTRY(0x11d41884, "AD1884", patch_ad1884), + HDA_CODEC_ENTRY(0x11d4194a, "AD1984A", patch_ad1884), + HDA_CODEC_ENTRY(0x11d4194b, "AD1984B", patch_ad1884), + HDA_CODEC_ENTRY(0x11d41981, "AD1981", patch_ad1981), + HDA_CODEC_ENTRY(0x11d41983, "AD1983", patch_ad1983), + HDA_CODEC_ENTRY(0x11d41984, "AD1984", patch_ad1884), + HDA_CODEC_ENTRY(0x11d41986, "AD1986A", patch_ad1986a), + HDA_CODEC_ENTRY(0x11d41988, "AD1988", patch_ad1988), + HDA_CODEC_ENTRY(0x11d4198b, "AD1988B", patch_ad1988), + HDA_CODEC_ENTRY(0x11d4882a, "AD1882A", patch_ad1882), + HDA_CODEC_ENTRY(0x11d4989a, "AD1989A", patch_ad1988), + HDA_CODEC_ENTRY(0x11d4989b, "AD1989B", patch_ad1988), {} /* terminator */ }; - -MODULE_ALIAS("snd-hda-codec-id:11d4*"); +MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_analog); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Analog Devices HD-audio codec"); static struct hda_codec_driver analog_driver = { - .preset = snd_hda_preset_analog, + .id = snd_hda_id_analog, }; module_hda_codec_driver(analog_driver); diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 484bbf4134cd..c2d9ee9cfdc0 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -83,22 +83,19 @@ static int patch_ca0110(struct hda_codec *codec) /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_ca0110[] = { - { .id = 0x1102000a, .name = "CA0110-IBG", .patch = patch_ca0110 }, - { .id = 0x1102000b, .name = "CA0110-IBG", .patch = patch_ca0110 }, - { .id = 0x1102000d, .name = "SB0880 X-Fi", .patch = patch_ca0110 }, +static const struct hda_device_id snd_hda_id_ca0110[] = { + HDA_CODEC_ENTRY(0x1102000a, "CA0110-IBG", patch_ca0110), + HDA_CODEC_ENTRY(0x1102000b, "CA0110-IBG", patch_ca0110), + HDA_CODEC_ENTRY(0x1102000d, "SB0880 X-Fi", patch_ca0110), {} /* terminator */ }; - -MODULE_ALIAS("snd-hda-codec-id:1102000a"); -MODULE_ALIAS("snd-hda-codec-id:1102000b"); -MODULE_ALIAS("snd-hda-codec-id:1102000d"); +MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_ca0110); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Creative CA0110-IBG HD-audio codec"); static struct hda_codec_driver ca0110_driver = { - .preset = snd_hda_preset_ca0110, + .id = snd_hda_id_ca0110, }; module_hda_codec_driver(ca0110_driver); diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 186792fe226e..f8a12ca477f1 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2673,13 +2673,13 @@ static bool dspload_wait_loaded(struct hda_codec *codec) do { if (dspload_is_loaded(codec)) { - pr_info("ca0132 DOWNLOAD OK :-) DSP IS RUNNING.\n"); + codec_info(codec, "ca0132 DSP downloaded and running\n"); return true; } msleep(20); } while (time_before(jiffies, timeout)); - pr_err("ca0132 DOWNLOAD FAILED!!! DSP IS NOT RUNNING.\n"); + codec_err(codec, "ca0132 failed to download DSP\n"); return false; } @@ -4375,7 +4375,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) { - pr_err("ca0132 dspload_image failed.\n"); + codec_err(codec, "ca0132 DSP load image failed\n"); goto exit_download; } @@ -4778,18 +4778,17 @@ static int patch_ca0132(struct hda_codec *codec) /* * patch entries */ -static struct hda_codec_preset snd_hda_preset_ca0132[] = { - { .id = 0x11020011, .name = "CA0132", .patch = patch_ca0132 }, +static struct hda_device_id snd_hda_id_ca0132[] = { + HDA_CODEC_ENTRY(0x11020011, "CA0132", patch_ca0132), {} /* terminator */ }; - -MODULE_ALIAS("snd-hda-codec-id:11020011"); +MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_ca0132); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Creative Sound Core3D codec"); static struct hda_codec_driver ca0132_driver = { - .preset = snd_hda_preset_ca0132, + .id = snd_hda_id_ca0132, }; module_hda_codec_driver(ca0132_driver); diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 85813de26da8..a12ae8ac0914 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -570,6 +570,7 @@ static struct cs_spec *cs_alloc_spec(struct hda_codec *codec, int vendor_nid) return NULL; codec->spec = spec; spec->vendor_nid = vendor_nid; + codec->power_save_node = 1; snd_hda_gen_spec_init(&spec->gen); return spec; @@ -1200,26 +1201,21 @@ static int patch_cs4213(struct hda_codec *codec) /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_cirrus[] = { - { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, - { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, - { .id = 0x10134208, .name = "CS4208", .patch = patch_cs4208 }, - { .id = 0x10134210, .name = "CS4210", .patch = patch_cs4210 }, - { .id = 0x10134213, .name = "CS4213", .patch = patch_cs4213 }, +static const struct hda_device_id snd_hda_id_cirrus[] = { + HDA_CODEC_ENTRY(0x10134206, "CS4206", patch_cs420x), + HDA_CODEC_ENTRY(0x10134207, "CS4207", patch_cs420x), + HDA_CODEC_ENTRY(0x10134208, "CS4208", patch_cs4208), + HDA_CODEC_ENTRY(0x10134210, "CS4210", patch_cs4210), + HDA_CODEC_ENTRY(0x10134213, "CS4213", patch_cs4213), {} /* terminator */ }; - -MODULE_ALIAS("snd-hda-codec-id:10134206"); -MODULE_ALIAS("snd-hda-codec-id:10134207"); -MODULE_ALIAS("snd-hda-codec-id:10134208"); -MODULE_ALIAS("snd-hda-codec-id:10134210"); -MODULE_ALIAS("snd-hda-codec-id:10134213"); +MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_cirrus); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); static struct hda_codec_driver cirrus_driver = { - .preset = snd_hda_preset_cirrus, + .id = snd_hda_id_cirrus, }; module_hda_codec_driver(cirrus_driver); diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index f5ed078710f8..1b2195dd2b26 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -123,22 +123,19 @@ static int patch_cmi8888(struct hda_codec *codec) /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_cmedia[] = { - { .id = 0x13f68888, .name = "CMI8888", .patch = patch_cmi8888 }, - { .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 }, - { .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 }, +static const struct hda_device_id snd_hda_id_cmedia[] = { + HDA_CODEC_ENTRY(0x13f68888, "CMI8888", patch_cmi8888), + HDA_CODEC_ENTRY(0x13f69880, "CMI9880", patch_cmi9880), + HDA_CODEC_ENTRY(0x434d4980, "CMI9880", patch_cmi9880), {} /* terminator */ }; - -MODULE_ALIAS("snd-hda-codec-id:13f68888"); -MODULE_ALIAS("snd-hda-codec-id:13f69880"); -MODULE_ALIAS("snd-hda-codec-id:434d4980"); +MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_cmedia); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("C-Media HD-audio codec"); static struct hda_codec_driver cmedia_driver = { - .preset = snd_hda_preset_cmedia, + .id = snd_hda_id_cmedia, }; module_hda_codec_driver(cmedia_driver); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 2f0ec7c45fc7..c8b8ef5246a6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -954,100 +954,44 @@ static int patch_conexant_auto(struct hda_codec *codec) /* */ -static const struct hda_codec_preset snd_hda_preset_conexant[] = { - { .id = 0x14f15045, .name = "CX20549 (Venice)", - .patch = patch_conexant_auto }, - { .id = 0x14f15047, .name = "CX20551 (Waikiki)", - .patch = patch_conexant_auto }, - { .id = 0x14f15051, .name = "CX20561 (Hermosa)", - .patch = patch_conexant_auto }, - { .id = 0x14f15066, .name = "CX20582 (Pebble)", - .patch = patch_conexant_auto }, - { .id = 0x14f15067, .name = "CX20583 (Pebble HSF)", - .patch = patch_conexant_auto }, - { .id = 0x14f15068, .name = "CX20584", - .patch = patch_conexant_auto }, - { .id = 0x14f15069, .name = "CX20585", - .patch = patch_conexant_auto }, - { .id = 0x14f1506c, .name = "CX20588", - .patch = patch_conexant_auto }, - { .id = 0x14f1506e, .name = "CX20590", - .patch = patch_conexant_auto }, - { .id = 0x14f15097, .name = "CX20631", - .patch = patch_conexant_auto }, - { .id = 0x14f15098, .name = "CX20632", - .patch = patch_conexant_auto }, - { .id = 0x14f150a1, .name = "CX20641", - .patch = patch_conexant_auto }, - { .id = 0x14f150a2, .name = "CX20642", - .patch = patch_conexant_auto }, - { .id = 0x14f150ab, .name = "CX20651", - .patch = patch_conexant_auto }, - { .id = 0x14f150ac, .name = "CX20652", - .patch = patch_conexant_auto }, - { .id = 0x14f150b8, .name = "CX20664", - .patch = patch_conexant_auto }, - { .id = 0x14f150b9, .name = "CX20665", - .patch = patch_conexant_auto }, - { .id = 0x14f150f1, .name = "CX20721", - .patch = patch_conexant_auto }, - { .id = 0x14f150f2, .name = "CX20722", - .patch = patch_conexant_auto }, - { .id = 0x14f150f3, .name = "CX20723", - .patch = patch_conexant_auto }, - { .id = 0x14f150f4, .name = "CX20724", - .patch = patch_conexant_auto }, - { .id = 0x14f1510f, .name = "CX20751/2", - .patch = patch_conexant_auto }, - { .id = 0x14f15110, .name = "CX20751/2", - .patch = patch_conexant_auto }, - { .id = 0x14f15111, .name = "CX20753/4", - .patch = patch_conexant_auto }, - { .id = 0x14f15113, .name = "CX20755", - .patch = patch_conexant_auto }, - { .id = 0x14f15114, .name = "CX20756", - .patch = patch_conexant_auto }, - { .id = 0x14f15115, .name = "CX20757", - .patch = patch_conexant_auto }, - { .id = 0x14f151d7, .name = "CX20952", - .patch = patch_conexant_auto }, +static const struct hda_device_id snd_hda_id_conexant[] = { + HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15066, "CX20582 (Pebble)", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15067, "CX20583 (Pebble HSF)", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15068, "CX20584", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15069, "CX20585", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f1506c, "CX20588", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f1506e, "CX20590", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15097, "CX20631", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15098, "CX20632", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150a1, "CX20641", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150a2, "CX20642", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150ab, "CX20651", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150ac, "CX20652", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150b8, "CX20664", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150b9, "CX20665", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150f1, "CX20721", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150f2, "CX20722", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150f3, "CX20723", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150f4, "CX20724", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f1510f, "CX20751/2", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15110, "CX20751/2", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15111, "CX20753/4", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15113, "CX20755", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15114, "CX20756", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f15115, "CX20757", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f151d7, "CX20952", patch_conexant_auto), {} /* terminator */ }; - -MODULE_ALIAS("snd-hda-codec-id:14f15045"); -MODULE_ALIAS("snd-hda-codec-id:14f15047"); -MODULE_ALIAS("snd-hda-codec-id:14f15051"); -MODULE_ALIAS("snd-hda-codec-id:14f15066"); -MODULE_ALIAS("snd-hda-codec-id:14f15067"); -MODULE_ALIAS("snd-hda-codec-id:14f15068"); -MODULE_ALIAS("snd-hda-codec-id:14f15069"); -MODULE_ALIAS("snd-hda-codec-id:14f1506c"); -MODULE_ALIAS("snd-hda-codec-id:14f1506e"); -MODULE_ALIAS("snd-hda-codec-id:14f15097"); -MODULE_ALIAS("snd-hda-codec-id:14f15098"); -MODULE_ALIAS("snd-hda-codec-id:14f150a1"); -MODULE_ALIAS("snd-hda-codec-id:14f150a2"); -MODULE_ALIAS("snd-hda-codec-id:14f150ab"); -MODULE_ALIAS("snd-hda-codec-id:14f150ac"); -MODULE_ALIAS("snd-hda-codec-id:14f150b8"); -MODULE_ALIAS("snd-hda-codec-id:14f150b9"); -MODULE_ALIAS("snd-hda-codec-id:14f150f1"); -MODULE_ALIAS("snd-hda-codec-id:14f150f2"); -MODULE_ALIAS("snd-hda-codec-id:14f150f3"); -MODULE_ALIAS("snd-hda-codec-id:14f150f4"); -MODULE_ALIAS("snd-hda-codec-id:14f1510f"); -MODULE_ALIAS("snd-hda-codec-id:14f15110"); -MODULE_ALIAS("snd-hda-codec-id:14f15111"); -MODULE_ALIAS("snd-hda-codec-id:14f15113"); -MODULE_ALIAS("snd-hda-codec-id:14f15114"); -MODULE_ALIAS("snd-hda-codec-id:14f15115"); -MODULE_ALIAS("snd-hda-codec-id:14f151d7"); +MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_conexant); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); static struct hda_codec_driver conexant_driver = { - .preset = snd_hda_preset_conexant, + .id = snd_hda_id_conexant, }; module_hda_codec_driver(conexant_driver); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index acbfbe087ee8..f503a883bef3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1775,6 +1775,16 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) return non_pcm; } +/* There is a fixed mapping between audio pin node and display port + * on current Intel platforms: + * Pin Widget 5 - PORT B (port = 1 in i915 driver) + * Pin Widget 6 - PORT C (port = 2 in i915 driver) + * Pin Widget 7 - PORT D (port = 3 in i915 driver) + */ +static int intel_pin2port(hda_nid_t pin_nid) +{ + return pin_nid - 4; +} /* * HDMI callbacks @@ -1791,6 +1801,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, int pin_idx = hinfo_to_pin_index(codec, hinfo); struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; + struct snd_pcm_runtime *runtime = substream->runtime; + struct i915_audio_component *acomp = codec->bus->core.audio_component; bool non_pcm; int pinctl; @@ -1807,6 +1819,13 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx); } + /* Call sync_audio_rate to set the N/CTS/M manually if necessary */ + /* Todo: add DP1.2 MST audio support later */ + if (acomp && acomp->ops && acomp->ops->sync_audio_rate) + acomp->ops->sync_audio_rate(acomp->dev, + intel_pin2port(pin_nid), + runtime->rate); + non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); per_pin->channels = substream->runtime->channels; @@ -2561,7 +2580,7 @@ static int simple_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; struct snd_pcm_hw_constraint_list *hw_constraints_channels = NULL; - switch (codec->preset->id) { + switch (codec->preset->vendor_id) { case 0x10de0002: case 0x10de0003: case 0x10de0005: @@ -2879,7 +2898,7 @@ static int nvhdmi_7x_8ch_build_controls(struct hda_codec *codec) snd_pcm_alt_chmaps, 8, 0, &chmap); if (err < 0) return err; - switch (codec->preset->id) { + switch (codec->preset->vendor_id) { case 0x10de0002: case 0x10de0003: case 0x10de0005: @@ -3487,138 +3506,77 @@ static int patch_via_hdmi(struct hda_codec *codec) /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_hdmi[] = { -{ .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi }, -{ .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi }, -{ .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi }, -{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi }, -{ .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x10de0002, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, -{ .id = 0x10de0003, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, -{ .id = 0x10de0005, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, -{ .id = 0x10de0006, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, -{ .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x }, -{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_nvhdmi }, -{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_nvhdmi }, +static const struct hda_device_id snd_hda_id_hdmi[] = { +HDA_CODEC_ENTRY(0x1002793c, "RS600 HDMI", patch_atihdmi), +HDA_CODEC_ENTRY(0x10027919, "RS600 HDMI", patch_atihdmi), +HDA_CODEC_ENTRY(0x1002791a, "RS690/780 HDMI", patch_atihdmi), +HDA_CODEC_ENTRY(0x1002aa01, "R6xx HDMI", patch_atihdmi), +HDA_CODEC_ENTRY(0x10951390, "SiI1390 HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x10951392, "SiI1392 HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x17e80047, "Chrontel HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x10de0002, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x), +HDA_CODEC_ENTRY(0x10de0003, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x), +HDA_CODEC_ENTRY(0x10de0005, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x), +HDA_CODEC_ENTRY(0x10de0006, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x), +HDA_CODEC_ENTRY(0x10de0007, "MCP79/7A HDMI", patch_nvhdmi_8ch_7x), +HDA_CODEC_ENTRY(0x10de000a, "GPU 0a HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de000b, "GPU 0b HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de000c, "MCP89 HDMI", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de000d, "GPU 0d HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0010, "GPU 10 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0011, "GPU 11 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0012, "GPU 12 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0013, "GPU 13 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0014, "GPU 14 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0015, "GPU 15 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0016, "GPU 16 HDMI/DP", patch_nvhdmi), /* 17 is known to be absent */ -{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0020, .name = "Tegra30 HDMI", .patch = patch_tegra_hdmi }, -{ .id = 0x10de0022, .name = "Tegra114 HDMI", .patch = patch_tegra_hdmi }, -{ .id = 0x10de0028, .name = "Tegra124 HDMI", .patch = patch_tegra_hdmi }, -{ .id = 0x10de0029, .name = "Tegra210 HDMI/DP", .patch = patch_tegra_hdmi }, -{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, -{ .id = 0x10de0070, .name = "GPU 70 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de0072, .name = "GPU 72 HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de007d, .name = "GPU 7d HDMI/DP", .patch = patch_nvhdmi }, -{ .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, -{ .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, -{ .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, -{ .id = 0x11069f84, .name = "VX11 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x11069f85, .name = "VX11 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x80860054, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862801, .name = "Bearlake HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862802, .name = "Cantiga HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862803, .name = "Eaglelake HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862807, .name = "Haswell HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862808, .name = "Broadwell HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862809, .name = "Skylake HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x8086280a, .name = "Broxton HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862883, .name = "Braswell HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, +HDA_CODEC_ENTRY(0x10de0018, "GPU 18 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0019, "GPU 19 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de001a, "GPU 1a HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de001b, "GPU 1b HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de001c, "GPU 1c HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0020, "Tegra30 HDMI", patch_tegra_hdmi), +HDA_CODEC_ENTRY(0x10de0022, "Tegra114 HDMI", patch_tegra_hdmi), +HDA_CODEC_ENTRY(0x10de0028, "Tegra124 HDMI", patch_tegra_hdmi), +HDA_CODEC_ENTRY(0x10de0029, "Tegra210 HDMI/DP", patch_tegra_hdmi), +HDA_CODEC_ENTRY(0x10de0040, "GPU 40 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0041, "GPU 41 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0042, "GPU 42 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0043, "GPU 43 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0044, "GPU 44 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0051, "GPU 51 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0060, "GPU 60 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0067, "MCP67 HDMI", patch_nvhdmi_2ch), +HDA_CODEC_ENTRY(0x10de0070, "GPU 70 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0071, "GPU 71 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0072, "GPU 72 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de007d, "GPU 7d HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), +HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), +HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), +HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_generic_hdmi), /* special ID for generic HDMI */ -{ .id = HDA_CODEC_ID_GENERIC_HDMI, .patch = patch_generic_hdmi }, +HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC_HDMI, "Generic HDMI", patch_generic_hdmi), {} /* terminator */ }; - -MODULE_ALIAS("snd-hda-codec-id:1002793c"); -MODULE_ALIAS("snd-hda-codec-id:10027919"); -MODULE_ALIAS("snd-hda-codec-id:1002791a"); -MODULE_ALIAS("snd-hda-codec-id:1002aa01"); -MODULE_ALIAS("snd-hda-codec-id:10951390"); -MODULE_ALIAS("snd-hda-codec-id:10951392"); -MODULE_ALIAS("snd-hda-codec-id:10de0002"); -MODULE_ALIAS("snd-hda-codec-id:10de0003"); -MODULE_ALIAS("snd-hda-codec-id:10de0005"); -MODULE_ALIAS("snd-hda-codec-id:10de0006"); -MODULE_ALIAS("snd-hda-codec-id:10de0007"); -MODULE_ALIAS("snd-hda-codec-id:10de000a"); -MODULE_ALIAS("snd-hda-codec-id:10de000b"); -MODULE_ALIAS("snd-hda-codec-id:10de000c"); -MODULE_ALIAS("snd-hda-codec-id:10de000d"); -MODULE_ALIAS("snd-hda-codec-id:10de0010"); -MODULE_ALIAS("snd-hda-codec-id:10de0011"); -MODULE_ALIAS("snd-hda-codec-id:10de0012"); -MODULE_ALIAS("snd-hda-codec-id:10de0013"); -MODULE_ALIAS("snd-hda-codec-id:10de0014"); -MODULE_ALIAS("snd-hda-codec-id:10de0015"); -MODULE_ALIAS("snd-hda-codec-id:10de0016"); -MODULE_ALIAS("snd-hda-codec-id:10de0018"); -MODULE_ALIAS("snd-hda-codec-id:10de0019"); -MODULE_ALIAS("snd-hda-codec-id:10de001a"); -MODULE_ALIAS("snd-hda-codec-id:10de001b"); -MODULE_ALIAS("snd-hda-codec-id:10de001c"); -MODULE_ALIAS("snd-hda-codec-id:10de0028"); -MODULE_ALIAS("snd-hda-codec-id:10de0040"); -MODULE_ALIAS("snd-hda-codec-id:10de0041"); -MODULE_ALIAS("snd-hda-codec-id:10de0042"); -MODULE_ALIAS("snd-hda-codec-id:10de0043"); -MODULE_ALIAS("snd-hda-codec-id:10de0044"); -MODULE_ALIAS("snd-hda-codec-id:10de0051"); -MODULE_ALIAS("snd-hda-codec-id:10de0060"); -MODULE_ALIAS("snd-hda-codec-id:10de0067"); -MODULE_ALIAS("snd-hda-codec-id:10de0070"); -MODULE_ALIAS("snd-hda-codec-id:10de0071"); -MODULE_ALIAS("snd-hda-codec-id:10de0072"); -MODULE_ALIAS("snd-hda-codec-id:10de007d"); -MODULE_ALIAS("snd-hda-codec-id:10de8001"); -MODULE_ALIAS("snd-hda-codec-id:11069f80"); -MODULE_ALIAS("snd-hda-codec-id:11069f81"); -MODULE_ALIAS("snd-hda-codec-id:11069f84"); -MODULE_ALIAS("snd-hda-codec-id:11069f85"); -MODULE_ALIAS("snd-hda-codec-id:17e80047"); -MODULE_ALIAS("snd-hda-codec-id:80860054"); -MODULE_ALIAS("snd-hda-codec-id:80862801"); -MODULE_ALIAS("snd-hda-codec-id:80862802"); -MODULE_ALIAS("snd-hda-codec-id:80862803"); -MODULE_ALIAS("snd-hda-codec-id:80862804"); -MODULE_ALIAS("snd-hda-codec-id:80862805"); -MODULE_ALIAS("snd-hda-codec-id:80862806"); -MODULE_ALIAS("snd-hda-codec-id:80862807"); -MODULE_ALIAS("snd-hda-codec-id:80862808"); -MODULE_ALIAS("snd-hda-codec-id:80862809"); -MODULE_ALIAS("snd-hda-codec-id:8086280a"); -MODULE_ALIAS("snd-hda-codec-id:80862880"); -MODULE_ALIAS("snd-hda-codec-id:80862882"); -MODULE_ALIAS("snd-hda-codec-id:80862883"); -MODULE_ALIAS("snd-hda-codec-id:808629fb"); +MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_hdmi); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("HDMI HD-audio codec"); @@ -3627,7 +3585,7 @@ MODULE_ALIAS("snd-hda-codec-nvhdmi"); MODULE_ALIAS("snd-hda-codec-atihdmi"); static struct hda_codec_driver hdmi_driver = { - .preset = snd_hda_preset_hdmi, + .id = snd_hda_id_hdmi, }; module_hda_codec_driver(hdmi_driver); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 720a9fb32e20..2f7b065f9ac4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -822,17 +822,7 @@ static const struct hda_codec_ops alc_patch_ops = { }; -/* replace the codec chip_name with the given string */ -static int alc_codec_rename(struct hda_codec *codec, const char *name) -{ - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup(name, GFP_KERNEL); - if (!codec->core.chip_name) { - alc_free(codec); - return -ENOMEM; - } - return 0; -} +#define alc_codec_rename(codec, name) snd_hda_codec_set_name(codec, name) /* * Rename codecs appropriately from COEF value or subvendor id @@ -6640,78 +6630,70 @@ static int patch_alc680(struct hda_codec *codec) /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_realtek[] = { - { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 }, - { .id = 0x10ec0231, .name = "ALC231", .patch = patch_alc269 }, - { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 }, - { .id = 0x10ec0235, .name = "ALC233", .patch = patch_alc269 }, - { .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 }, - { .id = 0x10ec0256, .name = "ALC256", .patch = patch_alc269 }, - { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, - { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, - { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, - { .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 }, - { .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 }, - { .id = 0x10ec0270, .name = "ALC270", .patch = patch_alc269 }, - { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 }, - { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 }, - { .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 }, - { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 }, - { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 }, - { .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 }, - { .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 }, - { .id = 0x10ec0285, .name = "ALC285", .patch = patch_alc269 }, - { .id = 0x10ec0286, .name = "ALC286", .patch = patch_alc269 }, - { .id = 0x10ec0288, .name = "ALC288", .patch = patch_alc269 }, - { .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 }, - { .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 }, - { .id = 0x10ec0293, .name = "ALC293", .patch = patch_alc269 }, - { .id = 0x10ec0298, .name = "ALC298", .patch = patch_alc269 }, - { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", - .patch = patch_alc861 }, - { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, - { .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 }, - { .id = 0x10ec0862, .name = "ALC861-VD", .patch = patch_alc861vd }, - { .id = 0x10ec0662, .rev = 0x100002, .name = "ALC662 rev2", - .patch = patch_alc882 }, - { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1", - .patch = patch_alc662 }, - { .id = 0x10ec0662, .rev = 0x100300, .name = "ALC662 rev3", - .patch = patch_alc662 }, - { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, - { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, - { .id = 0x10ec0667, .name = "ALC667", .patch = patch_alc662 }, - { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 }, - { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, - { .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 }, - { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, - { .id = 0x10ec0867, .name = "ALC891", .patch = patch_alc882 }, - { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, - { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, - { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 }, - { .id = 0x10ec0885, .rev = 0x100101, .name = "ALC889A", - .patch = patch_alc882 }, - { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A", - .patch = patch_alc882 }, - { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, - { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 }, - { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", - .patch = patch_alc882 }, - { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 }, - { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 }, - { .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 }, - { .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 }, - { .id = 0x10ec0900, .name = "ALC1150", .patch = patch_alc882 }, +static const struct hda_device_id snd_hda_id_realtek[] = { + HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0260, "ALC260", patch_alc260), + HDA_CODEC_ENTRY(0x10ec0262, "ALC262", patch_alc262), + HDA_CODEC_ENTRY(0x10ec0267, "ALC267", patch_alc268), + HDA_CODEC_ENTRY(0x10ec0268, "ALC268", patch_alc268), + HDA_CODEC_ENTRY(0x10ec0269, "ALC269", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0270, "ALC270", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0272, "ALC272", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0275, "ALC275", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0276, "ALC276", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0280, "ALC280", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0282, "ALC282", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0283, "ALC283", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0284, "ALC284", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0285, "ALC285", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0286, "ALC286", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0288, "ALC288", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0292, "ALC292", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0293, "ALC293", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269), + HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), + HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), + HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861), + HDA_CODEC_ENTRY(0x10ec0862, "ALC861-VD", patch_alc861vd), + HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100002, "ALC662 rev2", patch_alc882), + HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100101, "ALC662 rev1", patch_alc662), + HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100300, "ALC662 rev3", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0663, "ALC663", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0665, "ALC665", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0667, "ALC667", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0668, "ALC668", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0670, "ALC670", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0671, "ALC671", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0680, "ALC680", patch_alc680), + HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc882), + HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880), + HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882), + HDA_CODEC_ENTRY(0x10ec0883, "ALC883", patch_alc882), + HDA_CODEC_REV_ENTRY(0x10ec0885, 0x100101, "ALC889A", patch_alc882), + HDA_CODEC_REV_ENTRY(0x10ec0885, 0x100103, "ALC889A", patch_alc882), + HDA_CODEC_ENTRY(0x10ec0885, "ALC885", patch_alc882), + HDA_CODEC_ENTRY(0x10ec0887, "ALC887", patch_alc882), + HDA_CODEC_REV_ENTRY(0x10ec0888, 0x100101, "ALC1200", patch_alc882), + HDA_CODEC_ENTRY(0x10ec0888, "ALC888", patch_alc882), + HDA_CODEC_ENTRY(0x10ec0889, "ALC889", patch_alc882), + HDA_CODEC_ENTRY(0x10ec0892, "ALC892", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0899, "ALC898", patch_alc882), + HDA_CODEC_ENTRY(0x10ec0900, "ALC1150", patch_alc882), {} /* terminator */ }; - -MODULE_ALIAS("snd-hda-codec-id:10ec*"); +MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Realtek HD-audio codec"); static struct hda_codec_driver realtek_driver = { - .preset = snd_hda_preset_realtek, + .id = snd_hda_id_realtek, }; module_hda_codec_driver(realtek_driver); diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 5104bebb2286..ffda38c45509 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -289,41 +289,30 @@ static int patch_si3054(struct hda_codec *codec) /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_si3054[] = { - { .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 }, - { .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 }, - { .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 }, - { .id = 0x11c13055, .name = "Si3054", .patch = patch_si3054 }, - { .id = 0x11c13155, .name = "Si3054", .patch = patch_si3054 }, - { .id = 0x10573055, .name = "Si3054", .patch = patch_si3054 }, - { .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 }, - { .id = 0x10573155, .name = "Si3054", .patch = patch_si3054 }, +static const struct hda_device_id snd_hda_id_si3054[] = { + HDA_CODEC_ENTRY(0x163c3055, "Si3054", patch_si3054), + HDA_CODEC_ENTRY(0x163c3155, "Si3054", patch_si3054), + HDA_CODEC_ENTRY(0x11c13026, "Si3054", patch_si3054), + HDA_CODEC_ENTRY(0x11c13055, "Si3054", patch_si3054), + HDA_CODEC_ENTRY(0x11c13155, "Si3054", patch_si3054), + HDA_CODEC_ENTRY(0x10573055, "Si3054", patch_si3054), + HDA_CODEC_ENTRY(0x10573057, "Si3054", patch_si3054), + HDA_CODEC_ENTRY(0x10573155, "Si3054", patch_si3054), /* VIA HDA on Clevo m540 */ - { .id = 0x11063288, .name = "Si3054", .patch = patch_si3054 }, + HDA_CODEC_ENTRY(0x11063288, "Si3054", patch_si3054), /* Asus A8J Modem (SM56) */ - { .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 }, + HDA_CODEC_ENTRY(0x15433155, "Si3054", patch_si3054), /* LG LW20 modem */ - { .id = 0x18540018, .name = "Si3054", .patch = patch_si3054 }, + HDA_CODEC_ENTRY(0x18540018, "Si3054", patch_si3054), {} }; - -MODULE_ALIAS("snd-hda-codec-id:163c3055"); -MODULE_ALIAS("snd-hda-codec-id:163c3155"); -MODULE_ALIAS("snd-hda-codec-id:11c13026"); -MODULE_ALIAS("snd-hda-codec-id:11c13055"); -MODULE_ALIAS("snd-hda-codec-id:11c13155"); -MODULE_ALIAS("snd-hda-codec-id:10573055"); -MODULE_ALIAS("snd-hda-codec-id:10573057"); -MODULE_ALIAS("snd-hda-codec-id:10573155"); -MODULE_ALIAS("snd-hda-codec-id:11063288"); -MODULE_ALIAS("snd-hda-codec-id:15433155"); -MODULE_ALIAS("snd-hda-codec-id:18540018"); +MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_si3054); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Si3054 HD-audio modem codec"); static struct hda_codec_driver si3054_driver = { - .preset = snd_hda_preset_si3054, + .id = snd_hda_id_si3054, }; module_hda_codec_driver(si3054_driver); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index def5cc8dff02..08a0f6a35cca 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5012,121 +5012,119 @@ static int patch_stac9872(struct hda_codec *codec) /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_sigmatel[] = { - { .id = 0x83847690, .name = "STAC9200", .patch = patch_stac9200 }, - { .id = 0x83847882, .name = "STAC9220 A1", .patch = patch_stac922x }, - { .id = 0x83847680, .name = "STAC9221 A1", .patch = patch_stac922x }, - { .id = 0x83847880, .name = "STAC9220 A2", .patch = patch_stac922x }, - { .id = 0x83847681, .name = "STAC9220D/9223D A2", .patch = patch_stac922x }, - { .id = 0x83847682, .name = "STAC9221 A2", .patch = patch_stac922x }, - { .id = 0x83847683, .name = "STAC9221D A2", .patch = patch_stac922x }, - { .id = 0x83847618, .name = "STAC9227", .patch = patch_stac927x }, - { .id = 0x83847619, .name = "STAC9227", .patch = patch_stac927x }, - { .id = 0x83847616, .name = "STAC9228", .patch = patch_stac927x }, - { .id = 0x83847617, .name = "STAC9228", .patch = patch_stac927x }, - { .id = 0x83847614, .name = "STAC9229", .patch = patch_stac927x }, - { .id = 0x83847615, .name = "STAC9229", .patch = patch_stac927x }, - { .id = 0x83847620, .name = "STAC9274", .patch = patch_stac927x }, - { .id = 0x83847621, .name = "STAC9274D", .patch = patch_stac927x }, - { .id = 0x83847622, .name = "STAC9273X", .patch = patch_stac927x }, - { .id = 0x83847623, .name = "STAC9273D", .patch = patch_stac927x }, - { .id = 0x83847624, .name = "STAC9272X", .patch = patch_stac927x }, - { .id = 0x83847625, .name = "STAC9272D", .patch = patch_stac927x }, - { .id = 0x83847626, .name = "STAC9271X", .patch = patch_stac927x }, - { .id = 0x83847627, .name = "STAC9271D", .patch = patch_stac927x }, - { .id = 0x83847628, .name = "STAC9274X5NH", .patch = patch_stac927x }, - { .id = 0x83847629, .name = "STAC9274D5NH", .patch = patch_stac927x }, - { .id = 0x83847632, .name = "STAC9202", .patch = patch_stac925x }, - { .id = 0x83847633, .name = "STAC9202D", .patch = patch_stac925x }, - { .id = 0x83847634, .name = "STAC9250", .patch = patch_stac925x }, - { .id = 0x83847635, .name = "STAC9250D", .patch = patch_stac925x }, - { .id = 0x83847636, .name = "STAC9251", .patch = patch_stac925x }, - { .id = 0x83847637, .name = "STAC9250D", .patch = patch_stac925x }, - { .id = 0x83847645, .name = "92HD206X", .patch = patch_stac927x }, - { .id = 0x83847646, .name = "92HD206D", .patch = patch_stac927x }, - /* The following does not take into account .id=0x83847661 when subsys = - * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are - * currently not fully supported. - */ - { .id = 0x83847661, .name = "CXD9872RD/K", .patch = patch_stac9872 }, - { .id = 0x83847662, .name = "STAC9872AK", .patch = patch_stac9872 }, - { .id = 0x83847664, .name = "CXD9872AKD", .patch = patch_stac9872 }, - { .id = 0x83847698, .name = "STAC9205", .patch = patch_stac9205 }, - { .id = 0x838476a0, .name = "STAC9205", .patch = patch_stac9205 }, - { .id = 0x838476a1, .name = "STAC9205D", .patch = patch_stac9205 }, - { .id = 0x838476a2, .name = "STAC9204", .patch = patch_stac9205 }, - { .id = 0x838476a3, .name = "STAC9204D", .patch = patch_stac9205 }, - { .id = 0x838476a4, .name = "STAC9255", .patch = patch_stac9205 }, - { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, - { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, - { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, - { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, - { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76d4, .name = "92HD83C1C5", .patch = patch_stac92hd83xxx}, - { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76d1, .name = "92HD87B1/3", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76d9, .name = "92HD87B2/4", .patch = patch_stac92hd83xxx}, - { .id = 0x111d7666, .name = "92HD88B3", .patch = patch_stac92hd83xxx}, - { .id = 0x111d7667, .name = "92HD88B1", .patch = patch_stac92hd83xxx}, - { .id = 0x111d7668, .name = "92HD88B2", .patch = patch_stac92hd83xxx}, - { .id = 0x111d7669, .name = "92HD88B4", .patch = patch_stac92hd83xxx}, - { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, - { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d7695, .name = "92HD95", .patch = patch_stac92hd95 }, - { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, - { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, - { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx }, - { .id = 0x111d76b3, .name = "92HD71B7X", .patch = patch_stac92hd71bxx }, - { .id = 0x111d76b4, .name = "92HD71B6X", .patch = patch_stac92hd71bxx }, - { .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx }, - { .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, - { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, - { .id = 0x111d76c0, .name = "92HD89C3", .patch = patch_stac92hd73xx }, - { .id = 0x111d76c1, .name = "92HD89C2", .patch = patch_stac92hd73xx }, - { .id = 0x111d76c2, .name = "92HD89C1", .patch = patch_stac92hd73xx }, - { .id = 0x111d76c3, .name = "92HD89B3", .patch = patch_stac92hd73xx }, - { .id = 0x111d76c4, .name = "92HD89B2", .patch = patch_stac92hd73xx }, - { .id = 0x111d76c5, .name = "92HD89B1", .patch = patch_stac92hd73xx }, - { .id = 0x111d76c6, .name = "92HD89E3", .patch = patch_stac92hd73xx }, - { .id = 0x111d76c7, .name = "92HD89E2", .patch = patch_stac92hd73xx }, - { .id = 0x111d76c8, .name = "92HD89E1", .patch = patch_stac92hd73xx }, - { .id = 0x111d76c9, .name = "92HD89D3", .patch = patch_stac92hd73xx }, - { .id = 0x111d76ca, .name = "92HD89D2", .patch = patch_stac92hd73xx }, - { .id = 0x111d76cb, .name = "92HD89D1", .patch = patch_stac92hd73xx }, - { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx }, - { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx }, - { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx }, - { .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76e7, .name = "92HD90BXX", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76e8, .name = "92HD66B1X5", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76e9, .name = "92HD66B2X5", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76ea, .name = "92HD66B3X5", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76eb, .name = "92HD66C1X5", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76ec, .name = "92HD66C2X5", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76ed, .name = "92HD66C3X5", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76ee, .name = "92HD66B1X3", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76ef, .name = "92HD66B2X3", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76f0, .name = "92HD66B3X3", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76f1, .name = "92HD66C1X3", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76f2, .name = "92HD66C2X3", .patch = patch_stac92hd83xxx}, - { .id = 0x111d76f3, .name = "92HD66C3/65", .patch = patch_stac92hd83xxx}, +static const struct hda_device_id snd_hda_id_sigmatel[] = { + HDA_CODEC_ENTRY(0x83847690, "STAC9200", patch_stac9200), + HDA_CODEC_ENTRY(0x83847882, "STAC9220 A1", patch_stac922x), + HDA_CODEC_ENTRY(0x83847680, "STAC9221 A1", patch_stac922x), + HDA_CODEC_ENTRY(0x83847880, "STAC9220 A2", patch_stac922x), + HDA_CODEC_ENTRY(0x83847681, "STAC9220D/9223D A2", patch_stac922x), + HDA_CODEC_ENTRY(0x83847682, "STAC9221 A2", patch_stac922x), + HDA_CODEC_ENTRY(0x83847683, "STAC9221D A2", patch_stac922x), + HDA_CODEC_ENTRY(0x83847618, "STAC9227", patch_stac927x), + HDA_CODEC_ENTRY(0x83847619, "STAC9227", patch_stac927x), + HDA_CODEC_ENTRY(0x83847616, "STAC9228", patch_stac927x), + HDA_CODEC_ENTRY(0x83847617, "STAC9228", patch_stac927x), + HDA_CODEC_ENTRY(0x83847614, "STAC9229", patch_stac927x), + HDA_CODEC_ENTRY(0x83847615, "STAC9229", patch_stac927x), + HDA_CODEC_ENTRY(0x83847620, "STAC9274", patch_stac927x), + HDA_CODEC_ENTRY(0x83847621, "STAC9274D", patch_stac927x), + HDA_CODEC_ENTRY(0x83847622, "STAC9273X", patch_stac927x), + HDA_CODEC_ENTRY(0x83847623, "STAC9273D", patch_stac927x), + HDA_CODEC_ENTRY(0x83847624, "STAC9272X", patch_stac927x), + HDA_CODEC_ENTRY(0x83847625, "STAC9272D", patch_stac927x), + HDA_CODEC_ENTRY(0x83847626, "STAC9271X", patch_stac927x), + HDA_CODEC_ENTRY(0x83847627, "STAC9271D", patch_stac927x), + HDA_CODEC_ENTRY(0x83847628, "STAC9274X5NH", patch_stac927x), + HDA_CODEC_ENTRY(0x83847629, "STAC9274D5NH", patch_stac927x), + HDA_CODEC_ENTRY(0x83847632, "STAC9202", patch_stac925x), + HDA_CODEC_ENTRY(0x83847633, "STAC9202D", patch_stac925x), + HDA_CODEC_ENTRY(0x83847634, "STAC9250", patch_stac925x), + HDA_CODEC_ENTRY(0x83847635, "STAC9250D", patch_stac925x), + HDA_CODEC_ENTRY(0x83847636, "STAC9251", patch_stac925x), + HDA_CODEC_ENTRY(0x83847637, "STAC9250D", patch_stac925x), + HDA_CODEC_ENTRY(0x83847645, "92HD206X", patch_stac927x), + HDA_CODEC_ENTRY(0x83847646, "92HD206D", patch_stac927x), + /* The following does not take into account .id=0x83847661 when subsys = + * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are + * currently not fully supported. + */ + HDA_CODEC_ENTRY(0x83847661, "CXD9872RD/K", patch_stac9872), + HDA_CODEC_ENTRY(0x83847662, "STAC9872AK", patch_stac9872), + HDA_CODEC_ENTRY(0x83847664, "CXD9872AKD", patch_stac9872), + HDA_CODEC_ENTRY(0x83847698, "STAC9205", patch_stac9205), + HDA_CODEC_ENTRY(0x838476a0, "STAC9205", patch_stac9205), + HDA_CODEC_ENTRY(0x838476a1, "STAC9205D", patch_stac9205), + HDA_CODEC_ENTRY(0x838476a2, "STAC9204", patch_stac9205), + HDA_CODEC_ENTRY(0x838476a3, "STAC9204D", patch_stac9205), + HDA_CODEC_ENTRY(0x838476a4, "STAC9255", patch_stac9205), + HDA_CODEC_ENTRY(0x838476a5, "STAC9255D", patch_stac9205), + HDA_CODEC_ENTRY(0x838476a6, "STAC9254", patch_stac9205), + HDA_CODEC_ENTRY(0x838476a7, "STAC9254D", patch_stac9205), + HDA_CODEC_ENTRY(0x111d7603, "92HD75B3X5", patch_stac92hd71bxx), + HDA_CODEC_ENTRY(0x111d7604, "92HD83C1X5", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76d4, "92HD83C1C5", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d7605, "92HD81B1X5", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76d5, "92HD81B1C5", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76d1, "92HD87B1/3", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76d9, "92HD87B2/4", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d7666, "92HD88B3", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d7667, "92HD88B1", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d7668, "92HD88B2", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d7669, "92HD88B4", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d7608, "92HD75B2X5", patch_stac92hd71bxx), + HDA_CODEC_ENTRY(0x111d7674, "92HD73D1X5", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d7675, "92HD73C1X5", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d7676, "92HD73E1X5", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d7695, "92HD95", patch_stac92hd95), + HDA_CODEC_ENTRY(0x111d76b0, "92HD71B8X", patch_stac92hd71bxx), + HDA_CODEC_ENTRY(0x111d76b1, "92HD71B8X", patch_stac92hd71bxx), + HDA_CODEC_ENTRY(0x111d76b2, "92HD71B7X", patch_stac92hd71bxx), + HDA_CODEC_ENTRY(0x111d76b3, "92HD71B7X", patch_stac92hd71bxx), + HDA_CODEC_ENTRY(0x111d76b4, "92HD71B6X", patch_stac92hd71bxx), + HDA_CODEC_ENTRY(0x111d76b5, "92HD71B6X", patch_stac92hd71bxx), + HDA_CODEC_ENTRY(0x111d76b6, "92HD71B5X", patch_stac92hd71bxx), + HDA_CODEC_ENTRY(0x111d76b7, "92HD71B5X", patch_stac92hd71bxx), + HDA_CODEC_ENTRY(0x111d76c0, "92HD89C3", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76c1, "92HD89C2", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76c2, "92HD89C1", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76c3, "92HD89B3", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76c4, "92HD89B2", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76c5, "92HD89B1", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76c6, "92HD89E3", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76c7, "92HD89E2", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76c8, "92HD89E1", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76c9, "92HD89D3", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76ca, "92HD89D2", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76cb, "92HD89D1", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76cc, "92HD89F3", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76cd, "92HD89F2", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76ce, "92HD89F1", patch_stac92hd73xx), + HDA_CODEC_ENTRY(0x111d76df, "92HD93BXX", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76e0, "92HD91BXX", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76e3, "92HD98BXX", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76e5, "92HD99BXX", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76e7, "92HD90BXX", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76e8, "92HD66B1X5", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76e9, "92HD66B2X5", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76ea, "92HD66B3X5", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76eb, "92HD66C1X5", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76ec, "92HD66C2X5", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76ed, "92HD66C3X5", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76ee, "92HD66B1X3", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76ef, "92HD66B2X3", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76f0, "92HD66B3X3", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76f1, "92HD66C1X3", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76f2, "92HD66C2X3", patch_stac92hd83xxx), + HDA_CODEC_ENTRY(0x111d76f3, "92HD66C3/65", patch_stac92hd83xxx), {} /* terminator */ }; - -MODULE_ALIAS("snd-hda-codec-id:8384*"); -MODULE_ALIAS("snd-hda-codec-id:111d*"); +MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_sigmatel); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("IDT/Sigmatel HD-audio codec"); static struct hda_codec_driver sigmatel_driver = { - .preset = snd_hda_preset_sigmatel, + .id = snd_hda_id_sigmatel, }; module_hda_codec_driver(sigmatel_driver); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index da5366405eda..fc30d1e8aa76 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -785,21 +785,11 @@ static int patch_vt1708S(struct hda_codec *codec) override_mic_boost(codec, 0x1e, 0, 3, 40); /* correct names for VT1708BCE */ - if (get_codec_type(codec) == VT1708BCE) { - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup("VT1708BCE", GFP_KERNEL); - snprintf(codec->card->mixername, - sizeof(codec->card->mixername), - "%s %s", codec->core.vendor_name, codec->core.chip_name); - } + if (get_codec_type(codec) == VT1708BCE) + snd_hda_codec_set_name(codec, "VT1708BCE"); /* correct names for VT1705 */ - if (codec->core.vendor_id == 0x11064397) { - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup("VT1705", GFP_KERNEL); - snprintf(codec->card->mixername, - sizeof(codec->card->mixername), - "%s %s", codec->core.vendor_name, codec->core.chip_name); - } + if (codec->core.vendor_id == 0x11064397) + snd_hda_codec_set_name(codec, "VT1705"); /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); @@ -1210,109 +1200,64 @@ static int patch_vt3476(struct hda_codec *codec) /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_via[] = { - { .id = 0x11061708, .name = "VT1708", .patch = patch_vt1708}, - { .id = 0x11061709, .name = "VT1708", .patch = patch_vt1708}, - { .id = 0x1106170a, .name = "VT1708", .patch = patch_vt1708}, - { .id = 0x1106170b, .name = "VT1708", .patch = patch_vt1708}, - { .id = 0x1106e710, .name = "VT1709 10-Ch", - .patch = patch_vt1709}, - { .id = 0x1106e711, .name = "VT1709 10-Ch", - .patch = patch_vt1709}, - { .id = 0x1106e712, .name = "VT1709 10-Ch", - .patch = patch_vt1709}, - { .id = 0x1106e713, .name = "VT1709 10-Ch", - .patch = patch_vt1709}, - { .id = 0x1106e714, .name = "VT1709 6-Ch", - .patch = patch_vt1709}, - { .id = 0x1106e715, .name = "VT1709 6-Ch", - .patch = patch_vt1709}, - { .id = 0x1106e716, .name = "VT1709 6-Ch", - .patch = patch_vt1709}, - { .id = 0x1106e717, .name = "VT1709 6-Ch", - .patch = patch_vt1709}, - { .id = 0x1106e720, .name = "VT1708B 8-Ch", - .patch = patch_vt1708B}, - { .id = 0x1106e721, .name = "VT1708B 8-Ch", - .patch = patch_vt1708B}, - { .id = 0x1106e722, .name = "VT1708B 8-Ch", - .patch = patch_vt1708B}, - { .id = 0x1106e723, .name = "VT1708B 8-Ch", - .patch = patch_vt1708B}, - { .id = 0x1106e724, .name = "VT1708B 4-Ch", - .patch = patch_vt1708B}, - { .id = 0x1106e725, .name = "VT1708B 4-Ch", - .patch = patch_vt1708B}, - { .id = 0x1106e726, .name = "VT1708B 4-Ch", - .patch = patch_vt1708B}, - { .id = 0x1106e727, .name = "VT1708B 4-Ch", - .patch = patch_vt1708B}, - { .id = 0x11060397, .name = "VT1708S", - .patch = patch_vt1708S}, - { .id = 0x11061397, .name = "VT1708S", - .patch = patch_vt1708S}, - { .id = 0x11062397, .name = "VT1708S", - .patch = patch_vt1708S}, - { .id = 0x11063397, .name = "VT1708S", - .patch = patch_vt1708S}, - { .id = 0x11064397, .name = "VT1705", - .patch = patch_vt1708S}, - { .id = 0x11065397, .name = "VT1708S", - .patch = patch_vt1708S}, - { .id = 0x11066397, .name = "VT1708S", - .patch = patch_vt1708S}, - { .id = 0x11067397, .name = "VT1708S", - .patch = patch_vt1708S}, - { .id = 0x11060398, .name = "VT1702", - .patch = patch_vt1702}, - { .id = 0x11061398, .name = "VT1702", - .patch = patch_vt1702}, - { .id = 0x11062398, .name = "VT1702", - .patch = patch_vt1702}, - { .id = 0x11063398, .name = "VT1702", - .patch = patch_vt1702}, - { .id = 0x11064398, .name = "VT1702", - .patch = patch_vt1702}, - { .id = 0x11065398, .name = "VT1702", - .patch = patch_vt1702}, - { .id = 0x11066398, .name = "VT1702", - .patch = patch_vt1702}, - { .id = 0x11067398, .name = "VT1702", - .patch = patch_vt1702}, - { .id = 0x11060428, .name = "VT1718S", - .patch = patch_vt1718S}, - { .id = 0x11064428, .name = "VT1718S", - .patch = patch_vt1718S}, - { .id = 0x11060441, .name = "VT2020", - .patch = patch_vt1718S}, - { .id = 0x11064441, .name = "VT1828S", - .patch = patch_vt1718S}, - { .id = 0x11060433, .name = "VT1716S", - .patch = patch_vt1716S}, - { .id = 0x1106a721, .name = "VT1716S", - .patch = patch_vt1716S}, - { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P}, - { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P}, - { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812}, - { .id = 0x11060440, .name = "VT1818S", - .patch = patch_vt1708S}, - { .id = 0x11060446, .name = "VT1802", - .patch = patch_vt2002P}, - { .id = 0x11068446, .name = "VT1802", - .patch = patch_vt2002P}, - { .id = 0x11064760, .name = "VT1705CF", - .patch = patch_vt3476}, - { .id = 0x11064761, .name = "VT1708SCE", - .patch = patch_vt3476}, - { .id = 0x11064762, .name = "VT1808", - .patch = patch_vt3476}, +static const struct hda_device_id snd_hda_id_via[] = { + HDA_CODEC_ENTRY(0x11061708, "VT1708", patch_vt1708), + HDA_CODEC_ENTRY(0x11061709, "VT1708", patch_vt1708), + HDA_CODEC_ENTRY(0x1106170a, "VT1708", patch_vt1708), + HDA_CODEC_ENTRY(0x1106170b, "VT1708", patch_vt1708), + HDA_CODEC_ENTRY(0x1106e710, "VT1709 10-Ch", patch_vt1709), + HDA_CODEC_ENTRY(0x1106e711, "VT1709 10-Ch", patch_vt1709), + HDA_CODEC_ENTRY(0x1106e712, "VT1709 10-Ch", patch_vt1709), + HDA_CODEC_ENTRY(0x1106e713, "VT1709 10-Ch", patch_vt1709), + HDA_CODEC_ENTRY(0x1106e714, "VT1709 6-Ch", patch_vt1709), + HDA_CODEC_ENTRY(0x1106e715, "VT1709 6-Ch", patch_vt1709), + HDA_CODEC_ENTRY(0x1106e716, "VT1709 6-Ch", patch_vt1709), + HDA_CODEC_ENTRY(0x1106e717, "VT1709 6-Ch", patch_vt1709), + HDA_CODEC_ENTRY(0x1106e720, "VT1708B 8-Ch", patch_vt1708B), + HDA_CODEC_ENTRY(0x1106e721, "VT1708B 8-Ch", patch_vt1708B), + HDA_CODEC_ENTRY(0x1106e722, "VT1708B 8-Ch", patch_vt1708B), + HDA_CODEC_ENTRY(0x1106e723, "VT1708B 8-Ch", patch_vt1708B), + HDA_CODEC_ENTRY(0x1106e724, "VT1708B 4-Ch", patch_vt1708B), + HDA_CODEC_ENTRY(0x1106e725, "VT1708B 4-Ch", patch_vt1708B), + HDA_CODEC_ENTRY(0x1106e726, "VT1708B 4-Ch", patch_vt1708B), + HDA_CODEC_ENTRY(0x1106e727, "VT1708B 4-Ch", patch_vt1708B), + HDA_CODEC_ENTRY(0x11060397, "VT1708S", patch_vt1708S), + HDA_CODEC_ENTRY(0x11061397, "VT1708S", patch_vt1708S), + HDA_CODEC_ENTRY(0x11062397, "VT1708S", patch_vt1708S), + HDA_CODEC_ENTRY(0x11063397, "VT1708S", patch_vt1708S), + HDA_CODEC_ENTRY(0x11064397, "VT1705", patch_vt1708S), + HDA_CODEC_ENTRY(0x11065397, "VT1708S", patch_vt1708S), + HDA_CODEC_ENTRY(0x11066397, "VT1708S", patch_vt1708S), + HDA_CODEC_ENTRY(0x11067397, "VT1708S", patch_vt1708S), + HDA_CODEC_ENTRY(0x11060398, "VT1702", patch_vt1702), + HDA_CODEC_ENTRY(0x11061398, "VT1702", patch_vt1702), + HDA_CODEC_ENTRY(0x11062398, "VT1702", patch_vt1702), + HDA_CODEC_ENTRY(0x11063398, "VT1702", patch_vt1702), + HDA_CODEC_ENTRY(0x11064398, "VT1702", patch_vt1702), + HDA_CODEC_ENTRY(0x11065398, "VT1702", patch_vt1702), + HDA_CODEC_ENTRY(0x11066398, "VT1702", patch_vt1702), + HDA_CODEC_ENTRY(0x11067398, "VT1702", patch_vt1702), + HDA_CODEC_ENTRY(0x11060428, "VT1718S", patch_vt1718S), + HDA_CODEC_ENTRY(0x11064428, "VT1718S", patch_vt1718S), + HDA_CODEC_ENTRY(0x11060441, "VT2020", patch_vt1718S), + HDA_CODEC_ENTRY(0x11064441, "VT1828S", patch_vt1718S), + HDA_CODEC_ENTRY(0x11060433, "VT1716S", patch_vt1716S), + HDA_CODEC_ENTRY(0x1106a721, "VT1716S", patch_vt1716S), + HDA_CODEC_ENTRY(0x11060438, "VT2002P", patch_vt2002P), + HDA_CODEC_ENTRY(0x11064438, "VT2002P", patch_vt2002P), + HDA_CODEC_ENTRY(0x11060448, "VT1812", patch_vt1812), + HDA_CODEC_ENTRY(0x11060440, "VT1818S", patch_vt1708S), + HDA_CODEC_ENTRY(0x11060446, "VT1802", patch_vt2002P), + HDA_CODEC_ENTRY(0x11068446, "VT1802", patch_vt2002P), + HDA_CODEC_ENTRY(0x11064760, "VT1705CF", patch_vt3476), + HDA_CODEC_ENTRY(0x11064761, "VT1708SCE", patch_vt3476), + HDA_CODEC_ENTRY(0x11064762, "VT1808", patch_vt3476), {} /* terminator */ }; - -MODULE_ALIAS("snd-hda-codec-id:1106*"); +MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_via); static struct hda_codec_driver via_driver = { - .preset = snd_hda_preset_via, + .id = snd_hda_id_via, }; MODULE_LICENSE("GPL"); diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 7acbc21d642a..9e1ad119a3ce 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -1394,7 +1394,9 @@ static int snd_korg1212_playback_open(struct snd_pcm_substream *substream) spin_unlock_irqrestore(&korg1212->lock, flags); - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, kPlayBufferFrames, kPlayBufferFrames); + snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + kPlayBufferFrames); + return 0; } @@ -1422,8 +1424,8 @@ static int snd_korg1212_capture_open(struct snd_pcm_substream *substream) spin_unlock_irqrestore(&korg1212->lock, flags); - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - kPlayBufferFrames, kPlayBufferFrames); + snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + kPlayBufferFrames); return 0; } diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index cba89beb2b38..8b8e2e54fba3 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -234,8 +234,8 @@ static int lx_pcm_open(struct snd_pcm_substream *substream) /* the clock rate cannot be changed */ board_rate = chip->board_sample_rate; - err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, - board_rate, board_rate); + err = snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_RATE, + board_rate); if (err < 0) { dev_warn(chip->card->dev, "could not constrain periods\n"); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 72e89cedc52d..17ae92613de4 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -1929,15 +1929,32 @@ snd_m3_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) return; snd_m3_outw(chip, val, CODEC_DATA); snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND); + /* + * Workaround for buggy ES1988 integrated AC'97 codec. It remains silent + * until the MASTER volume or mute is touched (alsactl restore does not + * work). + */ + if (ac97->id == 0x45838308 && reg == AC97_MASTER) { + snd_m3_ac97_wait(chip); + snd_m3_outw(chip, val, CODEC_DATA); + snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND); + } } -static void snd_m3_remote_codec_config(int io, int isremote) +static void snd_m3_remote_codec_config(struct snd_m3 *chip, int isremote) { + int io = chip->iobase; + u16 tmp; + isremote = isremote ? 1 : 0; - outw((inw(io + RING_BUS_CTRL_B) & ~SECOND_CODEC_ID_MASK) | isremote, - io + RING_BUS_CTRL_B); + tmp = inw(io + RING_BUS_CTRL_B) & ~SECOND_CODEC_ID_MASK; + /* enable dock on Dell Latitude C810 */ + if (chip->pci->subsystem_vendor == 0x1028 && + chip->pci->subsystem_device == 0x00e5) + tmp |= M3I_DOCK_ENABLE; + outw(tmp | isremote, io + RING_BUS_CTRL_B); outw((inw(io + SDO_OUT_DEST_CTRL) & ~COMMAND_ADDR_OUT) | isremote, io + SDO_OUT_DEST_CTRL); outw((inw(io + SDO_IN_DEST_CTRL) & ~STATUS_ADDR_IN) | isremote, @@ -1989,7 +2006,7 @@ static void snd_m3_ac97_reset(struct snd_m3 *chip) if (!chip->irda_workaround) dir |= 0x10; /* assuming pci bus master? */ - snd_m3_remote_codec_config(io, 0); + snd_m3_remote_codec_config(chip, 0); outw(IO_SRAM_ENABLE, io + RING_BUS_CTRL_A); udelay(20); diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 23d7f5d30c41..cd94ac548ba3 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -831,9 +831,9 @@ static struct snd_pcm_hw_constraint_list hw_constraints_period_bytes = { static void snd_rme32_set_buffer_constraint(struct rme32 *rme32, struct snd_pcm_runtime *runtime) { if (! rme32->fullduplex_mode) { - snd_pcm_hw_constraint_minmax(runtime, + snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - RME32_BUFFER_SIZE, RME32_BUFFER_SIZE); + RME32_BUFFER_SIZE); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes); diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 2306ccf7281e..714df906249e 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -1152,13 +1152,13 @@ rme96_set_buffer_size_constraint(struct rme96 *rme96, { unsigned int size; - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - RME96_BUFFER_SIZE, RME96_BUFFER_SIZE); + snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + RME96_BUFFER_SIZE); if ((size = rme96->playback_periodsize) != 0 || (size = rme96->capture_periodsize) != 0) - snd_pcm_hw_constraint_minmax(runtime, + snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, - size, size); + size); else snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 9bba275b4c9b..2875b4f6d8c9 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5112,6 +5112,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) dev_err(hdsp->card->dev, "too short firmware size %d (expected %d)\n", (int)fw->size, HDSP_FIRMWARE_SIZE); + release_firmware(fw); return -EINVAL; } diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index cb666c73712d..8bc8016c173d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6080,18 +6080,17 @@ static int snd_hdspm_open(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32, 4096); /* RayDAT & AIO have a fixed buffer of 16384 samples per channel */ - snd_pcm_hw_constraint_minmax(runtime, + snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - 16384, 16384); + 16384); break; default: snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 64, 8192); - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIODS, - 2, 2); + snd_pcm_hw_constraint_single(runtime, + SNDRV_PCM_HW_PARAM_PERIODS, 2); break; } diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 225bfda414e9..7ff7d88e46dd 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -9,7 +9,6 @@ menuconfig SND_SOC select SND_JACK if INPUT=y || INPUT=SND select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER - select SND_COMPRESS_OFFLOAD ---help--- If you want ASoC support, you should say Y here and also to the @@ -30,6 +29,10 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_COMPRESS + bool + select SND_COMPRESS_OFFLOAD + config SND_SOC_TOPOLOGY bool @@ -58,6 +61,7 @@ source "sound/soc/sh/Kconfig" source "sound/soc/sirf/Kconfig" source "sound/soc/spear/Kconfig" source "sound/soc/sti/Kconfig" +source "sound/soc/sunxi/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 134aca150a50..8eb06db32fa0 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,6 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o ifneq ($(CONFIG_SND_SOC_TOPOLOGY),) snd-soc-core-objs += soc-topology.o @@ -40,6 +41,7 @@ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += sirf/ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += sti/ +obj-$(CONFIG_SND_SOC) += sunxi/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 1489cd461aec..2d30464b81ce 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -59,4 +59,13 @@ config SND_AT91_SOC_SAM9X5_WM8731 help Say Y if you want to add support for audio SoC on an at91sam9x5 based board that is using WM8731 codec. + +config SND_ATMEL_SOC_CLASSD + tristate "Atmel ASoC driver for boards using CLASSD" + depends on ARCH_AT91 || COMPILE_TEST + select SND_ATMEL_SOC_DMA + select REGMAP_MMIO + help + Say Y if you want to add support for Atmel ASoC driver for boards using + CLASSD. endif diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index b327e5cc8de3..f6f7db428216 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -11,7 +11,9 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o snd-atmel-soc-wm8904-objs := atmel_wm8904.o snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o +snd-atmel-soc-classd-objs := atmel-classd.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o +obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c new file mode 100644 index 000000000000..8276675730ef --- /dev/null +++ b/sound/soc/atmel/atmel-classd.c @@ -0,0 +1,679 @@ +/* Atmel ALSA SoC Audio Class D Amplifier (CLASSD) driver + * + * Copyright (C) 2015 Atmel + * + * Author: Songjun Wu <songjun.wu@atmel.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 or later + * as published by the Free Software Foundation. + */ + +#include <linux/of.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include "atmel-classd.h" + +struct atmel_classd_pdata { + bool non_overlap_enable; + int non_overlap_time; + int pwm_type; + const char *card_name; +}; + +struct atmel_classd { + dma_addr_t phy_base; + struct regmap *regmap; + struct clk *pclk; + struct clk *gclk; + struct clk *aclk; + int irq; + const struct atmel_classd_pdata *pdata; +}; + +#ifdef CONFIG_OF +static const struct of_device_id atmel_classd_of_match[] = { + { + .compatible = "atmel,sama5d2-classd", + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(of, atmel_classd_of_match); + +static struct atmel_classd_pdata *atmel_classd_dt_init(struct device *dev) +{ + struct device_node *np = dev->of_node; + struct atmel_classd_pdata *pdata; + const char *pwm_type; + int ret; + + if (!np) { + dev_err(dev, "device node not found\n"); + return ERR_PTR(-EINVAL); + } + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return ERR_PTR(-ENOMEM); + + ret = of_property_read_string(np, "atmel,pwm-type", &pwm_type); + if ((ret == 0) && (strcmp(pwm_type, "diff") == 0)) + pdata->pwm_type = CLASSD_MR_PWMTYP_DIFF; + else + pdata->pwm_type = CLASSD_MR_PWMTYP_SINGLE; + + ret = of_property_read_u32(np, + "atmel,non-overlap-time", &pdata->non_overlap_time); + if (ret) + pdata->non_overlap_enable = false; + else + pdata->non_overlap_enable = true; + + ret = of_property_read_string(np, "atmel,model", &pdata->card_name); + if (ret) + pdata->card_name = "CLASSD"; + + return pdata; +} +#else +static inline struct atmel_classd_pdata * +atmel_classd_dt_init(struct device *dev) +{ + return ERR_PTR(-EINVAL); +} +#endif + +#define ATMEL_CLASSD_RATES (SNDRV_PCM_RATE_8000 \ + | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 \ + | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 \ + | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 \ + | SNDRV_PCM_RATE_96000) + +static const struct snd_pcm_hardware atmel_classd_hw = { + .info = SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_RESUME + | SNDRV_PCM_INFO_PAUSE, + .formats = (SNDRV_PCM_FMTBIT_S16_LE), + .rates = ATMEL_CLASSD_RATES, + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024, + .period_bytes_min = 256, + .period_bytes_max = 32 * 1024, + .periods_min = 2, + .periods_max = 256, +}; + +#define ATMEL_CLASSD_PREALLOC_BUF_SIZE (64 * 1024) + +/* cpu dai component */ +static int atmel_classd_cpu_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + + regmap_write(dd->regmap, CLASSD_THR, 0x0); + + return clk_prepare_enable(dd->pclk); +} + +static void atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + + clk_disable_unprepare(dd->pclk); +} + +static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { + .startup = atmel_classd_cpu_dai_startup, + .shutdown = atmel_classd_cpu_dai_shutdown, +}; + +static struct snd_soc_dai_driver atmel_classd_cpu_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = ATMEL_CLASSD_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &atmel_classd_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { + .name = "atmel-classd", +}; + +/* platform */ +static int +atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + + if (params_physical_width(params) != 16) { + dev_err(rtd->platform->dev, + "only supports 16-bit audio data\n"); + return -EINVAL; + } + + slave_config->direction = DMA_MEM_TO_DEV; + slave_config->dst_addr = dd->phy_base + CLASSD_THR; + slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + slave_config->dst_maxburst = 1; + slave_config->src_maxburst = 1; + slave_config->device_fc = false; + + return 0; +} + +static const struct snd_dmaengine_pcm_config +atmel_classd_dmaengine_pcm_config = { + .prepare_slave_config = atmel_classd_platform_configure_dma, + .pcm_hardware = &atmel_classd_hw, + .prealloc_buffer_size = ATMEL_CLASSD_PREALLOC_BUF_SIZE, +}; + +/* codec */ +static const char * const mono_mode_text[] = { + "mix", "sat", "left", "right" +}; + +static SOC_ENUM_SINGLE_DECL(classd_mono_mode_enum, + CLASSD_INTPMR, CLASSD_INTPMR_MONO_MODE_SHIFT, + mono_mode_text); + +static const char * const eqcfg_text[] = { + "Treble-12dB", "Treble-6dB", + "Medium-8dB", "Medium-3dB", + "Bass-12dB", "Bass-6dB", + "0 dB", + "Bass+6dB", "Bass+12dB", + "Medium+3dB", "Medium+8dB", + "Treble+6dB", "Treble+12dB", +}; + +static const unsigned int eqcfg_value[] = { + CLASSD_INTPMR_EQCFG_T_CUT_12, CLASSD_INTPMR_EQCFG_T_CUT_6, + CLASSD_INTPMR_EQCFG_M_CUT_8, CLASSD_INTPMR_EQCFG_M_CUT_3, + CLASSD_INTPMR_EQCFG_B_CUT_12, CLASSD_INTPMR_EQCFG_B_CUT_6, + CLASSD_INTPMR_EQCFG_FLAT, + CLASSD_INTPMR_EQCFG_B_BOOST_6, CLASSD_INTPMR_EQCFG_B_BOOST_12, + CLASSD_INTPMR_EQCFG_M_BOOST_3, CLASSD_INTPMR_EQCFG_M_BOOST_8, + CLASSD_INTPMR_EQCFG_T_BOOST_6, CLASSD_INTPMR_EQCFG_T_BOOST_12, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(classd_eqcfg_enum, + CLASSD_INTPMR, CLASSD_INTPMR_EQCFG_SHIFT, 0xf, + eqcfg_text, eqcfg_value); + +static const DECLARE_TLV_DB_SCALE(classd_digital_tlv, -7800, 100, 1); + +static const struct snd_kcontrol_new atmel_classd_snd_controls[] = { +SOC_DOUBLE_TLV("Playback Volume", CLASSD_INTPMR, + CLASSD_INTPMR_ATTL_SHIFT, CLASSD_INTPMR_ATTR_SHIFT, + 78, 1, classd_digital_tlv), + +SOC_SINGLE("Deemphasis Switch", CLASSD_INTPMR, + CLASSD_INTPMR_DEEMP_SHIFT, 1, 0), + +SOC_SINGLE("Mono Switch", CLASSD_INTPMR, CLASSD_INTPMR_MONO_SHIFT, 1, 0), + +SOC_SINGLE("Swap Switch", CLASSD_INTPMR, CLASSD_INTPMR_SWAP_SHIFT, 1, 0), + +SOC_ENUM("Mono Mode", classd_mono_mode_enum), + +SOC_ENUM("EQ", classd_eqcfg_enum), +}; + +static const char * const pwm_type[] = { + "Single ended", "Differential" +}; + +static int atmel_classd_codec_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_card *card = snd_soc_codec_get_drvdata(codec); + struct atmel_classd *dd = snd_soc_card_get_drvdata(card); + const struct atmel_classd_pdata *pdata = dd->pdata; + u32 mask, val; + + mask = CLASSD_MR_PWMTYP_MASK; + val = pdata->pwm_type << CLASSD_MR_PWMTYP_SHIFT; + + mask |= CLASSD_MR_NON_OVERLAP_MASK; + if (pdata->non_overlap_enable) { + val |= (CLASSD_MR_NON_OVERLAP_EN + << CLASSD_MR_NON_OVERLAP_SHIFT); + + mask |= CLASSD_MR_NOVR_VAL_MASK; + switch (pdata->non_overlap_time) { + case 5: + val |= (CLASSD_MR_NOVR_VAL_5NS + << CLASSD_MR_NOVR_VAL_SHIFT); + break; + case 10: + val |= (CLASSD_MR_NOVR_VAL_10NS + << CLASSD_MR_NOVR_VAL_SHIFT); + break; + case 15: + val |= (CLASSD_MR_NOVR_VAL_15NS + << CLASSD_MR_NOVR_VAL_SHIFT); + break; + case 20: + val |= (CLASSD_MR_NOVR_VAL_20NS + << CLASSD_MR_NOVR_VAL_SHIFT); + break; + default: + val |= (CLASSD_MR_NOVR_VAL_10NS + << CLASSD_MR_NOVR_VAL_SHIFT); + dev_warn(codec->dev, + "non-overlapping value %d is invalid, the default value 10 is specified\n", + pdata->non_overlap_time); + break; + } + } + + snd_soc_update_bits(codec, CLASSD_MR, mask, val); + + dev_info(codec->dev, + "PWM modulation type is %s, non-overlapping is %s\n", + pwm_type[pdata->pwm_type], + pdata->non_overlap_enable?"enabled":"disabled"); + + return 0; +} + +static struct regmap *atmel_classd_codec_get_remap(struct device *dev) +{ + return dev_get_regmap(dev, NULL); +} + +static struct snd_soc_codec_driver soc_codec_dev_classd = { + .probe = atmel_classd_codec_probe, + .controls = atmel_classd_snd_controls, + .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), + .get_regmap = atmel_classd_codec_get_remap, +}; + +/* codec dai component */ +static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + int ret; + + ret = clk_prepare_enable(dd->aclk); + if (ret) + return ret; + + return clk_prepare_enable(dd->gclk); +} + +static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai, + int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u32 mask, val; + + mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK; + + if (mute) + val = mask; + else + val = 0; + + snd_soc_update_bits(codec, CLASSD_MR, mask, val); + + return 0; +} + +#define CLASSD_ACLK_RATE_11M2896_MPY_8 (112896 * 100 * 8) +#define CLASSD_ACLK_RATE_12M288_MPY_8 (12228 * 1000 * 8) + +static struct { + int rate; + int sample_rate; + int dsp_clk; + unsigned long aclk_rate; +} const sample_rates[] = { + { 8000, CLASSD_INTPMR_FRAME_8K, + CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 }, + { 16000, CLASSD_INTPMR_FRAME_16K, + CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 }, + { 32000, CLASSD_INTPMR_FRAME_32K, + CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 }, + { 48000, CLASSD_INTPMR_FRAME_48K, + CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 }, + { 96000, CLASSD_INTPMR_FRAME_96K, + CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 }, + { 22050, CLASSD_INTPMR_FRAME_22K, + CLASSD_INTPMR_DSP_CLK_FREQ_11M2896, CLASSD_ACLK_RATE_11M2896_MPY_8 }, + { 44100, CLASSD_INTPMR_FRAME_44K, + CLASSD_INTPMR_DSP_CLK_FREQ_11M2896, CLASSD_ACLK_RATE_11M2896_MPY_8 }, + { 88200, CLASSD_INTPMR_FRAME_88K, + CLASSD_INTPMR_DSP_CLK_FREQ_11M2896, CLASSD_ACLK_RATE_11M2896_MPY_8 }, +}; + +static int +atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_codec *codec = codec_dai->codec; + int fs; + int i, best, best_val, cur_val, ret; + u32 mask, val; + + fs = params_rate(params); + + best = 0; + best_val = abs(fs - sample_rates[0].rate); + for (i = 1; i < ARRAY_SIZE(sample_rates); i++) { + /* Closest match */ + cur_val = abs(fs - sample_rates[i].rate); + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + + dev_dbg(codec->dev, + "Selected SAMPLE_RATE of %dHz, ACLK_RATE of %ldHz\n", + sample_rates[best].rate, sample_rates[best].aclk_rate); + + clk_disable_unprepare(dd->gclk); + clk_disable_unprepare(dd->aclk); + + ret = clk_set_rate(dd->aclk, sample_rates[best].aclk_rate); + if (ret) + return ret; + + mask = CLASSD_INTPMR_DSP_CLK_FREQ_MASK | CLASSD_INTPMR_FRAME_MASK; + val = (sample_rates[best].dsp_clk << CLASSD_INTPMR_DSP_CLK_FREQ_SHIFT) + | (sample_rates[best].sample_rate << CLASSD_INTPMR_FRAME_SHIFT); + + snd_soc_update_bits(codec, CLASSD_INTPMR, mask, val); + + ret = clk_prepare_enable(dd->aclk); + if (ret) + return ret; + + return clk_prepare_enable(dd->gclk); +} + +static void +atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + + clk_disable_unprepare(dd->gclk); + clk_disable_unprepare(dd->aclk); +} + +static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + snd_soc_update_bits(codec, CLASSD_MR, + CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK, + (CLASSD_MR_LEN_DIS << CLASSD_MR_LEN_SHIFT) + |(CLASSD_MR_REN_DIS << CLASSD_MR_REN_SHIFT)); + + return 0; +} + +static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u32 mask, val; + + mask = CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = mask; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = (CLASSD_MR_LEN_DIS << CLASSD_MR_LEN_SHIFT) + | (CLASSD_MR_REN_DIS << CLASSD_MR_REN_SHIFT); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, CLASSD_MR, mask, val); + + return 0; +} + +static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = { + .digital_mute = atmel_classd_codec_dai_digital_mute, + .startup = atmel_classd_codec_dai_startup, + .shutdown = atmel_classd_codec_dai_shutdown, + .hw_params = atmel_classd_codec_dai_hw_params, + .prepare = atmel_classd_codec_dai_prepare, + .trigger = atmel_classd_codec_dai_trigger, +}; + +#define ATMEL_CLASSD_CODEC_DAI_NAME "atmel-classd-hifi" + +static struct snd_soc_dai_driver atmel_classd_codec_dai = { + .name = ATMEL_CLASSD_CODEC_DAI_NAME, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ATMEL_CLASSD_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &atmel_classd_codec_dai_ops, +}; + +/* ASoC sound card */ +static int atmel_classd_asoc_card_init(struct device *dev, + struct snd_soc_card *card) +{ + struct snd_soc_dai_link *dai_link; + struct atmel_classd *dd = snd_soc_card_get_drvdata(card); + + dai_link = devm_kzalloc(dev, sizeof(*dai_link), GFP_KERNEL); + if (!dai_link) + return -ENOMEM; + + dai_link->name = "CLASSD"; + dai_link->stream_name = "CLASSD PCM"; + dai_link->codec_dai_name = ATMEL_CLASSD_CODEC_DAI_NAME; + dai_link->cpu_dai_name = dev_name(dev); + dai_link->codec_name = dev_name(dev); + dai_link->platform_name = dev_name(dev); + + card->dai_link = dai_link; + card->num_links = 1; + card->name = dd->pdata->card_name; + card->dev = dev; + + return 0; +}; + +/* regmap configuration */ +static const struct reg_default atmel_classd_reg_defaults[] = { + { CLASSD_INTPMR, 0x00301212 }, +}; + +#define ATMEL_CLASSD_REG_MAX 0xE4 +static const struct regmap_config atmel_classd_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = ATMEL_CLASSD_REG_MAX, + + .cache_type = REGCACHE_FLAT, + .reg_defaults = atmel_classd_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(atmel_classd_reg_defaults), +}; + +static int atmel_classd_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct atmel_classd *dd; + struct resource *res; + void __iomem *io_base; + const struct atmel_classd_pdata *pdata; + struct snd_soc_card *card; + int ret; + + pdata = dev_get_platdata(dev); + if (!pdata) { + pdata = atmel_classd_dt_init(dev); + if (IS_ERR(pdata)) + return PTR_ERR(pdata); + } + + dd = devm_kzalloc(dev, sizeof(*dd), GFP_KERNEL); + if (!dd) + return -ENOMEM; + + dd->pdata = pdata; + + dd->irq = platform_get_irq(pdev, 0); + if (dd->irq < 0) { + ret = dd->irq; + dev_err(dev, "failed to could not get irq: %d\n", ret); + return ret; + } + + dd->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(dd->pclk)) { + ret = PTR_ERR(dd->pclk); + dev_err(dev, "failed to get peripheral clock: %d\n", ret); + return ret; + } + + dd->gclk = devm_clk_get(dev, "gclk"); + if (IS_ERR(dd->gclk)) { + ret = PTR_ERR(dd->gclk); + dev_err(dev, "failed to get GCK clock: %d\n", ret); + return ret; + } + + dd->aclk = devm_clk_get(dev, "aclk"); + if (IS_ERR(dd->aclk)) { + ret = PTR_ERR(dd->aclk); + dev_err(dev, "failed to get audio clock: %d\n", ret); + return ret; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(dev, "no memory resource\n"); + return -ENXIO; + } + + io_base = devm_ioremap_resource(dev, res); + if (IS_ERR(io_base)) { + ret = PTR_ERR(io_base); + dev_err(dev, "failed to remap register memory: %d\n", ret); + return ret; + } + + dd->phy_base = res->start; + + dd->regmap = devm_regmap_init_mmio(dev, io_base, + &atmel_classd_regmap_config); + if (IS_ERR(dd->regmap)) { + ret = PTR_ERR(dd->regmap); + dev_err(dev, "failed to init register map: %d\n", ret); + return ret; + } + + ret = devm_snd_soc_register_component(dev, + &atmel_classd_cpu_dai_component, + &atmel_classd_cpu_dai, 1); + if (ret) { + dev_err(dev, "could not register CPU DAI: %d\n", ret); + return ret; + } + + ret = devm_snd_dmaengine_pcm_register(dev, + &atmel_classd_dmaengine_pcm_config, + 0); + if (ret) { + dev_err(dev, "could not register platform: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(dev, &soc_codec_dev_classd, + &atmel_classd_codec_dai, 1); + if (ret) { + dev_err(dev, "could not register codec: %d\n", ret); + return ret; + } + + /* register sound card */ + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + snd_soc_card_set_drvdata(card, dd); + platform_set_drvdata(pdev, card); + + ret = atmel_classd_asoc_card_init(dev, card); + if (ret) { + dev_err(dev, "failed to init sound card\n"); + return ret; + } + + ret = devm_snd_soc_register_card(dev, card); + if (ret) { + dev_err(dev, "failed to register sound card: %d\n", ret); + return ret; + } + + return 0; +} + +static int atmel_classd_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver atmel_classd_driver = { + .driver = { + .name = "atmel-classd", + .of_match_table = of_match_ptr(atmel_classd_of_match), + .pm = &snd_soc_pm_ops, + }, + .probe = atmel_classd_probe, + .remove = atmel_classd_remove, +}; +module_platform_driver(atmel_classd_driver); + +MODULE_DESCRIPTION("Atmel ClassD driver under ALSA SoC architecture"); +MODULE_AUTHOR("Songjun Wu <songjun.wu@atmel.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel-classd.h b/sound/soc/atmel/atmel-classd.h new file mode 100644 index 000000000000..73f8fdd1ca83 --- /dev/null +++ b/sound/soc/atmel/atmel-classd.h @@ -0,0 +1,120 @@ +#ifndef __ATMEL_CLASSD_H_ +#define __ATMEL_CLASSD_H_ + +#define CLASSD_CR 0x00000000 +#define CLASSD_CR_RESET 0x1 + +#define CLASSD_MR 0x00000004 + +#define CLASSD_MR_LEN_DIS 0x0 +#define CLASSD_MR_LEN_EN 0x1 +#define CLASSD_MR_LEN_MASK (0x1 << 0) +#define CLASSD_MR_LEN_SHIFT (0) + +#define CLASSD_MR_LMUTE_DIS 0x0 +#define CLASSD_MR_LMUTE_EN 0x1 +#define CLASSD_MR_LMUTE_SHIFT (0x1) +#define CLASSD_MR_LMUTE_MASK (0x1 << 1) + +#define CLASSD_MR_REN_DIS 0x0 +#define CLASSD_MR_REN_EN 0x1 +#define CLASSD_MR_REN_MASK (0x1 << 4) +#define CLASSD_MR_REN_SHIFT (4) + +#define CLASSD_MR_RMUTE_DIS 0x0 +#define CLASSD_MR_RMUTE_EN 0x1 +#define CLASSD_MR_RMUTE_SHIFT (0x5) +#define CLASSD_MR_RMUTE_MASK (0x1 << 5) + +#define CLASSD_MR_PWMTYP_SINGLE 0x0 +#define CLASSD_MR_PWMTYP_DIFF 0x1 +#define CLASSD_MR_PWMTYP_MASK (0x1 << 8) +#define CLASSD_MR_PWMTYP_SHIFT (8) + +#define CLASSD_MR_NON_OVERLAP_DIS 0x0 +#define CLASSD_MR_NON_OVERLAP_EN 0x1 +#define CLASSD_MR_NON_OVERLAP_MASK (0x1 << 16) +#define CLASSD_MR_NON_OVERLAP_SHIFT (16) + +#define CLASSD_MR_NOVR_VAL_5NS 0x0 +#define CLASSD_MR_NOVR_VAL_10NS 0x1 +#define CLASSD_MR_NOVR_VAL_15NS 0x2 +#define CLASSD_MR_NOVR_VAL_20NS 0x3 +#define CLASSD_MR_NOVR_VAL_MASK (0x3 << 20) +#define CLASSD_MR_NOVR_VAL_SHIFT (20) + +#define CLASSD_INTPMR 0x00000008 + +#define CLASSD_INTPMR_ATTL_MASK (0x3f << 0) +#define CLASSD_INTPMR_ATTL_SHIFT (0) +#define CLASSD_INTPMR_ATTR_MASK (0x3f << 8) +#define CLASSD_INTPMR_ATTR_SHIFT (8) + +#define CLASSD_INTPMR_DSP_CLK_FREQ_12M288 0x0 +#define CLASSD_INTPMR_DSP_CLK_FREQ_11M2896 0x1 +#define CLASSD_INTPMR_DSP_CLK_FREQ_MASK (0x1 << 16) +#define CLASSD_INTPMR_DSP_CLK_FREQ_SHIFT (16) + +#define CLASSD_INTPMR_DEEMP_DIS 0x0 +#define CLASSD_INTPMR_DEEMP_EN 0x1 +#define CLASSD_INTPMR_DEEMP_MASK (0x1 << 18) +#define CLASSD_INTPMR_DEEMP_SHIFT (18) + +#define CLASSD_INTPMR_SWAP_LEFT_ON_LSB 0x0 +#define CLASSD_INTPMR_SWAP_RIGHT_ON_LSB 0x1 +#define CLASSD_INTPMR_SWAP_MASK (0x1 << 19) +#define CLASSD_INTPMR_SWAP_SHIFT (19) + +#define CLASSD_INTPMR_FRAME_8K 0x0 +#define CLASSD_INTPMR_FRAME_16K 0x1 +#define CLASSD_INTPMR_FRAME_32K 0x2 +#define CLASSD_INTPMR_FRAME_48K 0x3 +#define CLASSD_INTPMR_FRAME_96K 0x4 +#define CLASSD_INTPMR_FRAME_22K 0x5 +#define CLASSD_INTPMR_FRAME_44K 0x6 +#define CLASSD_INTPMR_FRAME_88K 0x7 +#define CLASSD_INTPMR_FRAME_MASK (0x7 << 20) +#define CLASSD_INTPMR_FRAME_SHIFT (20) + +#define CLASSD_INTPMR_EQCFG_FLAT 0x0 +#define CLASSD_INTPMR_EQCFG_B_BOOST_12 0x1 +#define CLASSD_INTPMR_EQCFG_B_BOOST_6 0x2 +#define CLASSD_INTPMR_EQCFG_B_CUT_12 0x3 +#define CLASSD_INTPMR_EQCFG_B_CUT_6 0x4 +#define CLASSD_INTPMR_EQCFG_M_BOOST_3 0x5 +#define CLASSD_INTPMR_EQCFG_M_BOOST_8 0x6 +#define CLASSD_INTPMR_EQCFG_M_CUT_3 0x7 +#define CLASSD_INTPMR_EQCFG_M_CUT_8 0x8 +#define CLASSD_INTPMR_EQCFG_T_BOOST_12 0x9 +#define CLASSD_INTPMR_EQCFG_T_BOOST_6 0xa +#define CLASSD_INTPMR_EQCFG_T_CUT_12 0xb +#define CLASSD_INTPMR_EQCFG_T_CUT_6 0xc +#define CLASSD_INTPMR_EQCFG_SHIFT (24) + +#define CLASSD_INTPMR_MONO_DIS 0x0 +#define CLASSD_INTPMR_MONO_EN 0x1 +#define CLASSD_INTPMR_MONO_MASK (0x1 << 28) +#define CLASSD_INTPMR_MONO_SHIFT (28) + +#define CLASSD_INTPMR_MONO_MODE_MIX 0x0 +#define CLASSD_INTPMR_MONO_MODE_SAT 0x1 +#define CLASSD_INTPMR_MONO_MODE_LEFT 0x2 +#define CLASSD_INTPMR_MONO_MODE_RIGHT 0x3 +#define CLASSD_INTPMR_MONO_MODE_MASK (0x3 << 29) +#define CLASSD_INTPMR_MONO_MODE_SHIFT (29) + +#define CLASSD_INTSR 0x0000000c + +#define CLASSD_THR 0x00000010 + +#define CLASSD_IER 0x00000014 + +#define CLASSD_IDR 0x00000018 + +#define CLASSD_IMR 0x0000001c + +#define CLASSD_ISR 0x00000020 + +#define CLASSD_WPMR 0x000000e4 + +#endif diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index aa354e1c6ff7..1933bcd46cca 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -176,6 +176,7 @@ static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = { { .compatible = "atmel,asoc-wm8904", }, { } }; +MODULE_DEVICE_TABLE(of, atmel_asoc_wm8904_dt_ids); #endif static struct platform_driver atmel_asoc_wm8904_driver = { diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c index 452f404abfd2..e97c32798e98 100644 --- a/sound/soc/au1x/db1000.c +++ b/sound/soc/au1x/db1000.c @@ -38,14 +38,7 @@ static int db1000_audio_probe(struct platform_device *pdev) { struct snd_soc_card *card = &db1000_ac97; card->dev = &pdev->dev; - return snd_soc_register_card(card); -} - -static int db1000_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - snd_soc_unregister_card(card); - return 0; + return devm_snd_soc_register_card(&pdev->dev, card); } static struct platform_driver db1000_audio_driver = { @@ -54,7 +47,6 @@ static struct platform_driver db1000_audio_driver = { .pm = &snd_soc_pm_ops, }, .probe = db1000_audio_probe, - .remove = db1000_audio_remove, }; module_platform_driver(db1000_audio_driver); diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 8c907ebea189..5c73061d912a 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -178,14 +178,7 @@ static int db1200_audio_probe(struct platform_device *pdev) card = db1200_cards[pid->driver_data]; card->dev = &pdev->dev; - return snd_soc_register_card(card); -} - -static int db1200_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - snd_soc_unregister_card(card); - return 0; + return devm_snd_soc_register_card(&pdev->dev, card); } static struct platform_driver db1200_audio_driver = { @@ -195,7 +188,6 @@ static struct platform_driver db1200_audio_driver = { }, .id_table = db1200_pids, .probe = db1200_audio_probe, - .remove = db1200_audio_remove, }; module_platform_driver(db1200_audio_driver); diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index 5bf1501e5e3c..864df2616e10 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -87,27 +87,18 @@ static int bf5xx_ad1836_driver_probe(struct platform_device *pdev) card->dev = &pdev->dev; platform_set_drvdata(pdev, card); - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "Failed to register card\n"); return ret; } -static int bf5xx_ad1836_driver_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - static struct platform_driver bf5xx_ad1836_driver = { .driver = { .name = "bfin-snd-ad1836", .pm = &snd_soc_pm_ops, }, .probe = bf5xx_ad1836_driver_probe, - .remove = bf5xx_ad1836_driver_remove, }; module_platform_driver(bf5xx_ad1836_driver); diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c index 523baf5820d7..72ac78988426 100644 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -154,16 +154,7 @@ static int bfin_eval_adau1373_probe(struct platform_device *pdev) card->dev = &pdev->dev; - return snd_soc_register_card(&bfin_eval_adau1373); -} - -static int bfin_eval_adau1373_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; + return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1373); } static struct platform_driver bfin_eval_adau1373_driver = { @@ -172,7 +163,6 @@ static struct platform_driver bfin_eval_adau1373_driver = { .pm = &snd_soc_pm_ops, }, .probe = bfin_eval_adau1373_probe, - .remove = bfin_eval_adau1373_remove, }; module_platform_driver(bfin_eval_adau1373_driver); diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c index f9e926dfd4ef..5c67f72cf9a9 100644 --- a/sound/soc/blackfin/bfin-eval-adau1701.c +++ b/sound/soc/blackfin/bfin-eval-adau1701.c @@ -94,16 +94,7 @@ static int bfin_eval_adau1701_probe(struct platform_device *pdev) card->dev = &pdev->dev; - return snd_soc_register_card(&bfin_eval_adau1701); -} - -static int bfin_eval_adau1701_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; + return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1701); } static struct platform_driver bfin_eval_adau1701_driver = { @@ -112,7 +103,6 @@ static struct platform_driver bfin_eval_adau1701_driver = { .pm = &snd_soc_pm_ops, }, .probe = bfin_eval_adau1701_probe, - .remove = bfin_eval_adau1701_remove, }; module_platform_driver(bfin_eval_adau1701_driver); diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c index 27eee66afdb2..1037477d10b2 100644 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ b/sound/soc/blackfin/bfin-eval-adav80x.c @@ -119,16 +119,7 @@ static int bfin_eval_adav80x_probe(struct platform_device *pdev) card->dev = &pdev->dev; - return snd_soc_register_card(&bfin_eval_adav80x); -} - -static int bfin_eval_adav80x_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; + return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adav80x); } static const struct platform_device_id bfin_eval_adav80x_ids[] = { @@ -144,7 +135,6 @@ static struct platform_driver bfin_eval_adav80x_driver = { .pm = &snd_soc_pm_ops, }, .probe = bfin_eval_adav80x_probe, - .remove = bfin_eval_adav80x_remove, .id_table = bfin_eval_adav80x_ids, }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733ecd17f..cfdafc4c11ea 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4554 + select SND_SOC_AK4613 if I2C select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C @@ -57,6 +58,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C + select SND_SOC_DA7219 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DMIC @@ -79,7 +81,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C - select SND_SOC_HDMI_CODEC + select SND_SOC_NAU8825 if I2C select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 @@ -171,6 +173,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8996 if I2C select SND_SOC_WM8997 if MFD_WM8997 + select SND_SOC_WM8998 if MFD_WM8998 select SND_SOC_WM9081 if I2C select SND_SOC_WM9090 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS @@ -195,9 +198,11 @@ config SND_SOC_ARIZONA default y if SND_SOC_WM5102=y default y if SND_SOC_WM5110=y default y if SND_SOC_WM8997=y + default y if SND_SOC_WM8998=y default m if SND_SOC_WM5102=m default m if SND_SOC_WM5110=m default m if SND_SOC_WM8997=m + default m if SND_SOC_WM8998=m config SND_SOC_WM_HUBS tristate @@ -319,6 +324,10 @@ config SND_SOC_AK4535 config SND_SOC_AK4554 tristate "AKM AK4554 CODEC" +config SND_SOC_AK4613 + tristate "AKM AK4613 CODEC" + depends on I2C + config SND_SOC_AK4641 tristate @@ -430,6 +439,9 @@ config SND_SOC_DA7210 config SND_SOC_DA7213 tristate +config SND_SOC_DA7219 + tristate + config SND_SOC_DA732X tristate @@ -442,9 +454,6 @@ config SND_SOC_BT_SCO config SND_SOC_DMIC tristate -config SND_SOC_HDMI_CODEC - tristate "HDMI stub CODEC" - config SND_SOC_ES8328 tristate "Everest Semi ES8328 CODEC" @@ -865,6 +874,9 @@ config SND_SOC_WM8996 config SND_SOC_WM8997 tristate +config SND_SOC_WM8998 + tristate + config SND_SOC_WM9081 tristate @@ -896,6 +908,9 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8825 + tristate + config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a32077954ae..f632fc42f59f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -26,6 +26,7 @@ snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4554-objs := ak4554.o +snd-soc-ak4613-objs := ak4613.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -49,6 +50,7 @@ snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o +snd-soc-da7219-objs := da7219.o da7219-aad.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-bt-sco-objs := bt-sco.o @@ -72,7 +74,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o -snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-nau8825-objs := nau8825.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o @@ -176,6 +178,7 @@ snd-soc-wm8993-objs := wm8993.o snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o snd-soc-wm8995-objs := wm8995.o snd-soc-wm8997-objs := wm8997.o +snd-soc-wm8998-objs := wm8998.o snd-soc-wm9081-objs := wm9081.o snd-soc-wm9090-objs := wm9090.o snd-soc-wm9705-objs := wm9705.o @@ -216,6 +219,7 @@ obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o +obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o @@ -241,6 +245,7 @@ obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o +obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o @@ -264,7 +269,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o -obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o @@ -364,6 +369,7 @@ obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o obj-$(CONFIG_SND_SOC_WM8997) += snd-soc-wm8997.o +obj-$(CONFIG_SND_SOC_WM8998) += snd-soc-wm8998.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c index df3a1a415825..171313664bc8 100644 --- a/sound/soc/codecs/ad193x-i2c.c +++ b/sound/soc/codecs/ad193x-i2c.c @@ -15,8 +15,8 @@ #include "ad193x.h" static const struct i2c_device_id ad193x_id[] = { - { "ad1936", 0 }, - { "ad1937", 0 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, { } }; MODULE_DEVICE_TABLE(i2c, ad193x_id); @@ -30,7 +30,9 @@ static int ad193x_i2c_probe(struct i2c_client *client, config.val_bits = 8; config.reg_bits = 8; - return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config)); + return ad193x_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c index 390cef9b9dc2..431f95da1de1 100644 --- a/sound/soc/codecs/ad193x-spi.c +++ b/sound/soc/codecs/ad193x-spi.c @@ -16,6 +16,7 @@ static int ad193x_spi_probe(struct spi_device *spi) { + const struct spi_device_id *id = spi_get_device_id(spi); struct regmap_config config; config = ad193x_regmap_config; @@ -24,7 +25,8 @@ static int ad193x_spi_probe(struct spi_device *spi) config.read_flag_mask = 0x09; config.write_flag_mask = 0x08; - return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); + return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_spi_remove(struct spi_device *spi) @@ -33,6 +35,17 @@ static int ad193x_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id ad193x_spi_id[] = { + { "ad193x", AD193X }, + { "ad1933", AD1933 }, + { "ad1934", AD1934 }, + { "ad1938", AD193X }, + { "ad1939", AD193X }, + { "adau1328", AD193X }, + { } +}; +MODULE_DEVICE_TABLE(spi, ad193x_spi_id); + static struct spi_driver ad193x_spi_driver = { .driver = { .name = "ad193x", @@ -40,6 +53,7 @@ static struct spi_driver ad193x_spi_driver = { }, .probe = ad193x_spi_probe, .remove = ad193x_spi_remove, + .id_table = ad193x_spi_id, }; module_spi_driver(ad193x_spi_driver); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 17c953595660..3a3f3f2343d7 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,6 +23,7 @@ /* codec private data */ struct ad193x_priv { struct regmap *regmap; + enum ad193x_type type; int sysclk; }; @@ -47,12 +48,6 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL, AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv), - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, - AD193X_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, - AD193X_ADCR2_MUTE, 1, 1), - /* DAC switch control */ SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE, AD193X_DACR1_MUTE, 1, 1), @@ -63,26 +58,37 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE, AD193X_DACR4_MUTE, 1, 1), + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), +}; + +static const struct snd_kcontrol_new ad193x_adc_snd_controls[] = { + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, + AD193X_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, + AD193X_ADCR2_MUTE, 1, 1), + /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0, AD193X_ADC_HIGHPASS_FILTER, 1, 0), - - /* DAC de-emphasis */ - SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), SND_SOC_DAPM_OUTPUT("DAC4OUT"), +}; + +static const struct snd_soc_dapm_widget ad193x_adc_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_INPUT("ADC1IN"), SND_SOC_DAPM_INPUT("ADC2IN"), }; @@ -91,18 +97,33 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, { "DAC Output", NULL, "DAC" }, { "DAC Output", NULL, "VMID" }, - { "ADC", NULL, "SYSCLK" }, - { "DAC", NULL, "ADC_PWR" }, - { "ADC", NULL, "ADC_PWR" }, { "DAC1OUT", NULL, "DAC Output" }, { "DAC2OUT", NULL, "DAC Output" }, { "DAC3OUT", NULL, "DAC Output" }, { "DAC4OUT", NULL, "DAC Output" }, + { "SYSCLK", NULL, "PLL_PWR" }, +}; + +static const struct snd_soc_dapm_route ad193x_adc_audio_paths[] = { + { "ADC", NULL, "SYSCLK" }, + { "ADC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, - { "SYSCLK", NULL, "PLL_PWR" }, }; +static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x) +{ + switch (ad193x->type) { + case AD1933: + case AD1934: + return false; + default: + break; + } + + return true; +} + /* * DAI ops entries */ @@ -147,8 +168,10 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_CHAN_MASK, + channels << AD193X_ADC_CHAN_SHFT); return 0; } @@ -172,7 +195,9 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, adc_serfmt |= AD193X_ADC_SERFMT_AUX; break; default: - return -EINVAL; + if (ad193x_has_adc(ad193x)) + return -EINVAL; + break; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -217,10 +242,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_SERFMT_MASK, adc_serfmt); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_FMT_MASK, adc_fmt); + if (ad193x_has_adc(ad193x)) { + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_SERFMT_MASK, adc_serfmt); + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_FMT_MASK, adc_fmt); + } regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_FMT_MASK, dac_fmt); @@ -287,8 +314,9 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, AD193X_DAC_WORD_LEN_MASK, word_len << AD193X_DAC_WORD_LEN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_WORD_LEN_MASK, word_len); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_WORD_LEN_MASK, word_len); return 0; } @@ -326,6 +354,8 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int num, ret; /* default setting for ad193x */ @@ -335,14 +365,46 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); /* dac in tdm mode */ regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); - /* high-pass filter enable */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); - /* sata delay=1, adc aux mode */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* high-pass filter enable */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + } + /* pll input: mclki/xi */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* add adc controls */ + num = ARRAY_SIZE(ad193x_adc_snd_controls); + ret = snd_soc_add_codec_controls(codec, + ad193x_adc_snd_controls, + num); + if (ret) + return ret; + + /* add adc widgets */ + num = ARRAY_SIZE(ad193x_adc_widgets); + ret = snd_soc_dapm_new_controls(dapm, + ad193x_adc_widgets, + num); + if (ret) + return ret; + + /* add adc routes */ + num = ARRAY_SIZE(ad193x_adc_audio_paths); + ret = snd_soc_dapm_add_routes(dapm, + ad193x_adc_audio_paths, + num); + if (ret) + return ret; + } + return 0; } @@ -356,18 +418,13 @@ static struct snd_soc_codec_driver soc_codec_dev_ad193x = { .num_dapm_routes = ARRAY_SIZE(audio_paths), }; -static bool adau193x_reg_volatile(struct device *dev, unsigned int reg) -{ - return false; -} - const struct regmap_config ad193x_regmap_config = { .max_register = AD193X_NUM_REGS - 1, - .volatile_reg = adau193x_reg_volatile, }; EXPORT_SYMBOL_GPL(ad193x_regmap_config); -int ad193x_probe(struct device *dev, struct regmap *regmap) +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type) { struct ad193x_priv *ad193x; @@ -379,6 +436,7 @@ int ad193x_probe(struct device *dev, struct regmap *regmap) return -ENOMEM; ad193x->regmap = regmap; + ad193x->type = type; dev_set_drvdata(dev, ad193x); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index ab9a998f15be..8b1e65f928d2 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -13,8 +13,15 @@ struct device; +enum ad193x_type { + AD193X, + AD1933, + AD1934, +}; + extern const struct regmap_config ad193x_regmap_config; -int ad193x_probe(struct device *dev, struct regmap *regmap); +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type); #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 198c924551b7..acff8d62059c 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -728,8 +728,8 @@ static int adav80x_dai_startup(struct snd_pcm_substream *substream, if (!snd_soc_codec_is_active(codec) || !adav80x->rate) return 0; - return snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate); + return snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, adav80x->rate); } static void adav80x_dai_shutdown(struct snd_pcm_substream *substream, diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c new file mode 100644 index 000000000000..07a266460ec3 --- /dev/null +++ b/sound/soc/codecs/ak4613.c @@ -0,0 +1,497 @@ +/* + * ak4613.c -- Asahi Kasei ALSA Soc Audio driver + * + * Copyright (C) 2015 Renesas Electronics Corporation + * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + * + * Based on ak4642.c by Kuninori Morimoto + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#define PW_MGMT1 0x00 /* Power Management 1 */ +#define PW_MGMT2 0x01 /* Power Management 2 */ +#define PW_MGMT3 0x02 /* Power Management 3 */ +#define CTRL1 0x03 /* Control 1 */ +#define CTRL2 0x04 /* Control 2 */ +#define DEMP1 0x05 /* De-emphasis1 */ +#define DEMP2 0x06 /* De-emphasis2 */ +#define OFD 0x07 /* Overflow Detect */ +#define ZRD 0x08 /* Zero Detect */ +#define ICTRL 0x09 /* Input Control */ +#define OCTRL 0x0a /* Output Control */ +#define LOUT1 0x0b /* LOUT1 Volume Control */ +#define ROUT1 0x0c /* ROUT1 Volume Control */ +#define LOUT2 0x0d /* LOUT2 Volume Control */ +#define ROUT2 0x0e /* ROUT2 Volume Control */ +#define LOUT3 0x0f /* LOUT3 Volume Control */ +#define ROUT3 0x10 /* ROUT3 Volume Control */ +#define LOUT4 0x11 /* LOUT4 Volume Control */ +#define ROUT4 0x12 /* ROUT4 Volume Control */ +#define LOUT5 0x13 /* LOUT5 Volume Control */ +#define ROUT5 0x14 /* ROUT5 Volume Control */ +#define LOUT6 0x15 /* LOUT6 Volume Control */ +#define ROUT6 0x16 /* ROUT6 Volume Control */ + +/* PW_MGMT1 */ +#define RSTN BIT(0) +#define PMDAC BIT(1) +#define PMADC BIT(2) +#define PMVR BIT(3) + +/* PW_MGMT2 */ +#define PMAD_ALL 0x7 + +/* PW_MGMT3 */ +#define PMDA_ALL 0x3f + +/* CTRL1 */ +#define DIF0 BIT(3) +#define DIF1 BIT(4) +#define DIF2 BIT(5) +#define TDM0 BIT(6) +#define TDM1 BIT(7) +#define NO_FMT (0xff) +#define FMT_MASK (0xf8) + +/* CTRL2 */ +#define DFS_NORMAL_SPEED (0 << 2) +#define DFS_DOUBLE_SPEED (1 << 2) +#define DFS_QUAD_SPEED (2 << 2) + +struct ak4613_priv { + struct mutex lock; + + unsigned int fmt; + u8 fmt_ctrl; + int cnt; +}; + +struct ak4613_formats { + unsigned int width; + unsigned int fmt; +}; + +struct ak4613_interface { + struct ak4613_formats capture; + struct ak4613_formats playback; +}; + +/* + * Playback Volume + * + * max : 0x00 : 0 dB + * ( 0.5 dB step ) + * min : 0xFE : -127.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new ak4613_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume1", LOUT1, ROUT1, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume2", LOUT2, ROUT2, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume3", LOUT3, ROUT3, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume4", LOUT4, ROUT4, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume5", LOUT5, ROUT5, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume6", LOUT6, ROUT6, + 0, 0xFF, 1, out_tlv), +}; + +static const struct reg_default ak4613_reg[] = { + { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 }, + { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 }, + { 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 }, + { 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 }, + { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 }, + { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 }, +}; + +#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3) +#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } +static const struct ak4613_interface ak4613_iface[] = { + /* capture */ /* playback */ + [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, + [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, + [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, + [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, + [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, +}; + +static const struct regmap_config ak4613_regmap_cfg = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0x16, + .reg_defaults = ak4613_reg, + .num_reg_defaults = ARRAY_SIZE(ak4613_reg), +}; + +static const struct of_device_id ak4613_of_match[] = { + { .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg }, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4613_of_match); + +static const struct i2c_device_id ak4613_i2c_id[] = { + { "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id); + +static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = { + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("LOUT4"), + SND_SOC_DAPM_OUTPUT("LOUT5"), + SND_SOC_DAPM_OUTPUT("LOUT6"), + + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + SND_SOC_DAPM_OUTPUT("ROUT4"), + SND_SOC_DAPM_OUTPUT("ROUT5"), + SND_SOC_DAPM_OUTPUT("ROUT6"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("RIN2"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0), + SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0), + SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0), + SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0), + SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0), + SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0), + SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0), +}; + +static const struct snd_soc_dapm_route ak4613_intercon[] = { + {"LOUT1", NULL, "DAC1"}, + {"LOUT2", NULL, "DAC2"}, + {"LOUT3", NULL, "DAC3"}, + {"LOUT4", NULL, "DAC4"}, + {"LOUT5", NULL, "DAC5"}, + {"LOUT6", NULL, "DAC6"}, + + {"ROUT1", NULL, "DAC1"}, + {"ROUT2", NULL, "DAC2"}, + {"ROUT3", NULL, "DAC3"}, + {"ROUT4", NULL, "DAC4"}, + {"ROUT5", NULL, "DAC5"}, + {"ROUT6", NULL, "DAC6"}, + + {"DAC1", NULL, "Playback"}, + {"DAC2", NULL, "Playback"}, + {"DAC3", NULL, "Playback"}, + {"DAC4", NULL, "Playback"}, + {"DAC5", NULL, "Playback"}, + {"DAC6", NULL, "Playback"}, + + {"Capture", NULL, "ADC1"}, + {"Capture", NULL, "ADC2"}, + + {"ADC1", NULL, "LIN1"}, + {"ADC2", NULL, "LIN2"}, + + {"ADC1", NULL, "RIN1"}, + {"ADC2", NULL, "RIN2"}, +}; + +static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + + mutex_lock(&priv->lock); + priv->cnt--; + if (priv->cnt < 0) { + dev_err(dev, "unexpected counter error\n"); + priv->cnt = 0; + } + if (!priv->cnt) + priv->fmt_ctrl = NO_FMT; + mutex_unlock(&priv->lock); +} + +static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + fmt &= SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_I2S: + priv->fmt = fmt; + + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + const struct ak4613_formats *fmts; + struct device *dev = codec->dev; + unsigned int width = params_width(params); + unsigned int fmt = priv->fmt; + unsigned int rate; + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int i, ret; + u8 fmt_ctrl, ctrl2; + + rate = params_rate(params); + switch (rate) { + case 32000: + case 44100: + case 48000: + ctrl2 = DFS_NORMAL_SPEED; + break; + case 88200: + case 96000: + ctrl2 = DFS_DOUBLE_SPEED; + break; + case 176400: + case 192000: + ctrl2 = DFS_QUAD_SPEED; + break; + default: + return -EINVAL; + } + + /* + * FIXME + * + * It doesn't support TDM at this point + */ + fmt_ctrl = NO_FMT; + for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) { + fmts = (is_play) ? &ak4613_iface[i].playback : + &ak4613_iface[i].capture; + + if (fmts->fmt != fmt) + continue; + + if (fmt == SND_SOC_DAIFMT_RIGHT_J) { + if (fmts->width != width) + continue; + } else { + if (fmts->width < width) + continue; + } + + fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i); + break; + } + + ret = -EINVAL; + if (fmt_ctrl == NO_FMT) + goto hw_params_end; + + mutex_lock(&priv->lock); + if ((priv->fmt_ctrl == NO_FMT) || + (priv->fmt_ctrl == fmt_ctrl)) { + priv->fmt_ctrl = fmt_ctrl; + priv->cnt++; + ret = 0; + } + mutex_unlock(&priv->lock); + + if (ret < 0) + goto hw_params_end; + + snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl); + snd_soc_write(codec, CTRL2, ctrl2); + +hw_params_end: + if (ret < 0) + dev_warn(dev, "unsupported data width/format combination\n"); + + return ret; +} + +static int ak4613_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 mgmt1 = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + mgmt1 |= RSTN; + /* fall through */ + case SND_SOC_BIAS_PREPARE: + mgmt1 |= PMADC | PMDAC; + /* fall through */ + case SND_SOC_BIAS_STANDBY: + mgmt1 |= PMVR; + /* fall through */ + case SND_SOC_BIAS_OFF: + default: + break; + } + + snd_soc_write(codec, PW_MGMT1, mgmt1); + + return 0; +} + +static const struct snd_soc_dai_ops ak4613_dai_ops = { + .shutdown = ak4613_dai_shutdown, + .set_fmt = ak4613_dai_set_fmt, + .hw_params = ak4613_dai_hw_params, +}; + +#define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_64000 |\ + SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) +#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver ak4613_dai = { + .name = "ak4613-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .ops = &ak4613_dai_ops, + .symmetric_rates = 1, +}; + +static int ak4613_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + + regcache_mark_dirty(regmap); + return regcache_sync(regmap); +} + +static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { + .resume = ak4613_resume, + .set_bias_level = ak4613_set_bias_level, + .controls = ak4613_snd_controls, + .num_controls = ARRAY_SIZE(ak4613_snd_controls), + .dapm_widgets = ak4613_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets), + .dapm_routes = ak4613_intercon, + .num_dapm_routes = ARRAY_SIZE(ak4613_intercon), +}; + +static int ak4613_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct device_node *np = dev->of_node; + const struct regmap_config *regmap_cfg; + struct regmap *regmap; + struct ak4613_priv *priv; + + regmap_cfg = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4613_of_match, dev); + if (of_id) + regmap_cfg = of_id->data; + } else { + regmap_cfg = (const struct regmap_config *)id->driver_data; + } + + if (!regmap_cfg) + return -EINVAL; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->fmt_ctrl = NO_FMT; + priv->cnt = 0; + + mutex_init(&priv->lock); + + i2c_set_clientdata(i2c, priv); + + regmap = devm_regmap_init_i2c(i2c, regmap_cfg); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return snd_soc_register_codec(dev, &soc_codec_dev_ak4613, + &ak4613_dai, 1); +} + +static int ak4613_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver ak4613_i2c_driver = { + .driver = { + .name = "ak4613-codec", + .owner = THIS_MODULE, + .of_match_table = ak4613_of_match, + }, + .probe = ak4613_i2c_probe, + .remove = ak4613_i2c_remove, + .id_table = ak4613_i2c_id, +}; + +module_i2c_driver(ak4613_i2c_driver); + +MODULE_DESCRIPTION("Soc AK4613 driver"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 4a90143d0e90..cda27c22812a 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -23,6 +23,8 @@ * AK4648 is tested. */ +#include <linux/clk.h> +#include <linux/clk-provider.h> #include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> @@ -128,11 +130,8 @@ #define I2S (3 << 0) /* MD_CTL2 */ -#define FS0 (1 << 0) -#define FS1 (1 << 1) -#define FS2 (1 << 2) -#define FS3 (1 << 5) -#define FS_MASK (FS0 | FS1 | FS2 | FS3) +#define FSs(val) (((val & 0x7) << 0) | ((val & 0x8) << 2)) +#define PSs(val) ((val & 0x3) << 6) /* MD_CTL3 */ #define BST1 (1 << 3) @@ -147,6 +146,7 @@ struct ak4642_drvdata { struct ak4642_priv { const struct ak4642_drvdata *drvdata; + struct clk *mcko; }; /* @@ -430,56 +430,56 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int ak4642_set_mcko(struct snd_soc_codec *codec, + u32 frequency) +{ + u32 fs_list[] = { + [0] = 8000, + [1] = 12000, + [2] = 16000, + [3] = 24000, + [4] = 7350, + [5] = 11025, + [6] = 14700, + [7] = 22050, + [10] = 32000, + [11] = 48000, + [14] = 29400, + [15] = 44100, + }; + u32 ps_list[] = { + [0] = 256, + [1] = 128, + [2] = 64, + [3] = 32 + }; + int ps, fs; + + for (ps = 0; ps < ARRAY_SIZE(ps_list); ps++) { + for (fs = 0; fs < ARRAY_SIZE(fs_list); fs++) { + if (frequency == ps_list[ps] * fs_list[fs]) { + snd_soc_write(codec, MD_CTL2, + PSs(ps) | FSs(fs)); + return 0; + } + } + } + + return 0; +} + static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - u8 rate; + struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec); + u32 rate = clk_get_rate(priv->mcko); - switch (params_rate(params)) { - case 7350: - rate = FS2; - break; - case 8000: - rate = 0; - break; - case 11025: - rate = FS2 | FS0; - break; - case 12000: - rate = FS0; - break; - case 14700: - rate = FS2 | FS1; - break; - case 16000: - rate = FS1; - break; - case 22050: - rate = FS2 | FS1 | FS0; - break; - case 24000: - rate = FS1 | FS0; - break; - case 29400: - rate = FS3 | FS2 | FS1; - break; - case 32000: - rate = FS3 | FS1; - break; - case 44100: - rate = FS3 | FS2 | FS1 | FS0; - break; - case 48000: - rate = FS3 | FS1 | FS0; - break; - default: - return -EINVAL; - } - snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); + if (!rate) + rate = params_rate(params) * 256; - return 0; + return ak4642_set_mcko(codec, rate); } static int ak4642_set_bias_level(struct snd_soc_codec *codec, @@ -532,7 +532,18 @@ static int ak4642_resume(struct snd_soc_codec *codec) return 0; } +static int ak4642_probe(struct snd_soc_codec *codec) +{ + struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec); + + if (priv->mcko) + ak4642_set_mcko(codec, clk_get_rate(priv->mcko)); + + return 0; +} + static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { + .probe = ak4642_probe, .resume = ak4642_resume, .set_bias_level = ak4642_set_bias_level, .controls = ak4642_snd_controls, @@ -580,19 +591,54 @@ static const struct ak4642_drvdata ak4648_drvdata = { .extended_frequencies = 1, }; +#ifdef CONFIG_COMMON_CLK +static struct clk *ak4642_of_parse_mcko(struct device *dev) +{ + struct device_node *np = dev->of_node; + struct clk *clk; + const char *clk_name = np->name; + const char *parent_clk_name = NULL; + u32 rate; + + if (of_property_read_u32(np, "clock-frequency", &rate)) + return NULL; + + if (of_property_read_bool(np, "clocks")) + parent_clk_name = of_clk_get_parent_name(np, 0); + + of_property_read_string(np, "clock-output-names", &clk_name); + + clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, + (parent_clk_name) ? 0 : CLK_IS_ROOT, + rate); + if (!IS_ERR(clk)) + of_clk_add_provider(np, of_clk_src_simple_get, clk); + + return clk; +} +#else +#define ak4642_of_parse_mcko(d) 0 +#endif + static const struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct device_node *np = i2c->dev.of_node; + struct device *dev = &i2c->dev; + struct device_node *np = dev->of_node; const struct ak4642_drvdata *drvdata = NULL; struct regmap *regmap; struct ak4642_priv *priv; + struct clk *mcko = NULL; if (np) { const struct of_device_id *of_id; - of_id = of_match_device(ak4642_of_match, &i2c->dev); + mcko = ak4642_of_parse_mcko(dev); + if (IS_ERR(mcko)) + mcko = NULL; + + of_id = of_match_device(ak4642_of_match, dev); if (of_id) drvdata = of_id->data; } else { @@ -600,15 +646,16 @@ static int ak4642_i2c_probe(struct i2c_client *i2c, } if (!drvdata) { - dev_err(&i2c->dev, "Unknown device type\n"); + dev_err(dev, "Unknown device type\n"); return -EINVAL; } - priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; priv->drvdata = drvdata; + priv->mcko = mcko; i2c_set_clientdata(i2c, priv); @@ -616,7 +663,7 @@ static int ak4642_i2c_probe(struct i2c_client *i2c, if (IS_ERR(regmap)) return PTR_ERR(regmap); - return snd_soc_register_codec(&i2c->dev, + return snd_soc_register_codec(dev, &soc_codec_dev_ak4642, &ak4642_dai, 1); } diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8a2221ab3d10..9929efc6b9aa 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -147,6 +147,8 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, 0x4f5, 0x0da); } break; + default: + break; } return 0; @@ -314,6 +316,7 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "Tone Generator 2", "Haptics", "AEC", + "AEC2", "Mic Mute Mixer", "Noise Generator", "IN1L", @@ -421,6 +424,7 @@ int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { 0x05, 0x06, /* Haptics */ 0x08, /* AEC */ + 0x09, /* AEC2 */ 0x0c, /* Noise mixer */ 0x0d, /* Comfort noise */ 0x10, /* IN1L */ @@ -525,6 +529,32 @@ EXPORT_SYMBOL_GPL(arizona_mixer_values); const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0); EXPORT_SYMBOL_GPL(arizona_mixer_tlv); +const char * const arizona_sample_rate_text[ARIZONA_SAMPLE_RATE_ENUM_SIZE] = { + "12kHz", "24kHz", "48kHz", "96kHz", "192kHz", + "11.025kHz", "22.05kHz", "44.1kHz", "88.2kHz", "176.4kHz", + "4kHz", "8kHz", "16kHz", "32kHz", +}; +EXPORT_SYMBOL_GPL(arizona_sample_rate_text); + +const unsigned int arizona_sample_rate_val[ARIZONA_SAMPLE_RATE_ENUM_SIZE] = { + 0x01, 0x02, 0x03, 0x04, 0x05, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, + 0x10, 0x11, 0x12, 0x13, +}; +EXPORT_SYMBOL_GPL(arizona_sample_rate_val); + +const char *arizona_sample_rate_val_to_name(unsigned int rate_val) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(arizona_sample_rate_val); ++i) { + if (arizona_sample_rate_val[i] == rate_val) + return arizona_sample_rate_text[i]; + } + + return "Illegal"; +} +EXPORT_SYMBOL_GPL(arizona_sample_rate_val_to_name); + const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = { "SYNCCLK rate", "8kHz", "16kHz", "ASYNCCLK rate", }; @@ -689,6 +719,15 @@ static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena) ARIZONA_IN_VU, val); } +bool arizona_input_analog(struct snd_soc_codec *codec, int shift) +{ + unsigned int reg = ARIZONA_IN1L_CONTROL + ((shift / 2) * 8); + unsigned int val = snd_soc_read(codec, reg); + + return !(val & ARIZONA_IN1_MODE_MASK); +} +EXPORT_SYMBOL_GPL(arizona_input_analog); + int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -725,6 +764,9 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES); if (reg == 0) arizona_in_set_vu(codec, 0); + break; + default: + break; } return 0; @@ -806,6 +848,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, break; } break; + default: + break; } return 0; @@ -1868,6 +1912,11 @@ static int arizona_calc_fratio(struct arizona_fll *fll, if (fll->arizona->rev < 3 || sync) return init_ratio; break; + case WM8998: + case WM1814: + if (sync) + return init_ratio; + break; default: return init_ratio; } diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index ada0a418ff4b..fea8b8ae8e1a 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -93,12 +93,17 @@ struct arizona_priv { bool dvfs_cached; }; -#define ARIZONA_NUM_MIXER_INPUTS 103 +#define ARIZONA_NUM_MIXER_INPUTS 104 extern const unsigned int arizona_mixer_tlv[]; extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; +#define ARIZONA_GAINMUX_CONTROLS(name, base) \ + SOC_SINGLE_RANGE_TLV(name " Input Volume", base + 1, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv) + #define ARIZONA_MIXER_CONTROLS(name, base) \ SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1, \ ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ @@ -209,8 +214,12 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; .num_regs = 1 }) } #define ARIZONA_RATE_ENUM_SIZE 4 +#define ARIZONA_SAMPLE_RATE_ENUM_SIZE 14 + extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; +extern const char * const arizona_sample_rate_text[ARIZONA_SAMPLE_RATE_ENUM_SIZE]; +extern const unsigned int arizona_sample_rate_val[ARIZONA_SAMPLE_RATE_ENUM_SIZE]; extern const struct soc_enum arizona_isrc_fsl[]; extern const struct soc_enum arizona_isrc_fsh[]; @@ -294,4 +303,7 @@ extern int arizona_init_dai(struct arizona_priv *priv, int dai); int arizona_set_output_mode(struct snd_soc_codec *codec, int output, bool diff); +extern bool arizona_input_analog(struct snd_soc_codec *codec, int shift); + +extern const char *arizona_sample_rate_val_to_name(unsigned int rate_val); #endif diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index a9c86efb3187..7278f93460c1 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -12,6 +12,7 @@ * option) any later version. */ +#include <linux/clk.h> #include <linux/delay.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -1222,23 +1223,44 @@ static int da7213_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if ((da7213->clk_src == clk_id) && (da7213->mclk_rate == freq)) + return 0; + + if (((freq < 5000000) && (freq != 32768)) || (freq > 54000000)) { + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } switch (clk_id) { case DA7213_CLKSRC_MCLK: - if ((freq == 32768) || - ((freq >= 5000000) && (freq <= 54000000))) { - da7213->mclk_rate = freq; - return 0; - } else { - dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", - freq); - return -EINVAL; - } + da7213->mclk_squarer_en = false; + break; + case DA7213_CLKSRC_MCLK_SQR: + da7213->mclk_squarer_en = true; break; default: dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); return -EINVAL; } + + da7213->clk_src = clk_id; + + if (da7213->mclk) { + freq = clk_round_rate(da7213->mclk, freq); + ret = clk_set_rate(da7213->mclk, freq); + if (ret) { + dev_err(codec_dai->dev, "Failed to set clock rate %d\n", + freq); + return ret; + } + } + + da7213->mclk_rate = freq; + + return 0; } /* Supported PLL input frequencies are 5MHz - 54MHz. */ @@ -1366,12 +1388,25 @@ static struct snd_soc_dai_driver da7213_dai = { static int da7213_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + int ret; + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + /* MCLK */ + if (da7213->mclk) { + ret = clk_prepare_enable(da7213->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable mclk\n"); + return ret; + } + } + /* Enable VMID reference & master bias */ snd_soc_update_bits(codec, DA7213_REFERENCES, DA7213_VMID_EN | DA7213_BIAS_EN, @@ -1382,15 +1417,127 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec, /* Disable VMID reference & master bias */ snd_soc_update_bits(codec, DA7213_REFERENCES, DA7213_VMID_EN | DA7213_BIAS_EN, 0); + + /* MCLK */ + if (da7213->mclk) + clk_disable_unprepare(da7213->mclk); break; } return 0; } +/* DT */ +static const struct of_device_id da7213_of_match[] = { + { .compatible = "dlg,da7213", }, + { } +}; +MODULE_DEVICE_TABLE(of, da7213_of_match); + +static enum da7213_micbias_voltage + da7213_of_micbias_lvl(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1600: + return DA7213_MICBIAS_1_6V; + case 2200: + return DA7213_MICBIAS_2_2V; + case 2500: + return DA7213_MICBIAS_2_5V; + case 3000: + return DA7213_MICBIAS_3_0V; + default: + dev_warn(codec->dev, "Invalid micbias level\n"); + return DA7213_MICBIAS_2_2V; + } +} + +static enum da7213_dmic_data_sel + da7213_of_dmic_data_sel(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "lrise_rfall")) { + return DA7213_DMIC_DATA_LRISE_RFALL; + } else if (!strcmp(str, "lfall_rrise")) { + return DA7213_DMIC_DATA_LFALL_RRISE; + } else { + dev_warn(codec->dev, "Invalid DMIC data select type\n"); + return DA7213_DMIC_DATA_LRISE_RFALL; + } +} + +static enum da7213_dmic_samplephase + da7213_of_dmic_samplephase(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "on_clkedge")) { + return DA7213_DMIC_SAMPLE_ON_CLKEDGE; + } else if (!strcmp(str, "between_clkedge")) { + return DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE; + } else { + dev_warn(codec->dev, "Invalid DMIC sample phase\n"); + return DA7213_DMIC_SAMPLE_ON_CLKEDGE; + } +} + +static enum da7213_dmic_clk_rate + da7213_of_dmic_clkrate(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1500000: + return DA7213_DMIC_CLK_1_5MHZ; + case 3000000: + return DA7213_DMIC_CLK_3_0MHZ; + default: + dev_warn(codec->dev, "Invalid DMIC clock rate\n"); + return DA7213_DMIC_CLK_1_5MHZ; + } +} + +static struct da7213_platform_data + *da7213_of_to_pdata(struct snd_soc_codec *codec) +{ + struct device_node *np = codec->dev->of_node; + struct da7213_platform_data *pdata; + const char *of_str; + u32 of_val32; + + pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) { + dev_warn(codec->dev, "Failed to allocate memory for pdata\n"); + return NULL; + } + + if (of_property_read_u32(np, "dlg,micbias1-lvl", &of_val32) >= 0) + pdata->micbias1_lvl = da7213_of_micbias_lvl(codec, of_val32); + else + pdata->micbias1_lvl = DA7213_MICBIAS_2_2V; + + if (of_property_read_u32(np, "dlg,micbias2-lvl", &of_val32) >= 0) + pdata->micbias2_lvl = da7213_of_micbias_lvl(codec, of_val32); + else + pdata->micbias2_lvl = DA7213_MICBIAS_2_2V; + + if (!of_property_read_string(np, "dlg,dmic-data-sel", &of_str)) + pdata->dmic_data_sel = da7213_of_dmic_data_sel(codec, of_str); + else + pdata->dmic_data_sel = DA7213_DMIC_DATA_LRISE_RFALL; + + if (!of_property_read_string(np, "dlg,dmic-samplephase", &of_str)) + pdata->dmic_samplephase = + da7213_of_dmic_samplephase(codec, of_str); + else + pdata->dmic_samplephase = DA7213_DMIC_SAMPLE_ON_CLKEDGE; + + if (of_property_read_u32(np, "dlg,dmic-clkrate", &of_val32) >= 0) + pdata->dmic_clk_rate = da7213_of_dmic_clkrate(codec, of_val32); + else + pdata->dmic_clk_rate = DA7213_DMIC_CLK_3_0MHZ; + + return pdata; +} + + static int da7213_probe(struct snd_soc_codec *codec) { struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); - struct da7213_platform_data *pdata = da7213->pdata; /* Default to using ALC auto offset calibration mode. */ snd_soc_update_bits(codec, DA7213_ALC_CTRL1, @@ -1450,8 +1597,15 @@ static int da7213_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, DA7213_LINE_CTRL, DA7213_LINE_AMP_OE, DA7213_LINE_AMP_OE); + /* Handle DT/Platform data */ + if (codec->dev->of_node) + da7213->pdata = da7213_of_to_pdata(codec); + else + da7213->pdata = dev_get_platdata(codec->dev); + /* Set platform data values */ if (da7213->pdata) { + struct da7213_platform_data *pdata = da7213->pdata; u8 micbias_lvl = 0, dmic_cfg = 0; /* Set Mic Bias voltages */ @@ -1503,10 +1657,17 @@ static int da7213_probe(struct snd_soc_codec *codec) DA7213_DMIC_DATA_SEL_MASK | DA7213_DMIC_SAMPLEPHASE_MASK | DA7213_DMIC_CLK_RATE_MASK, dmic_cfg); + } - /* Set MCLK squaring */ - da7213->mclk_squarer_en = pdata->mclk_squaring; + /* Check if MCLK provided */ + da7213->mclk = devm_clk_get(codec->dev, "mclk"); + if (IS_ERR(da7213->mclk)) { + if (PTR_ERR(da7213->mclk) != -ENOENT) + return PTR_ERR(da7213->mclk); + else + da7213->mclk = NULL; } + return 0; } @@ -1537,7 +1698,6 @@ static int da7213_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct da7213_priv *da7213; - struct da7213_platform_data *pdata = dev_get_platdata(&i2c->dev); int ret; da7213 = devm_kzalloc(&i2c->dev, sizeof(struct da7213_priv), @@ -1545,9 +1705,6 @@ static int da7213_i2c_probe(struct i2c_client *i2c, if (!da7213) return -ENOMEM; - if (pdata) - da7213->pdata = pdata; - i2c_set_clientdata(i2c, da7213); da7213->regmap = devm_regmap_init_i2c(i2c, &da7213_regmap_config); @@ -1582,6 +1739,7 @@ MODULE_DEVICE_TABLE(i2c, da7213_i2c_id); static struct i2c_driver da7213_i2c_driver = { .driver = { .name = "da7213", + .of_match_table = of_match_ptr(da7213_of_match), }, .probe = da7213_i2c_probe, .remove = da7213_remove, diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 9cb9ddd01282..030fd691b076 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -13,6 +13,7 @@ #ifndef _DA7213_H #define _DA7213_H +#include <linux/clk.h> #include <linux/regmap.h> #include <sound/da7213.h> @@ -504,14 +505,17 @@ #define DA7213_PLL_INDIV_20_40_MHZ_VAL 8 #define DA7213_PLL_INDIV_40_54_MHZ_VAL 16 -enum clk_src { - DA7213_CLKSRC_MCLK +enum da7213_clk_src { + DA7213_CLKSRC_MCLK = 0, + DA7213_CLKSRC_MCLK_SQR, }; /* Codec private data */ struct da7213_priv { struct regmap *regmap; + struct clk *mclk; unsigned int mclk_rate; + int clk_src; bool master; bool mclk_squarer_en; bool srm_en; diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c new file mode 100644 index 000000000000..9459593eef13 --- /dev/null +++ b/sound/soc/codecs/da7219-aad.c @@ -0,0 +1,823 @@ +/* + * da7219-aad.c - Dialog DA7219 ALSA SoC AAD Driver + * + * Copyright (c) 2015 Dialog Semiconductor Ltd. + * + * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/of_device.h> +#include <linux/of_irq.h> +#include <linux/pm_wakeirq.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/workqueue.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/da7219.h> + +#include "da7219.h" +#include "da7219-aad.h" + + +/* + * Detection control + */ + +void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + da7219->aad->jack = jack; + da7219->aad->jack_inserted = false; + + /* Send an initial empty report */ + snd_soc_jack_report(jack, 0, DA7219_AAD_REPORT_ALL_MASK); + + /* Enable/Disable jack detection */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_ACCDET_EN_MASK, + (jack ? DA7219_ACCDET_EN_MASK : 0)); +} +EXPORT_SYMBOL_GPL(da7219_aad_jack_det); + +/* + * Button/HPTest work + */ + +static void da7219_aad_btn_det_work(struct work_struct *work) +{ + struct da7219_aad_priv *da7219_aad = + container_of(work, struct da7219_aad_priv, btn_det_work); + struct snd_soc_codec *codec = da7219_aad->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + u8 statusa, micbias_ctrl; + bool micbias_up = false; + int retries = 0; + + /* Drive headphones/lineout */ + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK, + DA7219_HP_L_AMP_OE_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK, + DA7219_HP_R_AMP_OE_MASK); + + /* Make sure mic bias is up */ + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + snd_soc_dapm_sync(dapm); + + do { + statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A); + if (statusa & DA7219_MICBIAS_UP_STS_MASK) + micbias_up = true; + else if (retries++ < DA7219_AAD_MICBIAS_CHK_RETRIES) + msleep(DA7219_AAD_MICBIAS_CHK_DELAY); + } while ((!micbias_up) && (retries < DA7219_AAD_MICBIAS_CHK_RETRIES)); + + if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES) + dev_warn(codec->dev, "Mic bias status check timed out"); + + /* + * Mic bias pulse required to enable mic, must be done before enabling + * button detection to prevent erroneous button readings. + */ + if (da7219_aad->micbias_pulse_lvl && da7219_aad->micbias_pulse_time) { + /* Pulse higher level voltage */ + micbias_ctrl = snd_soc_read(codec, DA7219_MICBIAS_CTRL); + snd_soc_update_bits(codec, DA7219_MICBIAS_CTRL, + DA7219_MICBIAS1_LEVEL_MASK, + da7219_aad->micbias_pulse_lvl); + msleep(da7219_aad->micbias_pulse_time); + snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_ctrl); + + } + + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_BUTTON_CONFIG_MASK, + da7219_aad->btn_cfg); +} + +static void da7219_aad_hptest_work(struct work_struct *work) +{ + struct da7219_aad_priv *da7219_aad = + container_of(work, struct da7219_aad_priv, hptest_work); + struct snd_soc_codec *codec = da7219_aad->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + u16 tonegen_freq_hptest; + u8 accdet_cfg8; + int report = 0; + + /* Lock DAPM and any Kcontrols that are affected by this test */ + snd_soc_dapm_mutex_lock(dapm); + mutex_lock(&da7219->lock); + + /* Bypass cache so it saves current settings */ + regcache_cache_bypass(da7219->regmap, true); + + /* Make sure Tone Generator is disabled */ + snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0); + + /* Enable HPTest block, 1KOhms check */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8, + DA7219_HPTEST_EN_MASK | DA7219_HPTEST_RES_SEL_MASK, + DA7219_HPTEST_EN_MASK | + DA7219_HPTEST_RES_SEL_1KOHMS); + + /* Set gains to 0db */ + snd_soc_write(codec, DA7219_DAC_L_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB); + snd_soc_write(codec, DA7219_DAC_R_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB); + snd_soc_write(codec, DA7219_HP_L_GAIN, DA7219_HP_AMP_GAIN_0DB); + snd_soc_write(codec, DA7219_HP_R_GAIN, DA7219_HP_AMP_GAIN_0DB); + + /* Disable DAC filters, EQs and soft mute */ + snd_soc_update_bits(codec, DA7219_DAC_FILTERS1, DA7219_HPF_MODE_MASK, + 0); + snd_soc_update_bits(codec, DA7219_DAC_FILTERS4, DA7219_DAC_EQ_EN_MASK, + 0); + snd_soc_update_bits(codec, DA7219_DAC_FILTERS5, + DA7219_DAC_SOFTMUTE_EN_MASK, 0); + + /* Enable HP left & right paths */ + snd_soc_update_bits(codec, DA7219_CP_CTRL, DA7219_CP_EN_MASK, + DA7219_CP_EN_MASK); + snd_soc_update_bits(codec, DA7219_DIG_ROUTING_DAC, + DA7219_DAC_L_SRC_MASK | DA7219_DAC_R_SRC_MASK, + DA7219_DAC_L_SRC_TONEGEN | + DA7219_DAC_R_SRC_TONEGEN); + snd_soc_update_bits(codec, DA7219_DAC_L_CTRL, + DA7219_DAC_L_EN_MASK | DA7219_DAC_L_MUTE_EN_MASK, + DA7219_DAC_L_EN_MASK); + snd_soc_update_bits(codec, DA7219_DAC_R_CTRL, + DA7219_DAC_R_EN_MASK | DA7219_DAC_R_MUTE_EN_MASK, + DA7219_DAC_R_EN_MASK); + snd_soc_update_bits(codec, DA7219_MIXOUT_L_SELECT, + DA7219_MIXOUT_L_MIX_SELECT_MASK, + DA7219_MIXOUT_L_MIX_SELECT_MASK); + snd_soc_update_bits(codec, DA7219_MIXOUT_R_SELECT, + DA7219_MIXOUT_R_MIX_SELECT_MASK, + DA7219_MIXOUT_R_MIX_SELECT_MASK); + snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1L, + DA7219_OUTFILT_ST_1L_SRC_MASK, + DA7219_DMIX_ST_SRC_OUTFILT1L); + snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1R, + DA7219_OUTFILT_ST_1R_SRC_MASK, + DA7219_DMIX_ST_SRC_OUTFILT1R); + snd_soc_update_bits(codec, DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_L_AMP_EN_MASK, + DA7219_MIXOUT_L_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_MIXOUT_R_CTRL, + DA7219_MIXOUT_R_AMP_EN_MASK, + DA7219_MIXOUT_R_AMP_EN_MASK); + snd_soc_write(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK); + snd_soc_write(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK); + + /* Configure & start Tone Generator */ + snd_soc_write(codec, DA7219_TONE_GEN_ON_PER, DA7219_BEEP_ON_PER_MASK); + tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ); + regmap_raw_write(da7219->regmap, DA7219_TONE_GEN_FREQ1_L, + &tonegen_freq_hptest, sizeof(tonegen_freq_hptest)); + snd_soc_update_bits(codec, DA7219_TONE_GEN_CFG2, + DA7219_SWG_SEL_MASK | DA7219_TONE_GEN_GAIN_MASK, + DA7219_SWG_SEL_SRAMP | + DA7219_TONE_GEN_GAIN_MINUS_15DB); + snd_soc_write(codec, DA7219_TONE_GEN_CFG1, DA7219_START_STOPN_MASK); + + msleep(DA7219_AAD_HPTEST_PERIOD); + + /* Grab comparator reading */ + accdet_cfg8 = snd_soc_read(codec, DA7219_ACCDET_CONFIG_8); + if (accdet_cfg8 & DA7219_HPTEST_COMP_MASK) + report |= SND_JACK_HEADPHONE; + else + report |= SND_JACK_LINEOUT; + + /* Stop tone generator */ + snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0); + + msleep(DA7219_AAD_HPTEST_PERIOD); + + /* Restore original settings from cache */ + regcache_mark_dirty(da7219->regmap); + regcache_sync_region(da7219->regmap, DA7219_HP_L_CTRL, + DA7219_HP_R_CTRL); + regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_R_CTRL); + regcache_sync_region(da7219->regmap, DA7219_DROUTING_ST_OUTFILT_1L, + DA7219_DROUTING_ST_OUTFILT_1R); + regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_SELECT, + DA7219_MIXOUT_R_SELECT); + regcache_sync_region(da7219->regmap, DA7219_DAC_L_CTRL, + DA7219_DAC_R_CTRL); + regcache_sync_region(da7219->regmap, DA7219_DIG_ROUTING_DAC, + DA7219_DIG_ROUTING_DAC); + regcache_sync_region(da7219->regmap, DA7219_CP_CTRL, DA7219_CP_CTRL); + regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS5, + DA7219_DAC_FILTERS5); + regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS4, + DA7219_DAC_FILTERS1); + regcache_sync_region(da7219->regmap, DA7219_HP_L_GAIN, + DA7219_HP_R_GAIN); + regcache_sync_region(da7219->regmap, DA7219_DAC_L_GAIN, + DA7219_DAC_R_GAIN); + regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_ON_PER, + DA7219_TONE_GEN_ON_PER); + regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_FREQ1_L, + DA7219_TONE_GEN_FREQ1_U); + regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_CFG1, + DA7219_TONE_GEN_CFG2); + + regcache_cache_bypass(da7219->regmap, false); + + /* Disable HPTest block */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8, + DA7219_HPTEST_EN_MASK, 0); + + /* Drive Headphones/lineout */ + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_OE_MASK, + DA7219_HP_L_AMP_OE_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_OE_MASK, + DA7219_HP_R_AMP_OE_MASK); + + mutex_unlock(&da7219->lock); + snd_soc_dapm_mutex_unlock(dapm); + + /* + * Only send report if jack hasn't been removed during process, + * otherwise it's invalid and we drop it. + */ + if (da7219_aad->jack_inserted) + snd_soc_jack_report(da7219_aad->jack, report, + SND_JACK_HEADSET | SND_JACK_LINEOUT); +} + + +/* + * IRQ + */ + +static irqreturn_t da7219_aad_irq_thread(int irq, void *data) +{ + struct da7219_aad_priv *da7219_aad = data; + struct snd_soc_codec *codec = da7219_aad->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 events[DA7219_AAD_IRQ_REG_MAX]; + u8 statusa; + int i, report = 0, mask = 0; + + /* Read current IRQ events */ + regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + + if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) + return IRQ_NONE; + + /* Read status register for jack insertion & type status */ + statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A); + + /* Clear events */ + regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + + dev_dbg(codec->dev, "IRQ events = 0x%x|0x%x, status = 0x%x\n", + events[DA7219_AAD_IRQ_REG_A], events[DA7219_AAD_IRQ_REG_B], + statusa); + + if (statusa & DA7219_JACK_INSERTION_STS_MASK) { + /* Jack Insertion */ + if (events[DA7219_AAD_IRQ_REG_A] & + DA7219_E_JACK_INSERTED_MASK) { + report |= SND_JACK_MECHANICAL; + mask |= SND_JACK_MECHANICAL; + da7219_aad->jack_inserted = true; + } + + /* Jack type detection */ + if (events[DA7219_AAD_IRQ_REG_A] & + DA7219_E_JACK_DETECT_COMPLETE_MASK) { + /* + * If 4-pole, then enable button detection, else perform + * HP impedance test to determine output type to report. + * + * We schedule work here as the tasks themselves can + * take time to complete, and in particular for hptest + * we want to be able to check if the jack was removed + * during the procedure as this will invalidate the + * result. By doing this as work, the IRQ thread can + * handle a removal, and we can check at the end of + * hptest if we have a valid result or not. + */ + if (statusa & DA7219_JACK_TYPE_STS_MASK) { + report |= SND_JACK_HEADSET; + mask |= SND_JACK_HEADSET | SND_JACK_LINEOUT; + schedule_work(&da7219_aad->btn_det_work); + } else { + schedule_work(&da7219_aad->hptest_work); + } + } + + /* Button support for 4-pole jack */ + if (statusa & DA7219_JACK_TYPE_STS_MASK) { + for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) { + /* Button Press */ + if (events[DA7219_AAD_IRQ_REG_B] & + (DA7219_E_BUTTON_A_PRESSED_MASK << i)) { + report |= SND_JACK_BTN_0 >> i; + mask |= SND_JACK_BTN_0 >> i; + } + } + snd_soc_jack_report(da7219_aad->jack, report, mask); + + for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) { + /* Button Release */ + if (events[DA7219_AAD_IRQ_REG_B] & + (DA7219_E_BUTTON_A_RELEASED_MASK >> i)) { + report &= ~(SND_JACK_BTN_0 >> i); + mask |= SND_JACK_BTN_0 >> i; + } + } + } + } else { + /* Jack removal */ + if (events[DA7219_AAD_IRQ_REG_A] & DA7219_E_JACK_REMOVED_MASK) { + report = 0; + mask |= DA7219_AAD_REPORT_ALL_MASK; + da7219_aad->jack_inserted = false; + + /* Un-drive headphones/lineout */ + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK, 0); + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK, 0); + + /* Ensure button detection disabled */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_BUTTON_CONFIG_MASK, 0); + + /* Disable mic bias */ + snd_soc_dapm_disable_pin(dapm, "Mic Bias"); + snd_soc_dapm_sync(dapm); + + /* Cancel any pending work */ + cancel_work_sync(&da7219_aad->btn_det_work); + cancel_work_sync(&da7219_aad->hptest_work); + } + } + + snd_soc_jack_report(da7219_aad->jack, report, mask); + + return IRQ_HANDLED; +} + +/* + * DT to pdata conversion + */ + +static enum da7219_aad_micbias_pulse_lvl + da7219_aad_of_micbias_pulse_lvl(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 2800: + return DA7219_AAD_MICBIAS_PULSE_LVL_2_8V; + case 2900: + return DA7219_AAD_MICBIAS_PULSE_LVL_2_9V; + default: + dev_warn(codec->dev, "Invalid micbias pulse level"); + return DA7219_AAD_MICBIAS_PULSE_LVL_OFF; + } +} + +static enum da7219_aad_btn_cfg + da7219_aad_of_btn_cfg(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 2: + return DA7219_AAD_BTN_CFG_2MS; + case 5: + return DA7219_AAD_BTN_CFG_5MS; + case 10: + return DA7219_AAD_BTN_CFG_10MS; + case 50: + return DA7219_AAD_BTN_CFG_50MS; + case 100: + return DA7219_AAD_BTN_CFG_100MS; + case 200: + return DA7219_AAD_BTN_CFG_200MS; + case 500: + return DA7219_AAD_BTN_CFG_500MS; + default: + dev_warn(codec->dev, "Invalid button config"); + return DA7219_AAD_BTN_CFG_10MS; + } +} + +static enum da7219_aad_mic_det_thr + da7219_aad_of_mic_det_thr(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 200: + return DA7219_AAD_MIC_DET_THR_200_OHMS; + case 500: + return DA7219_AAD_MIC_DET_THR_500_OHMS; + case 750: + return DA7219_AAD_MIC_DET_THR_750_OHMS; + case 1000: + return DA7219_AAD_MIC_DET_THR_1000_OHMS; + default: + dev_warn(codec->dev, "Invalid mic detect threshold"); + return DA7219_AAD_MIC_DET_THR_500_OHMS; + } +} + +static enum da7219_aad_jack_ins_deb + da7219_aad_of_jack_ins_deb(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 5: + return DA7219_AAD_JACK_INS_DEB_5MS; + case 10: + return DA7219_AAD_JACK_INS_DEB_10MS; + case 20: + return DA7219_AAD_JACK_INS_DEB_20MS; + case 50: + return DA7219_AAD_JACK_INS_DEB_50MS; + case 100: + return DA7219_AAD_JACK_INS_DEB_100MS; + case 200: + return DA7219_AAD_JACK_INS_DEB_200MS; + case 500: + return DA7219_AAD_JACK_INS_DEB_500MS; + case 1000: + return DA7219_AAD_JACK_INS_DEB_1S; + default: + dev_warn(codec->dev, "Invalid jack insert debounce"); + return DA7219_AAD_JACK_INS_DEB_20MS; + } +} + +static enum da7219_aad_jack_det_rate + da7219_aad_of_jack_det_rate(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "32ms_64ms")) { + return DA7219_AAD_JACK_DET_RATE_32_64MS; + } else if (!strcmp(str, "64ms_128ms")) { + return DA7219_AAD_JACK_DET_RATE_64_128MS; + } else if (!strcmp(str, "128ms_256ms")) { + return DA7219_AAD_JACK_DET_RATE_128_256MS; + } else if (!strcmp(str, "256ms_512ms")) { + return DA7219_AAD_JACK_DET_RATE_256_512MS; + } else { + dev_warn(codec->dev, "Invalid jack detect rate"); + return DA7219_AAD_JACK_DET_RATE_256_512MS; + } +} + +static enum da7219_aad_jack_rem_deb + da7219_aad_of_jack_rem_deb(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1: + return DA7219_AAD_JACK_REM_DEB_1MS; + case 5: + return DA7219_AAD_JACK_REM_DEB_5MS; + case 10: + return DA7219_AAD_JACK_REM_DEB_10MS; + case 20: + return DA7219_AAD_JACK_REM_DEB_20MS; + default: + dev_warn(codec->dev, "Invalid jack removal debounce"); + return DA7219_AAD_JACK_REM_DEB_1MS; + } +} + +static enum da7219_aad_btn_avg + da7219_aad_of_btn_avg(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1: + return DA7219_AAD_BTN_AVG_1; + case 2: + return DA7219_AAD_BTN_AVG_2; + case 4: + return DA7219_AAD_BTN_AVG_4; + case 8: + return DA7219_AAD_BTN_AVG_8; + default: + dev_warn(codec->dev, "Invalid button average value"); + return DA7219_AAD_BTN_AVG_2; + } +} + +static enum da7219_aad_adc_1bit_rpt + da7219_aad_of_adc_1bit_rpt(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1: + return DA7219_AAD_ADC_1BIT_RPT_1; + case 2: + return DA7219_AAD_ADC_1BIT_RPT_2; + case 4: + return DA7219_AAD_ADC_1BIT_RPT_4; + case 8: + return DA7219_AAD_ADC_1BIT_RPT_8; + default: + dev_warn(codec->dev, "Invalid ADC 1-bit repeat value"); + return DA7219_AAD_ADC_1BIT_RPT_1; + } +} + +static struct da7219_aad_pdata *da7219_aad_of_to_pdata(struct snd_soc_codec *codec) +{ + struct device_node *np = codec->dev->of_node; + struct device_node *aad_np = of_find_node_by_name(np, "da7219_aad"); + struct da7219_aad_pdata *aad_pdata; + const char *of_str; + u32 of_val32; + + if (!aad_np) + return NULL; + + aad_pdata = devm_kzalloc(codec->dev, sizeof(*aad_pdata), GFP_KERNEL); + if (!aad_pdata) + goto out; + + aad_pdata->irq = irq_of_parse_and_map(np, 0); + + if (of_property_read_u32(aad_np, "dlg,micbias-pulse-lvl", + &of_val32) >= 0) + aad_pdata->micbias_pulse_lvl = + da7219_aad_of_micbias_pulse_lvl(codec, of_val32); + else + aad_pdata->micbias_pulse_lvl = DA7219_AAD_MICBIAS_PULSE_LVL_OFF; + + if (of_property_read_u32(aad_np, "dlg,micbias-pulse-time", + &of_val32) >= 0) + aad_pdata->micbias_pulse_time = of_val32; + + if (of_property_read_u32(aad_np, "dlg,btn-cfg", &of_val32) >= 0) + aad_pdata->btn_cfg = da7219_aad_of_btn_cfg(codec, of_val32); + else + aad_pdata->btn_cfg = DA7219_AAD_BTN_CFG_10MS; + + if (of_property_read_u32(aad_np, "dlg,mic-det-thr", &of_val32) >= 0) + aad_pdata->mic_det_thr = + da7219_aad_of_mic_det_thr(codec, of_val32); + else + aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_500_OHMS; + + if (of_property_read_u32(aad_np, "dlg,jack-ins-deb", &of_val32) >= 0) + aad_pdata->jack_ins_deb = + da7219_aad_of_jack_ins_deb(codec, of_val32); + else + aad_pdata->jack_ins_deb = DA7219_AAD_JACK_INS_DEB_20MS; + + if (!of_property_read_string(aad_np, "dlg,jack-det-rate", &of_str)) + aad_pdata->jack_det_rate = + da7219_aad_of_jack_det_rate(codec, of_str); + else + aad_pdata->jack_det_rate = DA7219_AAD_JACK_DET_RATE_256_512MS; + + if (of_property_read_u32(aad_np, "dlg,jack-rem-deb", &of_val32) >= 0) + aad_pdata->jack_rem_deb = + da7219_aad_of_jack_rem_deb(codec, of_val32); + else + aad_pdata->jack_rem_deb = DA7219_AAD_JACK_REM_DEB_1MS; + + if (of_property_read_u32(aad_np, "dlg,a-d-btn-thr", &of_val32) >= 0) + aad_pdata->a_d_btn_thr = (u8) of_val32; + else + aad_pdata->a_d_btn_thr = 0xA; + + if (of_property_read_u32(aad_np, "dlg,d-b-btn-thr", &of_val32) >= 0) + aad_pdata->d_b_btn_thr = (u8) of_val32; + else + aad_pdata->d_b_btn_thr = 0x16; + + if (of_property_read_u32(aad_np, "dlg,b-c-btn-thr", &of_val32) >= 0) + aad_pdata->b_c_btn_thr = (u8) of_val32; + else + aad_pdata->b_c_btn_thr = 0x21; + + if (of_property_read_u32(aad_np, "dlg,c-mic-btn-thr", &of_val32) >= 0) + aad_pdata->c_mic_btn_thr = (u8) of_val32; + else + aad_pdata->c_mic_btn_thr = 0x3E; + + if (of_property_read_u32(aad_np, "dlg,btn-avg", &of_val32) >= 0) + aad_pdata->btn_avg = da7219_aad_of_btn_avg(codec, of_val32); + else + aad_pdata->btn_avg = DA7219_AAD_BTN_AVG_2; + + if (of_property_read_u32(aad_np, "dlg,adc-1bit-rpt", &of_val32) >= 0) + aad_pdata->adc_1bit_rpt = + da7219_aad_of_adc_1bit_rpt(codec, of_val32); + else + aad_pdata->adc_1bit_rpt = DA7219_AAD_ADC_1BIT_RPT_1; + +out: + of_node_put(aad_np); + + return aad_pdata; +} + +static void da7219_aad_handle_pdata(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad = da7219->aad; + struct da7219_pdata *pdata = da7219->pdata; + + if ((pdata) && (pdata->aad_pdata)) { + struct da7219_aad_pdata *aad_pdata = pdata->aad_pdata; + u8 cfg, mask; + + da7219_aad->irq = aad_pdata->irq; + + switch (aad_pdata->micbias_pulse_lvl) { + case DA7219_AAD_MICBIAS_PULSE_LVL_2_8V: + case DA7219_AAD_MICBIAS_PULSE_LVL_2_9V: + da7219_aad->micbias_pulse_lvl = + (aad_pdata->micbias_pulse_lvl << + DA7219_MICBIAS1_LEVEL_SHIFT); + break; + default: + break; + } + + da7219_aad->micbias_pulse_time = aad_pdata->micbias_pulse_time; + + switch (aad_pdata->btn_cfg) { + case DA7219_AAD_BTN_CFG_2MS: + case DA7219_AAD_BTN_CFG_5MS: + case DA7219_AAD_BTN_CFG_10MS: + case DA7219_AAD_BTN_CFG_50MS: + case DA7219_AAD_BTN_CFG_100MS: + case DA7219_AAD_BTN_CFG_200MS: + case DA7219_AAD_BTN_CFG_500MS: + da7219_aad->btn_cfg = (aad_pdata->btn_cfg << + DA7219_BUTTON_CONFIG_SHIFT); + } + + cfg = 0; + mask = 0; + switch (aad_pdata->mic_det_thr) { + case DA7219_AAD_MIC_DET_THR_200_OHMS: + case DA7219_AAD_MIC_DET_THR_500_OHMS: + case DA7219_AAD_MIC_DET_THR_750_OHMS: + case DA7219_AAD_MIC_DET_THR_1000_OHMS: + cfg |= (aad_pdata->mic_det_thr << + DA7219_MIC_DET_THRESH_SHIFT); + mask |= DA7219_MIC_DET_THRESH_MASK; + } + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, mask, cfg); + + cfg = 0; + mask = 0; + switch (aad_pdata->jack_ins_deb) { + case DA7219_AAD_JACK_INS_DEB_5MS: + case DA7219_AAD_JACK_INS_DEB_10MS: + case DA7219_AAD_JACK_INS_DEB_20MS: + case DA7219_AAD_JACK_INS_DEB_50MS: + case DA7219_AAD_JACK_INS_DEB_100MS: + case DA7219_AAD_JACK_INS_DEB_200MS: + case DA7219_AAD_JACK_INS_DEB_500MS: + case DA7219_AAD_JACK_INS_DEB_1S: + cfg |= (aad_pdata->jack_ins_deb << + DA7219_JACKDET_DEBOUNCE_SHIFT); + mask |= DA7219_JACKDET_DEBOUNCE_MASK; + } + switch (aad_pdata->jack_det_rate) { + case DA7219_AAD_JACK_DET_RATE_32_64MS: + case DA7219_AAD_JACK_DET_RATE_64_128MS: + case DA7219_AAD_JACK_DET_RATE_128_256MS: + case DA7219_AAD_JACK_DET_RATE_256_512MS: + cfg |= (aad_pdata->jack_det_rate << + DA7219_JACK_DETECT_RATE_SHIFT); + mask |= DA7219_JACK_DETECT_RATE_MASK; + } + switch (aad_pdata->jack_rem_deb) { + case DA7219_AAD_JACK_REM_DEB_1MS: + case DA7219_AAD_JACK_REM_DEB_5MS: + case DA7219_AAD_JACK_REM_DEB_10MS: + case DA7219_AAD_JACK_REM_DEB_20MS: + cfg |= (aad_pdata->jack_rem_deb << + DA7219_JACKDET_REM_DEB_SHIFT); + mask |= DA7219_JACKDET_REM_DEB_MASK; + } + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_2, mask, cfg); + + snd_soc_write(codec, DA7219_ACCDET_CONFIG_3, + aad_pdata->a_d_btn_thr); + snd_soc_write(codec, DA7219_ACCDET_CONFIG_4, + aad_pdata->d_b_btn_thr); + snd_soc_write(codec, DA7219_ACCDET_CONFIG_5, + aad_pdata->b_c_btn_thr); + snd_soc_write(codec, DA7219_ACCDET_CONFIG_6, + aad_pdata->c_mic_btn_thr); + + cfg = 0; + mask = 0; + switch (aad_pdata->btn_avg) { + case DA7219_AAD_BTN_AVG_1: + case DA7219_AAD_BTN_AVG_2: + case DA7219_AAD_BTN_AVG_4: + case DA7219_AAD_BTN_AVG_8: + cfg |= (aad_pdata->btn_avg << + DA7219_BUTTON_AVERAGE_SHIFT); + mask |= DA7219_BUTTON_AVERAGE_MASK; + } + switch (aad_pdata->adc_1bit_rpt) { + case DA7219_AAD_ADC_1BIT_RPT_1: + case DA7219_AAD_ADC_1BIT_RPT_2: + case DA7219_AAD_ADC_1BIT_RPT_4: + case DA7219_AAD_ADC_1BIT_RPT_8: + cfg |= (aad_pdata->adc_1bit_rpt << + DA7219_ADC_1_BIT_REPEAT_SHIFT); + mask |= DA7219_ADC_1_BIT_REPEAT_MASK; + } + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_7, mask, cfg); + } +} + + +/* + * Init/Exit + */ + +int da7219_aad_init(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad; + u8 mask[DA7219_AAD_IRQ_REG_MAX]; + int ret; + + da7219_aad = devm_kzalloc(codec->dev, sizeof(*da7219_aad), GFP_KERNEL); + if (!da7219_aad) + return -ENOMEM; + + da7219->aad = da7219_aad; + da7219_aad->codec = codec; + + /* Handle any DT/platform data */ + if ((codec->dev->of_node) && (da7219->pdata)) + da7219->pdata->aad_pdata = da7219_aad_of_to_pdata(codec); + + da7219_aad_handle_pdata(codec); + + /* Disable button detection */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_BUTTON_CONFIG_MASK, 0); + + INIT_WORK(&da7219_aad->btn_det_work, da7219_aad_btn_det_work); + INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work); + + ret = request_threaded_irq(da7219_aad->irq, NULL, + da7219_aad_irq_thread, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "da7219-aad", da7219_aad); + if (ret) { + dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + return ret; + } + + /* Unmask AAD IRQs */ + memset(mask, 0, DA7219_AAD_IRQ_REG_MAX); + regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A, + &mask, DA7219_AAD_IRQ_REG_MAX); + + return 0; +} +EXPORT_SYMBOL_GPL(da7219_aad_init); + +void da7219_aad_exit(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad = da7219->aad; + u8 mask[DA7219_AAD_IRQ_REG_MAX]; + + /* Mask off AAD IRQs */ + memset(mask, DA7219_BYTE_MASK, DA7219_AAD_IRQ_REG_MAX); + regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A, + mask, DA7219_AAD_IRQ_REG_MAX); + + free_irq(da7219_aad->irq, da7219_aad); + + cancel_work_sync(&da7219_aad->btn_det_work); + cancel_work_sync(&da7219_aad->hptest_work); +} +EXPORT_SYMBOL_GPL(da7219_aad_exit); + +MODULE_DESCRIPTION("ASoC DA7219 AAD Driver"); +MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource@diasemi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h new file mode 100644 index 000000000000..4fccf677cd06 --- /dev/null +++ b/sound/soc/codecs/da7219-aad.h @@ -0,0 +1,212 @@ +/* + * da7219-aad.h - DA7322 ASoC AAD Driver + * + * Copyright (c) 2015 Dialog Semiconductor Ltd. + * + * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef __DA7219_AAD_H +#define __DA7219_AAD_H + +#include <linux/timer.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/da7219-aad.h> + +/* + * Registers + */ + +#define DA7219_ACCDET_STATUS_A 0xC0 +#define DA7219_ACCDET_STATUS_B 0xC1 +#define DA7219_ACCDET_IRQ_EVENT_A 0xC2 +#define DA7219_ACCDET_IRQ_EVENT_B 0xC3 +#define DA7219_ACCDET_IRQ_MASK_A 0xC4 +#define DA7219_ACCDET_IRQ_MASK_B 0xC5 +#define DA7219_ACCDET_CONFIG_1 0xC6 +#define DA7219_ACCDET_CONFIG_2 0xC7 +#define DA7219_ACCDET_CONFIG_3 0xC8 +#define DA7219_ACCDET_CONFIG_4 0xC9 +#define DA7219_ACCDET_CONFIG_5 0xCA +#define DA7219_ACCDET_CONFIG_6 0xCB +#define DA7219_ACCDET_CONFIG_7 0xCC +#define DA7219_ACCDET_CONFIG_8 0xCD + + +/* + * Bit Fields + */ + +/* DA7219_ACCDET_STATUS_A = 0xC0 */ +#define DA7219_JACK_INSERTION_STS_SHIFT 0 +#define DA7219_JACK_INSERTION_STS_MASK (0x1 << 0) +#define DA7219_JACK_TYPE_STS_SHIFT 1 +#define DA7219_JACK_TYPE_STS_MASK (0x1 << 1) +#define DA7219_JACK_PIN_ORDER_STS_SHIFT 2 +#define DA7219_JACK_PIN_ORDER_STS_MASK (0x1 << 2) +#define DA7219_MICBIAS_UP_STS_SHIFT 3 +#define DA7219_MICBIAS_UP_STS_MASK (0x1 << 3) + +/* DA7219_ACCDET_STATUS_B = 0xC1 */ +#define DA7219_BUTTON_TYPE_STS_SHIFT 0 +#define DA7219_BUTTON_TYPE_STS_MASK (0xFF << 0) + +/* DA7219_ACCDET_IRQ_EVENT_A = 0xC2 */ +#define DA7219_E_JACK_INSERTED_SHIFT 0 +#define DA7219_E_JACK_INSERTED_MASK (0x1 << 0) +#define DA7219_E_JACK_REMOVED_SHIFT 1 +#define DA7219_E_JACK_REMOVED_MASK (0x1 << 1) +#define DA7219_E_JACK_DETECT_COMPLETE_SHIFT 2 +#define DA7219_E_JACK_DETECT_COMPLETE_MASK (0x1 << 2) + +/* DA7219_ACCDET_IRQ_EVENT_B = 0xC3 */ +#define DA7219_E_BUTTON_A_PRESSED_SHIFT 0 +#define DA7219_E_BUTTON_A_PRESSED_MASK (0x1 << 0) +#define DA7219_E_BUTTON_B_PRESSED_SHIFT 1 +#define DA7219_E_BUTTON_B_PRESSED_MASK (0x1 << 1) +#define DA7219_E_BUTTON_C_PRESSED_SHIFT 2 +#define DA7219_E_BUTTON_C_PRESSED_MASK (0x1 << 2) +#define DA7219_E_BUTTON_D_PRESSED_SHIFT 3 +#define DA7219_E_BUTTON_D_PRESSED_MASK (0x1 << 3) +#define DA7219_E_BUTTON_D_RELEASED_SHIFT 4 +#define DA7219_E_BUTTON_D_RELEASED_MASK (0x1 << 4) +#define DA7219_E_BUTTON_C_RELEASED_SHIFT 5 +#define DA7219_E_BUTTON_C_RELEASED_MASK (0x1 << 5) +#define DA7219_E_BUTTON_B_RELEASED_SHIFT 6 +#define DA7219_E_BUTTON_B_RELEASED_MASK (0x1 << 6) +#define DA7219_E_BUTTON_A_RELEASED_SHIFT 7 +#define DA7219_E_BUTTON_A_RELEASED_MASK (0x1 << 7) + +/* DA7219_ACCDET_IRQ_MASK_A = 0xC4 */ +#define DA7219_M_JACK_INSERTED_SHIFT 0 +#define DA7219_M_JACK_INSERTED_MASK (0x1 << 0) +#define DA7219_M_JACK_REMOVED_SHIFT 1 +#define DA7219_M_JACK_REMOVED_MASK (0x1 << 1) +#define DA7219_M_JACK_DETECT_COMPLETE_SHIFT 2 +#define DA7219_M_JACK_DETECT_COMPLETE_MASK (0x1 << 2) + +/* DA7219_ACCDET_IRQ_MASK_B = 0xC5 */ +#define DA7219_M_BUTTON_A_PRESSED_SHIFT 0 +#define DA7219_M_BUTTON_A_PRESSED_MASK (0x1 << 0) +#define DA7219_M_BUTTON_B_PRESSED_SHIFT 1 +#define DA7219_M_BUTTON_B_PRESSED_MASK (0x1 << 1) +#define DA7219_M_BUTTON_C_PRESSED_SHIFT 2 +#define DA7219_M_BUTTON_C_PRESSED_MASK (0x1 << 2) +#define DA7219_M_BUTTON_D_PRESSED_SHIFT 3 +#define DA7219_M_BUTTON_D_PRESSED_MASK (0x1 << 3) +#define DA7219_M_BUTTON_D_RELEASED_SHIFT 4 +#define DA7219_M_BUTTON_D_RELEASED_MASK (0x1 << 4) +#define DA7219_M_BUTTON_C_RELEASED_SHIFT 5 +#define DA7219_M_BUTTON_C_RELEASED_MASK (0x1 << 5) +#define DA7219_M_BUTTON_B_RELEASED_SHIFT 6 +#define DA7219_M_BUTTON_B_RELEASED_MASK (0x1 << 6) +#define DA7219_M_BUTTON_A_RELEASED_SHIFT 7 +#define DA7219_M_BUTTON_A_RELEASED_MASK (0x1 << 7) + +/* DA7219_ACCDET_CONFIG_1 = 0xC6 */ +#define DA7219_ACCDET_EN_SHIFT 0 +#define DA7219_ACCDET_EN_MASK (0x1 << 0) +#define DA7219_BUTTON_CONFIG_SHIFT 1 +#define DA7219_BUTTON_CONFIG_MASK (0x7 << 1) +#define DA7219_MIC_DET_THRESH_SHIFT 4 +#define DA7219_MIC_DET_THRESH_MASK (0x3 << 4) +#define DA7219_JACK_TYPE_DET_EN_SHIFT 6 +#define DA7219_JACK_TYPE_DET_EN_MASK (0x1 << 6) +#define DA7219_PIN_ORDER_DET_EN_SHIFT 7 +#define DA7219_PIN_ORDER_DET_EN_MASK (0x1 << 7) + +/* DA7219_ACCDET_CONFIG_2 = 0xC7 */ +#define DA7219_ACCDET_PAUSE_SHIFT 0 +#define DA7219_ACCDET_PAUSE_MASK (0x1 << 0) +#define DA7219_JACKDET_DEBOUNCE_SHIFT 1 +#define DA7219_JACKDET_DEBOUNCE_MASK (0x7 << 1) +#define DA7219_JACK_DETECT_RATE_SHIFT 4 +#define DA7219_JACK_DETECT_RATE_MASK (0x3 << 4) +#define DA7219_JACKDET_REM_DEB_SHIFT 6 +#define DA7219_JACKDET_REM_DEB_MASK (0x3 << 6) + +/* DA7219_ACCDET_CONFIG_3 = 0xC8 */ +#define DA7219_A_D_BUTTON_THRESH_SHIFT 0 +#define DA7219_A_D_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_4 = 0xC9 */ +#define DA7219_D_B_BUTTON_THRESH_SHIFT 0 +#define DA7219_D_B_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_5 = 0xCA */ +#define DA7219_B_C_BUTTON_THRESH_SHIFT 0 +#define DA7219_B_C_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_6 = 0xCB */ +#define DA7219_C_MIC_BUTTON_THRESH_SHIFT 0 +#define DA7219_C_MIC_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_7 = 0xCC */ +#define DA7219_BUTTON_AVERAGE_SHIFT 0 +#define DA7219_BUTTON_AVERAGE_MASK (0x3 << 0) +#define DA7219_ADC_1_BIT_REPEAT_SHIFT 2 +#define DA7219_ADC_1_BIT_REPEAT_MASK (0x3 << 2) +#define DA7219_PIN_ORDER_FORCE_SHIFT 4 +#define DA7219_PIN_ORDER_FORCE_MASK (0x1 << 4) +#define DA7219_JACK_TYPE_FORCE_SHIFT 5 +#define DA7219_JACK_TYPE_FORCE_MASK (0x1 << 5) + +/* DA7219_ACCDET_CONFIG_8 = 0xCD */ +#define DA7219_HPTEST_EN_SHIFT 0 +#define DA7219_HPTEST_EN_MASK (0x1 << 0) +#define DA7219_HPTEST_RES_SEL_SHIFT 1 +#define DA7219_HPTEST_RES_SEL_MASK (0x3 << 1) +#define DA7219_HPTEST_RES_SEL_1KOHMS (0x0 << 1) +#define DA7219_HPTEST_COMP_SHIFT 4 +#define DA7219_HPTEST_COMP_MASK (0x1 << 4) + + +#define DA7219_AAD_MAX_BUTTONS 4 +#define DA7219_AAD_REPORT_ALL_MASK (SND_JACK_MECHANICAL | \ + SND_JACK_HEADSET | SND_JACK_LINEOUT | \ + SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +#define DA7219_AAD_MICBIAS_CHK_DELAY 10 +#define DA7219_AAD_MICBIAS_CHK_RETRIES 5 + +#define DA7219_AAD_HPTEST_RAMP_FREQ 0x28 +#define DA7219_AAD_HPTEST_PERIOD 65 + +enum da7219_aad_event_regs { + DA7219_AAD_IRQ_REG_A = 0, + DA7219_AAD_IRQ_REG_B, + DA7219_AAD_IRQ_REG_MAX, +}; + +/* Private data */ +struct da7219_aad_priv { + struct snd_soc_codec *codec; + int irq; + + u8 micbias_pulse_lvl; + u32 micbias_pulse_time; + + u8 btn_cfg; + + struct work_struct btn_det_work; + struct work_struct hptest_work; + + struct snd_soc_jack *jack; + bool jack_inserted; +}; + +/* AAD control */ +void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +/* Init/Exit */ +int da7219_aad_init(struct snd_soc_codec *codec); +void da7219_aad_exit(struct snd_soc_codec *codec); + +#endif /* __DA7219_AAD_H */ diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c new file mode 100644 index 000000000000..f238c1e8a69c --- /dev/null +++ b/sound/soc/codecs/da7219.c @@ -0,0 +1,1955 @@ +/* + * da7219.c - DA7219 ALSA SoC Codec Driver + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/pm.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> + +#include <sound/da7219.h> +#include "da7219.h" +#include "da7219-aad.h" + + +/* + * TLVs and Enums + */ + +/* Input TLVs */ +static const DECLARE_TLV_DB_SCALE(da7219_mic_gain_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(da7219_adc_dig_gain_tlv, -8325, 75, 0); +static const DECLARE_TLV_DB_SCALE(da7219_alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(da7219_alc_gain_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_alc_ana_gain_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_sidetone_gain_tlv, -4200, 300, 0); +static const DECLARE_TLV_DB_SCALE(da7219_tonegen_gain_tlv, -4500, 300, 0); + +/* Output TLVs */ +static const DECLARE_TLV_DB_SCALE(da7219_dac_eq_band_tlv, -1050, 150, 0); + +static const DECLARE_TLV_DB_RANGE(da7219_dac_dig_gain_tlv, + 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -77.25dB to 12dB */ + 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7725, 75, 0) +); + +static const DECLARE_TLV_DB_SCALE(da7219_dac_ng_threshold_tlv, -10200, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_hp_gain_tlv, -5700, 100, 0); + +/* Input Enums */ +static const char * const da7219_alc_attack_rate_txt[] = { + "7.33/fs", "14.66/fs", "29.32/fs", "58.64/fs", "117.3/fs", "234.6/fs", + "469.1/fs", "938.2/fs", "1876/fs", "3753/fs", "7506/fs", "15012/fs", + "30024/fs" +}; + +static const struct soc_enum da7219_alc_attack_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_ATTACK_SHIFT, + DA7219_ALC_ATTACK_MAX, da7219_alc_attack_rate_txt); + +static const char * const da7219_alc_release_rate_txt[] = { + "28.66/fs", "57.33/fs", "114.6/fs", "229.3/fs", "458.6/fs", "917.1/fs", + "1834/fs", "3668/fs", "7337/fs", "14674/fs", "29348/fs" +}; + +static const struct soc_enum da7219_alc_release_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_RELEASE_SHIFT, + DA7219_ALC_RELEASE_MAX, da7219_alc_release_rate_txt); + +static const char * const da7219_alc_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da7219_alc_hold_time = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_HOLD_SHIFT, + DA7219_ALC_HOLD_MAX, da7219_alc_hold_time_txt); + +static const char * const da7219_alc_env_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da7219_alc_env_attack_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_ATTACK_SHIFT, + DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt); + +static const struct soc_enum da7219_alc_env_release_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_RELEASE_SHIFT, + DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt); + +static const char * const da7219_alc_anticlip_step_txt[] = { + "0.034dB/fs", "0.068dB/fs", "0.136dB/fs", "0.272dB/fs" +}; + +static const struct soc_enum da7219_alc_anticlip_step = + SOC_ENUM_SINGLE(DA7219_ALC_ANTICLIP_CTRL, + DA7219_ALC_ANTICLIP_STEP_SHIFT, + DA7219_ALC_ANTICLIP_STEP_MAX, + da7219_alc_anticlip_step_txt); + +/* Input/Output Enums */ +static const char * const da7219_gain_ramp_rate_txt[] = { + "Nominal Rate * 8", "Nominal Rate", "Nominal Rate / 8", + "Nominal Rate / 16" +}; + +static const struct soc_enum da7219_gain_ramp_rate = + SOC_ENUM_SINGLE(DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_SHIFT, + DA7219_GAIN_RAMP_RATE_MAX, da7219_gain_ramp_rate_txt); + +static const char * const da7219_hpf_mode_txt[] = { + "Disabled", "Audio", "Voice" +}; + +static const unsigned int da7219_hpf_mode_val[] = { + DA7219_HPF_DISABLED, DA7219_HPF_AUDIO_EN, DA7219_HPF_VOICE_EN, +}; + +static const struct soc_enum da7219_adc_hpf_mode = + SOC_VALUE_ENUM_SINGLE(DA7219_ADC_FILTERS1, DA7219_HPF_MODE_SHIFT, + DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX, + da7219_hpf_mode_txt, da7219_hpf_mode_val); + +static const struct soc_enum da7219_dac_hpf_mode = + SOC_VALUE_ENUM_SINGLE(DA7219_DAC_FILTERS1, DA7219_HPF_MODE_SHIFT, + DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX, + da7219_hpf_mode_txt, da7219_hpf_mode_val); + +static const char * const da7219_audio_hpf_corner_txt[] = { + "2Hz", "4Hz", "8Hz", "16Hz" +}; + +static const struct soc_enum da7219_adc_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1, + DA7219_ADC_AUDIO_HPF_CORNER_SHIFT, + DA7219_AUDIO_HPF_CORNER_MAX, + da7219_audio_hpf_corner_txt); + +static const struct soc_enum da7219_dac_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1, + DA7219_DAC_AUDIO_HPF_CORNER_SHIFT, + DA7219_AUDIO_HPF_CORNER_MAX, + da7219_audio_hpf_corner_txt); + +static const char * const da7219_voice_hpf_corner_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7219_adc_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1, + DA7219_ADC_VOICE_HPF_CORNER_SHIFT, + DA7219_VOICE_HPF_CORNER_MAX, + da7219_voice_hpf_corner_txt); + +static const struct soc_enum da7219_dac_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1, + DA7219_DAC_VOICE_HPF_CORNER_SHIFT, + DA7219_VOICE_HPF_CORNER_MAX, + da7219_voice_hpf_corner_txt); + +static const char * const da7219_tonegen_dtmf_key_txt[] = { + "0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "A", "B", "C", "D", + "*", "#" +}; + +static const struct soc_enum da7219_tonegen_dtmf_key = + SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG1, DA7219_DTMF_REG_SHIFT, + DA7219_DTMF_REG_MAX, da7219_tonegen_dtmf_key_txt); + +static const char * const da7219_tonegen_swg_sel_txt[] = { + "Sum", "SWG1", "SWG2", "SWG1_1-Cos" +}; + +static const struct soc_enum da7219_tonegen_swg_sel = + SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG2, DA7219_SWG_SEL_SHIFT, + DA7219_SWG_SEL_MAX, da7219_tonegen_swg_sel_txt); + +/* Output Enums */ +static const char * const da7219_dac_softmute_rate_txt[] = { + "1 Sample", "2 Samples", "4 Samples", "8 Samples", "16 Samples", + "32 Samples", "64 Samples" +}; + +static const struct soc_enum da7219_dac_softmute_rate = + SOC_ENUM_SINGLE(DA7219_DAC_FILTERS5, DA7219_DAC_SOFTMUTE_RATE_SHIFT, + DA7219_DAC_SOFTMUTE_RATE_MAX, + da7219_dac_softmute_rate_txt); + +static const char * const da7219_dac_ng_setup_time_txt[] = { + "256 Samples", "512 Samples", "1024 Samples", "2048 Samples" +}; + +static const struct soc_enum da7219_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME, + DA7219_DAC_NG_SETUP_TIME_SHIFT, + DA7219_DAC_NG_SETUP_TIME_MAX, + da7219_dac_ng_setup_time_txt); + +static const char * const da7219_dac_ng_rampup_txt[] = { + "0.22ms/dB", "0.0138ms/dB" +}; + +static const struct soc_enum da7219_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME, + DA7219_DAC_NG_RAMPUP_RATE_SHIFT, + DA7219_DAC_NG_RAMP_RATE_MAX, + da7219_dac_ng_rampup_txt); + +static const char * const da7219_dac_ng_rampdown_txt[] = { + "0.88ms/dB", "14.08ms/dB" +}; + +static const struct soc_enum da7219_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME, + DA7219_DAC_NG_RAMPDN_RATE_SHIFT, + DA7219_DAC_NG_RAMP_RATE_MAX, + da7219_dac_ng_rampdown_txt); + + +static const char * const da7219_cp_track_mode_txt[] = { + "Largest Volume", "DAC Volume", "Signal Magnitude" +}; + +static const unsigned int da7219_cp_track_mode_val[] = { + DA7219_CP_MCHANGE_LARGEST_VOL, DA7219_CP_MCHANGE_DAC_VOL, + DA7219_CP_MCHANGE_SIG_MAG +}; + +static const struct soc_enum da7219_cp_track_mode = + SOC_VALUE_ENUM_SINGLE(DA7219_CP_CTRL, DA7219_CP_MCHANGE_SHIFT, + DA7219_CP_MCHANGE_REL_MASK, DA7219_CP_MCHANGE_MAX, + da7219_cp_track_mode_txt, + da7219_cp_track_mode_val); + + +/* + * Control Functions + */ + +/* Locked Kcontrol calls */ +static int da7219_volsw_locked_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_get_volsw(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +static int da7219_volsw_locked_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_put_volsw(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +static int da7219_enum_locked_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_get_enum_double(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +static int da7219_enum_locked_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_put_enum_double(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +/* ALC */ +static void da7219_alc_calib(struct snd_soc_codec *codec) +{ + u8 mic_ctrl, mixin_ctrl, adc_ctrl, calib_ctrl; + + /* Save current state of mic control register */ + mic_ctrl = snd_soc_read(codec, DA7219_MIC_1_CTRL); + + /* Save current state of input mixer control register */ + mixin_ctrl = snd_soc_read(codec, DA7219_MIXIN_L_CTRL); + + /* Save current state of input ADC control register */ + adc_ctrl = snd_soc_read(codec, DA7219_ADC_L_CTRL); + + /* Enable then Mute MIC PGAs */ + snd_soc_update_bits(codec, DA7219_MIC_1_CTRL, DA7219_MIC_1_AMP_EN_MASK, + DA7219_MIC_1_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_MUTE_EN_MASK, + DA7219_MIC_1_AMP_MUTE_EN_MASK); + + /* Enable input mixers unmuted */ + snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_MASK | + DA7219_MIXIN_L_AMP_MUTE_EN_MASK, + DA7219_MIXIN_L_AMP_EN_MASK); + + /* Enable input filters unmuted */ + snd_soc_update_bits(codec, DA7219_ADC_L_CTRL, + DA7219_ADC_L_MUTE_EN_MASK | DA7219_ADC_L_EN_MASK, + DA7219_ADC_L_EN_MASK); + + /* Perform auto calibration */ + snd_soc_update_bits(codec, DA7219_ALC_CTRL1, + DA7219_ALC_AUTO_CALIB_EN_MASK, + DA7219_ALC_AUTO_CALIB_EN_MASK); + do { + calib_ctrl = snd_soc_read(codec, DA7219_ALC_CTRL1); + } while (calib_ctrl & DA7219_ALC_AUTO_CALIB_EN_MASK); + + /* If auto calibration fails, disable DC offset, hybrid ALC */ + if (calib_ctrl & DA7219_ALC_CALIB_OVERFLOW_MASK) { + dev_warn(codec->dev, + "ALC auto calibration failed with overflow\n"); + snd_soc_update_bits(codec, DA7219_ALC_CTRL1, + DA7219_ALC_OFFSET_EN_MASK | + DA7219_ALC_SYNC_MODE_MASK, 0); + } else { + /* Enable DC offset cancellation, hybrid mode */ + snd_soc_update_bits(codec, DA7219_ALC_CTRL1, + DA7219_ALC_OFFSET_EN_MASK | + DA7219_ALC_SYNC_MODE_MASK, + DA7219_ALC_OFFSET_EN_MASK | + DA7219_ALC_SYNC_MODE_MASK); + } + + /* Restore input filter control register to original state */ + snd_soc_write(codec, DA7219_ADC_L_CTRL, adc_ctrl); + + /* Restore input mixer control registers to original state */ + snd_soc_write(codec, DA7219_MIXIN_L_CTRL, mixin_ctrl); + + /* Restore MIC control registers to original states */ + snd_soc_write(codec, DA7219_MIC_1_CTRL, mic_ctrl); +} + +static int da7219_mixin_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + + /* + * If ALC in operation and value of control has been updated, + * make sure calibrated offsets are updated. + */ + if ((ret == 1) && (da7219->alc_en)) + da7219_alc_calib(codec); + + return ret; +} + +static int da7219_alc_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + + /* Force ALC offset calibration if enabling ALC */ + if ((ucontrol->value.integer.value[0]) && (!da7219->alc_en)) { + da7219_alc_calib(codec); + da7219->alc_en = true; + } else { + da7219->alc_en = false; + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* ToneGen */ +static int da7219_tonegen_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int reg = mixer_ctrl->reg; + u16 val; + int ret; + + mutex_lock(&da7219->lock); + ret = regmap_raw_read(da7219->regmap, reg, &val, sizeof(val)); + mutex_unlock(&da7219->lock); + + if (ret) + return ret; + + /* + * Frequency value spans two 8-bit registers, lower then upper byte. + * Therefore we need to convert to host endianness here. + */ + ucontrol->value.integer.value[0] = le16_to_cpu(val); + + return 0; +} + +static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int reg = mixer_ctrl->reg; + u16 val; + int ret; + + /* + * Frequency value spans two 8-bit registers, lower then upper byte. + * Therefore we need to convert to little endian here to align with + * HW registers. + */ + val = cpu_to_le16(ucontrol->value.integer.value[0]); + + mutex_lock(&da7219->lock); + ret = regmap_raw_write(da7219->regmap, reg, &val, sizeof(val)); + mutex_unlock(&da7219->lock); + + return ret; +} + + +/* + * KControls + */ + +static const struct snd_kcontrol_new da7219_snd_controls[] = { + /* Mics */ + SOC_SINGLE_TLV("Mic Volume", DA7219_MIC_1_GAIN, + DA7219_MIC_1_AMP_GAIN_SHIFT, DA7219_MIC_1_AMP_GAIN_MAX, + DA7219_NO_INVERT, da7219_mic_gain_tlv), + SOC_SINGLE("Mic Switch", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + + /* Mixer Input */ + SOC_SINGLE_EXT_TLV("Mixin Volume", DA7219_MIXIN_L_GAIN, + DA7219_MIXIN_L_AMP_GAIN_SHIFT, + DA7219_MIXIN_L_AMP_GAIN_MAX, DA7219_NO_INVERT, + snd_soc_get_volsw, da7219_mixin_gain_put, + da7219_mixin_gain_tlv), + SOC_SINGLE("Mixin Switch", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + SOC_SINGLE("Mixin Gain Ramp Switch", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + SOC_SINGLE("Mixin ZC Gain Switch", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_ZC_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + + /* ADC */ + SOC_SINGLE_TLV("Capture Digital Volume", DA7219_ADC_L_GAIN, + DA7219_ADC_L_DIGITAL_GAIN_SHIFT, + DA7219_ADC_L_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, + da7219_adc_dig_gain_tlv), + SOC_SINGLE("Capture Digital Switch", DA7219_ADC_L_CTRL, + DA7219_ADC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + SOC_SINGLE("Capture Digital Gain Ramp Switch", DA7219_ADC_L_CTRL, + DA7219_ADC_L_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + + /* ALC */ + SOC_ENUM("ALC Attack Rate", da7219_alc_attack_rate), + SOC_ENUM("ALC Release Rate", da7219_alc_release_rate), + SOC_ENUM("ALC Hold Time", da7219_alc_hold_time), + SOC_ENUM("ALC Envelope Attack Rate", da7219_alc_env_attack_rate), + SOC_ENUM("ALC Envelope Release Rate", da7219_alc_env_release_rate), + SOC_SINGLE_TLV("ALC Noise Threshold", DA7219_ALC_NOISE, + DA7219_ALC_NOISE_SHIFT, DA7219_ALC_THRESHOLD_MAX, + DA7219_INVERT, da7219_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Min Threshold", DA7219_ALC_TARGET_MIN, + DA7219_ALC_THRESHOLD_MIN_SHIFT, DA7219_ALC_THRESHOLD_MAX, + DA7219_INVERT, da7219_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold", DA7219_ALC_TARGET_MAX, + DA7219_ALC_THRESHOLD_MAX_SHIFT, DA7219_ALC_THRESHOLD_MAX, + DA7219_INVERT, da7219_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation", DA7219_ALC_GAIN_LIMITS, + DA7219_ALC_ATTEN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Volume", DA7219_ALC_GAIN_LIMITS, + DA7219_ALC_GAIN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_gain_tlv), + SOC_SINGLE_RANGE_TLV("ALC Min Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS, + DA7219_ALC_ANA_GAIN_MIN_SHIFT, + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), + SOC_SINGLE_RANGE_TLV("ALC Max Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS, + DA7219_ALC_ANA_GAIN_MAX_SHIFT, + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), + SOC_ENUM("ALC Anticlip Step", da7219_alc_anticlip_step), + SOC_SINGLE("ALC Anticlip Switch", DA7219_ALC_ANTICLIP_CTRL, + DA7219_ALC_ANTIPCLIP_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + SOC_SINGLE_EXT("ALC Switch", DA7219_ALC_CTRL1, DA7219_ALC_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT, + snd_soc_get_volsw, da7219_alc_sw_put), + + /* Input High-Pass Filters */ + SOC_ENUM("ADC HPF Mode", da7219_adc_hpf_mode), + SOC_ENUM("ADC HPF Corner Audio", da7219_adc_audio_hpf_corner), + SOC_ENUM("ADC HPF Corner Voice", da7219_adc_voice_hpf_corner), + + /* Sidetone Filter */ + SOC_SINGLE_TLV("Sidetone Volume", DA7219_SIDETONE_GAIN, + DA7219_SIDETONE_GAIN_SHIFT, DA7219_SIDETONE_GAIN_MAX, + DA7219_NO_INVERT, da7219_sidetone_gain_tlv), + SOC_SINGLE("Sidetone Switch", DA7219_SIDETONE_CTRL, + DA7219_SIDETONE_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + + /* Tone Generator */ + SOC_SINGLE_EXT_TLV("ToneGen Volume", DA7219_TONE_GEN_CFG2, + DA7219_TONE_GEN_GAIN_SHIFT, DA7219_TONE_GEN_GAIN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put, da7219_tonegen_gain_tlv), + SOC_ENUM_EXT("ToneGen DTMF Key", da7219_tonegen_dtmf_key, + da7219_enum_locked_get, da7219_enum_locked_put), + SOC_SINGLE_EXT("ToneGen DTMF Switch", DA7219_TONE_GEN_CFG1, + DA7219_DTMF_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + SOC_ENUM_EXT("ToneGen Sinewave Gen Type", da7219_tonegen_swg_sel, + da7219_enum_locked_get, da7219_enum_locked_put), + SOC_SINGLE_EXT("ToneGen Sinewave1 Freq", DA7219_TONE_GEN_FREQ1_L, + DA7219_FREQ1_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT, + da7219_tonegen_freq_get, da7219_tonegen_freq_put), + SOC_SINGLE_EXT("ToneGen Sinewave2 Freq", DA7219_TONE_GEN_FREQ2_L, + DA7219_FREQ2_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT, + da7219_tonegen_freq_get, da7219_tonegen_freq_put), + SOC_SINGLE_EXT("ToneGen On Time", DA7219_TONE_GEN_ON_PER, + DA7219_BEEP_ON_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + SOC_SINGLE("ToneGen Off Time", DA7219_TONE_GEN_OFF_PER, + DA7219_BEEP_OFF_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX, + DA7219_NO_INVERT), + + /* Gain ramping */ + SOC_ENUM("Gain Ramp Rate", da7219_gain_ramp_rate), + + /* DAC High-Pass Filter */ + SOC_ENUM_EXT("DAC HPF Mode", da7219_dac_hpf_mode, + da7219_enum_locked_get, da7219_enum_locked_put), + SOC_ENUM("DAC HPF Corner Audio", da7219_dac_audio_hpf_corner), + SOC_ENUM("DAC HPF Corner Voice", da7219_dac_voice_hpf_corner), + + /* DAC 5-Band Equaliser */ + SOC_SINGLE_TLV("DAC EQ Band1 Volume", DA7219_DAC_FILTERS2, + DA7219_DAC_EQ_BAND1_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band2 Volume", DA7219_DAC_FILTERS2, + DA7219_DAC_EQ_BAND2_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band3 Volume", DA7219_DAC_FILTERS3, + DA7219_DAC_EQ_BAND3_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band4 Volume", DA7219_DAC_FILTERS3, + DA7219_DAC_EQ_BAND4_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band5 Volume", DA7219_DAC_FILTERS4, + DA7219_DAC_EQ_BAND5_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_EXT("DAC EQ Switch", DA7219_DAC_FILTERS4, + DA7219_DAC_EQ_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + + /* DAC Softmute */ + SOC_ENUM("DAC Soft Mute Rate", da7219_dac_softmute_rate), + SOC_SINGLE_EXT("DAC Soft Mute Switch", DA7219_DAC_FILTERS5, + DA7219_DAC_SOFTMUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + + /* DAC Noise Gate */ + SOC_ENUM("DAC NG Setup Time", da7219_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da7219_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da7219_dac_ng_rampdown_rate), + SOC_SINGLE_TLV("DAC NG Off Threshold", DA7219_DAC_NG_OFF_THRESH, + DA7219_DAC_NG_OFF_THRESHOLD_SHIFT, + DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT, + da7219_dac_ng_threshold_tlv), + SOC_SINGLE_TLV("DAC NG On Threshold", DA7219_DAC_NG_ON_THRESH, + DA7219_DAC_NG_ON_THRESHOLD_SHIFT, + DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT, + da7219_dac_ng_threshold_tlv), + SOC_SINGLE("DAC NG Switch", DA7219_DAC_NG_CTRL, DA7219_DAC_NG_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), + + /* DACs */ + SOC_DOUBLE_R_EXT_TLV("Playback Digital Volume", DA7219_DAC_L_GAIN, + DA7219_DAC_R_GAIN, DA7219_DAC_L_DIGITAL_GAIN_SHIFT, + DA7219_DAC_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, + da7219_volsw_locked_get, da7219_volsw_locked_put, + da7219_dac_dig_gain_tlv), + SOC_DOUBLE_R_EXT("Playback Digital Switch", DA7219_DAC_L_CTRL, + DA7219_DAC_R_CTRL, DA7219_DAC_L_MUTE_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_INVERT, + da7219_volsw_locked_get, da7219_volsw_locked_put), + SOC_DOUBLE_R("Playback Digital Gain Ramp Switch", DA7219_DAC_L_CTRL, + DA7219_DAC_R_CTRL, DA7219_DAC_L_RAMP_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), + + /* CP */ + SOC_ENUM("Charge Pump Track Mode", da7219_cp_track_mode), + SOC_SINGLE("Charge Pump Threshold", DA7219_CP_VOL_THRESHOLD1, + DA7219_CP_THRESH_VDD2_SHIFT, DA7219_CP_THRESH_VDD2_MAX, + DA7219_NO_INVERT), + + /* Headphones */ + SOC_DOUBLE_R_EXT_TLV("Headphone Volume", DA7219_HP_L_GAIN, + DA7219_HP_R_GAIN, DA7219_HP_L_AMP_GAIN_SHIFT, + DA7219_HP_AMP_GAIN_MAX, DA7219_NO_INVERT, + da7219_volsw_locked_get, da7219_volsw_locked_put, + da7219_hp_gain_tlv), + SOC_DOUBLE_R_EXT("Headphone Switch", DA7219_HP_L_CTRL, DA7219_HP_R_CTRL, + DA7219_HP_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + SOC_DOUBLE_R("Headphone Gain Ramp Switch", DA7219_HP_L_CTRL, + DA7219_HP_R_CTRL, DA7219_HP_L_AMP_RAMP_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), + SOC_DOUBLE_R("Headphone ZC Gain Switch", DA7219_HP_L_CTRL, + DA7219_HP_R_CTRL, DA7219_HP_L_AMP_ZC_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + + +/* + * DAPM Mux Controls + */ + +static const char * const da7219_out_sel_txt[] = { + "ADC", "Tone Generator", "DAIL", "DAIR" +}; + +static const struct soc_enum da7219_out_dail_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI, + DA7219_DAI_L_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dail_sel_mux = + SOC_DAPM_ENUM("Out DAIL Mux", da7219_out_dail_sel); + +static const struct soc_enum da7219_out_dair_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI, + DA7219_DAI_R_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dair_sel_mux = + SOC_DAPM_ENUM("Out DAIR Mux", da7219_out_dair_sel); + +static const struct soc_enum da7219_out_dacl_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC, + DA7219_DAC_L_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dacl_sel_mux = + SOC_DAPM_ENUM("Out DACL Mux", da7219_out_dacl_sel); + +static const struct soc_enum da7219_out_dacr_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC, + DA7219_DAC_R_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dacr_sel_mux = + SOC_DAPM_ENUM("Out DACR Mux", da7219_out_dacr_sel); + + +/* + * DAPM Mixer Controls + */ + +static const struct snd_kcontrol_new da7219_mixin_controls[] = { + SOC_DAPM_SINGLE("Mic Switch", DA7219_MIXIN_L_SELECT, + DA7219_MIXIN_L_MIX_SELECT_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + +static const struct snd_kcontrol_new da7219_mixout_l_controls[] = { + SOC_DAPM_SINGLE("DACL Switch", DA7219_MIXOUT_L_SELECT, + DA7219_MIXOUT_L_MIX_SELECT_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + +static const struct snd_kcontrol_new da7219_mixout_r_controls[] = { + SOC_DAPM_SINGLE("DACR Switch", DA7219_MIXOUT_R_SELECT, + DA7219_MIXOUT_R_MIX_SELECT_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + +#define DA7219_DMIX_ST_CTRLS(reg) \ + SOC_DAPM_SINGLE("Out FilterL Switch", reg, \ + DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT, \ + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), \ + SOC_DAPM_SINGLE("Out FilterR Switch", reg, \ + DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT, \ + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), \ + SOC_DAPM_SINGLE("Sidetone Switch", reg, \ + DA7219_DMIX_ST_SRC_SIDETONE_SHIFT, \ + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT) \ + +static const struct snd_kcontrol_new da7219_st_out_filtl_mix_controls[] = { + DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1L), +}; + +static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = { + DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1R), +}; + + +/* + * DAPM Events + */ + +static int da7219_dai_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 pll_ctrl, pll_status; + int i = 0; + bool srm_lock = false; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (da7219->master) + /* Enable DAI clks for master mode */ + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_CLK_EN_MASK, + DA7219_DAI_CLK_EN_MASK); + + /* PC synchronised to DAI */ + snd_soc_update_bits(codec, DA7219_PC_COUNT, + DA7219_PC_FREERUN_MASK, 0); + + /* Slave mode, if SRM not enabled no need for status checks */ + pll_ctrl = snd_soc_read(codec, DA7219_PLL_CTRL); + if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM) + return 0; + + /* Check SRM has locked */ + do { + pll_status = snd_soc_read(codec, DA7219_PLL_SRM_STS); + if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) { + srm_lock = true; + } else { + ++i; + msleep(50); + } + } while ((i < DA7219_SRM_CHECK_RETRIES) & (!srm_lock)); + + if (!srm_lock) + dev_warn(codec->dev, "SRM failed to lock\n"); + + return 0; + case SND_SOC_DAPM_POST_PMD: + /* PC free-running */ + snd_soc_update_bits(codec, DA7219_PC_COUNT, + DA7219_PC_FREERUN_MASK, + DA7219_PC_FREERUN_MASK); + + /* Disable DAI clks if in master mode */ + if (da7219->master) + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_CLK_EN_MASK, 0); + return 0; + default: + return -EINVAL; + } +} + + +/* + * DAPM Widgets + */ + +static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = { + /* Input Supplies */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7219_MICBIAS_CTRL, + DA7219_MICBIAS1_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + + /* Input Filters */ + SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT, + DA7219_NO_INVERT), + + /* Tone Generator */ + SND_SOC_DAPM_SIGGEN("TONE"), + SND_SOC_DAPM_PGA("Tone Generator", DA7219_TONE_GEN_CFG1, + DA7219_START_STOPN_SHIFT, DA7219_NO_INVERT, NULL, 0), + + /* Sidetone Input */ + SND_SOC_DAPM_ADC("Sidetone Filter", NULL, DA7219_SIDETONE_CTRL, + DA7219_SIDETONE_EN_SHIFT, DA7219_NO_INVERT), + + /* Input Mixer Supply */ + SND_SOC_DAPM_SUPPLY("Mixer In Supply", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_MIX_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + + /* Input Mixer */ + SND_SOC_DAPM_MIXER("Mixer In", SND_SOC_NOPM, 0, 0, + da7219_mixin_controls, + ARRAY_SIZE(da7219_mixin_controls)), + + /* Input Muxes */ + SND_SOC_DAPM_MUX("Out DAIL Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dail_sel_mux), + SND_SOC_DAPM_MUX("Out DAIR Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dair_sel_mux), + + /* DAI Supply */ + SND_SOC_DAPM_SUPPLY("DAI", DA7219_DAI_CTRL, DA7219_DAI_EN_SHIFT, + DA7219_NO_INVERT, da7219_dai_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* DAI */ + SND_SOC_DAPM_AIF_OUT("DAIOUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DAIIN", "Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Muxes */ + SND_SOC_DAPM_MUX("Out DACL Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dacl_sel_mux), + SND_SOC_DAPM_MUX("Out DACR Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dacr_sel_mux), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Mixer Out FilterL", SND_SOC_NOPM, 0, 0, + da7219_mixout_l_controls, + ARRAY_SIZE(da7219_mixout_l_controls)), + SND_SOC_DAPM_MIXER("Mixer Out FilterR", SND_SOC_NOPM, 0, 0, + da7219_mixout_r_controls, + ARRAY_SIZE(da7219_mixout_r_controls)), + + /* Sidetone Mixers */ + SND_SOC_DAPM_MIXER("ST Mixer Out FilterL", SND_SOC_NOPM, 0, 0, + da7219_st_out_filtl_mix_controls, + ARRAY_SIZE(da7219_st_out_filtl_mix_controls)), + SND_SOC_DAPM_MIXER("ST Mixer Out FilterR", SND_SOC_NOPM, 0, + 0, da7219_st_out_filtr_mix_controls, + ARRAY_SIZE(da7219_st_out_filtr_mix_controls)), + + /* DACs */ + SND_SOC_DAPM_DAC("DACL", NULL, DA7219_DAC_L_CTRL, DA7219_DAC_L_EN_SHIFT, + DA7219_NO_INVERT), + SND_SOC_DAPM_DAC("DACR", NULL, DA7219_DAC_R_CTRL, DA7219_DAC_R_EN_SHIFT, + DA7219_NO_INVERT), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("Mixout Left PGA", DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Mixout Right PGA", DA7219_MIXOUT_R_CTRL, + DA7219_MIXOUT_R_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left PGA", DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right PGA", DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0), + + /* Output Supplies */ + SND_SOC_DAPM_SUPPLY("Charge Pump", DA7219_CP_CTRL, DA7219_CP_EN_SHIFT, + DA7219_NO_INVERT, NULL, 0), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), +}; + + +/* + * DAPM Mux Routes + */ + +#define DA7219_OUT_DAI_MUX_ROUTES(name) \ + {name, "ADC", "Mixer In"}, \ + {name, "Tone Generator", "Tone Generator"}, \ + {name, "DAIL", "DAIOUT"}, \ + {name, "DAIR", "DAIOUT"} + +#define DA7219_OUT_DAC_MUX_ROUTES(name) \ + {name, "ADC", "Mixer In"}, \ + {name, "Tone Generator", "Tone Generator"}, \ + {name, "DAIL", "DAIIN"}, \ + {name, "DAIR", "DAIIN"} + +/* + * DAPM Mixer Routes + */ + +#define DA7219_DMIX_ST_ROUTES(name) \ + {name, "Out FilterL Switch", "Mixer Out FilterL"}, \ + {name, "Out FilterR Switch", "Mixer Out FilterR"}, \ + {name, "Sidetone Switch", "Sidetone Filter"} + + +/* + * DAPM audio route definition + */ + +static const struct snd_soc_dapm_route da7219_audio_map[] = { + /* Input paths */ + {"MIC", NULL, "Mic Bias"}, + {"Mic PGA", NULL, "MIC"}, + {"Mixin PGA", NULL, "Mic PGA"}, + {"ADC", NULL, "Mixin PGA"}, + + {"Sidetone Filter", NULL, "ADC"}, + {"Mixer In", NULL, "Mixer In Supply"}, + {"Mixer In", "Mic Switch", "ADC"}, + + {"Tone Generator", NULL, "TONE"}, + + DA7219_OUT_DAI_MUX_ROUTES("Out DAIL Mux"), + DA7219_OUT_DAI_MUX_ROUTES("Out DAIR Mux"), + + {"DAIOUT", NULL, "Out DAIL Mux"}, + {"DAIOUT", NULL, "Out DAIR Mux"}, + {"DAIOUT", NULL, "DAI"}, + + /* Output paths */ + {"DAIIN", NULL, "DAI"}, + + DA7219_OUT_DAC_MUX_ROUTES("Out DACL Mux"), + DA7219_OUT_DAC_MUX_ROUTES("Out DACR Mux"), + + {"Mixer Out FilterL", "DACL Switch", "Out DACL Mux"}, + {"Mixer Out FilterR", "DACR Switch", "Out DACR Mux"}, + + DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterL"), + DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterR"), + + {"DACL", NULL, "ST Mixer Out FilterL"}, + {"DACR", NULL, "ST Mixer Out FilterR"}, + + {"Mixout Left PGA", NULL, "DACL"}, + {"Mixout Right PGA", NULL, "DACR"}, + + {"Headphone Left PGA", NULL, "Mixout Left PGA"}, + {"Headphone Right PGA", NULL, "Mixout Right PGA"}, + + {"HPL", NULL, "Headphone Left PGA"}, + {"HPR", NULL, "Headphone Right PGA"}, + + {"HPL", NULL, "Charge Pump"}, + {"HPR", NULL, "Charge Pump"}, +}; + + +/* + * DAI operations + */ + +static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) + return 0; + + if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) { + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + + switch (clk_id) { + case DA7219_CLKSRC_MCLK_SQR: + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_MCLK_SQR_EN_MASK, + DA7219_PLL_MCLK_SQR_EN_MASK); + break; + case DA7219_CLKSRC_MCLK: + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_MCLK_SQR_EN_MASK, 0); + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } + + da7219->clk_src = clk_id; + + if (da7219->mclk) { + freq = clk_round_rate(da7219->mclk, freq); + ret = clk_set_rate(da7219->mclk, freq); + if (ret) { + dev_err(codec_dai->dev, "Failed to set clock rate %d\n", + freq); + return ret; + } + } + + da7219->mclk_rate = freq; + + return 0; +} + +static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + u8 pll_ctrl, indiv_bits, indiv; + u8 pll_frac_top, pll_frac_bot, pll_integer; + u32 freq_ref; + u64 frac_div; + + /* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */ + if (da7219->mclk_rate == 32768) { + indiv_bits = DA7219_PLL_INDIV_2_5_MHZ; + indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL; + } else if (da7219->mclk_rate < 2000000) { + dev_err(codec->dev, "PLL input clock %d below valid range\n", + da7219->mclk_rate); + return -EINVAL; + } else if (da7219->mclk_rate <= 5000000) { + indiv_bits = DA7219_PLL_INDIV_2_5_MHZ; + indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL; + } else if (da7219->mclk_rate <= 10000000) { + indiv_bits = DA7219_PLL_INDIV_5_10_MHZ; + indiv = DA7219_PLL_INDIV_5_10_MHZ_VAL; + } else if (da7219->mclk_rate <= 20000000) { + indiv_bits = DA7219_PLL_INDIV_10_20_MHZ; + indiv = DA7219_PLL_INDIV_10_20_MHZ_VAL; + } else if (da7219->mclk_rate <= 40000000) { + indiv_bits = DA7219_PLL_INDIV_20_40_MHZ; + indiv = DA7219_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7219->mclk_rate <= 54000000) { + indiv_bits = DA7219_PLL_INDIV_40_54_MHZ; + indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL; + } else { + dev_err(codec->dev, "PLL input clock %d above valid range\n", + da7219->mclk_rate); + return -EINVAL; + } + freq_ref = (da7219->mclk_rate / indiv); + pll_ctrl = indiv_bits; + + /* Configure PLL */ + switch (source) { + case DA7219_SYSCLK_MCLK: + pll_ctrl |= DA7219_PLL_MODE_BYPASS; + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_INDIV_MASK | + DA7219_PLL_MODE_MASK, pll_ctrl); + return 0; + case DA7219_SYSCLK_PLL: + pll_ctrl |= DA7219_PLL_MODE_NORMAL; + break; + case DA7219_SYSCLK_PLL_SRM: + pll_ctrl |= DA7219_PLL_MODE_SRM; + break; + case DA7219_SYSCLK_PLL_32KHZ: + pll_ctrl |= DA7219_PLL_MODE_32KHZ; + break; + default: + dev_err(codec->dev, "Invalid PLL config\n"); + return -EINVAL; + } + + /* Calculate dividers for PLL */ + pll_integer = fout / freq_ref; + frac_div = (u64)(fout % freq_ref) * 8192ULL; + do_div(frac_div, freq_ref); + pll_frac_top = (frac_div >> DA7219_BYTE_SHIFT) & DA7219_BYTE_MASK; + pll_frac_bot = (frac_div) & DA7219_BYTE_MASK; + + /* Write PLL config & dividers */ + snd_soc_write(codec, DA7219_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA7219_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA7219_PLL_INTEGER, pll_integer); + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_INDIV_MASK | DA7219_PLL_MODE_MASK, + pll_ctrl); + + return 0; +} + +static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 dai_clk_mode = 0, dai_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + da7219->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + da7219->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7219_DAI_WCLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + dai_clk_mode |= DA7219_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7219_DAI_WCLK_POL_INV | + DA7219_DAI_CLK_POL_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dai_ctrl |= DA7219_DAI_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + dai_ctrl |= DA7219_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_ctrl |= DA7219_DAI_FORMAT_RIGHT_J; + break; + case SND_SOC_DAIFMT_DSP_B: + dai_ctrl |= DA7219_DAI_FORMAT_DSP; + break; + default: + return -EINVAL; + } + + /* By default 64 BCLKs per WCLK is supported */ + dai_clk_mode |= DA7219_DAI_BCLKS_PER_WCLK_64; + + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK | + DA7219_DAI_CLK_POL_MASK | DA7219_DAI_WCLK_POL_MASK, + dai_clk_mode); + snd_soc_update_bits(codec, DA7219_DAI_CTRL, DA7219_DAI_FORMAT_MASK, + dai_ctrl); + + return 0; +} + +static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 dai_bclks_per_wclk; + u16 offset; + u32 frame_size; + + /* No channels enabled so disable TDM, revert to 64-bit frames */ + if (!tx_mask) { + snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL, + DA7219_DAI_TDM_CH_EN_MASK | + DA7219_DAI_TDM_MODE_EN_MASK, 0); + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK, + DA7219_DAI_BCLKS_PER_WCLK_64); + return 0; + } + + /* Check we have valid slots */ + if (fls(tx_mask) > DA7219_DAI_TDM_MAX_SLOTS) { + dev_err(codec->dev, "Invalid number of slots, max = %d\n", + DA7219_DAI_TDM_MAX_SLOTS); + return -EINVAL; + } + + /* Check we have a valid offset given */ + if (rx_mask > DA7219_DAI_OFFSET_MAX) { + dev_err(codec->dev, "Invalid slot offset, max = %d\n", + DA7219_DAI_OFFSET_MAX); + return -EINVAL; + } + + /* Calculate & validate frame size based on slot info provided. */ + frame_size = slots * slot_width; + switch (frame_size) { + case 32: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32; + break; + case 64: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64; + break; + case 128: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128; + break; + case 256: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256; + break; + default: + dev_err(codec->dev, "Invalid frame size %d\n", frame_size); + return -EINVAL; + } + + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK, + dai_bclks_per_wclk); + + offset = cpu_to_le16(rx_mask); + regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER, + &offset, sizeof(offset)); + + snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL, + DA7219_DAI_TDM_CH_EN_MASK | + DA7219_DAI_TDM_MODE_EN_MASK, + (tx_mask << DA7219_DAI_TDM_CH_EN_SHIFT) | + DA7219_DAI_TDM_MODE_EN_MASK); + + return 0; +} + +static int da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u8 dai_ctrl = 0, fs; + unsigned int channels; + + switch (params_width(params)) { + case 16: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S16_LE; + break; + case 20: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S20_LE; + break; + case 24: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S24_LE; + break; + case 32: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S32_LE; + break; + default: + return -EINVAL; + } + + channels = params_channels(params); + if ((channels < 1) | (channels > DA7219_DAI_CH_NUM_MAX)) { + dev_err(codec->dev, + "Invalid number of channels, only 1 to %d supported\n", + DA7219_DAI_CH_NUM_MAX); + return -EINVAL; + } + dai_ctrl |= channels << DA7219_DAI_CH_NUM_SHIFT; + + switch (params_rate(params)) { + case 8000: + fs = DA7219_SR_8000; + break; + case 11025: + fs = DA7219_SR_11025; + break; + case 12000: + fs = DA7219_SR_12000; + break; + case 16000: + fs = DA7219_SR_16000; + break; + case 22050: + fs = DA7219_SR_22050; + break; + case 24000: + fs = DA7219_SR_24000; + break; + case 32000: + fs = DA7219_SR_32000; + break; + case 44100: + fs = DA7219_SR_44100; + break; + case 48000: + fs = DA7219_SR_48000; + break; + case 88200: + fs = DA7219_SR_88200; + break; + case 96000: + fs = DA7219_SR_96000; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, DA7219_DAI_CTRL, + DA7219_DAI_WORD_LENGTH_MASK | + DA7219_DAI_CH_NUM_MASK, + dai_ctrl); + snd_soc_write(codec, DA7219_SR, fs); + + return 0; +} + +static const struct snd_soc_dai_ops da7219_dai_ops = { + .hw_params = da7219_hw_params, + .set_sysclk = da7219_set_dai_sysclk, + .set_pll = da7219_set_dai_pll, + .set_fmt = da7219_set_dai_fmt, + .set_tdm_slot = da7219_set_dai_tdm_slot, +}; + +#define DA7219_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver da7219_dai = { + .name = "da7219-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = DA7219_DAI_CH_NUM_MAX, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7219_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = DA7219_DAI_CH_NUM_MAX, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7219_FORMATS, + }, + .ops = &da7219_dai_ops, + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, +}; + + +/* + * DT + */ + +static const struct of_device_id da7219_of_match[] = { + { .compatible = "dlg,da7219", }, + { } +}; +MODULE_DEVICE_TABLE(of, da7219_of_match); + +static enum da7219_ldo_lvl_sel da7219_of_ldo_lvl(struct snd_soc_codec *codec, + u32 val) +{ + switch (val) { + case 1050: + return DA7219_LDO_LVL_SEL_1_05V; + case 1100: + return DA7219_LDO_LVL_SEL_1_10V; + case 1200: + return DA7219_LDO_LVL_SEL_1_20V; + case 1400: + return DA7219_LDO_LVL_SEL_1_40V; + default: + dev_warn(codec->dev, "Invalid LDO level"); + return DA7219_LDO_LVL_SEL_1_05V; + } +} + +static enum da7219_micbias_voltage + da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1800: + return DA7219_MICBIAS_1_8V; + case 2000: + return DA7219_MICBIAS_2_0V; + case 2200: + return DA7219_MICBIAS_2_2V; + case 2400: + return DA7219_MICBIAS_2_4V; + case 2600: + return DA7219_MICBIAS_2_6V; + default: + dev_warn(codec->dev, "Invalid micbias level"); + return DA7219_MICBIAS_2_2V; + } +} + +static enum da7219_mic_amp_in_sel + da7219_of_mic_amp_in_sel(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "diff")) { + return DA7219_MIC_AMP_IN_SEL_DIFF; + } else if (!strcmp(str, "se_p")) { + return DA7219_MIC_AMP_IN_SEL_SE_P; + } else if (!strcmp(str, "se_n")) { + return DA7219_MIC_AMP_IN_SEL_SE_N; + } else { + dev_warn(codec->dev, "Invalid mic input type selection"); + return DA7219_MIC_AMP_IN_SEL_DIFF; + } +} + +static struct da7219_pdata *da7219_of_to_pdata(struct snd_soc_codec *codec) +{ + struct device_node *np = codec->dev->of_node; + struct da7219_pdata *pdata; + const char *of_str; + u32 of_val32; + + pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return NULL; + + if (of_property_read_u32(np, "dlg,ldo-lvl", &of_val32) >= 0) + pdata->ldo_lvl_sel = da7219_of_ldo_lvl(codec, of_val32); + + if (of_property_read_u32(np, "dlg,micbias-lvl", &of_val32) >= 0) + pdata->micbias_lvl = da7219_of_micbias_lvl(codec, of_val32); + else + pdata->micbias_lvl = DA7219_MICBIAS_2_2V; + + if (!of_property_read_string(np, "dlg,mic-amp-in-sel", &of_str)) + pdata->mic_amp_in_sel = da7219_of_mic_amp_in_sel(codec, of_str); + else + pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF; + + return pdata; +} + + +/* + * Codec driver functions + */ + +static int da7219_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + /* MCLK */ + if (da7219->mclk) { + ret = clk_prepare_enable(da7219->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable mclk\n"); + return ret; + } + } + + /* Master bias */ + snd_soc_update_bits(codec, DA7219_REFERENCES, + DA7219_BIAS_EN_MASK, + DA7219_BIAS_EN_MASK); + + /* Enable Internal Digital LDO */ + snd_soc_update_bits(codec, DA7219_LDO_CTRL, + DA7219_LDO_EN_MASK, + DA7219_LDO_EN_MASK); + } + break; + case SND_SOC_BIAS_OFF: + /* Only disable if jack detection not active */ + if (!da7219->aad->jack) { + /* Bypass Internal Digital LDO */ + snd_soc_update_bits(codec, DA7219_LDO_CTRL, + DA7219_LDO_EN_MASK, 0); + + /* Master bias */ + snd_soc_update_bits(codec, DA7219_REFERENCES, + DA7219_BIAS_EN_MASK, 0); + } + + /* MCLK */ + if (da7219->mclk) + clk_disable_unprepare(da7219->mclk); + break; + } + + return 0; +} + +static const char *da7219_supply_names[DA7219_NUM_SUPPLIES] = { + [DA7219_SUPPLY_VDD] = "VDD", + [DA7219_SUPPLY_VDDMIC] = "VDDMIC", + [DA7219_SUPPLY_VDDIO] = "VDDIO", +}; + +static int da7219_handle_supplies(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct regulator *vddio; + u8 io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V; + int i, ret; + + /* Get required supplies */ + for (i = 0; i < DA7219_NUM_SUPPLIES; ++i) + da7219->supplies[i].supply = da7219_supply_names[i]; + + ret = devm_regulator_bulk_get(codec->dev, DA7219_NUM_SUPPLIES, + da7219->supplies); + if (ret) { + dev_err(codec->dev, "Failed to get supplies"); + return ret; + } + + /* Determine VDDIO voltage provided */ + vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; + ret = regulator_get_voltage(vddio); + if (ret < 1200000) + dev_warn(codec->dev, "Invalid VDDIO voltage\n"); + else if (ret < 2800000) + io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V; + + /* Enable main supplies */ + ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies); + if (ret) { + dev_err(codec->dev, "Failed to enable supplies"); + return ret; + } + + /* Ensure device in active mode */ + snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK); + + /* Update IO voltage level range */ + snd_soc_write(codec, DA7219_IO_CTRL, io_voltage_lvl); + + return 0; +} + +static void da7219_handle_pdata(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_pdata *pdata = da7219->pdata; + + if (pdata) { + u8 micbias_lvl = 0; + + /* Internal LDO */ + switch (pdata->ldo_lvl_sel) { + case DA7219_LDO_LVL_SEL_1_05V: + case DA7219_LDO_LVL_SEL_1_10V: + case DA7219_LDO_LVL_SEL_1_20V: + case DA7219_LDO_LVL_SEL_1_40V: + snd_soc_update_bits(codec, DA7219_LDO_CTRL, + DA7219_LDO_LEVEL_SELECT_MASK, + (pdata->ldo_lvl_sel << + DA7219_LDO_LEVEL_SELECT_SHIFT)); + break; + } + + /* Mic Bias voltages */ + switch (pdata->micbias_lvl) { + case DA7219_MICBIAS_1_8V: + case DA7219_MICBIAS_2_0V: + case DA7219_MICBIAS_2_2V: + case DA7219_MICBIAS_2_4V: + case DA7219_MICBIAS_2_6V: + micbias_lvl |= (pdata->micbias_lvl << + DA7219_MICBIAS1_LEVEL_SHIFT); + break; + } + + snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_lvl); + + /* Mic */ + switch (pdata->mic_amp_in_sel) { + case DA7219_MIC_AMP_IN_SEL_DIFF: + case DA7219_MIC_AMP_IN_SEL_SE_P: + case DA7219_MIC_AMP_IN_SEL_SE_N: + snd_soc_write(codec, DA7219_MIC_1_SELECT, + pdata->mic_amp_in_sel); + break; + } + } +} + +static int da7219_probe(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_init(&da7219->lock); + + /* Regulator configuration */ + ret = da7219_handle_supplies(codec); + if (ret) + return ret; + + /* Handle DT/Platform data */ + if (codec->dev->of_node) + da7219->pdata = da7219_of_to_pdata(codec); + else + da7219->pdata = dev_get_platdata(codec->dev); + + da7219_handle_pdata(codec); + + /* Check if MCLK provided */ + da7219->mclk = devm_clk_get(codec->dev, "mclk"); + if (IS_ERR(da7219->mclk)) { + if (PTR_ERR(da7219->mclk) != -ENOENT) + return PTR_ERR(da7219->mclk); + else + da7219->mclk = NULL; + } + + /* Default PC counter to free-running */ + snd_soc_update_bits(codec, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK, + DA7219_PC_FREERUN_MASK); + + /* Default gain ramping */ + snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_RAMP_EN_MASK, + DA7219_MIXIN_L_AMP_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_ADC_L_CTRL, DA7219_ADC_L_RAMP_EN_MASK, + DA7219_ADC_L_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_DAC_L_CTRL, DA7219_DAC_L_RAMP_EN_MASK, + DA7219_DAC_L_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_DAC_R_CTRL, DA7219_DAC_R_RAMP_EN_MASK, + DA7219_DAC_R_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_RAMP_EN_MASK, + DA7219_HP_L_AMP_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_RAMP_EN_MASK, + DA7219_HP_R_AMP_RAMP_EN_MASK); + + /* Default infinite tone gen, start/stop by Kcontrol */ + snd_soc_write(codec, DA7219_TONE_GEN_CYCLES, DA7219_BEEP_CYCLES_MASK); + + /* Initialise AAD block */ + return da7219_aad_init(codec); +} + +static int da7219_remove(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + da7219_aad_exit(codec); + + /* Supplies */ + return regulator_bulk_disable(DA7219_NUM_SUPPLIES, da7219->supplies); +} + +#ifdef CONFIG_PM +static int da7219_suspend(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + + /* Put device into standby mode if jack detection disabled */ + if (!da7219->aad->jack) + snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, 0); + + return 0; +} + +static int da7219_resume(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + /* Put device into active mode if previously pushed to standby */ + if (!da7219->aad->jack) + snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, + DA7219_SYSTEM_ACTIVE_MASK); + + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define da7219_suspend NULL +#define da7219_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_da7219 = { + .probe = da7219_probe, + .remove = da7219_remove, + .suspend = da7219_suspend, + .resume = da7219_resume, + .set_bias_level = da7219_set_bias_level, + + .controls = da7219_snd_controls, + .num_controls = ARRAY_SIZE(da7219_snd_controls), + + .dapm_widgets = da7219_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7219_dapm_widgets), + .dapm_routes = da7219_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7219_audio_map), +}; + + +/* + * Regmap configs + */ + +static struct reg_default da7219_reg_defaults[] = { + { DA7219_MIC_1_SELECT, 0x00 }, + { DA7219_CIF_TIMEOUT_CTRL, 0x01 }, + { DA7219_SR_24_48, 0x00 }, + { DA7219_SR, 0x0A }, + { DA7219_CIF_I2C_ADDR_CFG, 0x02 }, + { DA7219_PLL_CTRL, 0x10 }, + { DA7219_PLL_FRAC_TOP, 0x00 }, + { DA7219_PLL_FRAC_BOT, 0x00 }, + { DA7219_PLL_INTEGER, 0x20 }, + { DA7219_DIG_ROUTING_DAI, 0x10 }, + { DA7219_DAI_CLK_MODE, 0x01 }, + { DA7219_DAI_CTRL, 0x28 }, + { DA7219_DAI_TDM_CTRL, 0x40 }, + { DA7219_DIG_ROUTING_DAC, 0x32 }, + { DA7219_DAI_OFFSET_LOWER, 0x00 }, + { DA7219_DAI_OFFSET_UPPER, 0x00 }, + { DA7219_REFERENCES, 0x00 }, + { DA7219_MIXIN_L_SELECT, 0x00 }, + { DA7219_MIXIN_L_GAIN, 0x03 }, + { DA7219_ADC_L_GAIN, 0x6F }, + { DA7219_ADC_FILTERS1, 0x80 }, + { DA7219_MIC_1_GAIN, 0x01 }, + { DA7219_SIDETONE_CTRL, 0x40 }, + { DA7219_SIDETONE_GAIN, 0x0E }, + { DA7219_DROUTING_ST_OUTFILT_1L, 0x01 }, + { DA7219_DROUTING_ST_OUTFILT_1R, 0x02 }, + { DA7219_DAC_FILTERS5, 0x00 }, + { DA7219_DAC_FILTERS2, 0x88 }, + { DA7219_DAC_FILTERS3, 0x88 }, + { DA7219_DAC_FILTERS4, 0x08 }, + { DA7219_DAC_FILTERS1, 0x80 }, + { DA7219_DAC_L_GAIN, 0x6F }, + { DA7219_DAC_R_GAIN, 0x6F }, + { DA7219_CP_CTRL, 0x20 }, + { DA7219_HP_L_GAIN, 0x39 }, + { DA7219_HP_R_GAIN, 0x39 }, + { DA7219_MIXOUT_L_SELECT, 0x00 }, + { DA7219_MIXOUT_R_SELECT, 0x00 }, + { DA7219_MICBIAS_CTRL, 0x03 }, + { DA7219_MIC_1_CTRL, 0x40 }, + { DA7219_MIXIN_L_CTRL, 0x40 }, + { DA7219_ADC_L_CTRL, 0x40 }, + { DA7219_DAC_L_CTRL, 0x40 }, + { DA7219_DAC_R_CTRL, 0x40 }, + { DA7219_HP_L_CTRL, 0x40 }, + { DA7219_HP_R_CTRL, 0x40 }, + { DA7219_MIXOUT_L_CTRL, 0x10 }, + { DA7219_MIXOUT_R_CTRL, 0x10 }, + { DA7219_CHIP_ID1, 0x23 }, + { DA7219_CHIP_ID2, 0x93 }, + { DA7219_CHIP_REVISION, 0x00 }, + { DA7219_LDO_CTRL, 0x00 }, + { DA7219_IO_CTRL, 0x00 }, + { DA7219_GAIN_RAMP_CTRL, 0x00 }, + { DA7219_PC_COUNT, 0x02 }, + { DA7219_CP_VOL_THRESHOLD1, 0x0E }, + { DA7219_DIG_CTRL, 0x00 }, + { DA7219_ALC_CTRL2, 0x00 }, + { DA7219_ALC_CTRL3, 0x00 }, + { DA7219_ALC_NOISE, 0x3F }, + { DA7219_ALC_TARGET_MIN, 0x3F }, + { DA7219_ALC_TARGET_MAX, 0x00 }, + { DA7219_ALC_GAIN_LIMITS, 0xFF }, + { DA7219_ALC_ANA_GAIN_LIMITS, 0x71 }, + { DA7219_ALC_ANTICLIP_CTRL, 0x00 }, + { DA7219_ALC_ANTICLIP_LEVEL, 0x00 }, + { DA7219_DAC_NG_SETUP_TIME, 0x00 }, + { DA7219_DAC_NG_OFF_THRESH, 0x00 }, + { DA7219_DAC_NG_ON_THRESH, 0x00 }, + { DA7219_DAC_NG_CTRL, 0x00 }, + { DA7219_TONE_GEN_CFG1, 0x00 }, + { DA7219_TONE_GEN_CFG2, 0x00 }, + { DA7219_TONE_GEN_CYCLES, 0x00 }, + { DA7219_TONE_GEN_FREQ1_L, 0x55 }, + { DA7219_TONE_GEN_FREQ1_U, 0x15 }, + { DA7219_TONE_GEN_FREQ2_L, 0x00 }, + { DA7219_TONE_GEN_FREQ2_U, 0x40 }, + { DA7219_TONE_GEN_ON_PER, 0x02 }, + { DA7219_TONE_GEN_OFF_PER, 0x01 }, + { DA7219_ACCDET_IRQ_MASK_A, 0x00 }, + { DA7219_ACCDET_IRQ_MASK_B, 0x00 }, + { DA7219_ACCDET_CONFIG_1, 0xD6 }, + { DA7219_ACCDET_CONFIG_2, 0x34 }, + { DA7219_ACCDET_CONFIG_3, 0x0A }, + { DA7219_ACCDET_CONFIG_4, 0x16 }, + { DA7219_ACCDET_CONFIG_5, 0x21 }, + { DA7219_ACCDET_CONFIG_6, 0x3E }, + { DA7219_ACCDET_CONFIG_7, 0x01 }, + { DA7219_SYSTEM_ACTIVE, 0x00 }, +}; + +static bool da7219_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA7219_MIC_1_GAIN_STATUS: + case DA7219_MIXIN_L_GAIN_STATUS: + case DA7219_ADC_L_GAIN_STATUS: + case DA7219_DAC_L_GAIN_STATUS: + case DA7219_DAC_R_GAIN_STATUS: + case DA7219_HP_L_GAIN_STATUS: + case DA7219_HP_R_GAIN_STATUS: + case DA7219_CIF_CTRL: + case DA7219_PLL_SRM_STS: + case DA7219_ALC_CTRL1: + case DA7219_SYSTEM_MODES_INPUT: + case DA7219_SYSTEM_MODES_OUTPUT: + case DA7219_ALC_OFFSET_AUTO_M_L: + case DA7219_ALC_OFFSET_AUTO_U_L: + case DA7219_TONE_GEN_CFG1: + case DA7219_ACCDET_STATUS_A: + case DA7219_ACCDET_STATUS_B: + case DA7219_ACCDET_IRQ_EVENT_A: + case DA7219_ACCDET_IRQ_EVENT_B: + case DA7219_ACCDET_CONFIG_8: + case DA7219_SYSTEM_STATUS: + return 1; + default: + return 0; + } +} + +static const struct regmap_config da7219_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = DA7219_SYSTEM_ACTIVE, + .reg_defaults = da7219_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7219_reg_defaults), + .volatile_reg = da7219_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + + +/* + * I2C layer + */ + +static int da7219_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da7219_priv *da7219; + int ret; + + da7219 = devm_kzalloc(&i2c->dev, sizeof(struct da7219_priv), + GFP_KERNEL); + if (!da7219) + return -ENOMEM; + + i2c_set_clientdata(i2c, da7219); + + da7219->regmap = devm_regmap_init_i2c(i2c, &da7219_regmap_config); + if (IS_ERR(da7219->regmap)) { + ret = PTR_ERR(da7219->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7219, + &da7219_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da7219 codec: %d\n", + ret); + } + return ret; +} + +static int da7219_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id da7219_i2c_id[] = { + { "da7219", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da7219_i2c_id); + +static struct i2c_driver da7219_i2c_driver = { + .driver = { + .name = "da7219", + .of_match_table = of_match_ptr(da7219_of_match), + }, + .probe = da7219_i2c_probe, + .remove = da7219_i2c_remove, + .id_table = da7219_i2c_id, +}; + +module_i2c_driver(da7219_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA7219 Codec Driver"); +MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource@diasemi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h new file mode 100644 index 000000000000..b514268c6c56 --- /dev/null +++ b/sound/soc/codecs/da7219.h @@ -0,0 +1,820 @@ +/* + * da7219.h - DA7219 ALSA SoC Codec Driver + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef __DA7219_H +#define __DA7219_H + +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/da7219.h> + +/* + * Registers + */ + +#define DA7219_MIC_1_GAIN_STATUS 0x6 +#define DA7219_MIXIN_L_GAIN_STATUS 0x8 +#define DA7219_ADC_L_GAIN_STATUS 0xA +#define DA7219_DAC_L_GAIN_STATUS 0xC +#define DA7219_DAC_R_GAIN_STATUS 0xD +#define DA7219_HP_L_GAIN_STATUS 0xE +#define DA7219_HP_R_GAIN_STATUS 0xF +#define DA7219_MIC_1_SELECT 0x10 +#define DA7219_CIF_TIMEOUT_CTRL 0x12 +#define DA7219_CIF_CTRL 0x13 +#define DA7219_SR_24_48 0x16 +#define DA7219_SR 0x17 +#define DA7219_CIF_I2C_ADDR_CFG 0x1B +#define DA7219_PLL_CTRL 0x20 +#define DA7219_PLL_FRAC_TOP 0x22 +#define DA7219_PLL_FRAC_BOT 0x23 +#define DA7219_PLL_INTEGER 0x24 +#define DA7219_PLL_SRM_STS 0x25 +#define DA7219_DIG_ROUTING_DAI 0x2A +#define DA7219_DAI_CLK_MODE 0x2B +#define DA7219_DAI_CTRL 0x2C +#define DA7219_DAI_TDM_CTRL 0x2D +#define DA7219_DIG_ROUTING_DAC 0x2E +#define DA7219_ALC_CTRL1 0x2F +#define DA7219_DAI_OFFSET_LOWER 0x30 +#define DA7219_DAI_OFFSET_UPPER 0x31 +#define DA7219_REFERENCES 0x32 +#define DA7219_MIXIN_L_SELECT 0x33 +#define DA7219_MIXIN_L_GAIN 0x34 +#define DA7219_ADC_L_GAIN 0x36 +#define DA7219_ADC_FILTERS1 0x38 +#define DA7219_MIC_1_GAIN 0x39 +#define DA7219_SIDETONE_CTRL 0x3A +#define DA7219_SIDETONE_GAIN 0x3B +#define DA7219_DROUTING_ST_OUTFILT_1L 0x3C +#define DA7219_DROUTING_ST_OUTFILT_1R 0x3D +#define DA7219_DAC_FILTERS5 0x40 +#define DA7219_DAC_FILTERS2 0x41 +#define DA7219_DAC_FILTERS3 0x42 +#define DA7219_DAC_FILTERS4 0x43 +#define DA7219_DAC_FILTERS1 0x44 +#define DA7219_DAC_L_GAIN 0x45 +#define DA7219_DAC_R_GAIN 0x46 +#define DA7219_CP_CTRL 0x47 +#define DA7219_HP_L_GAIN 0x48 +#define DA7219_HP_R_GAIN 0x49 +#define DA7219_MIXOUT_L_SELECT 0x4B +#define DA7219_MIXOUT_R_SELECT 0x4C +#define DA7219_SYSTEM_MODES_INPUT 0x50 +#define DA7219_SYSTEM_MODES_OUTPUT 0x51 +#define DA7219_MICBIAS_CTRL 0x62 +#define DA7219_MIC_1_CTRL 0x63 +#define DA7219_MIXIN_L_CTRL 0x65 +#define DA7219_ADC_L_CTRL 0x67 +#define DA7219_DAC_L_CTRL 0x69 +#define DA7219_DAC_R_CTRL 0x6A +#define DA7219_HP_L_CTRL 0x6B +#define DA7219_HP_R_CTRL 0x6C +#define DA7219_MIXOUT_L_CTRL 0x6E +#define DA7219_MIXOUT_R_CTRL 0x6F +#define DA7219_CHIP_ID1 0x81 +#define DA7219_CHIP_ID2 0x82 +#define DA7219_CHIP_REVISION 0x83 +#define DA7219_LDO_CTRL 0x90 +#define DA7219_IO_CTRL 0x91 +#define DA7219_GAIN_RAMP_CTRL 0x92 +#define DA7219_PC_COUNT 0x94 +#define DA7219_CP_VOL_THRESHOLD1 0x95 +#define DA7219_CP_DELAY 0x96 +#define DA7219_DIG_CTRL 0x99 +#define DA7219_ALC_CTRL2 0x9A +#define DA7219_ALC_CTRL3 0x9B +#define DA7219_ALC_NOISE 0x9C +#define DA7219_ALC_TARGET_MIN 0x9D +#define DA7219_ALC_TARGET_MAX 0x9E +#define DA7219_ALC_GAIN_LIMITS 0x9F +#define DA7219_ALC_ANA_GAIN_LIMITS 0xA0 +#define DA7219_ALC_ANTICLIP_CTRL 0xA1 +#define DA7219_ALC_ANTICLIP_LEVEL 0xA2 +#define DA7219_ALC_OFFSET_AUTO_M_L 0xA3 +#define DA7219_ALC_OFFSET_AUTO_U_L 0xA4 +#define DA7219_DAC_NG_SETUP_TIME 0xAF +#define DA7219_DAC_NG_OFF_THRESH 0xB0 +#define DA7219_DAC_NG_ON_THRESH 0xB1 +#define DA7219_DAC_NG_CTRL 0xB2 +#define DA7219_TONE_GEN_CFG1 0xB4 +#define DA7219_TONE_GEN_CFG2 0xB5 +#define DA7219_TONE_GEN_CYCLES 0xB6 +#define DA7219_TONE_GEN_FREQ1_L 0xB7 +#define DA7219_TONE_GEN_FREQ1_U 0xB8 +#define DA7219_TONE_GEN_FREQ2_L 0xB9 +#define DA7219_TONE_GEN_FREQ2_U 0xBA +#define DA7219_TONE_GEN_ON_PER 0xBB +#define DA7219_TONE_GEN_OFF_PER 0xBC +#define DA7219_SYSTEM_STATUS 0xE0 +#define DA7219_SYSTEM_ACTIVE 0xFD + + +/* + * Bit Fields + */ + +#define DA7219_SWITCH_EN_MAX 0x1 + +/* DA7219_MIC_1_GAIN_STATUS = 0x6 */ +#define DA7219_MIC_1_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_MIC_1_AMP_GAIN_STATUS_MASK (0x7 << 0) +#define DA7219_MIC_1_AMP_GAIN_MAX 0x7 + +/* DA7219_MIXIN_L_GAIN_STATUS = 0x8 */ +#define DA7219_MIXIN_L_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_MIXIN_L_AMP_GAIN_STATUS_MASK (0xF << 0) + +/* DA7219_ADC_L_GAIN_STATUS = 0xA */ +#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_SHIFT 0 +#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_MASK (0x7F << 0) + +/* DA7219_DAC_L_GAIN_STATUS = 0xC */ +#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_SHIFT 0 +#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_MASK (0x7F << 0) + +/* DA7219_DAC_R_GAIN_STATUS = 0xD */ +#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_SHIFT 0 +#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_MASK (0x7F << 0) + +/* DA7219_HP_L_GAIN_STATUS = 0xE */ +#define DA7219_HP_L_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_HP_L_AMP_GAIN_STATUS_MASK (0x3F << 0) + +/* DA7219_HP_R_GAIN_STATUS = 0xF */ +#define DA7219_HP_R_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_HP_R_AMP_GAIN_STATUS_MASK (0x3F << 0) + +/* DA7219_MIC_1_SELECT = 0x10 */ +#define DA7219_MIC_1_AMP_IN_SEL_SHIFT 0 +#define DA7219_MIC_1_AMP_IN_SEL_MASK (0x3 << 0) + +/* DA7219_CIF_TIMEOUT_CTRL = 0x12 */ +#define DA7219_I2C_TIMEOUT_EN_SHIFT 0 +#define DA7219_I2C_TIMEOUT_EN_MASK (0x1 << 0) + +/* DA7219_CIF_CTRL = 0x13 */ +#define DA7219_CIF_I2C_WRITE_MODE_SHIFT 0 +#define DA7219_CIF_I2C_WRITE_MODE_MASK (0x1 << 0) +#define DA7219_CIF_REG_SOFT_RESET_SHIFT 7 +#define DA7219_CIF_REG_SOFT_RESET_MASK (0x1 << 7) + +/* DA7219_SR_24_48 = 0x16 */ +#define DA7219_SR_24_48_SHIFT 0 +#define DA7219_SR_24_48_MASK (0x1 << 0) + +/* DA7219_SR = 0x17 */ +#define DA7219_SR_SHIFT 0 +#define DA7219_SR_MASK (0xF << 0) +#define DA7219_SR_8000 (0x01 << 0) +#define DA7219_SR_11025 (0x02 << 0) +#define DA7219_SR_12000 (0x03 << 0) +#define DA7219_SR_16000 (0x05 << 0) +#define DA7219_SR_22050 (0x06 << 0) +#define DA7219_SR_24000 (0x07 << 0) +#define DA7219_SR_32000 (0x09 << 0) +#define DA7219_SR_44100 (0x0A << 0) +#define DA7219_SR_48000 (0x0B << 0) +#define DA7219_SR_88200 (0x0E << 0) +#define DA7219_SR_96000 (0x0F << 0) + +/* DA7219_CIF_I2C_ADDR_CFG = 0x1B */ +#define DA7219_CIF_I2C_ADDR_CFG_SHIFT 0 +#define DA7219_CIF_I2C_ADDR_CFG_MASK (0x3 << 0) + +/* DA7219_PLL_CTRL = 0x20 */ +#define DA7219_PLL_INDIV_SHIFT 2 +#define DA7219_PLL_INDIV_MASK (0x7 << 2) +#define DA7219_PLL_INDIV_2_5_MHZ (0x0 << 2) +#define DA7219_PLL_INDIV_5_10_MHZ (0x1 << 2) +#define DA7219_PLL_INDIV_10_20_MHZ (0x2 << 2) +#define DA7219_PLL_INDIV_20_40_MHZ (0x3 << 2) +#define DA7219_PLL_INDIV_40_54_MHZ (0x4 << 2) +#define DA7219_PLL_MCLK_SQR_EN_SHIFT 5 +#define DA7219_PLL_MCLK_SQR_EN_MASK (0x1 << 5) +#define DA7219_PLL_MODE_SHIFT 6 +#define DA7219_PLL_MODE_MASK (0x3 << 6) +#define DA7219_PLL_MODE_BYPASS (0x0 << 6) +#define DA7219_PLL_MODE_NORMAL (0x1 << 6) +#define DA7219_PLL_MODE_SRM (0x2 << 6) +#define DA7219_PLL_MODE_32KHZ (0x3 << 6) + +/* DA7219_PLL_FRAC_TOP = 0x22 */ +#define DA7219_PLL_FBDIV_FRAC_TOP_SHIFT 0 +#define DA7219_PLL_FBDIV_FRAC_TOP_MASK (0x1F << 0) + +/* DA7219_PLL_FRAC_BOT = 0x23 */ +#define DA7219_PLL_FBDIV_FRAC_BOT_SHIFT 0 +#define DA7219_PLL_FBDIV_FRAC_BOT_MASK (0xFF << 0) + +/* DA7219_PLL_INTEGER = 0x24 */ +#define DA7219_PLL_FBDIV_INTEGER_SHIFT 0 +#define DA7219_PLL_FBDIV_INTEGER_MASK (0x7F << 0) + +/* DA7219_PLL_SRM_STS = 0x25 */ +#define DA7219_PLL_SRM_STATE_SHIFT 0 +#define DA7219_PLL_SRM_STATE_MASK (0xF << 0) +#define DA7219_PLL_SRM_STATUS_SHIFT 4 +#define DA7219_PLL_SRM_STATUS_MASK (0xF << 4) +#define DA7219_PLL_SRM_STS_SRM_LOCK (0x1 << 7) + +/* DA7219_DIG_ROUTING_DAI = 0x2A */ +#define DA7219_DAI_L_SRC_SHIFT 0 +#define DA7219_DAI_L_SRC_MASK (0x3 << 0) +#define DA7219_DAI_R_SRC_SHIFT 4 +#define DA7219_DAI_R_SRC_MASK (0x3 << 4) +#define DA7219_OUT_SRC_MAX 4 + +/* DA7219_DAI_CLK_MODE = 0x2B */ +#define DA7219_DAI_BCLKS_PER_WCLK_SHIFT 0 +#define DA7219_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_32 (0x0 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_64 (0x1 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_128 (0x2 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_256 (0x3 << 0) +#define DA7219_DAI_CLK_POL_SHIFT 2 +#define DA7219_DAI_CLK_POL_MASK (0x1 << 2) +#define DA7219_DAI_CLK_POL_INV (0x1 << 2) +#define DA7219_DAI_WCLK_POL_SHIFT 3 +#define DA7219_DAI_WCLK_POL_MASK (0x1 << 3) +#define DA7219_DAI_WCLK_POL_INV (0x1 << 3) +#define DA7219_DAI_WCLK_TRI_STATE_SHIFT 4 +#define DA7219_DAI_WCLK_TRI_STATE_MASK (0x1 << 4) +#define DA7219_DAI_CLK_EN_SHIFT 7 +#define DA7219_DAI_CLK_EN_MASK (0x1 << 7) + +/* DA7219_DAI_CTRL = 0x2C */ +#define DA7219_DAI_FORMAT_SHIFT 0 +#define DA7219_DAI_FORMAT_MASK (0x3 << 0) +#define DA7219_DAI_FORMAT_I2S (0x0 << 0) +#define DA7219_DAI_FORMAT_LEFT_J (0x1 << 0) +#define DA7219_DAI_FORMAT_RIGHT_J (0x2 << 0) +#define DA7219_DAI_FORMAT_DSP (0x3 << 0) +#define DA7219_DAI_WORD_LENGTH_SHIFT 2 +#define DA7219_DAI_WORD_LENGTH_MASK (0x3 << 2) +#define DA7219_DAI_WORD_LENGTH_S16_LE (0x0 << 2) +#define DA7219_DAI_WORD_LENGTH_S20_LE (0x1 << 2) +#define DA7219_DAI_WORD_LENGTH_S24_LE (0x2 << 2) +#define DA7219_DAI_WORD_LENGTH_S32_LE (0x3 << 2) +#define DA7219_DAI_CH_NUM_SHIFT 4 +#define DA7219_DAI_CH_NUM_MASK (0x3 << 4) +#define DA7219_DAI_CH_NUM_MAX 2 +#define DA7219_DAI_EN_SHIFT 7 +#define DA7219_DAI_EN_MASK (0x1 << 7) + +/* DA7219_DAI_TDM_CTRL = 0x2D */ +#define DA7219_DAI_TDM_CH_EN_SHIFT 0 +#define DA7219_DAI_TDM_CH_EN_MASK (0x3 << 0) +#define DA7219_DAI_OE_SHIFT 6 +#define DA7219_DAI_OE_MASK (0x1 << 6) +#define DA7219_DAI_TDM_MODE_EN_SHIFT 7 +#define DA7219_DAI_TDM_MODE_EN_MASK (0x1 << 7) +#define DA7219_DAI_TDM_MAX_SLOTS 2 + +/* DA7219_DIG_ROUTING_DAC = 0x2E */ +#define DA7219_DAC_L_SRC_SHIFT 0 +#define DA7219_DAC_L_SRC_MASK (0x3 << 0) +#define DA7219_DAC_L_SRC_TONEGEN (0x1 << 0) +#define DA7219_DAC_L_MONO_SHIFT 3 +#define DA7219_DAC_L_MONO_MASK (0x1 << 3) +#define DA7219_DAC_R_SRC_SHIFT 4 +#define DA7219_DAC_R_SRC_MASK (0x3 << 4) +#define DA7219_DAC_R_SRC_TONEGEN (0x1 << 4) +#define DA7219_DAC_R_MONO_SHIFT 7 +#define DA7219_DAC_R_MONO_MASK (0x1 << 7) + +/* DA7219_ALC_CTRL1 = 0x2F */ +#define DA7219_ALC_OFFSET_EN_SHIFT 0 +#define DA7219_ALC_OFFSET_EN_MASK (0x1 << 0) +#define DA7219_ALC_SYNC_MODE_SHIFT 1 +#define DA7219_ALC_SYNC_MODE_MASK (0x1 << 1) +#define DA7219_ALC_EN_SHIFT 3 +#define DA7219_ALC_EN_MASK (0x1 << 3) +#define DA7219_ALC_AUTO_CALIB_EN_SHIFT 4 +#define DA7219_ALC_AUTO_CALIB_EN_MASK (0x1 << 4) +#define DA7219_ALC_CALIB_OVERFLOW_SHIFT 5 +#define DA7219_ALC_CALIB_OVERFLOW_MASK (0x1 << 5) + +/* DA7219_DAI_OFFSET_LOWER = 0x30 */ +#define DA7219_DAI_OFFSET_LOWER_SHIFT 0 +#define DA7219_DAI_OFFSET_LOWER_MASK (0xFF << 0) + +/* DA7219_DAI_OFFSET_UPPER = 0x31 */ +#define DA7219_DAI_OFFSET_UPPER_SHIFT 0 +#define DA7219_DAI_OFFSET_UPPER_MASK (0x7 << 0) +#define DA7219_DAI_OFFSET_MAX 0x2FF + +/* DA7219_REFERENCES = 0x32 */ +#define DA7219_BIAS_EN_SHIFT 3 +#define DA7219_BIAS_EN_MASK (0x1 << 3) +#define DA7219_VMID_FAST_CHARGE_SHIFT 4 +#define DA7219_VMID_FAST_CHARGE_MASK (0x1 << 4) + +/* DA7219_MIXIN_L_SELECT = 0x33 */ +#define DA7219_MIXIN_L_MIX_SELECT_SHIFT 0 +#define DA7219_MIXIN_L_MIX_SELECT_MASK (0x1 << 0) + +/* DA7219_MIXIN_L_GAIN = 0x34 */ +#define DA7219_MIXIN_L_AMP_GAIN_SHIFT 0 +#define DA7219_MIXIN_L_AMP_GAIN_MASK (0xF << 0) +#define DA7219_MIXIN_L_AMP_GAIN_MAX 0xF + +/* DA7219_ADC_L_GAIN = 0x36 */ +#define DA7219_ADC_L_DIGITAL_GAIN_SHIFT 0 +#define DA7219_ADC_L_DIGITAL_GAIN_MASK (0x7F << 0) +#define DA7219_ADC_L_DIGITAL_GAIN_MAX 0x7F + +/* DA7219_ADC_FILTERS1 = 0x38 */ +#define DA7219_ADC_VOICE_HPF_CORNER_SHIFT 0 +#define DA7219_ADC_VOICE_HPF_CORNER_MASK (0x7 << 0) +#define DA7219_VOICE_HPF_CORNER_MAX 8 +#define DA7219_ADC_VOICE_EN_SHIFT 3 +#define DA7219_ADC_VOICE_EN_MASK (0x1 << 3) +#define DA7219_ADC_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7219_ADC_AUDIO_HPF_CORNER_MASK (0x3 << 4) +#define DA7219_AUDIO_HPF_CORNER_MAX 4 +#define DA7219_ADC_HPF_EN_SHIFT 7 +#define DA7219_ADC_HPF_EN_MASK (0x1 << 7) +#define DA7219_HPF_MODE_SHIFT 0 +#define DA7219_HPF_DISABLED ((0x0 << 3) | (0x0 << 7)) +#define DA7219_HPF_AUDIO_EN ((0x0 << 3) | (0x1 << 7)) +#define DA7219_HPF_VOICE_EN ((0x1 << 3) | (0x1 << 7)) +#define DA7219_HPF_MODE_MASK ((0x1 << 3) | (0x1 << 7)) +#define DA7219_HPF_MODE_MAX 3 + +/* DA7219_MIC_1_GAIN = 0x39 */ +#define DA7219_MIC_1_AMP_GAIN_SHIFT 0 +#define DA7219_MIC_1_AMP_GAIN_MASK (0x7 << 0) + +/* DA7219_SIDETONE_CTRL = 0x3A */ +#define DA7219_SIDETONE_MUTE_EN_SHIFT 6 +#define DA7219_SIDETONE_MUTE_EN_MASK (0x1 << 6) +#define DA7219_SIDETONE_EN_SHIFT 7 +#define DA7219_SIDETONE_EN_MASK (0x1 << 7) + +/* DA7219_SIDETONE_GAIN = 0x3B */ +#define DA7219_SIDETONE_GAIN_SHIFT 0 +#define DA7219_SIDETONE_GAIN_MASK (0xF << 0) +#define DA7219_SIDETONE_GAIN_MAX 0xE + +/* DA7219_DROUTING_ST_OUTFILT_1L = 0x3C */ +#define DA7219_OUTFILT_ST_1L_SRC_SHIFT 0 +#define DA7219_OUTFILT_ST_1L_SRC_MASK (0x7 << 0) +#define DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT 0 +#define DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT 1 +#define DA7219_DMIX_ST_SRC_SIDETONE_SHIFT 2 +#define DA7219_DMIX_ST_SRC_OUTFILT1L (0x1 << 0) +#define DA7219_DMIX_ST_SRC_OUTFILT1R (0x1 << 1) + +/* DA7219_DROUTING_ST_OUTFILT_1R = 0x3D */ +#define DA7219_OUTFILT_ST_1R_SRC_SHIFT 0 +#define DA7219_OUTFILT_ST_1R_SRC_MASK (0x7 << 0) + +/* DA7219_DAC_FILTERS5 = 0x40 */ +#define DA7219_DAC_SOFTMUTE_RATE_SHIFT 4 +#define DA7219_DAC_SOFTMUTE_RATE_MASK (0x7 << 4) +#define DA7219_DAC_SOFTMUTE_RATE_MAX 7 +#define DA7219_DAC_SOFTMUTE_EN_SHIFT 7 +#define DA7219_DAC_SOFTMUTE_EN_MASK (0x1 << 7) + +/* DA7219_DAC_FILTERS2 = 0x41 */ +#define DA7219_DAC_EQ_BAND1_SHIFT 0 +#define DA7219_DAC_EQ_BAND1_MASK (0xF << 0) +#define DA7219_DAC_EQ_BAND2_SHIFT 4 +#define DA7219_DAC_EQ_BAND2_MASK (0xF << 4) +#define DA7219_DAC_EQ_BAND_MAX 0xF + +/* DA7219_DAC_FILTERS3 = 0x42 */ +#define DA7219_DAC_EQ_BAND3_SHIFT 0 +#define DA7219_DAC_EQ_BAND3_MASK (0xF << 0) +#define DA7219_DAC_EQ_BAND4_SHIFT 4 +#define DA7219_DAC_EQ_BAND4_MASK (0xF << 4) + +/* DA7219_DAC_FILTERS4 = 0x43 */ +#define DA7219_DAC_EQ_BAND5_SHIFT 0 +#define DA7219_DAC_EQ_BAND5_MASK (0xF << 0) +#define DA7219_DAC_EQ_EN_SHIFT 7 +#define DA7219_DAC_EQ_EN_MASK (0x1 << 7) + +/* DA7219_DAC_FILTERS1 = 0x44 */ +#define DA7219_DAC_VOICE_HPF_CORNER_SHIFT 0 +#define DA7219_DAC_VOICE_HPF_CORNER_MASK (0x7 << 0) +#define DA7219_DAC_VOICE_EN_SHIFT 3 +#define DA7219_DAC_VOICE_EN_MASK (0x1 << 3) +#define DA7219_DAC_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7219_DAC_AUDIO_HPF_CORNER_MASK (0x3 << 4) +#define DA7219_DAC_HPF_EN_SHIFT 7 +#define DA7219_DAC_HPF_EN_MASK (0x1 << 7) + +/* DA7219_DAC_L_GAIN = 0x45 */ +#define DA7219_DAC_L_DIGITAL_GAIN_SHIFT 0 +#define DA7219_DAC_L_DIGITAL_GAIN_MASK (0x7F << 0) +#define DA7219_DAC_DIGITAL_GAIN_MAX 0x7F +#define DA7219_DAC_DIGITAL_GAIN_0DB (0x6F << 0) + +/* DA7219_DAC_R_GAIN = 0x46 */ +#define DA7219_DAC_R_DIGITAL_GAIN_SHIFT 0 +#define DA7219_DAC_R_DIGITAL_GAIN_MASK (0x7F << 0) + +/* DA7219_CP_CTRL = 0x47 */ +#define DA7219_CP_MCHANGE_SHIFT 4 +#define DA7219_CP_MCHANGE_MASK (0x3 << 4) +#define DA7219_CP_MCHANGE_REL_MASK 0x3 +#define DA7219_CP_MCHANGE_MAX 3 +#define DA7219_CP_MCHANGE_LARGEST_VOL 0x1 +#define DA7219_CP_MCHANGE_DAC_VOL 0x2 +#define DA7219_CP_MCHANGE_SIG_MAG 0x3 +#define DA7219_CP_EN_SHIFT 7 +#define DA7219_CP_EN_MASK (0x1 << 7) + +/* DA7219_HP_L_GAIN = 0x48 */ +#define DA7219_HP_L_AMP_GAIN_SHIFT 0 +#define DA7219_HP_L_AMP_GAIN_MASK (0x3F << 0) +#define DA7219_HP_AMP_GAIN_MAX 0x3F +#define DA7219_HP_AMP_GAIN_0DB (0x39 << 0) + +/* DA7219_HP_R_GAIN = 0x49 */ +#define DA7219_HP_R_AMP_GAIN_SHIFT 0 +#define DA7219_HP_R_AMP_GAIN_MASK (0x3F << 0) + +/* DA7219_MIXOUT_L_SELECT = 0x4B */ +#define DA7219_MIXOUT_L_MIX_SELECT_SHIFT 0 +#define DA7219_MIXOUT_L_MIX_SELECT_MASK (0x1 << 0) + +/* DA7219_MIXOUT_R_SELECT = 0x4C */ +#define DA7219_MIXOUT_R_MIX_SELECT_SHIFT 0 +#define DA7219_MIXOUT_R_MIX_SELECT_MASK (0x1 << 0) + +/* DA7219_SYSTEM_MODES_INPUT = 0x50 */ +#define DA7219_MODE_SUBMIT_SHIFT 0 +#define DA7219_MODE_SUBMIT_MASK (0x1 << 0) +#define DA7219_ADC_MODE_SHIFT 1 +#define DA7219_ADC_MODE_MASK (0x7F << 1) + +/* DA7219_SYSTEM_MODES_OUTPUT = 0x51 */ +#define DA7219_MODE_SUBMIT_SHIFT 0 +#define DA7219_MODE_SUBMIT_MASK (0x1 << 0) +#define DA7219_DAC_MODE_SHIFT 1 +#define DA7219_DAC_MODE_MASK (0x7F << 1) + +/* DA7219_MICBIAS_CTRL = 0x62 */ +#define DA7219_MICBIAS1_LEVEL_SHIFT 0 +#define DA7219_MICBIAS1_LEVEL_MASK (0x7 << 0) +#define DA7219_MICBIAS1_EN_SHIFT 3 +#define DA7219_MICBIAS1_EN_MASK (0x1 << 3) + +/* DA7219_MIC_1_CTRL = 0x63 */ +#define DA7219_MIC_1_AMP_RAMP_EN_SHIFT 5 +#define DA7219_MIC_1_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_MIC_1_AMP_MUTE_EN_SHIFT 6 +#define DA7219_MIC_1_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_MIC_1_AMP_EN_SHIFT 7 +#define DA7219_MIC_1_AMP_EN_MASK (0x1 << 7) + +/* DA7219_MIXIN_L_CTRL = 0x65 */ +#define DA7219_MIXIN_L_MIX_EN_SHIFT 3 +#define DA7219_MIXIN_L_MIX_EN_MASK (0x1 << 3) +#define DA7219_MIXIN_L_AMP_ZC_EN_SHIFT 4 +#define DA7219_MIXIN_L_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT 5 +#define DA7219_MIXIN_L_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT 6 +#define DA7219_MIXIN_L_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_MIXIN_L_AMP_EN_SHIFT 7 +#define DA7219_MIXIN_L_AMP_EN_MASK (0x1 << 7) + +/* DA7219_ADC_L_CTRL = 0x67 */ +#define DA7219_ADC_L_BIAS_SHIFT 0 +#define DA7219_ADC_L_BIAS_MASK (0x3 << 0) +#define DA7219_ADC_L_RAMP_EN_SHIFT 5 +#define DA7219_ADC_L_RAMP_EN_MASK (0x1 << 5) +#define DA7219_ADC_L_MUTE_EN_SHIFT 6 +#define DA7219_ADC_L_MUTE_EN_MASK (0x1 << 6) +#define DA7219_ADC_L_EN_SHIFT 7 +#define DA7219_ADC_L_EN_MASK (0x1 << 7) + +/* DA7219_DAC_L_CTRL = 0x69 */ +#define DA7219_DAC_L_RAMP_EN_SHIFT 5 +#define DA7219_DAC_L_RAMP_EN_MASK (0x1 << 5) +#define DA7219_DAC_L_MUTE_EN_SHIFT 6 +#define DA7219_DAC_L_MUTE_EN_MASK (0x1 << 6) +#define DA7219_DAC_L_EN_SHIFT 7 +#define DA7219_DAC_L_EN_MASK (0x1 << 7) + +/* DA7219_DAC_R_CTRL = 0x6A */ +#define DA7219_DAC_R_RAMP_EN_SHIFT 5 +#define DA7219_DAC_R_RAMP_EN_MASK (0x1 << 5) +#define DA7219_DAC_R_MUTE_EN_SHIFT 6 +#define DA7219_DAC_R_MUTE_EN_MASK (0x1 << 6) +#define DA7219_DAC_R_EN_SHIFT 7 +#define DA7219_DAC_R_EN_MASK (0x1 << 7) + +/* DA7219_HP_L_CTRL = 0x6B */ +#define DA7219_HP_L_AMP_MIN_GAIN_EN_SHIFT 2 +#define DA7219_HP_L_AMP_MIN_GAIN_EN_MASK (0x1 << 2) +#define DA7219_HP_L_AMP_OE_SHIFT 3 +#define DA7219_HP_L_AMP_OE_MASK (0x1 << 3) +#define DA7219_HP_L_AMP_ZC_EN_SHIFT 4 +#define DA7219_HP_L_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7219_HP_L_AMP_RAMP_EN_SHIFT 5 +#define DA7219_HP_L_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_HP_L_AMP_MUTE_EN_SHIFT 6 +#define DA7219_HP_L_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_HP_L_AMP_EN_SHIFT 7 +#define DA7219_HP_L_AMP_EN_MASK (0x1 << 7) + +/* DA7219_HP_R_CTRL = 0x6C */ +#define DA7219_HP_R_AMP_MIN_GAIN_EN_SHIFT 2 +#define DA7219_HP_R_AMP_MIN_GAIN_EN_MASK (0x1 << 2) +#define DA7219_HP_R_AMP_OE_SHIFT 3 +#define DA7219_HP_R_AMP_OE_MASK (0x1 << 3) +#define DA7219_HP_R_AMP_ZC_EN_SHIFT 4 +#define DA7219_HP_R_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7219_HP_R_AMP_RAMP_EN_SHIFT 5 +#define DA7219_HP_R_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_HP_R_AMP_MUTE_EN_SHIFT 6 +#define DA7219_HP_R_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_HP_R_AMP_EN_SHIFT 7 +#define DA7219_HP_R_AMP_EN_MASK (0x1 << 7) + +/* DA7219_MIXOUT_L_CTRL = 0x6E */ +#define DA7219_MIXOUT_L_AMP_EN_SHIFT 7 +#define DA7219_MIXOUT_L_AMP_EN_MASK (0x1 << 7) + +/* DA7219_MIXOUT_R_CTRL = 0x6F */ +#define DA7219_MIXOUT_R_AMP_EN_SHIFT 7 +#define DA7219_MIXOUT_R_AMP_EN_MASK (0x1 << 7) + +/* DA7219_CHIP_ID1 = 0x81 */ +#define DA7219_CHIP_ID1_SHIFT 0 +#define DA7219_CHIP_ID1_MASK (0xFF << 0) + +/* DA7219_CHIP_ID2 = 0x82 */ +#define DA7219_CHIP_ID2_SHIFT 0 +#define DA7219_CHIP_ID2_MASK (0xFF << 0) + +/* DA7219_CHIP_REVISION = 0x83 */ +#define DA7219_CHIP_MINOR_SHIFT 0 +#define DA7219_CHIP_MINOR_MASK (0xF << 0) +#define DA7219_CHIP_MAJOR_SHIFT 4 +#define DA7219_CHIP_MAJOR_MASK (0xF << 4) + +/* DA7219_LDO_CTRL = 0x90 */ +#define DA7219_LDO_LEVEL_SELECT_SHIFT 4 +#define DA7219_LDO_LEVEL_SELECT_MASK (0x3 << 4) +#define DA7219_LDO_EN_SHIFT 7 +#define DA7219_LDO_EN_MASK (0x1 << 7) + +/* DA7219_IO_CTRL = 0x91 */ +#define DA7219_IO_VOLTAGE_LEVEL_SHIFT 0 +#define DA7219_IO_VOLTAGE_LEVEL_MASK (0x1 << 0) +#define DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V 0 +#define DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V 1 + +/* DA7219_GAIN_RAMP_CTRL = 0x92 */ +#define DA7219_GAIN_RAMP_RATE_SHIFT 0 +#define DA7219_GAIN_RAMP_RATE_MASK (0x3 << 0) +#define DA7219_GAIN_RAMP_RATE_MAX 4 + +/* DA7219_PC_COUNT = 0x94 */ +#define DA7219_PC_FREERUN_SHIFT 0 +#define DA7219_PC_FREERUN_MASK (0x1 << 0) +#define DA7219_PC_RESYNC_AUTO_SHIFT 1 +#define DA7219_PC_RESYNC_AUTO_MASK (0x1 << 1) + +/* DA7219_CP_VOL_THRESHOLD1 = 0x95 */ +#define DA7219_CP_THRESH_VDD2_SHIFT 0 +#define DA7219_CP_THRESH_VDD2_MASK (0x3F << 0) +#define DA7219_CP_THRESH_VDD2_MAX 0x3F + +/* DA7219_DIG_CTRL = 0x99 */ +#define DA7219_DAC_L_INV_SHIFT 3 +#define DA7219_DAC_L_INV_MASK (0x1 << 3) +#define DA7219_DAC_R_INV_SHIFT 7 +#define DA7219_DAC_R_INV_MASK (0x1 << 7) + +/* DA7219_ALC_CTRL2 = 0x9A */ +#define DA7219_ALC_ATTACK_SHIFT 0 +#define DA7219_ALC_ATTACK_MASK (0xF << 0) +#define DA7219_ALC_ATTACK_MAX 13 +#define DA7219_ALC_RELEASE_SHIFT 4 +#define DA7219_ALC_RELEASE_MASK (0xF << 4) +#define DA7219_ALC_RELEASE_MAX 11 + +/* DA7219_ALC_CTRL3 = 0x9B */ +#define DA7219_ALC_HOLD_SHIFT 0 +#define DA7219_ALC_HOLD_MASK (0xF << 0) +#define DA7219_ALC_HOLD_MAX 16 +#define DA7219_ALC_INTEG_ATTACK_SHIFT 4 +#define DA7219_ALC_INTEG_ATTACK_MASK (0x3 << 4) +#define DA7219_ALC_INTEG_RELEASE_SHIFT 6 +#define DA7219_ALC_INTEG_RELEASE_MASK (0x3 << 6) +#define DA7219_ALC_INTEG_MAX 4 + +/* DA7219_ALC_NOISE = 0x9C */ +#define DA7219_ALC_NOISE_SHIFT 0 +#define DA7219_ALC_NOISE_MASK (0x3F << 0) +#define DA7219_ALC_THRESHOLD_MAX 0x3F + +/* DA7219_ALC_TARGET_MIN = 0x9D */ +#define DA7219_ALC_THRESHOLD_MIN_SHIFT 0 +#define DA7219_ALC_THRESHOLD_MIN_MASK (0x3F << 0) + +/* DA7219_ALC_TARGET_MAX = 0x9E */ +#define DA7219_ALC_THRESHOLD_MAX_SHIFT 0 +#define DA7219_ALC_THRESHOLD_MAX_MASK (0x3F << 0) + +/* DA7219_ALC_GAIN_LIMITS = 0x9F */ +#define DA7219_ALC_ATTEN_MAX_SHIFT 0 +#define DA7219_ALC_ATTEN_MAX_MASK (0xF << 0) +#define DA7219_ALC_GAIN_MAX_SHIFT 4 +#define DA7219_ALC_GAIN_MAX_MASK (0xF << 4) +#define DA7219_ALC_ATTEN_GAIN_MAX 0xF + +/* DA7219_ALC_ANA_GAIN_LIMITS = 0xA0 */ +#define DA7219_ALC_ANA_GAIN_MIN_SHIFT 0 +#define DA7219_ALC_ANA_GAIN_MIN_MASK (0x7 << 0) +#define DA7219_ALC_ANA_GAIN_MIN 0x1 +#define DA7219_ALC_ANA_GAIN_MAX_SHIFT 4 +#define DA7219_ALC_ANA_GAIN_MAX_MASK (0x7 << 4) +#define DA7219_ALC_ANA_GAIN_MAX 0x7 + +/* DA7219_ALC_ANTICLIP_CTRL = 0xA1 */ +#define DA7219_ALC_ANTICLIP_STEP_SHIFT 0 +#define DA7219_ALC_ANTICLIP_STEP_MASK (0x3 << 0) +#define DA7219_ALC_ANTICLIP_STEP_MAX 4 +#define DA7219_ALC_ANTIPCLIP_EN_SHIFT 7 +#define DA7219_ALC_ANTIPCLIP_EN_MASK (0x1 << 7) + +/* DA7219_ALC_ANTICLIP_LEVEL = 0xA2 */ +#define DA7219_ALC_ANTICLIP_LEVEL_SHIFT 0 +#define DA7219_ALC_ANTICLIP_LEVEL_MASK (0x7F << 0) + +/* DA7219_ALC_OFFSET_AUTO_M_L = 0xA3 */ +#define DA7219_ALC_OFFSET_AUTO_M_L_SHIFT 0 +#define DA7219_ALC_OFFSET_AUTO_M_L_MASK (0xFF << 0) + +/* DA7219_ALC_OFFSET_AUTO_U_L = 0xA4 */ +#define DA7219_ALC_OFFSET_AUTO_U_L_SHIFT 0 +#define DA7219_ALC_OFFSET_AUTO_U_L_MASK (0xF << 0) + +/* DA7219_DAC_NG_SETUP_TIME = 0xAF */ +#define DA7219_DAC_NG_SETUP_TIME_SHIFT 0 +#define DA7219_DAC_NG_SETUP_TIME_MASK (0x3 << 0) +#define DA7219_DAC_NG_SETUP_TIME_MAX 4 +#define DA7219_DAC_NG_RAMPUP_RATE_SHIFT 2 +#define DA7219_DAC_NG_RAMPUP_RATE_MASK (0x1 << 2) +#define DA7219_DAC_NG_RAMPDN_RATE_SHIFT 3 +#define DA7219_DAC_NG_RAMPDN_RATE_MASK (0x1 << 3) +#define DA7219_DAC_NG_RAMP_RATE_MAX 2 + +/* DA7219_DAC_NG_OFF_THRESH = 0xB0 */ +#define DA7219_DAC_NG_OFF_THRESHOLD_SHIFT 0 +#define DA7219_DAC_NG_OFF_THRESHOLD_MASK (0x7 << 0) +#define DA7219_DAC_NG_THRESHOLD_MAX 0x7 + +/* DA7219_DAC_NG_ON_THRESH = 0xB1 */ +#define DA7219_DAC_NG_ON_THRESHOLD_SHIFT 0 +#define DA7219_DAC_NG_ON_THRESHOLD_MASK (0x7 << 0) + +/* DA7219_DAC_NG_CTRL = 0xB2 */ +#define DA7219_DAC_NG_EN_SHIFT 7 +#define DA7219_DAC_NG_EN_MASK (0x1 << 7) + +/* DA7219_TONE_GEN_CFG1 = 0xB4 */ +#define DA7219_DTMF_REG_SHIFT 0 +#define DA7219_DTMF_REG_MASK (0xF << 0) +#define DA7219_DTMF_REG_MAX 16 +#define DA7219_DTMF_EN_SHIFT 4 +#define DA7219_DTMF_EN_MASK (0x1 << 4) +#define DA7219_START_STOPN_SHIFT 7 +#define DA7219_START_STOPN_MASK (0x1 << 7) + +/* DA7219_TONE_GEN_CFG2 = 0xB5 */ +#define DA7219_SWG_SEL_SHIFT 0 +#define DA7219_SWG_SEL_MASK (0x3 << 0) +#define DA7219_SWG_SEL_MAX 4 +#define DA7219_SWG_SEL_SRAMP (0x3 << 0) +#define DA7219_TONE_GEN_GAIN_SHIFT 4 +#define DA7219_TONE_GEN_GAIN_MASK (0xF << 4) +#define DA7219_TONE_GEN_GAIN_MAX 0xF +#define DA7219_TONE_GEN_GAIN_MINUS_9DB (0x3 << 4) +#define DA7219_TONE_GEN_GAIN_MINUS_15DB (0x5 << 4) + +/* DA7219_TONE_GEN_CYCLES = 0xB6 */ +#define DA7219_BEEP_CYCLES_SHIFT 0 +#define DA7219_BEEP_CYCLES_MASK (0x7 << 0) + +/* DA7219_TONE_GEN_FREQ1_L = 0xB7 */ +#define DA7219_FREQ1_L_SHIFT 0 +#define DA7219_FREQ1_L_MASK (0xFF << 0) +#define DA7219_FREQ_MAX 0xFFFF + +/* DA7219_TONE_GEN_FREQ1_U = 0xB8 */ +#define DA7219_FREQ1_U_SHIFT 0 +#define DA7219_FREQ1_U_MASK (0xFF << 0) + +/* DA7219_TONE_GEN_FREQ2_L = 0xB9 */ +#define DA7219_FREQ2_L_SHIFT 0 +#define DA7219_FREQ2_L_MASK (0xFF << 0) + +/* DA7219_TONE_GEN_FREQ2_U = 0xBA */ +#define DA7219_FREQ2_U_SHIFT 0 +#define DA7219_FREQ2_U_MASK (0xFF << 0) + +/* DA7219_TONE_GEN_ON_PER = 0xBB */ +#define DA7219_BEEP_ON_PER_SHIFT 0 +#define DA7219_BEEP_ON_PER_MASK (0x3F << 0) +#define DA7219_BEEP_ON_OFF_MAX 0x3F + +/* DA7219_TONE_GEN_OFF_PER = 0xBC */ +#define DA7219_BEEP_OFF_PER_SHIFT 0 +#define DA7219_BEEP_OFF_PER_MASK (0x3F << 0) + +/* DA7219_SYSTEM_STATUS = 0xE0 */ +#define DA7219_SC1_BUSY_SHIFT 0 +#define DA7219_SC1_BUSY_MASK (0x1 << 0) +#define DA7219_SC2_BUSY_SHIFT 1 +#define DA7219_SC2_BUSY_MASK (0x1 << 1) + +/* DA7219_SYSTEM_ACTIVE = 0xFD */ +#define DA7219_SYSTEM_ACTIVE_SHIFT 0 +#define DA7219_SYSTEM_ACTIVE_MASK (0x1 << 0) + + +/* + * General defines & data + */ + +/* Register inversion */ +#define DA7219_NO_INVERT 0 +#define DA7219_INVERT 1 + +/* Byte related defines */ +#define DA7219_BYTE_SHIFT 8 +#define DA7219_BYTE_MASK 0xFF + +/* PLL Output Frequencies */ +#define DA7219_PLL_FREQ_OUT_90316 90316800 +#define DA7219_PLL_FREQ_OUT_98304 98304000 + +/* PLL Frequency Dividers */ +#define DA7219_PLL_INDIV_2_5_MHZ_VAL 1 +#define DA7219_PLL_INDIV_5_10_MHZ_VAL 2 +#define DA7219_PLL_INDIV_10_20_MHZ_VAL 4 +#define DA7219_PLL_INDIV_20_40_MHZ_VAL 8 +#define DA7219_PLL_INDIV_40_54_MHZ_VAL 16 + +/* SRM */ +#define DA7219_SRM_CHECK_RETRIES 8 + +enum da7219_clk_src { + DA7219_CLKSRC_MCLK = 0, + DA7219_CLKSRC_MCLK_SQR, +}; + +enum da7219_sys_clk { + DA7219_SYSCLK_MCLK = 0, + DA7219_SYSCLK_PLL, + DA7219_SYSCLK_PLL_SRM, + DA7219_SYSCLK_PLL_32KHZ +}; + +/* Regulators */ +enum da7219_supplies { + DA7219_SUPPLY_VDD = 0, + DA7219_SUPPLY_VDDMIC, + DA7219_SUPPLY_VDDIO, + DA7219_NUM_SUPPLIES, +}; + +struct da7219_aad_priv; + +/* Private data */ +struct da7219_priv { + struct da7219_aad_priv *aad; + struct da7219_pdata *pdata; + + struct regulator_bulk_data supplies[DA7219_NUM_SUPPLIES]; + struct regmap *regmap; + struct mutex lock; + + struct clk *mclk; + unsigned int mclk_rate; + int clk_src; + + bool master; + bool alc_en; +}; + +#endif /* __DA7219_H */ diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 6a091016e0fc..969e337dc17c 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -129,7 +129,7 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; int ret; if (deemph > 1) diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c deleted file mode 100644 index bd42ad34e004..000000000000 --- a/sound/soc/codecs/hdmi.c +++ /dev/null @@ -1,109 +0,0 @@ -/* - * ALSA SoC codec driver for HDMI audio codecs. - * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/ - * Author: Ricardo Neri <ricardo.neri@ti.com> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ -#include <linux/module.h> -#include <sound/soc.h> -#include <linux/of.h> -#include <linux/of_device.h> - -#define DRV_NAME "hdmi-audio-codec" - -static const struct snd_soc_dapm_widget hdmi_widgets[] = { - SND_SOC_DAPM_INPUT("RX"), - SND_SOC_DAPM_OUTPUT("TX"), -}; - -static const struct snd_soc_dapm_route hdmi_routes[] = { - { "Capture", NULL, "RX" }, - { "TX", NULL, "Playback" }, -}; - -static struct snd_soc_dai_driver hdmi_codec_dai = { - .name = "hdmi-hifi", - .playback = { - .stream_name = "Playback", - .channels_min = 2, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, - .sig_bits = 24, - }, - .capture = { - .stream_name = "Capture", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE, - }, - -}; - -#ifdef CONFIG_OF -static const struct of_device_id hdmi_audio_codec_ids[] = { - { .compatible = "linux,hdmi-audio", }, - { } -}; -MODULE_DEVICE_TABLE(of, hdmi_audio_codec_ids); -#endif - -static struct snd_soc_codec_driver hdmi_codec = { - .dapm_widgets = hdmi_widgets, - .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), - .dapm_routes = hdmi_routes, - .num_dapm_routes = ARRAY_SIZE(hdmi_routes), - .ignore_pmdown_time = true, -}; - -static int hdmi_codec_probe(struct platform_device *pdev) -{ - return snd_soc_register_codec(&pdev->dev, &hdmi_codec, - &hdmi_codec_dai, 1); -} - -static int hdmi_codec_remove(struct platform_device *pdev) -{ - snd_soc_unregister_codec(&pdev->dev); - return 0; -} - -static struct platform_driver hdmi_codec_driver = { - .driver = { - .name = DRV_NAME, - .of_match_table = of_match_ptr(hdmi_audio_codec_ids), - }, - - .probe = hdmi_codec_probe, - .remove = hdmi_codec_remove, -}; - -module_platform_driver(hdmi_codec_driver); - -MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>"); -MODULE_DESCRIPTION("ASoC generic HDMI codec driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 000000000000..7fc7b4e3f444 --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,1309 @@ +/* + * Nuvoton NAU8825 audio codec driver + * + * Copyright 2015 Google Chromium project. + * Author: Anatol Pomozov <anatol@chromium.org> + * Copyright 2015 Nuvoton Technology Corp. + * Co-author: Meng-Huang Kuo <mhkuo@nuvoton.com> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/acpi.h> +#include <linux/math64.h> + +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> + + +#include "nau8825.h" + +#define NAU_FREF_MAX 13500000 +#define NAU_FVCO_MAX 100000000 +#define NAU_FVCO_MIN 90000000 + +struct nau8825_fll { + int mclk_src; + int ratio; + int fll_frac; + int fll_int; + int clk_ref_div; +}; + +struct nau8825_fll_attr { + unsigned int param; + unsigned int val; +}; + +/* scaling for mclk from sysclk_src output */ +static const struct nau8825_fll_attr mclk_src_scaling[] = { + { 1, 0x0 }, + { 2, 0x2 }, + { 4, 0x3 }, + { 8, 0x4 }, + { 16, 0x5 }, + { 32, 0x6 }, + { 3, 0x7 }, + { 6, 0xa }, + { 12, 0xb }, + { 24, 0xc }, + { 48, 0xd }, + { 96, 0xe }, + { 5, 0xf }, +}; + +/* ratio for input clk freq */ +static const struct nau8825_fll_attr fll_ratio[] = { + { 512000, 0x01 }, + { 256000, 0x02 }, + { 128000, 0x04 }, + { 64000, 0x08 }, + { 32000, 0x10 }, + { 8000, 0x20 }, + { 4000, 0x40 }, +}; + +static const struct nau8825_fll_attr fll_pre_scalar[] = { + { 1, 0x0 }, + { 2, 0x1 }, + { 4, 0x2 }, + { 8, 0x3 }, +}; + +static const struct reg_default nau8825_reg_defaults[] = { + { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_CLK_DIVIDER, 0x0050 }, + { NAU8825_REG_FLL1, 0x0 }, + { NAU8825_REG_FLL2, 0x3126 }, + { NAU8825_REG_FLL3, 0x0008 }, + { NAU8825_REG_FLL4, 0x0010 }, + { NAU8825_REG_FLL5, 0x0 }, + { NAU8825_REG_FLL6, 0x6000 }, + { NAU8825_REG_FLL_VCO_RSV, 0xf13c }, + { NAU8825_REG_HSD_CTRL, 0x000c }, + { NAU8825_REG_JACK_DET_CTRL, 0x0 }, + { NAU8825_REG_INTERRUPT_MASK, 0x0 }, + { NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8825_REG_SAR_CTRL, 0x0015 }, + { NAU8825_REG_KEYDET_CTRL, 0x0110 }, + { NAU8825_REG_VDET_THRESHOLD_1, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_2, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_3, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_4, 0x0 }, + { NAU8825_REG_GPIO34_CTRL, 0x0 }, + { NAU8825_REG_GPIO12_CTRL, 0x0 }, + { NAU8825_REG_TDM_CTRL, 0x0 }, + { NAU8825_REG_I2S_PCM_CTRL1, 0x000b }, + { NAU8825_REG_I2S_PCM_CTRL2, 0x8010 }, + { NAU8825_REG_LEFT_TIME_SLOT, 0x0 }, + { NAU8825_REG_RIGHT_TIME_SLOT, 0x0 }, + { NAU8825_REG_BIQ_CTRL, 0x0 }, + { NAU8825_REG_BIQ_COF1, 0x0 }, + { NAU8825_REG_BIQ_COF2, 0x0 }, + { NAU8825_REG_BIQ_COF3, 0x0 }, + { NAU8825_REG_BIQ_COF4, 0x0 }, + { NAU8825_REG_BIQ_COF5, 0x0 }, + { NAU8825_REG_BIQ_COF6, 0x0 }, + { NAU8825_REG_BIQ_COF7, 0x0 }, + { NAU8825_REG_BIQ_COF8, 0x0 }, + { NAU8825_REG_BIQ_COF9, 0x0 }, + { NAU8825_REG_BIQ_COF10, 0x0 }, + { NAU8825_REG_ADC_RATE, 0x0010 }, + { NAU8825_REG_DAC_CTRL1, 0x0001 }, + { NAU8825_REG_DAC_CTRL2, 0x0 }, + { NAU8825_REG_DAC_DGAIN_CTRL, 0x0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf }, + { NAU8825_REG_MUTE_CTRL, 0x0 }, + { NAU8825_REG_HSVOL_CTRL, 0x0 }, + { NAU8825_REG_DACL_CTRL, 0x02cf }, + { NAU8825_REG_DACR_CTRL, 0x00cf }, + { NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_ADC_DRC_SLOPES, 0x25ff }, + { NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_IMM_MODE_CTRL, 0x0 }, + { NAU8825_REG_CLASSG_CTRL, 0x0 }, + { NAU8825_REG_OPT_EFUSE_CTRL, 0x0 }, + { NAU8825_REG_MISC_CTRL, 0x0 }, + { NAU8825_REG_BIAS_ADJ, 0x0 }, + { NAU8825_REG_TRIM_SETTINGS, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_1, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_2, 0x0 }, + { NAU8825_REG_ANALOG_ADC_1, 0x0011 }, + { NAU8825_REG_ANALOG_ADC_2, 0x0020 }, + { NAU8825_REG_RDAC, 0x0008 }, + { NAU8825_REG_MIC_BIAS, 0x0006 }, + { NAU8825_REG_BOOST, 0x0 }, + { NAU8825_REG_FEPGA, 0x0 }, + { NAU8825_REG_POWER_UP_CONTROL, 0x0 }, + { NAU8825_REG_CHARGE_PUMP, 0x0 }, +}; + +static bool nau8825_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } + +} + +static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK: + case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET: + case NAU8825_REG_IRQ_STATUS: + case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_IMM_RMS_L: + case NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_I2C_DEVICE_ID: + case NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_CHARGE_PUMP_INPUT_READ: + case NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static int nau8825_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Prevent startup click by letting charge pump to ramp up */ + msleep(10); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const char * const nau8825_adc_decimation[] = { + "32", "64", "128", "256" +}; + +static const struct soc_enum nau8825_adc_decimation_enum = + SOC_ENUM_SINGLE(NAU8825_REG_ADC_RATE, NAU8825_ADC_SYNC_DOWN_SFT, + ARRAY_SIZE(nau8825_adc_decimation), nau8825_adc_decimation); + +static const char * const nau8825_dac_oversampl[] = { + "64", "256", "128", "", "32" +}; + +static const struct soc_enum nau8825_dac_oversampl_enum = + SOC_ENUM_SINGLE(NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_SFT, + ARRAY_SIZE(nau8825_dac_oversampl), nau8825_dac_oversampl); + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); +static const DECLARE_TLV_DB_MINMAX_MUTE(crosstalk_vol_tlv, -9600, 2400); + +static const struct snd_kcontrol_new nau8825_controls[] = { + SOC_SINGLE_TLV("Mic Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 0, 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("Headphone Bypass Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("Headphone Volume", NAU8825_REG_HSVOL_CTRL, + 6, 0, 0x3f, 1, dac_vol_tlv), + SOC_SINGLE_TLV("Frontend PGA Volume", NAU8825_REG_POWER_UP_CONTROL, + 8, 37, 0, fepga_gain_tlv), + SOC_DOUBLE_TLV("Headphone Crosstalk Volume", NAU8825_REG_DAC_DGAIN_CTRL, + 0, 8, 0xff, 0, crosstalk_vol_tlv), + + SOC_ENUM("ADC Decimation Rate", nau8825_adc_decimation_enum), + SOC_ENUM("DAC Oversampling Rate", nau8825_dac_oversampl_enum), +}; + +/* DAC Mux 0x33[9] and 0x34[9] */ +static const char * const nau8825_dac_src[] = { + "DACL", "DACR", +}; + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); + +static const struct snd_kcontrol_new nau8825_dacl_mux = + SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); + +static const struct snd_kcontrol_new nau8825_dacr_mux = + SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); + + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2, + 15, 1), + + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), + + SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0), + + SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, + 0), + + /* ADC for button press detection */ + SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0), + + SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), + SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), + SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACL_SFT, 0), + SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacl_mux), + SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacr_mux), + + SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("Charge Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, + nau8825_pump_event, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_PGA("Output Driver R Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver L Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver R Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver L Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 1, + NAU8825_REG_POWER_UP_CONTROL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 1, + NAU8825_REG_POWER_UP_CONTROL, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA_S("Output DACL", 2, NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, 0), + SND_SOC_DAPM_PGA_S("Output DACR", 2, NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"Frontend PGA", NULL, "MIC"}, + {"ADC", NULL, "Frontend PGA"}, + {"ADC", NULL, "ADC Clock"}, + {"ADC", NULL, "ADC Power"}, + {"AIFTX", NULL, "ADC"}, + + {"DDACL", NULL, "Playback"}, + {"DDACR", NULL, "Playback"}, + {"DDACL", NULL, "DDAC Clock"}, + {"DDACR", NULL, "DDAC Clock"}, + {"DACL Mux", "DACL", "DDACL"}, + {"DACL Mux", "DACR", "DDACR"}, + {"DACR Mux", "DACL", "DDACL"}, + {"DACR Mux", "DACR", "DDACR"}, + {"HP amp L", NULL, "DACL Mux"}, + {"HP amp R", NULL, "DACR Mux"}, + {"HP amp L", NULL, "HP amp power"}, + {"HP amp R", NULL, "HP amp power"}, + {"ADACL", NULL, "HP amp L"}, + {"ADACR", NULL, "HP amp R"}, + {"ADACL", NULL, "ADACL Clock"}, + {"ADACR", NULL, "ADACR Clock"}, + {"Output Driver L Stage 1", NULL, "ADACL"}, + {"Output Driver R Stage 1", NULL, "ADACR"}, + {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, + {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, + {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, + {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, + {"Output DACL", NULL, "Output Driver L Stage 3"}, + {"Output DACR", NULL, "Output Driver R Stage 3"}, + {"HPOL", NULL, "Output DACL"}, + {"HPOR", NULL, "Output DACR"}, + {"HPOL", NULL, "Charge Pump"}, + {"HPOR", NULL, "Charge Pump"}, +}; + +static int nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + val_len |= NAU8825_I2S_DL_16; + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK, val_len); + + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8825_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8825_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8825_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + ctrl1_val |= NAU8825_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK | + NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + ctrl1_val); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, ctrl2_val); + + return 0; +} + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .hw_params = nau8825_hw_params, + .set_fmt = nau8825_set_dai_fmt, +}; + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8825_dai = { + .name = "nau8825-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, +}; + +/** + * nau8825_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8825->regmap; + + nau8825->jack = jack; + + /* Ground HP Outputs[1:0], needed for headset auto detection + * Enable Automatic Mic/Gnd switching reading on insert interrupt[6] + */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, + NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, + NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); + + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); + + +static bool nau8825_is_jack_inserted(struct regmap *regmap) +{ + int status; + + regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); + return !(status & NAU8825_GPIO2JD1); +} + +static void nau8825_restart_jack_detection(struct regmap *regmap) +{ + /* this will restart the entire jack detection process including MIC/GND + * switching and create interrupts. We have to go from 0 to 1 and back + * to 0 to restart. + */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, 0); +} + +static void nau8825_eject_jack(struct nau8825 *nau8825) +{ + struct snd_soc_dapm_context *dapm = nau8825->dapm; + struct regmap *regmap = nau8825->regmap; + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); + + snd_soc_dapm_sync(dapm); +} + +static int nau8825_button_decode(int value) +{ + int buttons = 0; + + /* The chip supports up to 8 buttons, but ALSA defines only 6 buttons */ + if (value & BIT(0)) + buttons |= SND_JACK_BTN_0; + if (value & BIT(1)) + buttons |= SND_JACK_BTN_1; + if (value & BIT(2)) + buttons |= SND_JACK_BTN_2; + if (value & BIT(3)) + buttons |= SND_JACK_BTN_3; + if (value & BIT(4)) + buttons |= SND_JACK_BTN_4; + if (value & BIT(5)) + buttons |= SND_JACK_BTN_5; + + return buttons; +} + +static int nau8825_jack_insert(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + struct snd_soc_dapm_context *dapm = nau8825->dapm; + int jack_status_reg, mic_detected; + int type = 0; + + regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); + mic_detected = (jack_status_reg >> 10) & 3; + + switch (mic_detected) { + case 0: + /* no mic */ + type = SND_JACK_HEADPHONE; + break; + case 1: + dev_dbg(nau8825->dev, "OMTP (micgnd1) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 1 << 2); + /* Attach 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKR2); + /* Attach SARADC to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKR2); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + case 2: + case 3: + dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 2 << 2); + /* Attach 2kOhm Resistor from MICBIAS to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKSLV); + /* Attach SARADC to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKSLV); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + } + + if (type & SND_JACK_HEADPHONE) { + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + } + + return type; +} + +#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +static irqreturn_t nau8825_interrupt(int irq, void *data) +{ + struct nau8825 *nau8825 = (struct nau8825 *)data; + struct regmap *regmap = nau8825->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + + if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == + NAU8825_JACK_EJECTION_DETECTED) { + + nau8825_eject_jack(nau8825); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK; + } else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) { + int key_status; + + regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, + &key_status); + + /* upper 8 bits of the register are for short pressed keys, + * lower 8 bits - for long pressed buttons + */ + nau8825->button_pressed = nau8825_button_decode( + key_status >> 8); + + event |= nau8825->button_pressed; + event_mask |= NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8825_KEY_RELEASE_IRQ) { + event_mask = NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_RELEASE_IRQ; + } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { + if (nau8825_is_jack_inserted(regmap)) { + event |= nau8825_jack_insert(nau8825); + } else { + dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); + nau8825_eject_jack(nau8825); + } + + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8825->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static void nau8825_setup_buttons(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_TRACKING_GAIN_MASK, + nau8825->sar_voltage << NAU8825_SAR_TRACKING_GAIN_SFT); + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_COMPARE_TIME_MASK, + nau8825->sar_compare_time << NAU8825_SAR_COMPARE_TIME_SFT); + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_SAMPLING_TIME_MASK, + nau8825->sar_sampling_time << NAU8825_SAR_SAMPLING_TIME_SFT); + + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_LEVELS_NR_MASK, + (nau8825->sar_threshold_num - 1) << NAU8825_KEYDET_LEVELS_NR_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_HYSTERESIS_MASK, + nau8825->sar_hysteresis << NAU8825_KEYDET_HYSTERESIS_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK, + nau8825->key_debounce << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT); + + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1, + (nau8825->sar_threshold[0] << 8) | nau8825->sar_threshold[1]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2, + (nau8825->sar_threshold[2] << 8) | nau8825->sar_threshold[3]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_3, + (nau8825->sar_threshold[4] << 8) | nau8825->sar_threshold[5]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_4, + (nau8825->sar_threshold[6] << 8) | nau8825->sar_threshold[7]); + + /* Enable short press and release interruptions */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_KEY_SHORT_PRESS_EN | NAU8825_IRQ_KEY_RELEASE_EN, + 0); +} + +static void nau8825_init_regs(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Enable Bias/Vmid */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_VMID, NAU8825_BIAS_VMID); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_GLOBAL_BIAS_EN, NAU8825_GLOBAL_BIAS_EN); + + /* VMID Tieoff */ + regmap_update_bits(regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_VMID_SEL_MASK, + nau8825->vref_impedance << NAU8825_BIAS_VMID_SEL_SFT); + /* Disable Boost Driver, Automatic Short circuit protection enable */ + regmap_update_bits(regmap, NAU8825_REG_BOOST, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | + NAU8825_SHORT_SHUTDOWN_EN, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | + NAU8825_SHORT_SHUTDOWN_EN); + + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_OUTPUT_EN, + nau8825->jkdet_enable ? 0 : NAU8825_JKDET_OUTPUT_EN); + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_EN, + nau8825->jkdet_pull_enable ? 0 : NAU8825_JKDET_PULL_EN); + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_UP, + nau8825->jkdet_pull_up ? NAU8825_JKDET_PULL_UP : 0); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_POLARITY, + /* jkdet_polarity - 1 is for active-low */ + nau8825->jkdet_polarity ? 0 : NAU8825_JACK_POLARITY); + + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_INSERT_DEBOUNCE_MASK, + nau8825->jack_insert_debounce << NAU8825_JACK_INSERT_DEBOUNCE_SFT); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_EJECT_DEBOUNCE_MASK, + nau8825->jack_eject_debounce << NAU8825_JACK_EJECT_DEBOUNCE_SFT); + + /* Mask unneeded IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, 0x7ff, 0x7ff); + + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_VOLTAGE_MASK, nau8825->micbias_voltage); + + if (nau8825->sar_threshold_num) + nau8825_setup_buttons(nau8825); + + /* Default oversampling/decimations settings are unusable + * (audible hiss). Set it to something better. + */ + regmap_update_bits(regmap, NAU8825_REG_ADC_RATE, + NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_128); + regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, + NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_128); +} + +static const struct regmap_config nau8825_regmap_config = { + .val_bits = 16, + .reg_bits = 16, + + .max_register = NAU8825_REG_MAX, + .readable_reg = nau8825_readable_reg, + .writeable_reg = nau8825_writeable_reg, + .volatile_reg = nau8825_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults), +}; + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + + nau8825->dapm = dapm; + + /* The interrupt clock is gated by x1[10:8], + * one of them needs to be enabled all the time for + * interrupts to happen. + */ + snd_soc_dapm_force_enable_pin(dapm, "DDACR"); + snd_soc_dapm_sync(dapm); + + /* Unmask interruptions. Handler uses dapm object so we can enable + * interruptions only after dapm is fully initialized. + */ + regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} + +/** + * nau8825_calc_fll_param - Calculate FLL parameters. + * @fll_in: external clock provided to codec. + * @fs: sampling rate. + * @fll_param: Pointer to structure of FLL parameters. + * + * Calculate FLL parameters to configure codec. + * + * Returns 0 for success or negative error code. + */ +static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, + struct nau8825_fll *fll_param) +{ + u64 fvco; + unsigned int fref, i; + + /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing + * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. + * FREF = freq_in / NAU8825_FLL_REF_DIV_MASK + */ + for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) { + fref = fll_in / fll_pre_scalar[i].param; + if (fref <= NAU_FREF_MAX) + break; + } + if (i == ARRAY_SIZE(fll_pre_scalar)) + return -EINVAL; + fll_param->clk_ref_div = fll_pre_scalar[i].val; + + /* Choose the FLL ratio based on FREF */ + for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) { + if (fref >= fll_ratio[i].param) + break; + } + if (i == ARRAY_SIZE(fll_ratio)) + return -EINVAL; + fll_param->ratio = fll_ratio[i].val; + + /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. + * FDCO must be within the 90MHz - 100MHz or the FFL cannot be + * guaranteed across the full range of operation. + * FDCO = freq_out * 2 * mclk_src_scaling + */ + for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { + fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + if (NAU_FVCO_MIN < fvco && fvco < NAU_FVCO_MAX) + break; + } + if (i == ARRAY_SIZE(mclk_src_scaling)) + return -EINVAL; + fll_param->mclk_src = mclk_src_scaling[i].val; + + /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional + * input based on FDCO, FREF and FLL ratio. + */ + fvco = div_u64(fvco << 16, fref * fll_param->ratio); + fll_param->fll_int = (fvco >> 16) & 0x3FF; + fll_param->fll_frac = fvco & 0xFFFF; + return 0; +} + +static void nau8825_fll_apply(struct nau8825 *nau8825, + struct nau8825_fll *fll_param) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, + NAU8825_FLL_RATIO_MASK, fll_param->ratio); + /* FLL 16-bit fractional input */ + regmap_write(nau8825->regmap, NAU8825_REG_FLL2, fll_param->fll_frac); + /* FLL 10-bit integer input */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL3, + NAU8825_FLL_INTEGER_MASK, fll_param->fll_int); + /* FLL pre-scaler */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL4, + NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); + /* select divided VCO input */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_FILTER_SW_MASK, 0x0000); + /* FLL sigma delta modulator enable */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); +} + +/* freq_out must be 256*Fs in order to achieve the best performance */ +static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct nau8825_fll fll_param; + int ret, fs; + + fs = freq_out / 256; + ret = nau8825_calc_fll_param(freq_in, fs, &fll_param); + if (ret < 0) { + dev_err(codec->dev, "Unsupported input clock %d\n", freq_in); + return ret; + } + dev_dbg(codec->dev, "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n", + fll_param.mclk_src, fll_param.ratio, fll_param.fll_frac, + fll_param.fll_int, fll_param.clk_ref_div); + + nau8825_fll_apply(nau8825, &fll_param); + mdelay(2); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + return 0; +} + +static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, + unsigned int freq) +{ + struct regmap *regmap = nau8825->regmap; + int ret; + + switch (clk_id) { + case NAU8825_CLK_MCLK: + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + + /* We selected MCLK source but the clock itself managed externally */ + if (!nau8825->mclk) + break; + + if (!nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + return ret; + } + } + + if (nau8825->mclk_freq != freq) { + nau8825->mclk_freq = freq; + + freq = clk_round_rate(nau8825->mclk, freq); + ret = clk_set_rate(nau8825->mclk, freq); + if (ret) { + dev_err(nau8825->dev, "Unable to set mclk rate\n"); + return ret; + } + } + + break; + case NAU8825_CLK_INTERNAL: + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, + NAU8825_DCO_EN); + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; + default: + dev_err(nau8825->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + dev_dbg(nau8825->dev, "Sysclk is %dHz and clock id is %d\n", freq, + clk_id); + return 0; +} + +static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + return nau8825_configure_sysclk(nau8825, clk_id, freq); +} + +static int nau8825_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + if (nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + return ret; + } + } + + ret = regcache_sync(nau8825->regmap); + if (ret) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + + break; + + case SND_SOC_BIAS_OFF: + if (nau8825->mclk_freq) + clk_disable_unprepare(nau8825->mclk); + + regcache_mark_dirty(nau8825->regmap); + break; + } + return 0; +} + +static struct snd_soc_codec_driver nau8825_codec_driver = { + .probe = nau8825_codec_probe, + .set_sysclk = nau8825_set_sysclk, + .set_pll = nau8825_set_pll, + .set_bias_level = nau8825_set_bias_level, + .suspend_bias_off = true, + + .controls = nau8825_controls, + .num_controls = ARRAY_SIZE(nau8825_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static void nau8825_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8825_REG_RESET, 0x00); + regmap_write(regmap, NAU8825_REG_RESET, 0x00); +} + +static void nau8825_print_device_properties(struct nau8825 *nau8825) +{ + int i; + struct device *dev = nau8825->dev; + + dev_dbg(dev, "jkdet-enable: %d\n", nau8825->jkdet_enable); + dev_dbg(dev, "jkdet-pull-enable: %d\n", nau8825->jkdet_pull_enable); + dev_dbg(dev, "jkdet-pull-up: %d\n", nau8825->jkdet_pull_up); + dev_dbg(dev, "jkdet-polarity: %d\n", nau8825->jkdet_polarity); + dev_dbg(dev, "micbias-voltage: %d\n", nau8825->micbias_voltage); + dev_dbg(dev, "vref-impedance: %d\n", nau8825->vref_impedance); + + dev_dbg(dev, "sar-threshold-num: %d\n", nau8825->sar_threshold_num); + for (i = 0; i < nau8825->sar_threshold_num; i++) + dev_dbg(dev, "sar-threshold[%d]=%d\n", i, + nau8825->sar_threshold[i]); + + dev_dbg(dev, "sar-hysteresis: %d\n", nau8825->sar_hysteresis); + dev_dbg(dev, "sar-voltage: %d\n", nau8825->sar_voltage); + dev_dbg(dev, "sar-compare-time: %d\n", nau8825->sar_compare_time); + dev_dbg(dev, "sar-sampling-time: %d\n", nau8825->sar_sampling_time); + dev_dbg(dev, "short-key-debounce: %d\n", nau8825->key_debounce); + dev_dbg(dev, "jack-insert-debounce: %d\n", + nau8825->jack_insert_debounce); + dev_dbg(dev, "jack-eject-debounce: %d\n", + nau8825->jack_eject_debounce); +} + +static int nau8825_read_device_properties(struct device *dev, + struct nau8825 *nau8825) { + + nau8825->jkdet_enable = device_property_read_bool(dev, + "nuvoton,jkdet-enable"); + nau8825->jkdet_pull_enable = device_property_read_bool(dev, + "nuvoton,jkdet-pull-enable"); + nau8825->jkdet_pull_up = device_property_read_bool(dev, + "nuvoton,jkdet-pull-up"); + device_property_read_u32(dev, "nuvoton,jkdet-polarity", + &nau8825->jkdet_polarity); + device_property_read_u32(dev, "nuvoton,micbias-voltage", + &nau8825->micbias_voltage); + device_property_read_u32(dev, "nuvoton,vref-impedance", + &nau8825->vref_impedance); + device_property_read_u32(dev, "nuvoton,sar-threshold-num", + &nau8825->sar_threshold_num); + device_property_read_u32_array(dev, "nuvoton,sar-threshold", + nau8825->sar_threshold, nau8825->sar_threshold_num); + device_property_read_u32(dev, "nuvoton,sar-hysteresis", + &nau8825->sar_hysteresis); + device_property_read_u32(dev, "nuvoton,sar-voltage", + &nau8825->sar_voltage); + device_property_read_u32(dev, "nuvoton,sar-compare-time", + &nau8825->sar_compare_time); + device_property_read_u32(dev, "nuvoton,sar-sampling-time", + &nau8825->sar_sampling_time); + device_property_read_u32(dev, "nuvoton,short-key-debounce", + &nau8825->key_debounce); + device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + &nau8825->jack_insert_debounce); + device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + &nau8825->jack_eject_debounce); + + nau8825->mclk = devm_clk_get(dev, "mclk"); + if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { + return -EPROBE_DEFER; + } else if (PTR_ERR(nau8825->mclk) == -ENOENT) { + /* The MCLK is managed externally or not used at all */ + nau8825->mclk = NULL; + dev_info(dev, "No 'mclk' clock found, assume MCLK is managed externally"); + } else if (IS_ERR(nau8825->mclk)) { + return -EINVAL; + } + + return 0; +} + +static int nau8825_setup_irq(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + int ret; + + /* IRQ Output Enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable DDACR needed for interrupts + * It is the same as force_enable_pin("DDACR") we do later + */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, + nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8825", nau8825); + + if (ret) { + dev_err(nau8825->dev, "Cannot request irq %d (%d)\n", + nau8825->irq, ret); + return ret; + } + + return 0; +} + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8825 *nau8825 = dev_get_platdata(&i2c->dev); + int ret, value; + + if (!nau8825) { + nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL); + if (!nau8825) + return -ENOMEM; + ret = nau8825_read_device_properties(dev, nau8825); + if (ret) + return ret; + } + + i2c_set_clientdata(i2c, nau8825); + + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config); + if (IS_ERR(nau8825->regmap)) + return PTR_ERR(nau8825->regmap); + nau8825->dev = dev; + nau8825->irq = i2c->irq; + + nau8825_print_device_properties(nau8825); + + nau8825_reset_chip(nau8825->regmap); + ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU8825: %d\n", + ret); + return ret; + } + if ((value & NAU8825_SOFTWARE_ID_MASK) != + NAU8825_SOFTWARE_ID_NAU8825) { + dev_err(dev, "Not a NAU8825 chip\n"); + return -ENODEV; + } + + nau8825_init_regs(nau8825); + + if (i2c->irq) + nau8825_setup_irq(nau8825); + + return snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver, + &nau8825_dai, 1); +} + +static int nau8825_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_ids[] = { + { "nau8825", 0 }, + { } +}; + +#ifdef CONFIG_OF +static const struct of_device_id nau8825_of_ids[] = { + { .compatible = "nuvoton,nau8825", }, + {} +}; +MODULE_DEVICE_TABLE(of, nau8825_of_ids); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id nau8825_acpi_match[] = { + { "10508825", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8825_acpi_match); +#endif + +static struct i2c_driver nau8825_driver = { + .driver = { + .name = "nau8825", + .of_match_table = of_match_ptr(nau8825_of_ids), + .acpi_match_table = ACPI_PTR(nau8825_acpi_match), + }, + .probe = nau8825_i2c_probe, + .remove = nau8825_i2c_remove, + .id_table = nau8825_i2c_ids, +}; +module_i2c_driver(nau8825_driver); + +MODULE_DESCRIPTION("ASoC nau8825 driver"); +MODULE_AUTHOR("Anatol Pomozov <anatol@chromium.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 000000000000..dff8edb83bfd --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,341 @@ +/* + * NAU8825 ALSA SoC audio driver + * + * Copyright 2015 Google Inc. + * Author: Anatol Pomozov <anatol.pomozov@chrominium.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8825_H__ +#define __NAU8825_H__ + +#define NAU8825_REG_RESET 0x00 +#define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_CLK_DIVIDER 0x03 +#define NAU8825_REG_FLL1 0x04 +#define NAU8825_REG_FLL2 0x05 +#define NAU8825_REG_FLL3 0x06 +#define NAU8825_REG_FLL4 0x07 +#define NAU8825_REG_FLL5 0x08 +#define NAU8825_REG_FLL6 0x09 +#define NAU8825_REG_FLL_VCO_RSV 0x0a +#define NAU8825_REG_HSD_CTRL 0x0c +#define NAU8825_REG_JACK_DET_CTRL 0x0d +#define NAU8825_REG_INTERRUPT_MASK 0x0f +#define NAU8825_REG_IRQ_STATUS 0x10 +#define NAU8825_REG_INT_CLR_KEY_STATUS 0x11 +#define NAU8825_REG_INTERRUPT_DIS_CTRL 0x12 +#define NAU8825_REG_SAR_CTRL 0x13 +#define NAU8825_REG_KEYDET_CTRL 0x14 +#define NAU8825_REG_VDET_THRESHOLD_1 0x15 +#define NAU8825_REG_VDET_THRESHOLD_2 0x16 +#define NAU8825_REG_VDET_THRESHOLD_3 0x17 +#define NAU8825_REG_VDET_THRESHOLD_4 0x18 +#define NAU8825_REG_GPIO34_CTRL 0x19 +#define NAU8825_REG_GPIO12_CTRL 0x1a +#define NAU8825_REG_TDM_CTRL 0x1b +#define NAU8825_REG_I2S_PCM_CTRL1 0x1c +#define NAU8825_REG_I2S_PCM_CTRL2 0x1d +#define NAU8825_REG_LEFT_TIME_SLOT 0x1e +#define NAU8825_REG_RIGHT_TIME_SLOT 0x1f +#define NAU8825_REG_BIQ_CTRL 0x20 +#define NAU8825_REG_BIQ_COF1 0x21 +#define NAU8825_REG_BIQ_COF2 0x22 +#define NAU8825_REG_BIQ_COF3 0x23 +#define NAU8825_REG_BIQ_COF4 0x24 +#define NAU8825_REG_BIQ_COF5 0x25 +#define NAU8825_REG_BIQ_COF6 0x26 +#define NAU8825_REG_BIQ_COF7 0x27 +#define NAU8825_REG_BIQ_COF8 0x28 +#define NAU8825_REG_BIQ_COF9 0x29 +#define NAU8825_REG_BIQ_COF10 0x2a +#define NAU8825_REG_ADC_RATE 0x2b +#define NAU8825_REG_DAC_CTRL1 0x2c +#define NAU8825_REG_DAC_CTRL2 0x2d +#define NAU8825_REG_DAC_DGAIN_CTRL 0x2f +#define NAU8825_REG_ADC_DGAIN_CTRL 0x30 +#define NAU8825_REG_MUTE_CTRL 0x31 +#define NAU8825_REG_HSVOL_CTRL 0x32 +#define NAU8825_REG_DACL_CTRL 0x33 +#define NAU8825_REG_DACR_CTRL 0x34 +#define NAU8825_REG_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_REG_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_REG_ADC_DRC_SLOPES 0x3a +#define NAU8825_REG_ADC_DRC_ATKDCY 0x3b +#define NAU8825_REG_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_REG_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_REG_DAC_DRC_SLOPES 0x47 +#define NAU8825_REG_DAC_DRC_ATKDCY 0x48 +#define NAU8825_REG_IMM_MODE_CTRL 0x4c +#define NAU8825_REG_IMM_RMS_L 0x4d +#define NAU8825_REG_IMM_RMS_R 0x4e +#define NAU8825_REG_CLASSG_CTRL 0x50 +#define NAU8825_REG_OPT_EFUSE_CTRL 0x51 +#define NAU8825_REG_MISC_CTRL 0x55 +#define NAU8825_REG_I2C_DEVICE_ID 0x58 +#define NAU8825_REG_SARDOUT_RAM_STATUS 0x59 +#define NAU8825_REG_BIAS_ADJ 0x66 +#define NAU8825_REG_TRIM_SETTINGS 0x68 +#define NAU8825_REG_ANALOG_CONTROL_1 0x69 +#define NAU8825_REG_ANALOG_CONTROL_2 0x6a +#define NAU8825_REG_ANALOG_ADC_1 0x71 +#define NAU8825_REG_ANALOG_ADC_2 0x72 +#define NAU8825_REG_RDAC 0x73 +#define NAU8825_REG_MIC_BIAS 0x74 +#define NAU8825_REG_BOOST 0x76 +#define NAU8825_REG_FEPGA 0x77 +#define NAU8825_REG_POWER_UP_CONTROL 0x7f +#define NAU8825_REG_CHARGE_PUMP 0x80 +#define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8825_REG_GENERAL_STATUS 0x82 +#define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS + +/* ENA_CTRL (0x1) */ +#define NAU8825_ENABLE_DACR_SFT 10 +#define NAU8825_ENABLE_DACR (1 << NAU8825_ENABLE_DACR_SFT) +#define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_SAR_SFT 1 + +/* CLK_DIVIDER (0x3) */ +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_MCLK_SRC_MASK (0xf << 0) + +/* FLL1 (0x04) */ +#define NAU8825_FLL_RATIO_MASK (0x7f << 0) + +/* FLL3 (0x06) */ +#define NAU8825_FLL_INTEGER_MASK (0x3ff << 0) + +/* FLL4 (0x07) */ +#define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) + +/* FLL5 (0x08) */ +#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) + +/* FLL6 (0x9) */ +#define NAU8825_DCO_EN_MASK (0x1 << 15) +#define NAU8825_DCO_EN (0x1 << 15) +#define NAU8825_DCO_DIS (0x0 << 15) +#define NAU8825_SDM_EN_MASK (0x1 << 14) +#define NAU8825_SDM_EN (0x1 << 14) +#define NAU8825_SDM_DIS (0x0 << 14) + +/* HSD_CTRL (0xc) */ +#define NAU8825_HSD_AUTO_MODE (1 << 6) +/* 0 - short to GND, 1 - open */ +#define NAU8825_SPKR_DWN1R (1 << 1) +#define NAU8825_SPKR_DWN1L (1 << 0) + +/* JACK_DET_CTRL (0xd) */ +#define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 +#define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) +#define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 +#define NAU8825_JACK_EJECT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_EJECT_DEBOUNCE_SFT) +#define NAU8825_JACK_POLARITY (1 << 1) /* 0 - active low, 1 - active high */ + +/* INTERRUPT_MASK (0xf) */ +#define NAU8825_IRQ_OUTPUT_EN (1 << 11) +#define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10) +#define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) +#define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) +#define NAU8825_IRQ_EJECT_EN (1 << 2) + +/* IRQ_STATUS (0x10) */ +#define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) +#define NAU8825_SHORT_CIRCUIT_IRQ (1 << 9) +#define NAU8825_IMPEDANCE_MEAS_IRQ (1 << 8) +#define NAU8825_KEY_IRQ_MASK (0x7 << 5) +#define NAU8825_KEY_RELEASE_IRQ (1 << 7) +#define NAU8825_KEY_LONG_PRESS_IRQ (1 << 6) +#define NAU8825_KEY_SHORT_PRESS_IRQ (1 << 5) +#define NAU8825_MIC_DETECTION_IRQ (1 << 4) +#define NAU8825_JACK_EJECTION_IRQ_MASK (3 << 2) +#define NAU8825_JACK_EJECTION_DETECTED (1 << 2) +#define NAU8825_JACK_INSERTION_IRQ_MASK (3 << 0) +#define NAU8825_JACK_INSERTION_DETECTED (1 << 0) + +/* INTERRUPT_DIS_CTRL (0x12) */ +#define NAU8825_IRQ_HEADSET_COMPLETE_DIS (1 << 10) +#define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7) +#define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5) +#define NAU8825_IRQ_EJECT_DIS (1 << 2) + +/* SAR_CTRL (0x13) */ +#define NAU8825_SAR_ADC_EN_SFT 12 +#define NAU8825_SAR_ADC_EN (1 << NAU8825_SAR_ADC_EN_SFT) +#define NAU8825_SAR_INPUT_MASK (1 << 11) +#define NAU8825_SAR_INPUT_JKSLV (1 << 11) +#define NAU8825_SAR_INPUT_JKR2 (0 << 11) +#define NAU8825_SAR_TRACKING_GAIN_SFT 8 +#define NAU8825_SAR_TRACKING_GAIN_MASK (0x7 << NAU8825_SAR_TRACKING_GAIN_SFT) +#define NAU8825_SAR_COMPARE_TIME_SFT 2 +#define NAU8825_SAR_COMPARE_TIME_MASK (3 << 2) +#define NAU8825_SAR_SAMPLING_TIME_SFT 0 +#define NAU8825_SAR_SAMPLING_TIME_MASK (3 << 0) + +/* KEYDET_CTRL (0x14) */ +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT 12 +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK (0x3 << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT) +#define NAU8825_KEYDET_LEVELS_NR_SFT 8 +#define NAU8825_KEYDET_LEVELS_NR_MASK (0x7 << 8) +#define NAU8825_KEYDET_HYSTERESIS_SFT 0 +#define NAU8825_KEYDET_HYSTERESIS_MASK 0xf + +/* GPIO12_CTRL (0x1a) */ +#define NAU8825_JKDET_PULL_UP (1 << 11) /* 0 - pull down, 1 - pull up */ +#define NAU8825_JKDET_PULL_EN (1 << 9) /* 0 - enable pull, 1 - disable */ +#define NAU8825_JKDET_OUTPUT_EN (1 << 8) /* 0 - enable input, 1 - enable output */ + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8825_I2S_BP_SFT 7 +#define NAU8825_I2S_BP_MASK (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_BP_INV (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_PCMB_SFT 6 +#define NAU8825_I2S_PCMB_MASK (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_PCMB_EN (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_DL_SFT 2 +#define NAU8825_I2S_DL_MASK (0x3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_16 (0 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_20 (1 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_24 (2 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_32 (3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DF_SFT 0 +#define NAU8825_I2S_DF_MASK (0x3 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_RIGTH (0 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_LEFT (1 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_I2S (2 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_PCM_AB (3 << NAU8825_I2S_DF_SFT) + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - normal mode, 1 - Hi-Z output */ +#define NAU8825_I2S_MS_SFT 3 +#define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SYNC_DOWN_SFT 0 +#define NAU8825_ADC_SYNC_DOWN_MASK 0x3 +#define NAU8825_ADC_SYNC_DOWN_32 0 +#define NAU8825_ADC_SYNC_DOWN_64 1 +#define NAU8825_ADC_SYNC_DOWN_128 2 +#define NAU8825_ADC_SYNC_DOWN_256 3 + +/* DAC_CTRL1 (0x2c) */ +#define NAU8825_DAC_CLIP_OFF (1 << 7) +#define NAU8825_DAC_OVERSAMPLE_SFT 0 +#define NAU8825_DAC_OVERSAMPLE_MASK 0x7 +#define NAU8825_DAC_OVERSAMPLE_64 0 +#define NAU8825_DAC_OVERSAMPLE_256 1 +#define NAU8825_DAC_OVERSAMPLE_128 2 +#define NAU8825_DAC_OVERSAMPLE_32 4 + +/* MUTE_CTRL (0x31) */ +#define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) +#define NAU8825_DAC_SOFT_MUTE (1 << 9) + +/* HSVOL_CTRL (0x32) */ +#define NAU8825_HP_MUTE (1 << 15) + +/* DACL_CTRL (0x33) */ +#define NAU8825_DACL_CH_SEL_SFT 9 + +/* DACR_CTRL (0x34) */ +#define NAU8825_DACR_CH_SEL_SFT 9 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (1 << 7) +#define NAU8825_SOFTWARE_ID_MASK 0x3 +#define NAU8825_SOFTWARE_ID_NAU8825 0x0 + +/* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_VMID (1 << 6) +#define NAU8825_BIAS_VMID_SEL_SFT 4 +#define NAU8825_BIAS_VMID_SEL_MASK (3 << NAU8825_BIAS_VMID_SEL_SFT) + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12) +#define NAU8825_DAC_CAPACITOR_MSB (1 << 1) +#define NAU8825_DAC_CAPACITOR_LSB (1 << 0) + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8825_ADC_VREFSEL_MASK (0x3 << 8) +#define NAU8825_ADC_VREFSEL_ANALOG (0 << 8) +#define NAU8825_ADC_VREFSEL_VMID (1 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB (2 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) +#define NAU8825_POWERUP_ADCL (1 << 6) + +/* MIC_BIAS (0x74) */ +#define NAU8825_MICBIAS_JKSLV (1 << 14) +#define NAU8825_MICBIAS_JKR2 (1 << 12) +#define NAU8825_MICBIAS_POWERUP_SFT 8 +#define NAU8825_MICBIAS_VOLTAGE_SFT 0 +#define NAU8825_MICBIAS_VOLTAGE_MASK 0x7 + +/* BOOST (0x76) */ +#define NAU8825_PRECHARGE_DIS (1 << 13) +#define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_G_DIS (1 << 8) +#define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8825_POWERUP_INTEGR_R (1 << 5) +#define NAU8825_POWERUP_INTEGR_L (1 << 4) +#define NAU8825_POWERUP_DRV_IN_R (1 << 3) +#define NAU8825_POWERUP_DRV_IN_L (1 << 2) +#define NAU8825_POWERUP_HP_DRV_R (1 << 1) +#define NAU8825_POWERUP_HP_DRV_L (1 << 0) + +/* CHARGE_PUMP (0x80) */ +#define NAU8825_JAMNODCLOW (1 << 10) +#define NAU8825_POWER_DOWN_DACR (1 << 9) +#define NAU8825_POWER_DOWN_DACL (1 << 8) +#define NAU8825_CHANRGE_PUMP_EN (1 << 5) + + +/* System Clock Source */ +enum { + NAU8825_CLK_MCLK = 0, + NAU8825_CLK_INTERNAL, +}; + +struct nau8825 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + struct snd_soc_jack *jack; + struct clk *mclk; + int irq; + int mclk_freq; /* 0 - mclk is disabled */ + int button_pressed; + int micbias_voltage; + int vref_impedance; + bool jkdet_enable; + bool jkdet_pull_enable; + bool jkdet_pull_up; + int jkdet_polarity; + int sar_threshold_num; + int sar_threshold[8]; + int sar_hysteresis; + int sar_voltage; + int sar_compare_time; + int sar_sampling_time; + int key_debounce; + int jack_insert_debounce; + int jack_eject_debounce; +}; + +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + + +#endif /* __NAU8825_H__ */ diff --git a/sound/soc/codecs/rl6347a.c b/sound/soc/codecs/rl6347a.c index 91d5166bd3a1..a4b910efbd45 100644 --- a/sound/soc/codecs/rl6347a.c +++ b/sound/soc/codecs/rl6347a.c @@ -11,25 +11,8 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> -#include <linux/delay.h> -#include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> -#include <linux/spi/spi.h> -#include <linux/dmi.h> -#include <linux/acpi.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/initval.h> -#include <sound/tlv.h> -#include <sound/jack.h> -#include <linux/workqueue.h> -#include <sound/hda_verbs.h> +#include <linux/regmap.h> #include "rl6347a.h" diff --git a/sound/soc/codecs/rl6347a.h b/sound/soc/codecs/rl6347a.h index 1cb56e50b7f3..e127919cb36b 100644 --- a/sound/soc/codecs/rl6347a.h +++ b/sound/soc/codecs/rl6347a.h @@ -12,6 +12,8 @@ #ifndef __RL6347A_H__ #define __RL6347A_H__ +#include <sound/hda_verbs.h> + #define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) #define RL6347A_VENDOR_REGISTERS 0x20 diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index bd9365885f73..af2ed774b552 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -29,7 +29,6 @@ #include <sound/jack.h> #include <linux/workqueue.h> #include <sound/rt286.h> -#include <sound/hda_verbs.h> #include "rl6347a.h" #include "rt286.h" @@ -38,7 +37,7 @@ #define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { - const struct reg_default *index_cache; + struct reg_default *index_cache; int index_cache_size; struct regmap *regmap; struct snd_soc_codec *codec; @@ -1161,7 +1160,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, return -ENODEV; } - rt286->index_cache = rt286_index_def; + rt286->index_cache = devm_kmemdup(&i2c->dev, rt286_index_def, + sizeof(rt286_index_def), GFP_KERNEL); + if (!rt286->index_cache) + return -ENOMEM; + rt286->index_cache_size = INDEX_CACHE_SIZE; rt286->i2c = i2c; i2c_set_clientdata(i2c, rt286); diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index f823eb502367..b3f795c60749 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -28,7 +28,6 @@ #include <sound/jack.h> #include <linux/workqueue.h> #include <sound/rt298.h> -#include <sound/hda_verbs.h> #include "rl6347a.h" #include "rt298.h" @@ -49,7 +48,7 @@ struct rt298_priv { int is_hp_in; }; -static struct reg_default rt298_index_def[] = { +static const struct reg_default rt298_index_def[] = { { 0x01, 0xa5a8 }, { 0x02, 0x8e95 }, { 0x03, 0x0002 }, @@ -129,7 +128,7 @@ static bool rt298_volatile_register(struct device *dev, unsigned int reg) case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_HP_OUT, 0): return true; default: - return true; + return false; } @@ -1165,7 +1164,11 @@ static int rt298_i2c_probe(struct i2c_client *i2c, return -ENODEV; } - rt298->index_cache = rt298_index_def; + rt298->index_cache = devm_kmemdup(&i2c->dev, rt298_index_def, + sizeof(rt298_index_def), GFP_KERNEL); + if (!rt298->index_cache) + return -ENOMEM; + rt298->index_cache_size = INDEX_CACHE_SIZE; rt298->i2c = i2c; i2c_set_clientdata(i2c, rt298); diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e1ceeb885f7d..f2beb1aa5763 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -405,11 +405,14 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), - /* IN1/IN2 Control */ + /* IN1/IN2/IN3 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, RT5640_BST_SFT1, 8, 0, bst_tlv), SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, RT5640_BST_SFT2, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN3 Boost", RT5640_IN1_IN2, + RT5640_BST_SFT2, 8, 0, bst_tlv), + /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5640_INL_INR_VOL, RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, @@ -598,6 +601,8 @@ static const struct snd_kcontrol_new rt5640_rec_l_mix[] = { RT5640_M_HP_L_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("INL Switch", RT5640_REC_L2_MIXER, RT5640_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_L2_MIXER, + RT5640_M_BST2_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_L2_MIXER, RT5640_M_BST4_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_L2_MIXER, @@ -611,6 +616,8 @@ static const struct snd_kcontrol_new rt5640_rec_r_mix[] = { RT5640_M_HP_R_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("INR Switch", RT5640_REC_R2_MIXER, RT5640_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_R2_MIXER, + RT5640_M_BST2_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_R2_MIXER, RT5640_M_BST4_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_R2_MIXER, @@ -1065,6 +1072,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1N"), SND_SOC_DAPM_INPUT("IN2P"), SND_SOC_DAPM_INPUT("IN2N"), + SND_SOC_DAPM_INPUT("IN3P"), + SND_SOC_DAPM_INPUT("IN3N"), SND_SOC_DAPM_PGA("DMIC L1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC R1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC L2", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1081,6 +1090,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { RT5640_PWR_BST1_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("BST2", RT5640_PWR_ANLG2, RT5640_PWR_BST4_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST3", RT5640_PWR_ANLG2, + RT5640_PWR_BST2_BIT, 0, NULL, 0), /* Input Volume */ SND_SOC_DAPM_PGA("INL VOL", RT5640_PWR_VOL, RT5640_PWR_IN_L_BIT, 0, NULL, 0), @@ -1310,6 +1321,7 @@ static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, + {"IN3P", NULL, "LDO2"}, {"DMIC L1", NULL, "DMIC1"}, {"DMIC R1", NULL, "DMIC1"}, @@ -1320,18 +1332,22 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"BST1", NULL, "IN1N"}, {"BST2", NULL, "IN2P"}, {"BST2", NULL, "IN2N"}, + {"BST3", NULL, "IN3P"}, + {"BST3", NULL, "IN3N"}, {"INL VOL", NULL, "IN2P"}, {"INR VOL", NULL, "IN2N"}, {"RECMIXL", "HPOL Switch", "HPOL"}, {"RECMIXL", "INL Switch", "INL VOL"}, + {"RECMIXL", "BST3 Switch", "BST3"}, {"RECMIXL", "BST2 Switch", "BST2"}, {"RECMIXL", "BST1 Switch", "BST1"}, {"RECMIXL", "OUT MIXL Switch", "OUT MIXL"}, {"RECMIXR", "HPOR Switch", "HPOR"}, {"RECMIXR", "INR Switch", "INR VOL"}, + {"RECMIXR", "BST3 Switch", "BST3"}, {"RECMIXR", "BST2 Switch", "BST2"}, {"RECMIXR", "BST1 Switch", "BST1"}, {"RECMIXR", "OUT MIXR Switch", "OUT MIXR"}, @@ -2260,6 +2276,10 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); + if (rt5640->pdata.in3_diff) + regmap_update_bits(rt5640->regmap, RT5640_IN1_IN2, + RT5640_IN_DF2, RT5640_IN_DF2); + rt5640->hp_mute = 1; return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 5c101af0ac63..28132375e427 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -42,6 +42,8 @@ #define RT5645_PR_BASE (RT5645_PR_RANGE_BASE + (0 * RT5645_PR_SPACING)) +#define RT5645_HWEQ_NUM 57 + static const struct regmap_range_cfg rt5645_ranges[] = { { .name = "PR", @@ -224,6 +226,11 @@ static const struct reg_default rt5645_reg[] = { { 0xff, 0x6308 }, }; +struct rt5645_eq_param_s { + unsigned short reg; + unsigned short val; +}; + static const char *const rt5645_supply_names[] = { "avdd", "cpvdd", @@ -240,6 +247,7 @@ struct rt5645_priv { struct snd_soc_jack *btn_jack; struct delayed_work jack_detect_work; struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; + struct rt5645_eq_param_s *eq_param; int codec_type; int sysclk; @@ -469,6 +477,94 @@ static const DECLARE_TLV_DB_RANGE(bst_tlv, 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0) ); +/* {-6, -4.5, -3, -1.5, 0, 0.82, 1.58, 2.28} dB */ +static const DECLARE_TLV_DB_RANGE(spk_clsd_tlv, + 0, 4, TLV_DB_SCALE_ITEM(-600, 150, 0), + 5, 5, TLV_DB_SCALE_ITEM(82, 0, 0), + 6, 6, TLV_DB_SCALE_ITEM(158, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(228, 0, 0) +); + +static int rt5645_hweq_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s); + + return 0; +} + +static int rt5645_hweq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + struct rt5645_eq_param_s *eq_param = + (struct rt5645_eq_param_s *)ucontrol->value.bytes.data; + int i; + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + eq_param[i].reg = cpu_to_be16(rt5645->eq_param[i].reg); + eq_param[i].val = cpu_to_be16(rt5645->eq_param[i].val); + } + + return 0; +} + +static bool rt5645_validate_hweq(unsigned short reg) +{ + if ((reg >= 0x1a4 && reg <= 0x1cd) | (reg >= 0x1e5 && reg <= 0x1f8) | + (reg == RT5645_EQ_CTRL2)) + return true; + + return false; +} + +static int rt5645_hweq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + struct rt5645_eq_param_s *eq_param = + (struct rt5645_eq_param_s *)ucontrol->value.bytes.data; + int i; + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + eq_param[i].reg = be16_to_cpu(eq_param[i].reg); + eq_param[i].val = be16_to_cpu(eq_param[i].val); + } + + /* The final setting of the table should be RT5645_EQ_CTRL2 */ + for (i = RT5645_HWEQ_NUM - 1; i >= 0; i--) { + if (eq_param[i].reg == 0) + continue; + else if (eq_param[i].reg != RT5645_EQ_CTRL2) + return 0; + else + break; + } + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + if (!rt5645_validate_hweq(eq_param[i].reg) && + eq_param[i].reg != 0) + return 0; + else if (eq_param[i].reg == 0) + break; + } + + memcpy(rt5645->eq_param, eq_param, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s)); + + return 0; +} + +#define RT5645_HWEQ(xname) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = rt5645_hweq_info, \ + .get = rt5645_hweq_get, \ + .put = rt5645_hweq_put \ +} + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -476,6 +572,10 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { SOC_DOUBLE_TLV("Speaker Playback Volume", RT5645_SPK_VOL, RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), + /* ClassD modulator Speaker Gain Ratio */ + SOC_SINGLE_TLV("Speaker ClassD Playback Volume", RT5645_SPO_CLSD_RATIO, + RT5645_SPK_G_CLSD_SFT, 7, 0, spk_clsd_tlv), + /* Headphone Output Volume */ SOC_DOUBLE("Headphone Channel Switch", RT5645_HP_VOL, RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1), @@ -529,6 +629,7 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* I2S2 function select */ SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT, 1, 1), + RT5645_HWEQ("Speaker HWEQ"), }; /** @@ -619,6 +720,22 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int rt5645_enable_hweq(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + if (rt5645_validate_hweq(rt5645->eq_param[i].reg)) + regmap_write(rt5645->regmap, rt5645->eq_param[i].reg, + rt5645->eq_param[i].val); + else + break; + } + + return 0; +} + /** * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters * @codec: SoC audio codec device. @@ -1523,6 +1640,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: + rt5645_enable_hweq(codec); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, @@ -1531,6 +1649,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT5645_EQ_CTRL2, 0); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, 0); @@ -2733,6 +2852,10 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_FV1 | RT5645_PWR_FV2, RT5645_PWR_FV1 | RT5645_PWR_FV2); + if (rt5645->en_button_func && + snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) + queue_delayed_work(system_power_efficient_wq, + &rt5645->jack_detect_work, msecs_to_jiffies(0)); break; case SND_SOC_BIAS_OFF: @@ -2829,6 +2952,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; } + if (rt5645->pdata.jd_invert) + regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, + RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); } else { /* jack out */ rt5645->jack_type = 0; @@ -2847,6 +2973,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_disable_pin(dapm, "LDO2"); snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); + if (rt5645->pdata.jd_invert) + regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, + RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR); } return rt5645->jack_type; @@ -3038,6 +3167,9 @@ static int rt5645_probe(struct snd_soc_codec *codec) snd_soc_dapm_sync(dapm); } + rt5645->eq_param = devm_kzalloc(codec->dev, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s), GFP_KERNEL); + return 0; } @@ -3098,7 +3230,7 @@ static struct snd_soc_dai_driver rt5645_dai[] = { .capture = { .stream_name = "AIF1 Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 4, .rates = RT5645_STEREO_RATES, .formats = RT5645_FORMATS, }, @@ -3209,9 +3341,42 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"), }, }, + { + .ident = "Google Reks", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Reks"), + }, + }, { } }; +static struct rt5645_platform_data buddy_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, + .jd_invert = true, +}; + +static int buddy_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &buddy_platform_data; + + return 1; +} + +static struct dmi_system_id dmi_platform_intel_broadwell[] = { + { + .ident = "Chrome Buddy", + .callback = buddy_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"), + }, + }, + { } +}; + + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3244,7 +3409,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else if (dmi_check_system(dmi_platform_intel_braswell)) + else if (dmi_check_system(dmi_platform_intel_braswell) || + dmi_check_system(dmi_platform_intel_broadwell)) rt5645->pdata = *rt5645_pdata; else rt5645_parse_dt(rt5645, &i2c->dev); @@ -3472,6 +3638,8 @@ static void rt5645_i2c_shutdown(struct i2c_client *i2c) RT5645_CBJ_MN_JD); regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN, 0); + msleep(20); + regmap_write(rt5645->regmap, RT5645_RESET, 0); } static struct i2c_driver rt5645_i2c_driver = { diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 8c964cfb120d..093e46d559fb 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -621,14 +621,14 @@ #define RT5645_G_OM_L_SM_L_SFT 6 #define RT5645_M_BST1_L_SM_L (0x1 << 5) #define RT5645_M_BST1_L_SM_L_SFT 5 +#define RT5645_M_BST3_L_SM_L (0x1 << 4) +#define RT5645_M_BST3_L_SM_L_SFT 4 #define RT5645_M_IN_L_SM_L (0x1 << 3) #define RT5645_M_IN_L_SM_L_SFT 3 -#define RT5645_M_DAC_L1_SM_L (0x1 << 1) -#define RT5645_M_DAC_L1_SM_L_SFT 1 #define RT5645_M_DAC_L2_SM_L (0x1 << 2) #define RT5645_M_DAC_L2_SM_L_SFT 2 -#define RT5645_M_BST3_L_SM_L (0x1 << 4) -#define RT5645_M_BST3_L_SM_L_SFT 4 +#define RT5645_M_DAC_L1_SM_L (0x1 << 1) +#define RT5645_M_DAC_L1_SM_L_SFT 1 /* SPK Right Mixer Control (0x47) */ #define RT5645_G_RM_R_SM_R_MASK (0x3 << 14) @@ -643,14 +643,14 @@ #define RT5645_G_OM_R_SM_R_SFT 6 #define RT5645_M_BST2_R_SM_R (0x1 << 5) #define RT5645_M_BST2_R_SM_R_SFT 5 +#define RT5645_M_BST3_R_SM_R (0x1 << 4) +#define RT5645_M_BST3_R_SM_R_SFT 4 #define RT5645_M_IN_R_SM_R (0x1 << 3) #define RT5645_M_IN_R_SM_R_SFT 3 -#define RT5645_M_DAC_R1_SM_R (0x1 << 1) -#define RT5645_M_DAC_R1_SM_R_SFT 1 #define RT5645_M_DAC_R2_SM_R (0x1 << 2) #define RT5645_M_DAC_R2_SM_R_SFT 2 -#define RT5645_M_BST3_R_SM_R (0x1 << 4) -#define RT5645_M_BST3_R_SM_R_SFT 4 +#define RT5645_M_DAC_R1_SM_R (0x1 << 1) +#define RT5645_M_DAC_R1_SM_R_SFT 1 /* SPOLMIX Control (0x48) */ #define RT5645_M_DAC_L1_SPM_L (0x1 << 15) @@ -670,13 +670,17 @@ #define RT5645_M_SV_R_SPM_R (0x1 << 0) #define RT5645_M_SV_R_SPM_R_SFT 0 +/* SPOMIX Ratio Control (0x4a) */ +#define RT5645_SPK_G_CLSD_MASK (0x7 << 0) +#define RT5645_SPK_G_CLSD_SFT 0 + /* Mono Output Mixer Control (0x4c) */ +#define RT5645_G_MONOMIX_MASK (0x1 << 10) +#define RT5645_G_MONOMIX_SFT 10 #define RT5645_M_OV_L_MM (0x1 << 9) #define RT5645_M_OV_L_MM_SFT 9 #define RT5645_M_DAC_L2_MA (0x1 << 8) #define RT5645_M_DAC_L2_MA_SFT 8 -#define RT5645_G_MONOMIX_MASK (0x1 << 10) -#define RT5645_G_MONOMIX_SFT 10 #define RT5645_M_BST2_MM (0x1 << 4) #define RT5645_M_BST2_MM_SFT 4 #define RT5645_M_DAC_R1_MM (0x1 << 3) @@ -779,8 +783,6 @@ #define RT5645_PWR_CLS_D_R_BIT 9 #define RT5645_PWR_CLS_D_L (0x1 << 8) #define RT5645_PWR_CLS_D_L_BIT 8 -#define RT5645_PWR_ADC_R (0x1 << 1) -#define RT5645_PWR_ADC_R_BIT 1 #define RT5645_PWR_DAC_L2 (0x1 << 7) #define RT5645_PWR_DAC_L2_BIT 7 #define RT5645_PWR_DAC_R2 (0x1 << 6) @@ -1628,6 +1630,10 @@ #define RT5645_OT_P_NOR (0x0 << 10) #define RT5645_OT_P_INV (0x1 << 10) #define RT5645_IRQ_JD_1_1_EN (0x1 << 9) +#define RT5645_JD_1_1_MASK (0x1 << 7) +#define RT5645_JD_1_1_SFT 7 +#define RT5645_JD_1_1_NOR (0x0 << 7) +#define RT5645_JD_1_1_INV (0x1 << 7) /* IRQ Control 2 (0xbe) */ #define RT5645_IRQ_MB1_OC_MASK (0x1 << 15) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index ddb0203fc649..86b81a60ac52 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -723,17 +723,11 @@ static struct snd_soc_codec_driver ssm2518_codec_driver = { .num_dapm_routes = ARRAY_SIZE(ssm2518_routes), }; -static bool ssm2518_register_volatile(struct device *dev, unsigned int reg) -{ - return false; -} - static const struct regmap_config ssm2518_regmap_config = { .val_bits = 8, .reg_bits = 8, .max_register = SSM2518_REG_DRC_9, - .volatile_reg = ssm2518_register_volatile, .cache_type = REGCACHE_RBTREE, .reg_defaults = ssm2518_reg_defaults, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 8739126a1f6f..a564759845f9 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -80,6 +80,7 @@ struct aic3x_priv { unsigned int sysclk; unsigned int dai_fmt; unsigned int tdm_delay; + unsigned int slot_width; struct list_head list; int master; int gpio_reset; @@ -1025,10 +1026,14 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; u16 d, pll_d = 1; int clk; + int width = aic3x->slot_width; + + if (!width) + width = params_width(params); /* select data word length */ data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); - switch (params_width(params)) { + switch (width) { case 16: break; case 20: @@ -1170,12 +1175,16 @@ static int aic3x_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); int delay = 0; + int width = aic3x->slot_width; + + if (!width) + width = substream->runtime->sample_bits; /* TDM slot selection only valid in DSP_A/_B mode */ if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_A) - delay += (aic3x->tdm_delay + 1); + delay += (aic3x->tdm_delay*width + 1); else if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_B) - delay += aic3x->tdm_delay; + delay += aic3x->tdm_delay*width; /* Configure data delay */ snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); @@ -1296,7 +1305,20 @@ static int aic3x_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, return -EINVAL; } - aic3x->tdm_delay = lsb * slot_width; + switch (slot_width) { + case 16: + case 20: + case 24: + case 32: + break; + default: + dev_err(codec->dev, "Unsupported slot width %d\n", slot_width); + return -EINVAL; + } + + + aic3x->tdm_delay = lsb; + aic3x->slot_width = slot_width; /* DOUT in high-impedance on inactive bit clocks */ snd_soc_update_bits(codec, AIC3X_ASD_INTF_CTRLA, diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2713e1845cbc..a5a4e9f75c57 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1612,19 +1612,16 @@ static void twl4030_constraints(struct twl4030_priv *twl4030, return; /* Set the constraints according to the already configured stream */ - snd_pcm_hw_constraint_minmax(slv_substream->runtime, + snd_pcm_hw_constraint_single(slv_substream->runtime, SNDRV_PCM_HW_PARAM_RATE, - twl4030->rate, twl4030->rate); - snd_pcm_hw_constraint_minmax(slv_substream->runtime, + snd_pcm_hw_constraint_single(slv_substream->runtime, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - twl4030->sample_bits, twl4030->sample_bits); - snd_pcm_hw_constraint_minmax(slv_substream->runtime, + snd_pcm_hw_constraint_single(slv_substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, - twl4030->channels, twl4030->channels); } @@ -1669,9 +1666,9 @@ static int twl4030_startup(struct snd_pcm_substream *substream, /* In option2 4 channel is not supported, set the * constraint for the first stream for channels, the * second stream will 'inherit' this cosntraint */ - snd_pcm_hw_constraint_minmax(substream->runtime, + snd_pcm_hw_constraint_single(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, - 2, 2); + 2); } twl4030->master_substream = substream; } diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index e19026380534..e4c694c758b8 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -150,14 +150,12 @@ static int uda134x_startup(struct snd_pcm_substream *substream, master_runtime->sample_bits, master_runtime->rate); - snd_pcm_hw_constraint_minmax(substream->runtime, + snd_pcm_hw_constraint_single(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, master_runtime->rate); - snd_pcm_hw_constraint_minmax(substream->runtime, + snd_pcm_hw_constraint_single(substream->runtime, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, master_runtime->sample_bits); uda134x->slave_substream = substream; diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 80fb1dc81f6c..7693c1129bab 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -307,11 +307,10 @@ static int wl1273_startup(struct snd_pcm_substream *substream, switch (wl1273->mode) { case WL1273_MODE_BT: - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - 8000, 8000); - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, 1, 1); + snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 8000); + snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, 1); break; case WL1273_MODE_FM_RX: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 786abd02b140..a67ea10f41a1 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -620,7 +620,7 @@ static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - int anc_active = ucontrol->value.integer.value[0]; + unsigned int anc_active = ucontrol->value.integer.value[0]; int ret; if (anc_active > 1) @@ -653,7 +653,7 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - int val = ucontrol->value.integer.value[0]; + unsigned int val = ucontrol->value.integer.value[0]; int ret; if (val > 1) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 9756578fc752..c04c0bc6f58a 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -38,6 +38,12 @@ struct wm5110_priv { struct arizona_priv core; struct arizona_fll fll[2]; + + unsigned int in_value; + int in_pre_pending; + int in_post_pending; + + unsigned int in_pga_cache[6]; }; static const struct wm_adsp_region wm5110_dsp1_regions[] = { @@ -428,6 +434,127 @@ err: return ret; } +static int wm5110_in_pga_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct snd_soc_card *card = dapm->card; + int ret; + + /* + * PGA Volume is also used as part of the enable sequence, so + * usage of it should be avoided whilst that is running. + */ + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_get_volsw_range(kcontrol, ucontrol); + + mutex_unlock(&card->dapm_mutex); + + return ret; +} + +static int wm5110_in_pga_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct snd_soc_card *card = dapm->card; + int ret; + + /* + * PGA Volume is also used as part of the enable sequence, so + * usage of it should be avoided whilst that is running. + */ + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_put_volsw_range(kcontrol, ucontrol); + + mutex_unlock(&card->dapm_mutex); + + return ret; +} + +static int wm5110_in_analog_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + unsigned int reg, mask; + struct reg_sequence analog_seq[] = { + { 0x80, 0x3 }, + { 0x35d, 0 }, + { 0x80, 0x0 }, + }; + + reg = ARIZONA_IN1L_CONTROL + ((w->shift ^ 0x1) * 4); + mask = ARIZONA_IN1L_PGA_VOL_MASK; + + switch (event) { + case SND_SOC_DAPM_WILL_PMU: + wm5110->in_value |= 0x3 << ((w->shift ^ 0x1) * 2); + wm5110->in_pre_pending++; + wm5110->in_post_pending++; + return 0; + case SND_SOC_DAPM_PRE_PMU: + wm5110->in_pga_cache[w->shift] = snd_soc_read(codec, reg); + + snd_soc_update_bits(codec, reg, mask, + 0x40 << ARIZONA_IN1L_PGA_VOL_SHIFT); + + wm5110->in_pre_pending--; + if (wm5110->in_pre_pending == 0) { + analog_seq[1].def = wm5110->in_value; + regmap_multi_reg_write_bypassed(arizona->regmap, + analog_seq, + ARRAY_SIZE(analog_seq)); + + msleep(55); + + wm5110->in_value = 0; + } + + break; + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, reg, mask, + wm5110->in_pga_cache[w->shift]); + + wm5110->in_post_pending--; + if (wm5110->in_post_pending == 0) + regmap_multi_reg_write_bypassed(arizona->regmap, + analog_seq, + ARRAY_SIZE(analog_seq)); + break; + default: + break; + } + + return 0; +} + +static int wm5110_in_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + + switch (arizona->rev) { + case 0 ... 4: + if (arizona_input_analog(codec, w->shift)) + wm5110_in_analog_ev(w, kcontrol, event); + + break; + default: + break; + } + + return arizona_in_ev(w, kcontrol, event); +} + static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); @@ -454,18 +581,24 @@ SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]), SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]), SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]), -SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, - ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, - ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, - ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum), @@ -896,29 +1029,35 @@ SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"), SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index ace8645245a0..07cf1bd7913a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -132,7 +132,7 @@ static int wm8731_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; int ret = 0; if (deemph > 1) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b011253459af..e4cc41e6c23e 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -452,7 +452,7 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; int ret = 0; if (deemph > 1) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index b783743dc97e..2aa23f1b9e3c 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -534,7 +534,7 @@ static int wm8904_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 12e4435f00f8..9db00d53abe7 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -402,7 +402,7 @@ static int wm8955_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index dbd88408861a..056375339ea3 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -201,7 +201,7 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c new file mode 100644 index 000000000000..8782dfb628ab --- /dev/null +++ b/sound/soc/codecs/wm8998.c @@ -0,0 +1,1430 @@ +/* + * wm8998.c -- ALSA SoC Audio driver for WM8998 codecs + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" +#include "wm8998.h" + +struct wm8998_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static int wm8998_asrc_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + val = snd_soc_read(codec, ARIZONA_ASRC_RATE1); + val &= ARIZONA_ASRC_RATE1_MASK; + val >>= ARIZONA_ASRC_RATE1_SHIFT; + + switch (val) { + case 0: + case 1: + case 2: + val = snd_soc_read(codec, + ARIZONA_SAMPLE_RATE_1 + val); + if (val >= 0x11) { + dev_warn(codec->dev, + "Unsupported ASRC rate1 (%s)\n", + arizona_sample_rate_val_to_name(val)); + return -EINVAL; + } + break; + default: + dev_err(codec->dev, + "Illegal ASRC rate1 selector (0x%x)\n", + val); + return -EINVAL; + } + + val = snd_soc_read(codec, ARIZONA_ASRC_RATE2); + val &= ARIZONA_ASRC_RATE2_MASK; + val >>= ARIZONA_ASRC_RATE2_SHIFT; + + switch (val) { + case 8: + case 9: + val -= 0x8; + val = snd_soc_read(codec, + ARIZONA_ASYNC_SAMPLE_RATE_1 + val); + if (val >= 0x11) { + dev_warn(codec->dev, + "Unsupported ASRC rate2 (%s)\n", + arizona_sample_rate_val_to_name(val)); + return -EINVAL; + } + break; + default: + dev_err(codec->dev, + "Illegal ASRC rate2 selector (0x%x)\n", + val); + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm8998_in1mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = wm8998->core.arizona; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int mux, inmode; + unsigned int mode_val, src_val; + + mux = ucontrol->value.enumerated.item[0]; + if (mux > 1) + return -EINVAL; + + /* L and R registers have same shift and mask */ + inmode = arizona->pdata.inmode[2 * mux]; + src_val = mux << ARIZONA_IN1L_SRC_SHIFT; + if (inmode & ARIZONA_INMODE_SE) + src_val |= 1 << ARIZONA_IN1L_SRC_SE_SHIFT; + + switch (arizona->pdata.inmode[0]) { + case ARIZONA_INMODE_DMIC: + if (mux) + mode_val = 0; /* B always analogue */ + else + mode_val = 1 << ARIZONA_IN1_MODE_SHIFT; + + snd_soc_update_bits(codec, ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_MODE_MASK, mode_val); + + /* IN1A is digital so L and R must change together */ + /* src_val setting same for both registers */ + snd_soc_update_bits(codec, + ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_SRC_MASK | + ARIZONA_IN1L_SRC_SE_MASK, src_val); + snd_soc_update_bits(codec, + ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_SRC_MASK | + ARIZONA_IN1R_SRC_SE_MASK, src_val); + break; + default: + /* both analogue */ + snd_soc_update_bits(codec, + e->reg, + ARIZONA_IN1L_SRC_MASK | + ARIZONA_IN1L_SRC_SE_MASK, + src_val); + break; + } + + return snd_soc_dapm_mux_update_power(dapm, kcontrol, + ucontrol->value.enumerated.item[0], + e, NULL); +} + +static int wm8998_in2mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = wm8998->core.arizona; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int mux, inmode, src_val, mode_val; + + mux = ucontrol->value.enumerated.item[0]; + if (mux > 1) + return -EINVAL; + + inmode = arizona->pdata.inmode[1 + (2 * mux)]; + if (inmode & ARIZONA_INMODE_DMIC) + mode_val = 1 << ARIZONA_IN2_MODE_SHIFT; + else + mode_val = 0; + + src_val = mux << ARIZONA_IN2L_SRC_SHIFT; + if (inmode & ARIZONA_INMODE_SE) + src_val |= 1 << ARIZONA_IN2L_SRC_SE_SHIFT; + + snd_soc_update_bits(codec, ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_MODE_MASK, mode_val); + + snd_soc_update_bits(codec, ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_SRC_MASK | ARIZONA_IN2L_SRC_SE_MASK, + src_val); + + return snd_soc_dapm_mux_update_power(dapm, kcontrol, + ucontrol->value.enumerated.item[0], + e, NULL); +} + +static const char * const wm8998_inmux_texts[] = { + "A", + "B", +}; + +static const SOC_ENUM_SINGLE_DECL(wm8998_in1muxl_enum, + ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_SRC_SHIFT, + wm8998_inmux_texts); + +static const SOC_ENUM_SINGLE_DECL(wm8998_in1muxr_enum, + ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_SRC_SHIFT, + wm8998_inmux_texts); + +static const SOC_ENUM_SINGLE_DECL(wm8998_in2mux_enum, + ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_SRC_SHIFT, + wm8998_inmux_texts); + +static const struct snd_kcontrol_new wm8998_in1mux[2] = { + SOC_DAPM_ENUM_EXT("IN1L Mux", wm8998_in1muxl_enum, + snd_soc_dapm_get_enum_double, wm8998_in1mux_put), + SOC_DAPM_ENUM_EXT("IN1R Mux", wm8998_in1muxr_enum, + snd_soc_dapm_get_enum_double, wm8998_in1mux_put), +}; + +static const struct snd_kcontrol_new wm8998_in2mux = + SOC_DAPM_ENUM_EXT("IN2 Mux", wm8998_in2mux_enum, + snd_soc_dapm_get_enum_double, wm8998_in2mux_put); + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); + +#define WM8998_NG_SRC(name, base) \ + SOC_SINGLE(name " NG HPOUTL Switch", base, 0, 1, 0), \ + SOC_SINGLE(name " NG HPOUTR Switch", base, 1, 1, 0), \ + SOC_SINGLE(name " NG LINEOUTL Switch", base, 2, 1, 0), \ + SOC_SINGLE(name " NG LINEOUTR Switch", base, 3, 1, 0), \ + SOC_SINGLE(name " NG EPOUT Switch", base, 4, 1, 0), \ + SOC_SINGLE(name " NG SPKOUTL Switch", base, 6, 1, 0), \ + SOC_SINGLE(name " NG SPKOUTR Switch", base, 7, 1, 0) + +static const struct snd_kcontrol_new wm8998_snd_controls[] = { +SOC_ENUM("IN1 OSR", arizona_in_dmic_osr[0]), +SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]), + +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum), + +SOC_SINGLE("IN1L HPF Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN1R HPF Switch", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN2 HPF Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_HPF_SHIFT, 1, 0), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), + +SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), +SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), + +ARIZONA_GAINMUX_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_GAINMUX_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_GAINMUX_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_GAINMUX_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE_SHIFT, 1, 0), +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE_SHIFT, 1, 0), +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE_SHIFT, 1, 0), +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE_SHIFT, 1, 0), +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_GAINMUX_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_GAINMUX_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), +SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), +SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +SOC_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), +SOC_ENUM("ISRC1 FSH", arizona_isrc_fsh[0]), +SOC_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]), +SOC_ENUM("ASRC RATE 1", arizona_asrc_rate1), + +ARIZONA_MIXER_CONTROLS("HPOUTL", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUTR", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LINEOUTL", ARIZONA_OUT2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LINEOUTR", ARIZONA_OUT2RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDATL", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDATR", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), + +SOC_DOUBLE_R("HPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("LINEOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("LINEOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_DOUBLE("SPKDAT Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), + +SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), +SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), + +SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_ENA_SHIFT, 1, 0), +SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), +SOC_ENUM("Noise Gate Hold", arizona_ng_hold), + +WM8998_NG_SRC("HPOUTL", ARIZONA_NOISE_GATE_SELECT_1L), +WM8998_NG_SRC("HPOUTR", ARIZONA_NOISE_GATE_SELECT_1R), +WM8998_NG_SRC("LINEOUTL", ARIZONA_NOISE_GATE_SELECT_2L), +WM8998_NG_SRC("LINEOUTR", ARIZONA_NOISE_GATE_SELECT_2R), +WM8998_NG_SRC("EPOUT", ARIZONA_NOISE_GATE_SELECT_3L), +WM8998_NG_SRC("SPKOUTL", ARIZONA_NOISE_GATE_SELECT_4L), +WM8998_NG_SRC("SPKOUTR", ARIZONA_NOISE_GATE_SELECT_4R), +WM8998_NG_SRC("SPKDATL", ARIZONA_NOISE_GATE_SELECT_5L), +WM8998_NG_SRC("SPKDATR", ARIZONA_NOISE_GATE_SELECT_5R), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX3", ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX4", ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX5", ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), + +ARIZONA_GAINMUX_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE), +ARIZONA_GAINMUX_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE), +ARIZONA_GAINMUX_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE), +ARIZONA_GAINMUX_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE), +ARIZONA_GAINMUX_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), +ARIZONA_GAINMUX_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), + +ARIZONA_GAINMUX_CONTROLS("SPDIFTX1", ARIZONA_SPDIFTX1MIX_INPUT_1_SOURCE), +ARIZONA_GAINMUX_CONTROLS("SPDIFTX2", ARIZONA_SPDIFTX2MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MUX_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDATL, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDATR, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX3, ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX4, ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX5, ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX6, ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(SPD1TX1, ARIZONA_SPDIFTX1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(SPD1TX2, ARIZONA_SPDIFTX2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT3, ARIZONA_ISRC1INT3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT4, ARIZONA_ISRC1INT4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC3, ARIZONA_ISRC1DEC3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC4, ARIZONA_ISRC1DEC4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE); + +static const char * const wm8998_aec_loopback_texts[] = { + "HPOUTL", "HPOUTR", "LINEOUTL", "LINEOUTR", "EPOUT", + "SPKOUTL", "SPKOUTR", "SPKDATL", "SPKDATR", +}; + +static const unsigned int wm8998_aec_loopback_values[] = { + 0, 1, 2, 3, 4, 6, 7, 8, 9, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL(wm8998_aec1_loopback, + ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, + wm8998_aec_loopback_texts, + wm8998_aec_loopback_values); + +static const SOC_VALUE_ENUM_SINGLE_DECL(wm8998_aec2_loopback, + ARIZONA_DAC_AEC_CONTROL_2, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, + wm8998_aec_loopback_texts, + wm8998_aec_loopback_values); + +static const struct snd_kcontrol_new wm8998_aec_loopback_mux[] = { + SOC_DAPM_ENUM("AEC1 Loopback", wm8998_aec1_loopback), + SOC_DAPM_ENUM("AEC2 Loopback", wm8998_aec2_loopback), +}; + +static const struct snd_soc_dapm_widget wm8998_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, + ARIZONA_SYSCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, + ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, + ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0, 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("HAPTICS"), + +SND_SOC_DAPM_INPUT("IN1AL"), +SND_SOC_DAPM_INPUT("IN1AR"), +SND_SOC_DAPM_INPUT("IN1BL"), +SND_SOC_DAPM_INPUT("IN1BR"), +SND_SOC_DAPM_INPUT("IN2A"), +SND_SOC_DAPM_INPUT("IN2B"), + +SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &wm8998_in1mux[0]), +SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &wm8998_in1mux[1]), +SND_SOC_DAPM_MUX("IN2 Mux", SND_SOC_NOPM, 0, 0, &wm8998_in2mux), + +SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2 PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA_E("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0, + NULL, 0, wm8998_asrc_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0, + NULL, 0, wm8998_asrc_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0, + NULL, 0, wm8998_asrc_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, + NULL, 0, wm8998_asrc_ev, SND_SOC_DAPM_PRE_PMU), + +SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT3", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT4", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC3", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC4", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_MUX("AEC1 Loopback", ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm8998_aec_loopback_mux[0]), + +SND_SOC_DAPM_MUX("AEC2 Loopback", ARIZONA_DAC_AEC_CONTROL_2, + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm8998_aec_loopback_mux[1]), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX3", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX4", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX5", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX6", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX3", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX4", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX5", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX6", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX4_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_PGA("SPD1TX1", ARIZONA_SPD1_TX_CONTROL, + ARIZONA_SPD1_VAL1_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("SPD1TX2", ARIZONA_SPD1_TX_CONTROL, + ARIZONA_SPD1_VAL2_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_OUT_DRV("SPD1", ARIZONA_SPD1_TX_CONTROL, + ARIZONA_SPD1_ENA_SHIFT, 0, NULL, 0), + +ARIZONA_MUX_WIDGETS(EQ1, "EQ1"), +ARIZONA_MUX_WIDGETS(EQ2, "EQ2"), +ARIZONA_MUX_WIDGETS(EQ3, "EQ3"), +ARIZONA_MUX_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MUX_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MUX_WIDGETS(DRC1R, "DRC1R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUTL"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUTR"), +ARIZONA_MIXER_WIDGETS(OUT2L, "LINEOUTL"), +ARIZONA_MIXER_WIDGETS(OUT2R, "LINEOUTR"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"), +ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"), +ARIZONA_MIXER_WIDGETS(SPKDATL, "SPKDATL"), +ARIZONA_MIXER_WIDGETS(SPKDATR, "SPKDATR"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), +ARIZONA_MIXER_WIDGETS(AIF2TX3, "AIF2TX3"), +ARIZONA_MIXER_WIDGETS(AIF2TX4, "AIF2TX4"), +ARIZONA_MIXER_WIDGETS(AIF2TX5, "AIF2TX5"), +ARIZONA_MIXER_WIDGETS(AIF2TX6, "AIF2TX6"), + +ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), +ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), + +ARIZONA_MUX_WIDGETS(SLIMTX1, "SLIMTX1"), +ARIZONA_MUX_WIDGETS(SLIMTX2, "SLIMTX2"), +ARIZONA_MUX_WIDGETS(SLIMTX3, "SLIMTX3"), +ARIZONA_MUX_WIDGETS(SLIMTX4, "SLIMTX4"), +ARIZONA_MUX_WIDGETS(SLIMTX5, "SLIMTX5"), +ARIZONA_MUX_WIDGETS(SLIMTX6, "SLIMTX6"), + +ARIZONA_MUX_WIDGETS(SPD1TX1, "SPDIFTX1"), +ARIZONA_MUX_WIDGETS(SPD1TX2, "SPDIFTX2"), + +ARIZONA_MUX_WIDGETS(ASRC1L, "ASRC1L"), +ARIZONA_MUX_WIDGETS(ASRC1R, "ASRC1R"), +ARIZONA_MUX_WIDGETS(ASRC2L, "ASRC2L"), +ARIZONA_MUX_WIDGETS(ASRC2R, "ASRC2R"), + +ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), +ARIZONA_MUX_WIDGETS(ISRC1DEC3, "ISRC1DEC3"), +ARIZONA_MUX_WIDGETS(ISRC1DEC4, "ISRC1DEC4"), + +ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), +ARIZONA_MUX_WIDGETS(ISRC1INT3, "ISRC1INT3"), +ARIZONA_MUX_WIDGETS(ISRC1INT4, "ISRC1INT4"), + +ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), + +SND_SOC_DAPM_OUTPUT("HPOUTL"), +SND_SOC_DAPM_OUTPUT("HPOUTR"), +SND_SOC_DAPM_OUTPUT("LINEOUTL"), +SND_SOC_DAPM_OUTPUT("LINEOUTR"), +SND_SOC_DAPM_OUTPUT("EPOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTLN"), +SND_SOC_DAPM_OUTPUT("SPKOUTLP"), +SND_SOC_DAPM_OUTPUT("SPKOUTRN"), +SND_SOC_DAPM_OUTPUT("SPKOUTRP"), +SND_SOC_DAPM_OUTPUT("SPKDATL"), +SND_SOC_DAPM_OUTPUT("SPKDATR"), +SND_SOC_DAPM_OUTPUT("SPDIF"), + +SND_SOC_DAPM_OUTPUT("MICSUPP"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "Haptics", "HAPTICS" }, \ + { name, "AEC", "AEC1 Loopback" }, \ + { name, "AEC2", "AEC2 Loopback" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2 PGA" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF2RX3", "AIF2RX3" }, \ + { name, "AIF2RX4", "AIF2RX4" }, \ + { name, "AIF2RX5", "AIF2RX5" }, \ + { name, "AIF2RX6", "AIF2RX6" }, \ + { name, "AIF3RX1", "AIF3RX1" }, \ + { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "SLIMRX1", "SLIMRX1" }, \ + { name, "SLIMRX2", "SLIMRX2" }, \ + { name, "SLIMRX3", "SLIMRX3" }, \ + { name, "SLIMRX4", "SLIMRX4" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ASRC1L", "ASRC1L" }, \ + { name, "ASRC1R", "ASRC1R" }, \ + { name, "ASRC2L", "ASRC2L" }, \ + { name, "ASRC2R", "ASRC2R" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1DEC3", "ISRC1DEC3" }, \ + { name, "ISRC1DEC4", "ISRC1DEC4" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC1INT3", "ISRC1INT3" }, \ + { name, "ISRC1INT4", "ISRC1INT4" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" } + +static const struct snd_soc_dapm_route wm8998_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "AIF3 Capture", NULL, "DBVDD3" }, + { "AIF3 Playback", NULL, "DBVDD3" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT2L", NULL, "CPVDD" }, + { "OUT2R", NULL, "CPVDD" }, + { "OUT3", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDDL" }, + { "OUT4R", NULL, "SPKVDDR" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT2L", NULL, "SYSCLK" }, + { "OUT2R", NULL, "SYSCLK" }, + { "OUT3", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + { "OUT4R", NULL, "SYSCLK" }, + { "OUT5L", NULL, "SYSCLK" }, + { "OUT5R", NULL, "SYSCLK" }, + + { "IN1AL", NULL, "SYSCLK" }, + { "IN1AR", NULL, "SYSCLK" }, + { "IN1BL", NULL, "SYSCLK" }, + { "IN1BR", NULL, "SYSCLK" }, + { "IN2A", NULL, "SYSCLK" }, + { "IN2B", NULL, "SYSCLK" }, + + { "SPD1", NULL, "SYSCLK" }, + { "SPD1", NULL, "SPD1TX1" }, + { "SPD1", NULL, "SPD1TX2" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + { "AIF2 Capture", NULL, "AIF2TX3" }, + { "AIF2 Capture", NULL, "AIF2TX4" }, + { "AIF2 Capture", NULL, "AIF2TX5" }, + { "AIF2 Capture", NULL, "AIF2TX6" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + { "AIF2RX3", NULL, "AIF2 Playback" }, + { "AIF2RX4", NULL, "AIF2 Playback" }, + { "AIF2RX5", NULL, "AIF2 Playback" }, + { "AIF2RX6", NULL, "AIF2 Playback" }, + + { "AIF3 Capture", NULL, "AIF3TX1" }, + { "AIF3 Capture", NULL, "AIF3TX2" }, + + { "AIF3RX1", NULL, "AIF3 Playback" }, + { "AIF3RX2", NULL, "AIF3 Playback" }, + + { "Slim1 Capture", NULL, "SLIMTX1" }, + { "Slim1 Capture", NULL, "SLIMTX2" }, + { "Slim1 Capture", NULL, "SLIMTX3" }, + { "Slim1 Capture", NULL, "SLIMTX4" }, + + { "Slim2 Capture", NULL, "SLIMTX5" }, + { "Slim2 Capture", NULL, "SLIMTX6" }, + + { "SLIMRX1", NULL, "Slim1 Playback" }, + { "SLIMRX2", NULL, "Slim1 Playback" }, + + { "SLIMRX3", NULL, "Slim2 Playback" }, + { "SLIMRX4", NULL, "Slim2 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "AIF3 Playback", NULL, "SYSCLK" }, + { "Slim1 Playback", NULL, "SYSCLK" }, + { "Slim2 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "AIF3 Capture", NULL, "SYSCLK" }, + { "Slim1 Capture", NULL, "SYSCLK" }, + { "Slim2 Capture", NULL, "SYSCLK" }, + + { "IN1L Mux", "A", "IN1AL" }, + { "IN1R Mux", "A", "IN1AR" }, + { "IN1L Mux", "B", "IN1BL" }, + { "IN1R Mux", "B", "IN1BR" }, + + { "IN2 Mux", "A", "IN2A" }, + { "IN2 Mux", "B", "IN2B" }, + + { "IN1L PGA", NULL, "IN1L Mux" }, + { "IN1R PGA", NULL, "IN1R Mux" }, + { "IN2 PGA", NULL, "IN2 Mux" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUTL"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUTR"), + ARIZONA_MIXER_ROUTES("OUT2L", "LINEOUTL"), + ARIZONA_MIXER_ROUTES("OUT2R", "LINEOUTR"), + ARIZONA_MIXER_ROUTES("OUT3", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"), + ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDATL"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDATR"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + ARIZONA_MIXER_ROUTES("AIF2TX3", "AIF2TX3"), + ARIZONA_MIXER_ROUTES("AIF2TX4", "AIF2TX4"), + ARIZONA_MIXER_ROUTES("AIF2TX5", "AIF2TX5"), + ARIZONA_MIXER_ROUTES("AIF2TX6", "AIF2TX6"), + + ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), + ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + + ARIZONA_MUX_ROUTES("SLIMTX1", "SLIMTX1"), + ARIZONA_MUX_ROUTES("SLIMTX2", "SLIMTX2"), + ARIZONA_MUX_ROUTES("SLIMTX3", "SLIMTX3"), + ARIZONA_MUX_ROUTES("SLIMTX4", "SLIMTX4"), + ARIZONA_MUX_ROUTES("SLIMTX5", "SLIMTX5"), + ARIZONA_MUX_ROUTES("SLIMTX6", "SLIMTX6"), + + ARIZONA_MUX_ROUTES("SPD1TX1", "SPDIFTX1"), + ARIZONA_MUX_ROUTES("SPD1TX2", "SPDIFTX2"), + + ARIZONA_MUX_ROUTES("EQ1", "EQ1"), + ARIZONA_MUX_ROUTES("EQ2", "EQ2"), + ARIZONA_MUX_ROUTES("EQ3", "EQ3"), + ARIZONA_MUX_ROUTES("EQ4", "EQ4"), + + ARIZONA_MUX_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MUX_ROUTES("DRC1R", "DRC1R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), + + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), + ARIZONA_MUX_ROUTES("ISRC1INT3", "ISRC1INT3"), + ARIZONA_MUX_ROUTES("ISRC1INT4", "ISRC1INT4"), + + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), + ARIZONA_MUX_ROUTES("ISRC1DEC3", "ISRC1DEC3"), + ARIZONA_MUX_ROUTES("ISRC1DEC4", "ISRC1DEC4"), + + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), + + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), + + { "AEC1 Loopback", "HPOUTL", "OUT1L" }, + { "AEC1 Loopback", "HPOUTR", "OUT1R" }, + { "AEC2 Loopback", "HPOUTL", "OUT1L" }, + { "AEC2 Loopback", "HPOUTR", "OUT1R" }, + { "HPOUTL", NULL, "OUT1L" }, + { "HPOUTR", NULL, "OUT1R" }, + + { "AEC1 Loopback", "LINEOUTL", "OUT2L" }, + { "AEC1 Loopback", "LINEOUTR", "OUT2R" }, + { "AEC2 Loopback", "LINEOUTL", "OUT2L" }, + { "AEC2 Loopback", "LINEOUTR", "OUT2R" }, + { "LINEOUTL", NULL, "OUT2L" }, + { "LINEOUTR", NULL, "OUT2R" }, + + { "AEC1 Loopback", "EPOUT", "OUT3" }, + { "AEC2 Loopback", "EPOUT", "OUT3" }, + { "EPOUT", NULL, "OUT3" }, + + { "AEC1 Loopback", "SPKOUTL", "OUT4L" }, + { "AEC2 Loopback", "SPKOUTL", "OUT4L" }, + { "SPKOUTLN", NULL, "OUT4L" }, + { "SPKOUTLP", NULL, "OUT4L" }, + + { "AEC1 Loopback", "SPKOUTR", "OUT4R" }, + { "AEC2 Loopback", "SPKOUTR", "OUT4R" }, + { "SPKOUTRN", NULL, "OUT4R" }, + { "SPKOUTRP", NULL, "OUT4R" }, + + { "SPDIF", NULL, "SPD1" }, + + { "AEC1 Loopback", "SPKDATL", "OUT5L" }, + { "AEC1 Loopback", "SPKDATR", "OUT5R" }, + { "AEC2 Loopback", "SPKDATL", "OUT5L" }, + { "AEC2 Loopback", "SPKDATR", "OUT5R" }, + { "SPKDATL", NULL, "OUT5L" }, + { "SPKDATR", NULL, "OUT5R" }, + + { "MICSUPP", NULL, "SYSCLK" }, + + { "DRC1 Signal Activity", NULL, "DRC1L" }, + { "DRC1 Signal Activity", NULL, "DRC1R" }, +}; + +#define WM8998_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM8998_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm8998_dai[] = { + { + .name = "wm8998-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 6, + .rates = WM8998_RATES, + .formats = WM8998_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 6, + .rates = WM8998_RATES, + .formats = WM8998_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + }, + { + .name = "wm8998-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 6, + .rates = WM8998_RATES, + .formats = WM8998_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 6, + .rates = WM8998_RATES, + .formats = WM8998_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + }, + { + .name = "wm8998-aif3", + .id = 3, + .base = ARIZONA_AIF3_BCLK_CTRL, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8998_RATES, + .formats = WM8998_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8998_RATES, + .formats = WM8998_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + }, + { + .name = "wm8998-slim1", + .id = 4, + .playback = { + .stream_name = "Slim1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8998_RATES, + .formats = WM8998_FORMATS, + }, + .capture = { + .stream_name = "Slim1 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = WM8998_RATES, + .formats = WM8998_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8998-slim2", + .id = 5, + .playback = { + .stream_name = "Slim2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8998_RATES, + .formats = WM8998_FORMATS, + }, + .capture = { + .stream_name = "Slim2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8998_RATES, + .formats = WM8998_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, +}; + +static int wm8998_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM8998_FLL1: + return arizona_set_fll(&wm8998->fll[0], source, Fref, Fout); + case WM8998_FLL2: + return arizona_set_fll(&wm8998->fll[1], source, Fref, Fout); + case WM8998_FLL1_REFCLK: + return arizona_set_fll_refclk(&wm8998->fll[0], source, Fref, + Fout); + case WM8998_FLL2_REFCLK: + return arizona_set_fll_refclk(&wm8998->fll[1], source, Fref, + Fout); + default: + return -EINVAL; + } +} + +static int wm8998_codec_probe(struct snd_soc_codec *codec) +{ + struct wm8998_priv *priv = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + + priv->core.arizona->dapm = dapm; + + arizona_init_spk(codec); + arizona_init_gpio(codec); + + snd_soc_dapm_disable_pin(dapm, "HAPTICS"); + + return 0; +} + +static int wm8998_codec_remove(struct snd_soc_codec *codec) +{ + struct wm8998_priv *priv = snd_soc_codec_get_drvdata(codec); + + priv->core.arizona->dapm = NULL; + + return 0; +} + +#define WM8998_DIG_VU 0x0200 + +static unsigned int wm8998_digital_vu[] = { + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct regmap *wm8998_get_regmap(struct device *dev) +{ + struct wm8998_priv *priv = dev_get_drvdata(dev); + + return priv->core.arizona->regmap; +} + +static struct snd_soc_codec_driver soc_codec_dev_wm8998 = { + .probe = wm8998_codec_probe, + .remove = wm8998_codec_remove, + .get_regmap = wm8998_get_regmap, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm8998_set_fll, + + .controls = wm8998_snd_controls, + .num_controls = ARRAY_SIZE(wm8998_snd_controls), + .dapm_widgets = wm8998_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8998_dapm_widgets), + .dapm_routes = wm8998_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8998_dapm_routes), +}; + +static int wm8998_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm8998_priv *wm8998; + int i; + + wm8998 = devm_kzalloc(&pdev->dev, sizeof(struct wm8998_priv), + GFP_KERNEL); + if (!wm8998) + return -ENOMEM; + platform_set_drvdata(pdev, wm8998); + + wm8998->core.arizona = arizona; + wm8998->core.num_inputs = 3; /* IN1L, IN1R, IN2 */ + + for (i = 0; i < ARRAY_SIZE(wm8998->fll); i++) + wm8998->fll[i].vco_mult = 1; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm8998->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm8998->fll[1]); + + for (i = 0; i < ARRAY_SIZE(wm8998_dai); i++) + arizona_init_dai(&wm8998->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm8998_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm8998_digital_vu[i], + WM8998_DIG_VU, WM8998_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8998, + wm8998_dai, ARRAY_SIZE(wm8998_dai)); +} + +static int wm8998_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm8998_codec_driver = { + .driver = { + .name = "wm8998-codec", + }, + .probe = wm8998_probe, + .remove = wm8998_remove, +}; + +module_platform_driver(wm8998_codec_driver); + +MODULE_DESCRIPTION("ASoC WM8998 driver"); +MODULE_AUTHOR("Richard Fitzgerald <rf@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:wm8998-codec"); diff --git a/sound/soc/codecs/wm8998.h b/sound/soc/codecs/wm8998.h new file mode 100644 index 000000000000..1e8647252162 --- /dev/null +++ b/sound/soc/codecs/wm8998.h @@ -0,0 +1,23 @@ +/* + * wm8998.h -- ALSA SoC Audio driver for WM8998 codecs + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8998_H +#define _WM8998_H + +#include "arizona.h" + +#define WM8998_FLL1 1 +#define WM8998_FLL2 2 +#define WM8998_FLL1_REFCLK 3 +#define WM8998_FLL2_REFCLK 4 + +#endif diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 7d45d98a861f..4495a40a9468 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -80,12 +80,13 @@ struct davinci_mcasp { /* McASP specific data */ int tdm_slots; + u32 tdm_mask[2]; + int slot_width; u8 op_mode; u8 num_serializer; u8 *serial_dir; u8 version; u8 bclk_div; - u16 bclk_lrclk_ratio; int streams; u32 irq_request[2]; int dma_request[2]; @@ -556,8 +557,21 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, mcasp->bclk_div = div; break; - case 2: /* BCLK/LRCLK ratio */ - mcasp->bclk_lrclk_ratio = div; + case 2: /* + * BCLK/LRCLK ratio descries how many bit-clock cycles + * fit into one frame. The clock ratio is given for a + * full period of data (for I2S format both left and + * right channels), so it has to be divided by number + * of tdm-slots (for I2S - divided by 2). + * Instead of storing this ratio, we calculate a new + * tdm_slot width by dividing the the ratio by the + * number of configured tdm slots. + */ + mcasp->slot_width = div / mcasp->tdm_slots; + if (div % mcasp->tdm_slots) + dev_warn(mcasp->dev, + "%s(): BCLK/LRCLK %d is not divisible by %d tdm slots", + __func__, div, mcasp->tdm_slots); break; default: @@ -596,12 +610,92 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, return 0; } +/* All serializers must have equal number of channels */ +static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream, + int serializers) +{ + struct snd_pcm_hw_constraint_list *cl = &mcasp->chconstr[stream]; + unsigned int *list = (unsigned int *) cl->list; + int slots = mcasp->tdm_slots; + int i, count = 0; + + if (mcasp->tdm_mask[stream]) + slots = hweight32(mcasp->tdm_mask[stream]); + + for (i = 2; i <= slots; i++) + list[count++] = i; + + for (i = 2; i <= serializers; i++) + list[count++] = i*slots; + + cl->count = count; + + return 0; +} + +static int davinci_mcasp_set_ch_constraints(struct davinci_mcasp *mcasp) +{ + int rx_serializers = 0, tx_serializers = 0, ret, i; + + for (i = 0; i < mcasp->num_serializer; i++) + if (mcasp->serial_dir[i] == TX_MODE) + tx_serializers++; + else if (mcasp->serial_dir[i] == RX_MODE) + rx_serializers++; + + ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_PLAYBACK, + tx_serializers); + if (ret) + return ret; + + ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_CAPTURE, + rx_serializers); + + return ret; +} + + +static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + + dev_dbg(mcasp->dev, + "%s() tx_mask 0x%08x rx_mask 0x%08x slots %d width %d\n", + __func__, tx_mask, rx_mask, slots, slot_width); + + if (tx_mask >= (1<<slots) || rx_mask >= (1<<slots)) { + dev_err(mcasp->dev, + "Bad tdm mask tx: 0x%08x rx: 0x%08x slots %d\n", + tx_mask, rx_mask, slots); + return -EINVAL; + } + + if (slot_width && + (slot_width < 8 || slot_width > 32 || slot_width % 4 != 0)) { + dev_err(mcasp->dev, "%s: Unsupported slot_width %d\n", + __func__, slot_width); + return -EINVAL; + } + + mcasp->tdm_slots = slots; + mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = rx_mask; + mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = tx_mask; + mcasp->slot_width = slot_width; + + return davinci_mcasp_set_ch_constraints(mcasp); +} + static int davinci_config_channel_size(struct davinci_mcasp *mcasp, - int word_length) + int sample_width) { u32 fmt; - u32 tx_rotate = (word_length / 4) & 0x7; - u32 mask = (1ULL << word_length) - 1; + u32 tx_rotate = (sample_width / 4) & 0x7; + u32 mask = (1ULL << sample_width) - 1; + u32 slot_width = sample_width; + /* * For captured data we should not rotate, inversion and masking is * enoguh to get the data to the right position: @@ -614,28 +708,23 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, u32 rx_rotate = 0; /* - * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() - * callback, take it into account here. That allows us to for example - * send 32 bits per channel to the codec, while only 16 of them carry - * audio payload. - * The clock ratio is given for a full period of data (for I2S format - * both left and right channels), so it has to be divided by number of - * tdm-slots (for I2S - divided by 2). + * Setting the tdm slot width either with set_clkdiv() or + * set_tdm_slot() allows us to for example send 32 bits per + * channel to the codec, while only 16 of them carry audio + * payload. */ - if (mcasp->bclk_lrclk_ratio) { - u32 slot_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots; - + if (mcasp->slot_width) { /* - * When we have more bclk then it is needed for the data, we - * need to use the rotation to move the received samples to have - * correct alignment. + * When we have more bclk then it is needed for the + * data, we need to use the rotation to move the + * received samples to have correct alignment. */ - rx_rotate = (slot_length - word_length) / 4; - word_length = slot_length; + slot_width = mcasp->slot_width; + rx_rotate = (slot_width - sample_width) / 4; } /* mapping of the XSSZ bit-field as described in the datasheet */ - fmt = (word_length >> 1) - 1; + fmt = (slot_width >> 1) - 1; if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt), @@ -776,33 +865,50 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, /* * If more than one serializer is needed, then use them with - * their specified tdm_slots count. Otherwise, one serializer - * can cope with the transaction using as many slots as channels - * in the stream, requires channels symmetry + * all the specified tdm_slots. Otherwise, one serializer can + * cope with the transaction using just as many slots as there + * are channels in the stream. */ - active_serializers = (channels + total_slots - 1) / total_slots; - if (active_serializers == 1) - active_slots = channels; - else - active_slots = total_slots; - - for (i = 0; i < active_slots; i++) - mask |= (1 << i); + if (mcasp->tdm_mask[stream]) { + active_slots = hweight32(mcasp->tdm_mask[stream]); + active_serializers = (channels + active_slots - 1) / + active_slots; + if (active_serializers == 1) { + active_slots = channels; + for (i = 0; i < total_slots; i++) { + if ((1 << i) & mcasp->tdm_mask[stream]) { + mask |= (1 << i); + if (--active_slots <= 0) + break; + } + } + } + } else { + active_serializers = (channels + total_slots - 1) / total_slots; + if (active_serializers == 1) + active_slots = channels; + else + active_slots = total_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + } mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); if (!mcasp->dat_port) busel = TXSEL; - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); - mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, - FSXMOD(total_slots), FSXMOD(0x1FF)); - - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); - mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); - mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, - FSRMOD(total_slots), FSRMOD(0x1FF)); + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(total_slots), FSXMOD(0x1FF)); + } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(total_slots), FSRMOD(0x1FF)); + } return 0; } @@ -922,6 +1028,9 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int sbits = params_width(params); int ppm, div; + if (mcasp->slot_width) + sbits = mcasp->slot_width; + div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*slots, &ppm); if (ppm) @@ -1027,6 +1136,9 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_interval range; int i; + if (rd->mcasp->slot_width) + sbits = rd->mcasp->slot_width; + snd_interval_any(&range); range.empty = 1; @@ -1069,10 +1181,14 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { if (snd_mask_test(fmt, i)) { - uint bclk_freq = snd_pcm_format_width(i)*slots*rate; + uint sbits = snd_pcm_format_width(i); int ppm; - davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (rd->mcasp->slot_width) + sbits = rd->mcasp->slot_width; + + davinci_mcasp_calc_clk_div(rd->mcasp, sbits*slots*rate, + &ppm); if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { snd_mask_set(&nfmt, i); count++; @@ -1094,6 +1210,10 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, &mcasp->ruledata[substream->stream]; u32 max_channels = 0; int i, dir; + int tdm_slots = mcasp->tdm_slots; + + if (mcasp->tdm_mask[substream->stream]) + tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); mcasp->substreams[substream->stream] = substream; @@ -1114,7 +1234,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, max_channels++; } ruledata->serializers = max_channels; - max_channels *= mcasp->tdm_slots; + max_channels *= tdm_slots; /* * If the already active stream has less channels than the calculated * limnit based on the seirializers * tdm_slots, we need to use that as @@ -1124,15 +1244,25 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, */ if (mcasp->channels && mcasp->channels < max_channels) max_channels = mcasp->channels; + /* + * But we can always allow channels upto the amount of + * the available tdm_slots. + */ + if (max_channels < tdm_slots) + max_channels = tdm_slots; snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, max_channels); - if (mcasp->chconstr[substream->stream].count) - snd_pcm_hw_constraint_list(substream->runtime, - 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &mcasp->chconstr[substream->stream]); + snd_pcm_hw_constraint_list(substream->runtime, + 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &mcasp->chconstr[substream->stream]); + + if (mcasp->slot_width) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + 8, mcasp->slot_width); /* * If we rely on implicit BCLK divider setting we should @@ -1184,6 +1314,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .set_fmt = davinci_mcasp_set_dai_fmt, .set_clkdiv = davinci_mcasp_set_clkdiv, .set_sysclk = davinci_mcasp_set_sysclk, + .set_tdm_slot = davinci_mcasp_set_tdm_slot, }; static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) @@ -1514,59 +1645,6 @@ nodata: return pdata; } -/* All serializers must have equal number of channels */ -static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, - struct snd_pcm_hw_constraint_list *cl, - int serializers) -{ - unsigned int *list; - int i, count = 0; - - if (serializers <= 1) - return 0; - - list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) * - (mcasp->tdm_slots + serializers - 2), - GFP_KERNEL); - if (!list) - return -ENOMEM; - - for (i = 2; i <= mcasp->tdm_slots; i++) - list[count++] = i; - - for (i = 2; i <= serializers; i++) - list[count++] = i*mcasp->tdm_slots; - - cl->count = count; - cl->list = list; - - return 0; -} - - -static int davinci_mcasp_init_ch_constraints(struct davinci_mcasp *mcasp) -{ - int rx_serializers = 0, tx_serializers = 0, ret, i; - - for (i = 0; i < mcasp->num_serializer; i++) - if (mcasp->serial_dir[i] == TX_MODE) - tx_serializers++; - else if (mcasp->serial_dir[i] == RX_MODE) - rx_serializers++; - - ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[ - SNDRV_PCM_STREAM_PLAYBACK], - tx_serializers); - if (ret) - return ret; - - ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[ - SNDRV_PCM_STREAM_CAPTURE], - rx_serializers); - - return ret; -} - enum { PCM_EDMA, PCM_SDMA, @@ -1783,7 +1861,28 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; } - ret = davinci_mcasp_init_ch_constraints(mcasp); + /* Allocate memory for long enough list for all possible + * scenarios. Maximum number tdm slots is 32 and there cannot + * be more serializers than given in the configuration. The + * serializer directions could be taken into account, but it + * would make code much more complex and save only couple of + * bytes. + */ + mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list = + devm_kzalloc(mcasp->dev, sizeof(unsigned int) * + (32 + mcasp->num_serializer - 2), + GFP_KERNEL); + + mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list = + devm_kzalloc(mcasp->dev, sizeof(unsigned int) * + (32 + mcasp->num_serializer - 2), + GFP_KERNEL); + + if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list || + !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list) + return -ENOMEM; + + ret = davinci_mcasp_set_ch_constraints(mcasp); if (ret) goto err; diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index ba34252b7bba..6e6a70c5c2bd 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -282,23 +282,25 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, config->sample_rate = params_rate(params); - if (dev->i2s_clk_cfg) { - ret = dev->i2s_clk_cfg(config); - if (ret < 0) { - dev_err(dev->dev, "runtime audio clk config fail\n"); - return ret; - } - } else { - u32 bitclk = config->sample_rate * config->data_width * 2; - - ret = clk_set_rate(dev->clk, bitclk); - if (ret) { - dev_err(dev->dev, "Can't set I2S clock rate: %d\n", - ret); - return ret; + if (dev->capability & DW_I2S_MASTER) { + if (dev->i2s_clk_cfg) { + ret = dev->i2s_clk_cfg(config); + if (ret < 0) { + dev_err(dev->dev, "runtime audio clk config fail\n"); + return ret; + } + } else { + u32 bitclk = config->sample_rate * + config->data_width * 2; + + ret = clk_set_rate(dev->clk, bitclk); + if (ret) { + dev_err(dev->dev, "Can't set I2S clock rate: %d\n", + ret); + return ret; + } } } - return 0; } @@ -348,12 +350,43 @@ static int dw_i2s_trigger(struct snd_pcm_substream *substream, return ret; } +static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + if (dev->capability & DW_I2S_SLAVE) + ret = 0; + else + ret = -EINVAL; + break; + case SND_SOC_DAIFMT_CBS_CFS: + if (dev->capability & DW_I2S_MASTER) + ret = 0; + else + ret = -EINVAL; + break; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + ret = -EINVAL; + break; + default: + dev_dbg(dev->dev, "dwc : Invalid master/slave format\n"); + ret = -EINVAL; + break; + } + return ret; +} + static struct snd_soc_dai_ops dw_i2s_dai_ops = { .startup = dw_i2s_startup, .shutdown = dw_i2s_shutdown, .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, + .set_fmt = dw_i2s_set_fmt, }; static const struct snd_soc_component_driver dw_i2s_component = { @@ -366,7 +399,8 @@ static int dw_i2s_suspend(struct snd_soc_dai *dai) { struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); - clk_disable(dev->clk); + if (dev->capability & DW_I2S_MASTER) + clk_disable(dev->clk); return 0; } @@ -374,7 +408,8 @@ static int dw_i2s_resume(struct snd_soc_dai *dai) { struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); - clk_enable(dev->clk); + if (dev->capability & DW_I2S_MASTER) + clk_enable(dev->clk); return 0; } @@ -452,6 +487,14 @@ static int dw_configure_dai(struct dw_i2s_dev *dev, dw_i2s_dai->capture.rates = rates; } + if (COMP1_MODE_EN(comp1)) { + dev_dbg(dev->dev, "designware: i2s master mode supported\n"); + dev->capability |= DW_I2S_MASTER; + } else { + dev_dbg(dev->dev, "designware: i2s slave mode supported\n"); + dev->capability |= DW_I2S_SLAVE; + } + return 0; } @@ -538,6 +581,7 @@ static int dw_i2s_probe(struct platform_device *pdev) struct resource *res; int ret; struct snd_soc_dai_driver *dw_i2s_dai; + const char *clk_id; dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); if (!dev) { @@ -559,32 +603,35 @@ static int dw_i2s_probe(struct platform_device *pdev) return PTR_ERR(dev->i2s_base); dev->dev = &pdev->dev; + if (pdata) { + dev->capability = pdata->cap; + clk_id = NULL; ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata); - if (ret < 0) - return ret; + } else { + clk_id = "i2sclk"; + ret = dw_configure_dai_by_dt(dev, dw_i2s_dai, res); + } + if (ret < 0) + return ret; - dev->capability = pdata->cap; - dev->i2s_clk_cfg = pdata->i2s_clk_cfg; - if (!dev->i2s_clk_cfg) { - dev_err(&pdev->dev, "no clock configure method\n"); - return -ENODEV; + if (dev->capability & DW_I2S_MASTER) { + if (pdata) { + dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + if (!dev->i2s_clk_cfg) { + dev_err(&pdev->dev, "no clock configure method\n"); + return -ENODEV; + } } + dev->clk = devm_clk_get(&pdev->dev, clk_id); - dev->clk = devm_clk_get(&pdev->dev, NULL); - } else { - ret = dw_configure_dai_by_dt(dev, dw_i2s_dai, res); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + ret = clk_prepare_enable(dev->clk); if (ret < 0) return ret; - - dev->clk = devm_clk_get(&pdev->dev, "i2sclk"); } - if (IS_ERR(dev->clk)) - return PTR_ERR(dev->clk); - - ret = clk_prepare_enable(dev->clk); - if (ret < 0) - return ret; dev_set_drvdata(&pdev->dev, dev); ret = devm_snd_soc_register_component(&pdev->dev, &dw_i2s_component, @@ -606,7 +653,8 @@ static int dw_i2s_probe(struct platform_device *pdev) return 0; err_clk_disable: - clk_disable_unprepare(dev->clk); + if (dev->capability & DW_I2S_MASTER) + clk_disable_unprepare(dev->clk); return ret; } @@ -614,7 +662,8 @@ static int dw_i2s_remove(struct platform_device *pdev) { struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev); - clk_disable_unprepare(dev->clk); + if (dev->capability & DW_I2S_MASTER) + clk_disable_unprepare(dev->clk); return 0; } diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 96f55ae75c71..1b05d1c5d9fd 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -14,6 +14,9 @@ #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_platform.h> +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include <sound/ac97_codec.h> +#endif #include <sound/pcm_params.h> #include <sound/soc.h> @@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; +static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) +{ + return priv->dai_fmt == SND_SOC_DAIFMT_AC97; +} + static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, * set_bias_level(), bypass the remaining settings in hw_params(). * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. */ - if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) + if ((priv->card.set_bias_level && + priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || + fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, ext_port--; /* - * Use asynchronous mode (6 wires) for all cases. + * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ @@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: - return -EINVAL; + if (!fsl_asoc_card_is_ac97(priv)) + return -EINVAL; + } + + if (fsl_asoc_card_is_ac97(priv)) { + int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ - ret = imx_audmux_v2_configure_port(int_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(dev, "audmux internal port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); + + ret = imx_audmux_v2_configure_port(int_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, @@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return ret; } - ret = imx_audmux_v2_configure_port(ext_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(dev, "audmux external port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); + + ret = imx_audmux_v2_configure_port(ext_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, @@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (fsl_asoc_card_is_ac97(priv)) { +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + /* + * Use slots 3/4 for S/PDIF so SSI won't try to enable + * other slots and send some samples there + * due to SLOTREQ bits for S/PDIF received from codec + */ + snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, + AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); +#endif + + return 0; + } + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -407,7 +454,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct i2c_client *codec_dev; - struct clk *codec_clk; const char *codec_dai_name; u32 width; int ret; @@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (!cpu_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); + if (!cpu_np) { + dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } @@ -434,22 +479,24 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) + codec_dev = of_find_i2c_device_by_node(codec_np); + else + codec_dev = NULL; asrc_np = of_parse_phandle(np, "audio-asrc", 0); if (asrc_np) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - codec_clk = clk_get(&codec_dev->dev, NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (codec_dev) { + struct clk *codec_clk = clk_get(&codec_dev->dev, NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } /* Default sample rate and format, will be updated in hw_params() */ @@ -486,12 +533,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name = "ac97-hifi"; + priv->card.set_bias_level = NULL; + priv->dai_fmt = SND_SOC_DAIFMT_AC97; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; goto asrc_fail; } + if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { + dev_err(&pdev->dev, "failed to find codec device\n"); + ret = -EINVAL; + goto asrc_fail; + } + /* Common settings for corresponding Freescale CPU DAI driver */ if (strstr(cpu_np->name, "ssi")) { /* Only SSI needs to configure AUDMUX */ @@ -508,7 +565,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } - sprintf(priv->name, "%s-audio", codec_dev->name); + snprintf(priv->name, sizeof(priv->name), "%s-audio", + fsl_asoc_card_is_ac97(priv) ? "ac97" : + codec_dev->name); /* Initialize sound card */ priv->pdev = pdev; @@ -532,8 +591,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; - priv->dai_link[0].codec_of_node = codec_np; priv->dai_link[0].codec_dai_name = codec_dai_name; + + if (!fsl_asoc_card_is_ac97(priv)) + priv->dai_link[0].codec_of_node = codec_np; + else { + u32 idx; + + ret = of_property_read_u32(cpu_np, "cell-index", &idx); + if (ret) { + dev_err(&pdev->dev, + "cannot get CPU index property\n"); + goto asrc_fail; + } + + priv->dai_link[0].codec_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, + "ac97-codec.%u", + (unsigned int)idx); + } + priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -544,6 +621,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[1].platform_of_node = asrc_np; priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].codec_name = + priv->dai_link[0].codec_name; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; @@ -579,20 +658,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); -fail: of_node_put(codec_np); +fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, {} }; +MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); static struct platform_driver fsl_asoc_card_driver = { .probe = fsl_asoc_card_probe, diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 837979ea5c92..59f234e51971 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -652,6 +652,24 @@ static const struct snd_soc_component_driver fsl_esai_component = { .name = "fsl-esai", }; +static const struct reg_default fsl_esai_reg_defaults[] = { + {0x8, 0x00000000}, + {0x10, 0x00000000}, + {0x18, 0x00000000}, + {0x98, 0x00000000}, + {0xd0, 0x00000000}, + {0xd4, 0x00000000}, + {0xd8, 0x00000000}, + {0xdc, 0x00000000}, + {0xe0, 0x00000000}, + {0xe4, 0x0000ffff}, + {0xe8, 0x0000ffff}, + {0xec, 0x0000ffff}, + {0xf0, 0x0000ffff}, + {0xf8, 0x00000000}, + {0xfc, 0x00000000}, +}; + static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -684,6 +702,31 @@ static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg) } } +static bool fsl_esai_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ESAI_ETDR: + case REG_ESAI_ERDR: + case REG_ESAI_ESR: + case REG_ESAI_TFSR: + case REG_ESAI_RFSR: + case REG_ESAI_TX0: + case REG_ESAI_TX1: + case REG_ESAI_TX2: + case REG_ESAI_TX3: + case REG_ESAI_TX4: + case REG_ESAI_TX5: + case REG_ESAI_RX0: + case REG_ESAI_RX1: + case REG_ESAI_RX2: + case REG_ESAI_RX3: + case REG_ESAI_SAISR: + return true; + default: + return false; + } +} + static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -721,8 +764,12 @@ static const struct regmap_config fsl_esai_regmap_config = { .val_bits = 32, .max_register = REG_ESAI_PCRC, + .reg_defaults = fsl_esai_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(fsl_esai_reg_defaults), .readable_reg = fsl_esai_readable_reg, + .volatile_reg = fsl_esai_volatile_reg, .writeable_reg = fsl_esai_writeable_reg, + .cache_type = REGCACHE_RBTREE, }; static int fsl_esai_probe(struct platform_device *pdev) @@ -853,10 +900,51 @@ static const struct of_device_id fsl_esai_dt_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); +#ifdef CONFIG_PM_SLEEP +static int fsl_esai_suspend(struct device *dev) +{ + struct fsl_esai *esai = dev_get_drvdata(dev); + + regcache_cache_only(esai->regmap, true); + regcache_mark_dirty(esai->regmap); + + return 0; +} + +static int fsl_esai_resume(struct device *dev) +{ + struct fsl_esai *esai = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(esai->regmap, false); + + /* FIFO reset for safety */ + regmap_update_bits(esai->regmap, REG_ESAI_TFCR, + ESAI_xFCR_xFR, ESAI_xFCR_xFR); + regmap_update_bits(esai->regmap, REG_ESAI_RFCR, + ESAI_xFCR_xFR, ESAI_xFCR_xFR); + + ret = regcache_sync(esai->regmap); + if (ret) + return ret; + + /* FIFO reset done */ + regmap_update_bits(esai->regmap, REG_ESAI_TFCR, ESAI_xFCR_xFR, 0); + regmap_update_bits(esai->regmap, REG_ESAI_RFCR, ESAI_xFCR_xFR, 0); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_esai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_esai_suspend, fsl_esai_resume) +}; + static struct platform_driver fsl_esai_driver = { .probe = fsl_esai_probe, .driver = { .name = "fsl-esai-dai", + .pm = &fsl_esai_pm_ops, .of_match_table = fsl_esai_dt_ids, }, }; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92c4a85..a4435f5e3be9 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -27,13 +27,13 @@ #define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\ FSL_SAI_CSR_FEIE) -static u32 fsl_sai_rates[] = { +static const unsigned int fsl_sai_rates[] = { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000 }; -static struct snd_pcm_hw_constraint_list fsl_sai_rate_constraints = { +static const struct snd_pcm_hw_constraint_list fsl_sai_rate_constraints = { .count = ARRAY_SIZE(fsl_sai_rates), .list = fsl_sai_rates, }; @@ -637,6 +637,8 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_RCSR: case FSL_SAI_TFR: case FSL_SAI_RFR: case FSL_SAI_TDR: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_FLAT, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -801,11 +804,42 @@ static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,imx6sx-sai", }, { /* sentinel */ } }; +MODULE_DEVICE_TABLE(of, fsl_sai_ids); + +#ifdef CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + msleep(1); + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) +}; static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, .driver = { .name = "fsl-sai", + .pm = &fsl_sai_pm_ops, .of_match_table = fsl_sai_ids, }, }; diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index ab729f2426fe..3d59bb6719f2 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -108,6 +108,8 @@ struct fsl_spdif_priv { struct clk *sysclk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; + /* regcache for SRPC */ + u32 regcache_srpc; }; /* DPLL locked and lock loss interrupt handler */ @@ -300,6 +302,8 @@ static int spdif_softreset(struct fsl_spdif_priv *spdif_priv) struct regmap *regmap = spdif_priv->regmap; u32 val, cycle = 1000; + regcache_cache_bypass(regmap, true); + regmap_write(regmap, REG_SPDIF_SCR, SCR_SOFT_RESET); /* @@ -310,6 +314,10 @@ static int spdif_softreset(struct fsl_spdif_priv *spdif_priv) regmap_read(regmap, REG_SPDIF_SCR, &val); } while ((val & SCR_SOFT_RESET) && cycle--); + regcache_cache_bypass(regmap, false); + regcache_mark_dirty(regmap); + regcache_sync(regmap); + if (cycle) return 0; else @@ -997,6 +1005,14 @@ static const struct snd_soc_component_driver fsl_spdif_component = { }; /* FSL SPDIF REGMAP */ +static const struct reg_default fsl_spdif_reg_defaults[] = { + {0x0, 0x00000400}, + {0x4, 0x00000000}, + {0xc, 0x00000000}, + {0x34, 0x00000000}, + {0x38, 0x00000000}, + {0x50, 0x00020f00}, +}; static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) { @@ -1022,6 +1038,26 @@ static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) } } +static bool fsl_spdif_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_SPDIF_SRPC: + case REG_SPDIF_SIS: + case REG_SPDIF_SRL: + case REG_SPDIF_SRR: + case REG_SPDIF_SRCSH: + case REG_SPDIF_SRCSL: + case REG_SPDIF_SRU: + case REG_SPDIF_SRQ: + case REG_SPDIF_STL: + case REG_SPDIF_STR: + case REG_SPDIF_SRFM: + return true; + default: + return false; + } +} + static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -1047,8 +1083,12 @@ static const struct regmap_config fsl_spdif_regmap_config = { .val_bits = 32, .max_register = REG_SPDIF_STC, + .reg_defaults = fsl_spdif_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(fsl_spdif_reg_defaults), .readable_reg = fsl_spdif_readable_reg, + .volatile_reg = fsl_spdif_volatile_reg, .writeable_reg = fsl_spdif_writeable_reg, + .cache_type = REGCACHE_RBTREE, }; static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, @@ -1271,6 +1311,38 @@ static int fsl_spdif_probe(struct platform_device *pdev) return ret; } +#ifdef CONFIG_PM_SLEEP +static int fsl_spdif_suspend(struct device *dev) +{ + struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev); + + regmap_read(spdif_priv->regmap, REG_SPDIF_SRPC, + &spdif_priv->regcache_srpc); + + regcache_cache_only(spdif_priv->regmap, true); + regcache_mark_dirty(spdif_priv->regmap); + + return 0; +} + +static int fsl_spdif_resume(struct device *dev) +{ + struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev); + + regcache_cache_only(spdif_priv->regmap, false); + + regmap_update_bits(spdif_priv->regmap, REG_SPDIF_SRPC, + SRPC_CLKSRC_SEL_MASK | SRPC_GAINSEL_MASK, + spdif_priv->regcache_srpc); + + return regcache_sync(spdif_priv->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_spdif_pm = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_spdif_suspend, fsl_spdif_resume) +}; + static const struct of_device_id fsl_spdif_dt_ids[] = { { .compatible = "fsl,imx35-spdif", }, { .compatible = "fsl,vf610-spdif", }, @@ -1282,6 +1354,7 @@ static struct platform_driver fsl_spdif_driver = { .driver = { .name = "fsl-spdif-dai", .of_match_table = fsl_spdif_dt_ids, + .pm = &fsl_spdif_pm, }, .probe = fsl_spdif_probe, }; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 37c5cd4d0e59..95d2392303eb 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -111,12 +111,75 @@ struct fsl_ssi_rxtx_reg_val { struct fsl_ssi_reg_val rx; struct fsl_ssi_reg_val tx; }; + +static const struct reg_default fsl_ssi_reg_defaults[] = { + {0x10, 0x00000000}, + {0x18, 0x00003003}, + {0x1c, 0x00000200}, + {0x20, 0x00000200}, + {0x24, 0x00040000}, + {0x28, 0x00040000}, + {0x38, 0x00000000}, + {0x48, 0x00000000}, + {0x4c, 0x00000000}, + {0x54, 0x00000000}, + {0x58, 0x00000000}, +}; + +static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CCSR_SSI_SACCEN: + case CCSR_SSI_SACCDIS: + return false; + default: + return true; + } +} + +static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CCSR_SSI_STX0: + case CCSR_SSI_STX1: + case CCSR_SSI_SRX0: + case CCSR_SSI_SRX1: + case CCSR_SSI_SISR: + case CCSR_SSI_SFCSR: + case CCSR_SSI_SACADD: + case CCSR_SSI_SACDAT: + case CCSR_SSI_SATAG: + case CCSR_SSI_SACCST: + return true; + default: + return false; + } +} + +static bool fsl_ssi_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CCSR_SSI_SRX0: + case CCSR_SSI_SRX1: + case CCSR_SSI_SACCST: + return false; + default: + return true; + } +} + static const struct regmap_config fsl_ssi_regconfig = { .max_register = CCSR_SSI_SACCDIS, .reg_bits = 32, .val_bits = 32, .reg_stride = 4, .val_format_endian = REGMAP_ENDIAN_NATIVE, + .reg_defaults = fsl_ssi_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults), + .readable_reg = fsl_ssi_readable_reg, + .volatile_reg = fsl_ssi_volatile_reg, + .writeable_reg = fsl_ssi_writeable_reg, + .cache_type = REGCACHE_RBTREE, }; struct fsl_ssi_soc_data { @@ -176,6 +239,9 @@ struct fsl_ssi_private { unsigned int baudclk_streams; unsigned int bitclk_freq; + /*regcache for SFCSR*/ + u32 regcache_sfcsr; + /* DMA params */ struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -1514,10 +1580,46 @@ static int fsl_ssi_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int fsl_ssi_suspend(struct device *dev) +{ + struct fsl_ssi_private *ssi_private = dev_get_drvdata(dev); + struct regmap *regs = ssi_private->regs; + + regmap_read(regs, CCSR_SSI_SFCSR, + &ssi_private->regcache_sfcsr); + + regcache_cache_only(regs, true); + regcache_mark_dirty(regs); + + return 0; +} + +static int fsl_ssi_resume(struct device *dev) +{ + struct fsl_ssi_private *ssi_private = dev_get_drvdata(dev); + struct regmap *regs = ssi_private->regs; + + regcache_cache_only(regs, false); + + regmap_update_bits(regs, CCSR_SSI_SFCSR, + CCSR_SSI_SFCSR_RFWM1_MASK | CCSR_SSI_SFCSR_TFWM1_MASK | + CCSR_SSI_SFCSR_RFWM0_MASK | CCSR_SSI_SFCSR_TFWM0_MASK, + ssi_private->regcache_sfcsr); + + return regcache_sync(regs); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_ssi_pm = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_ssi_suspend, fsl_ssi_resume) +}; + static struct platform_driver fsl_ssi_driver = { .driver = { .name = "fsl-ssi-dai", .of_match_table = fsl_ssi_ids, + .pm = &fsl_ssi_pm, }, .probe = fsl_ssi_probe, .remove = fsl_ssi_remove, diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 33da26a12457..a407e833c612 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -89,6 +89,7 @@ MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); static struct platform_driver imx_spdif_driver = { .driver = { .name = "imx-spdif", + .pm = &snd_soc_pm_ops, .of_match_table = imx_spdif_dt_ids, }, .probe = imx_spdif_audio_probe, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3ff76d419436..54c33204541f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -151,7 +151,9 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, } if (set->slots) { - ret = snd_soc_dai_set_tdm_slot(dai, 0, 0, + ret = snd_soc_dai_set_tdm_slot(dai, + set->tx_slot_mask, + set->rx_slot_mask, set->slots, set->slot_width); if (ret && ret != -ENOTSUPP) { @@ -243,7 +245,9 @@ asoc_simple_card_sub_parse_of(struct device_node *np, return ret; /* Parse TDM slot */ - ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width); + ret = snd_soc_of_parse_tdm_slot(np, &dai->tx_slot_mask, + &dai->rx_slot_mask, + &dai->slots, &dai->slot_width); if (ret) return ret; diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 05fde5e6e257..7b778ab85f8b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -12,6 +12,7 @@ config SND_MFLD_MACHINE config SND_SST_MFLD_PLATFORM tristate + select SND_SOC_COMPRESS config SND_SST_IPC tristate @@ -138,4 +139,18 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH config SND_SOC_INTEL_SKYLAKE tristate select SND_HDA_EXT_CORE + select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST + +config SND_SOC_INTEL_SKL_RT286_MACH + tristate "ASoC Audio driver for SKL with RT286 I2S mode" + depends on X86 && ACPI + select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT286 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Skylake platforms + with RT286 I2S audio codec. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 683e50116152..0487cfaac538 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -368,23 +368,6 @@ static void sst_media_close(struct snd_pcm_substream *substream, kfree(stream); } -static inline unsigned int get_current_pipe_id(struct snd_soc_dai *dai, - struct snd_pcm_substream *substream) -{ - struct sst_data *sst = snd_soc_dai_get_drvdata(dai); - struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map; - struct sst_runtime_stream *stream = - substream->runtime->private_data; - u32 str_id = stream->stream_info.str_id; - unsigned int pipe_id; - - pipe_id = map[str_id].device_id; - - dev_dbg(dai->dev, "got pipe_id = %#x for str_id = %d\n", - pipe_id, str_id); - return pipe_id; -} - static int sst_media_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -529,7 +512,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { }, { .name = "compress-cpu-dai", - .compress_dai = 1, + .compress_new = snd_soc_new_compress, .ops = &sst_compr_dai_ops, .playback = { .stream_name = "Compress Playback", diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index cb94895c9edb..371c4565cad8 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o +snd-soc-skl_rt286-objs := skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -15,3 +16,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o +obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 8bafaf6ceab1..3f8a1e10bed0 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -266,18 +266,11 @@ static int broadwell_audio_probe(struct platform_device *pdev) { broadwell_rt286.dev = &pdev->dev; - return snd_soc_register_card(&broadwell_rt286); -} - -static int broadwell_audio_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&broadwell_rt286); - return 0; + return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286); } static struct platform_driver broadwell_audio = { .probe = broadwell_audio_probe, - .remove = broadwell_audio_remove, .driver = { .name = "broadwell-audio", }, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index c4453120b11a..7a5c9a36c1db 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -117,20 +117,10 @@ static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static unsigned int rates_48000[] = { - 48000, -}; - -static struct snd_pcm_hw_constraint_list constraints_48000 = { - .count = ARRAY_SIZE(rates_48000), - .list = rates_48000, -}; - static int byt_aif1_startup(struct snd_pcm_substream *substream) { - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_48000); + return snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 48000); } static struct snd_soc_ops byt_aif1_ops = { diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 49f4869cec48..4e2fcf188dd1 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -193,20 +193,10 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static unsigned int rates_48000[] = { - 48000, -}; - -static struct snd_pcm_hw_constraint_list constraints_48000 = { - .count = ARRAY_SIZE(rates_48000), - .list = rates_48000, -}; - static int cht_aif1_startup(struct snd_pcm_substream *substream) { - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_48000); + return snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 48000); } static int cht_max98090_headset_init(struct snd_soc_component *component) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 7be8461e4d3b..38d65a3529c4 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -235,20 +235,10 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static unsigned int rates_48000[] = { - 48000, -}; - -static struct snd_pcm_hw_constraint_list constraints_48000 = { - .count = ARRAY_SIZE(rates_48000), - .list = rates_48000, -}; - static int cht_aif1_startup(struct snd_pcm_substream *substream) { - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_48000); + return snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 48000); } static struct snd_soc_ops cht_aif1_ops = { diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 23fe04075142..5621ccd92992 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -222,20 +222,10 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static unsigned int rates_48000[] = { - 48000, -}; - -static struct snd_pcm_hw_constraint_list constraints_48000 = { - .count = ARRAY_SIZE(rates_48000), - .list = rates_48000, -}; - static int cht_aif1_startup(struct snd_pcm_substream *substream) { - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_48000); + return snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 48000); } static struct snd_soc_ops cht_aif1_ops = { diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c new file mode 100644 index 000000000000..a73a431bd8b7 --- /dev/null +++ b/sound/soc/intel/boards/skl_rt286.c @@ -0,0 +1,259 @@ +/* + * Intel Skylake I2S Machine Driver + * + * Copyright (C) 2014-2015, Intel Corporation. All rights reserved. + * + * Modified from: + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include "../../codecs/rt286.h" + +static struct snd_soc_jack skylake_headset; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin skylake_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new skylake_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget skylake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route skylake_rt286_map[] = { + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphone Jack", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC2"}, + {"DMIC AIF", NULL, "SoC DMIC"}, + + /* CODEC BE connections */ + { "AIF1 Playback", NULL, "ssp0 Tx"}, + { "ssp0 Tx", NULL, "codec0_out"}, + { "ssp0 Tx", NULL, "codec1_out"}, + + { "codec0_in", NULL, "ssp0 Rx" }, + { "codec1_in", NULL, "ssp0 Rx" }, + { "ssp0 Rx", NULL, "AIF1 Capture" }, + + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "Capture" }, + + { "hif1", NULL, "iDisp Tx"}, + { "iDisp Tx", NULL, "iDisp_out"}, + +}; + +static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &skylake_headset, + skylake_headset_pins, ARRAY_SIZE(skylake_headset_pins)); + + if (ret) + return ret; + + rt286_mic_detect(codec, &skylake_headset); + + return 0; +} + + +static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The output is 48KHz, stereo, 16bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static int skylake_rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops skylake_rt286_ops = { + .hw_params = skylake_rt286_hw_params, +}; + +/* skylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link skylake_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "Skl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_playback = 1, + }, + { + .name = "Skl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_capture = 1, + }, + { + .name = "Skl Audio Reference cap", + .stream_name = "refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .ignore_suspend = 1, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .be_id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .init = skylake_rt286_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = skylake_ssp0_fixup, + .ops = &skylake_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .be_id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +/* skylake audio machine driver for SPT + RT286S */ +static struct snd_soc_card skylake_rt286 = { + .name = "skylake-rt286", + .owner = THIS_MODULE, + .dai_link = skylake_rt286_dais, + .num_links = ARRAY_SIZE(skylake_rt286_dais), + .controls = skylake_controls, + .num_controls = ARRAY_SIZE(skylake_controls), + .dapm_widgets = skylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), + .dapm_routes = skylake_rt286_map, + .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), + .fully_routed = true, +}; + +static int skylake_audio_probe(struct platform_device *pdev) +{ + skylake_rt286.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286); +} + +static struct platform_driver skylake_audio = { + .probe = skylake_audio_probe, + .driver = { + .name = "skl_alc286s_i2s", + }, +}; + +module_platform_driver(skylake_audio) + +/* Module information */ +MODULE_AUTHOR("Omair Mohammed Abdullah <omair.m.abdullah@intel.com>"); +MODULE_DESCRIPTION("Intel SST Audio for Skylake"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:skl_alc286s_i2s"); diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index f24154ca4e98..d9105584c51f 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -1,7 +1,11 @@ -snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o +snd-soc-sst-dsp-objs := sst-dsp.o snd-soc-sst-acpi-objs := sst-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o +ifneq ($(CONFIG_DW_DMAC_CORE),) +snd-soc-sst-dsp-objs += sst-firmware.o +endif + obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o diff --git a/sound/soc/intel/common/sst-dsp-priv.h b/sound/soc/intel/common/sst-dsp-priv.h index cbd568eac033..2151652d37b7 100644 --- a/sound/soc/intel/common/sst-dsp-priv.h +++ b/sound/soc/intel/common/sst-dsp-priv.h @@ -314,6 +314,7 @@ struct sst_dsp { int sst_state; struct skl_cl_dev cl_dev; u32 intr_status; + const struct firmware *fw; }; /* Size optimised DRAM/IRAM memcpy */ diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index a627236dd1f5..c9452e02e0dd 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -420,6 +420,7 @@ void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes) } EXPORT_SYMBOL_GPL(sst_dsp_inbox_read); +#if IS_ENABLED(CONFIG_DW_DMAC_CORE) struct sst_dsp *sst_dsp_new(struct device *dev, struct sst_dsp_device *sst_dev, struct sst_pdata *pdata) { @@ -484,6 +485,7 @@ void sst_dsp_free(struct sst_dsp *sst) sst_dma_free(sst->dma); } EXPORT_SYMBOL_GPL(sst_dsp_free); +#endif /* Module information */ MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/intel/common/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h index 1f45f18715c0..859f0de00339 100644 --- a/sound/soc/intel/common/sst-dsp.h +++ b/sound/soc/intel/common/sst-dsp.h @@ -216,10 +216,12 @@ struct sst_pdata { void *dsp; }; +#if IS_ENABLED(CONFIG_DW_DMAC_CORE) /* Initialization */ struct sst_dsp *sst_dsp_new(struct device *dev, struct sst_dsp_device *sst_dev, struct sst_pdata *pdata); void sst_dsp_free(struct sst_dsp *sst); +#endif /* SHIM Read / Write */ void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value); diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index ebcca6dc48d1..1636a1eeb002 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -26,7 +26,6 @@ #include <linux/acpi.h> /* supported DMA engine drivers */ -#include <linux/platform_data/dma-dw.h> #include <linux/dma/dw.h> #include <asm/page.h> @@ -169,12 +168,6 @@ err: return ret; } -static struct dw_dma_platform_data dw_pdata = { - .is_private = 1, - .chan_allocation_order = CHAN_ALLOCATION_ASCENDING, - .chan_priority = CHAN_PRIORITY_ASCENDING, -}; - static struct dw_dma_chip *dw_probe(struct device *dev, struct resource *mem, int irq) { @@ -195,7 +188,8 @@ static struct dw_dma_chip *dw_probe(struct device *dev, struct resource *mem, return ERR_PTR(err); chip->dev = dev; - err = dw_dma_probe(chip, &dw_pdata); + + err = dw_dma_probe(chip, NULL); if (err) return ERR_PTR(err); diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index 27db22178204..914b6dab9bea 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -1,4 +1,5 @@ -snd-soc-skl-objs := skl.o skl-pcm.o skl-nhlt.o skl-messages.o +snd-soc-skl-objs := skl.o skl-pcm.o skl-nhlt.o skl-messages.o \ +skl-topology.o obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 826d4fd8930a..50a109503a3f 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -54,6 +54,24 @@ static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab) return 0; } +#define NOTIFICATION_PARAM_ID 3 +#define NOTIFICATION_MASK 0xf + +/* disable notfication for underruns/overruns from firmware module */ +static void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable) +{ + struct notification_mask mask; + struct skl_ipc_large_config_msg msg = {0}; + + mask.notify = NOTIFICATION_MASK; + mask.enable = enable; + + msg.large_param_id = NOTIFICATION_PARAM_ID; + msg.param_data_size = sizeof(mask); + + skl_ipc_set_large_config(&ctx->ipc, &msg, (u32 *)&mask); +} + int skl_init_dsp(struct skl *skl) { void __iomem *mmio_base; @@ -79,7 +97,10 @@ int skl_init_dsp(struct skl *skl) ret = skl_sst_dsp_init(bus->dev, mmio_base, irq, loader_ops, &skl->skl_sst); + if (ret < 0) + return ret; + skl_dsp_enable_notification(skl->skl_sst, false); dev_dbg(bus->dev, "dsp registration status=%d\n", ret); return ret; @@ -122,6 +143,7 @@ int skl_suspend_dsp(struct skl *skl) int skl_resume_dsp(struct skl *skl) { struct skl_sst *ctx = skl->skl_sst; + int ret; /* if ppcap is not supported return 0 */ if (!skl->ebus.ppcap) @@ -131,7 +153,12 @@ int skl_resume_dsp(struct skl *skl) snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true); snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true); - return skl_dsp_wake(ctx->dsp); + ret = skl_dsp_wake(ctx->dsp); + if (ret < 0) + return ret; + + skl_dsp_enable_notification(skl->skl_sst, false); + return ret; } enum skl_bitdepth skl_get_bit_depth(int params) @@ -294,6 +321,7 @@ static void skl_copy_copier_caps(struct skl_module_cfg *mconfig, (mconfig->formats_config.caps_size) / 4; } +#define SKL_NON_GATEWAY_CPR_NODE_ID 0xFFFFFFFF /* * Calculate the gatewat settings required for copier module, type of * gateway and index of gateway to use @@ -303,6 +331,7 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx, struct skl_cpr_cfg *cpr_mconfig) { union skl_connector_node_id node_id = {0}; + union skl_ssp_dma_node ssp_node = {0}; struct skl_pipe_params *params = mconfig->pipe->p_params; switch (mconfig->dev_type) { @@ -320,9 +349,9 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx, (SKL_CONN_SOURCE == mconfig->hw_conn_type) ? SKL_DMA_I2S_LINK_OUTPUT_CLASS : SKL_DMA_I2S_LINK_INPUT_CLASS; - node_id.node.vindex = params->host_dma_id + - (mconfig->time_slot << 1) + - (mconfig->vbus_id << 3); + ssp_node.dma_node.time_slot_index = mconfig->time_slot; + ssp_node.dma_node.i2s_instance = mconfig->vbus_id; + node_id.node.vindex = ssp_node.val; break; case SKL_DEVICE_DMIC: @@ -339,13 +368,18 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx, node_id.node.vindex = params->link_dma_id; break; - default: + case SKL_DEVICE_HDAHOST: node_id.node.dma_type = (SKL_CONN_SOURCE == mconfig->hw_conn_type) ? SKL_DMA_HDA_HOST_OUTPUT_CLASS : SKL_DMA_HDA_HOST_INPUT_CLASS; node_id.node.vindex = params->host_dma_id; break; + + default: + cpr_mconfig->gtw_cfg.node_id = SKL_NON_GATEWAY_CPR_NODE_ID; + cpr_mconfig->cpr_feature_mask = 0; + return; } cpr_mconfig->gtw_cfg.node_id = node_id.val; diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 13036b19d7e5..b0c7bd113aac 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -25,7 +25,7 @@ static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45, #define DSDT_NHLT_PATH "\\_SB.PCI0.HDAS" -void __iomem *skl_nhlt_init(struct device *dev) +void *skl_nhlt_init(struct device *dev) { acpi_handle handle; union acpi_object *obj; @@ -40,17 +40,17 @@ void __iomem *skl_nhlt_init(struct device *dev) if (obj && obj->type == ACPI_TYPE_BUFFER) { nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; - return ioremap_cache(nhlt_ptr->min_addr, nhlt_ptr->length); + return memremap(nhlt_ptr->min_addr, nhlt_ptr->length, + MEMREMAP_WB); } dev_err(dev, "device specific method to extract NHLT blob failed\n"); return NULL; } -void skl_nhlt_free(void __iomem *addr) +void skl_nhlt_free(void *addr) { - iounmap(addr); - addr = NULL; + memunmap(addr); } static struct nhlt_specific_cfg *skl_get_specific_cfg( diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 7d617bf493bc..a2f94ce1679d 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -24,6 +24,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include "skl.h" +#include "skl-topology.h" #define HDA_MONO 1 #define HDA_STEREO 2 @@ -115,7 +116,7 @@ static int skl_pcm_open(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); ret = pm_runtime_get_sync(dai->dev); - if (ret) + if (ret < 0) return ret; stream = snd_hdac_ext_stream_assign(ebus, substream, @@ -214,6 +215,8 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream, struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct snd_pcm_runtime *runtime = substream->runtime; + struct skl_pipe_params p_params = {0}; + struct skl_module_cfg *m_cfg; int ret, dma_id; dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); @@ -228,6 +231,16 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream, dma_id = hdac_stream(stream)->stream_tag - 1; dev_dbg(dai->dev, "dma_id=%d\n", dma_id); + p_params.s_fmt = snd_pcm_format_width(params_format(params)); + p_params.ch = params_channels(params); + p_params.s_freq = params_rate(params); + p_params.host_dma_id = dma_id; + p_params.stream = substream->stream; + + m_cfg = skl_tplg_fe_get_cpr_module(dai, p_params.stream); + if (m_cfg) + skl_tplg_update_pipe_params(dai->dev, m_cfg, &p_params); + return 0; } @@ -268,6 +281,46 @@ static int skl_pcm_hw_free(struct snd_pcm_substream *substream, return skl_substream_free_pages(ebus_to_hbus(ebus), substream); } +static int skl_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct skl_pipe_params p_params = {0}; + + p_params.s_fmt = snd_pcm_format_width(params_format(params)); + p_params.ch = params_channels(params); + p_params.s_freq = params_rate(params); + p_params.stream = substream->stream; + skl_tplg_be_update_params(dai, &p_params); + + return 0; +} + +static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct skl *skl = get_skl_ctx(dai->dev); + struct skl_sst *ctx = skl->skl_sst; + struct skl_module_cfg *mconfig; + + mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); + if (!mconfig) + return -EIO; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + return skl_run_pipe(ctx, mconfig->pipe); + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + return skl_stop_pipe(ctx, mconfig->pipe); + + default: + return 0; + } +} + static int skl_link_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -277,9 +330,8 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct skl_dma_params *dma_params; struct snd_soc_dai *codec_dai = rtd->codec_dai; - int dma_id; + struct skl_pipe_params p_params = {0}; - pr_debug("%s\n", __func__); link_dev = snd_hdac_ext_stream_assign(ebus, substream, HDAC_EXT_STREAM_TYPE_LINK); if (!link_dev) @@ -293,7 +345,14 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, if (dma_params) dma_params->stream_tag = hdac_stream(link_dev)->stream_tag; snd_soc_dai_set_dma_data(codec_dai, substream, (void *)dma_params); - dma_id = hdac_stream(link_dev)->stream_tag - 1; + + p_params.s_fmt = snd_pcm_format_width(params_format(params)); + p_params.ch = params_channels(params); + p_params.s_freq = params_rate(params); + p_params.stream = substream->stream; + p_params.link_dma_id = hdac_stream(link_dev)->stream_tag - 1; + + skl_tplg_be_update_params(dai, &p_params); return 0; } @@ -308,27 +367,12 @@ static int skl_link_pcm_prepare(struct snd_pcm_substream *substream, unsigned int format_val = 0; struct skl_dma_params *dma_params; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_pcm_hw_params *params; - struct snd_interval *channels, *rate; struct hdac_ext_link *link; - dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); if (link_dev->link_prepared) { dev_dbg(dai->dev, "already stream is prepared - returning\n"); return 0; } - params = devm_kzalloc(dai->dev, sizeof(*params), GFP_KERNEL); - if (params == NULL) - return -ENOMEM; - - channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - channels->min = channels->max = substream->runtime->channels; - rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - rate->min = rate->max = substream->runtime->rate; - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - substream->runtime->format); - dma_params = (struct skl_dma_params *) snd_soc_dai_get_dma_data(codec_dai, substream); @@ -399,13 +443,13 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream, return 0; } -static int skl_hda_be_startup(struct snd_pcm_substream *substream, +static int skl_be_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { return pm_runtime_get_sync(dai->dev); } -static void skl_hda_be_shutdown(struct snd_pcm_substream *substream, +static void skl_be_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { pm_runtime_mark_last_busy(dai->dev); @@ -418,20 +462,28 @@ static struct snd_soc_dai_ops skl_pcm_dai_ops = { .prepare = skl_pcm_prepare, .hw_params = skl_pcm_hw_params, .hw_free = skl_pcm_hw_free, + .trigger = skl_pcm_trigger, }; static struct snd_soc_dai_ops skl_dmic_dai_ops = { - .startup = skl_hda_be_startup, - .shutdown = skl_hda_be_shutdown, + .startup = skl_be_startup, + .hw_params = skl_be_hw_params, + .shutdown = skl_be_shutdown, +}; + +static struct snd_soc_dai_ops skl_be_ssp_dai_ops = { + .startup = skl_be_startup, + .hw_params = skl_be_hw_params, + .shutdown = skl_be_shutdown, }; static struct snd_soc_dai_ops skl_link_dai_ops = { - .startup = skl_hda_be_startup, + .startup = skl_be_startup, .prepare = skl_link_pcm_prepare, .hw_params = skl_link_hw_params, .hw_free = skl_link_hw_free, .trigger = skl_link_pcm_trigger, - .shutdown = skl_hda_be_shutdown, + .shutdown = skl_be_shutdown, }; static struct snd_soc_dai_driver skl_platform_dai[] = { @@ -488,6 +540,24 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, /* BE CPU Dais */ { + .name = "SSP0 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp0 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp0 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ .name = "iDisp Pin", .ops = &skl_link_dai_ops, .playback = { @@ -510,17 +580,6 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }, { - .name = "DMIC23 Pin", - .ops = &skl_dmic_dai_ops, - .capture = { - .stream_name = "DMIC23 Rx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ .name = "HD-Codec Pin", .ops = &skl_link_dai_ops, .playback = { @@ -538,28 +597,6 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, -{ - .name = "HD-Codec-SPK Pin", - .ops = &skl_link_dai_ops, - .playback = { - .stream_name = "HD-Codec-SPK Tx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, -{ - .name = "HD-Codec-AMIC Pin", - .ops = &skl_link_dai_ops, - .capture = { - .stream_name = "HD-Codec-AMIC Rx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, }; static int skl_platform_open(struct snd_pcm_substream *substream) @@ -577,7 +614,7 @@ static int skl_platform_open(struct snd_pcm_substream *substream) return 0; } -static int skl_pcm_trigger(struct snd_pcm_substream *substream, +static int skl_coupled_trigger(struct snd_pcm_substream *substream, int cmd) { struct hdac_ext_bus *ebus = get_bus_ctx(substream); @@ -651,7 +688,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, return 0; } -static int skl_dsp_trigger(struct snd_pcm_substream *substream, +static int skl_decoupled_trigger(struct snd_pcm_substream *substream, int cmd) { struct hdac_ext_bus *ebus = get_bus_ctx(substream); @@ -708,9 +745,9 @@ static int skl_platform_pcm_trigger(struct snd_pcm_substream *substream, struct hdac_ext_bus *ebus = get_bus_ctx(substream); if (ebus->ppcap) - return skl_dsp_trigger(substream, cmd); + return skl_decoupled_trigger(substream, cmd); else - return skl_pcm_trigger(substream, cmd); + return skl_coupled_trigger(substream, cmd); } /* calculate runtime delay from LPIB */ @@ -877,7 +914,17 @@ static int skl_pcm_new(struct snd_soc_pcm_runtime *rtd) return retval; } +static int skl_platform_soc_probe(struct snd_soc_platform *platform) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(platform->dev); + + if (ebus->ppcap) + return skl_tplg_init(platform, ebus); + + return 0; +} static struct snd_soc_platform_driver skl_platform_drv = { + .probe = skl_platform_soc_probe, .ops = &skl_platform_ops, .pcm_new = skl_pcm_new, .pcm_free = skl_pcm_free, @@ -890,6 +937,11 @@ static const struct snd_soc_component_driver skl_component = { int skl_platform_register(struct device *dev) { int ret; + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct skl *skl = ebus_to_skl(ebus); + + INIT_LIST_HEAD(&skl->ppl_list); + INIT_LIST_HEAD(&skl->dapm_path_list); ret = snd_soc_register_platform(dev, &skl_platform_drv); if (ret) { diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c index 94875b008b0b..1bfb7f63b572 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.c +++ b/sound/soc/intel/skylake/skl-sst-dsp.c @@ -175,7 +175,7 @@ static int skl_dsp_core_power_down(struct sst_dsp *ctx) /* poll with timeout to check if operation successful */ return sst_dsp_register_poll(ctx, SKL_ADSP_REG_ADSPCS, - SKL_ADSPCS_SPA_MASK, + SKL_ADSPCS_CPA_MASK, 0, SKL_DSP_PD_TO, "Power down"); @@ -262,6 +262,11 @@ irqreturn_t skl_dsp_sst_interrupt(int irq, void *dev_id) val = sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPIS); ctx->intr_status = val; + if (val == 0xffffffff) { + spin_unlock(&ctx->spinlock); + return IRQ_NONE; + } + if (val & SKL_ADSPIS_IPC) { skl_ipc_int_disable(ctx); result = IRQ_WAKE_THREAD; diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 937a0a3a63a0..3345ea0d4414 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -464,6 +464,18 @@ void skl_ipc_op_int_enable(struct sst_dsp *ctx) SKL_ADSP_REG_HIPCCTL_BUSY, SKL_ADSP_REG_HIPCCTL_BUSY); } +void skl_ipc_op_int_disable(struct sst_dsp *ctx) +{ + /* disable IPC DONE interrupt */ + sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_HIPCCTL, + SKL_ADSP_REG_HIPCCTL_DONE, 0); + + /* Disable IPC BUSY interrupt */ + sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_HIPCCTL, + SKL_ADSP_REG_HIPCCTL_BUSY, 0); + +} + bool skl_ipc_int_status(struct sst_dsp *ctx) { return sst_dsp_shim_read_unlocked(ctx, diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h index 9f5f67202858..f1a154e45dc3 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -116,6 +116,7 @@ int skl_ipc_set_large_config(struct sst_generic_ipc *ipc, void skl_ipc_int_enable(struct sst_dsp *dsp); void skl_ipc_op_int_enable(struct sst_dsp *ctx); +void skl_ipc_op_int_disable(struct sst_dsp *ctx); void skl_ipc_int_disable(struct sst_dsp *dsp); bool skl_ipc_int_status(struct sst_dsp *dsp); diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index c18ea51b7484..3b83dc99f1d4 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -70,15 +70,31 @@ static int skl_transfer_firmware(struct sst_dsp *ctx, static int skl_load_base_firmware(struct sst_dsp *ctx) { int ret = 0, i; - const struct firmware *fw = NULL; struct skl_sst *skl = ctx->thread_context; u32 reg; - ret = request_firmware(&fw, "dsp_fw_release.bin", ctx->dev); + skl->boot_complete = false; + init_waitqueue_head(&skl->boot_wait); + + if (ctx->fw == NULL) { + ret = request_firmware(&ctx->fw, "dsp_fw_release.bin", ctx->dev); + if (ret < 0) { + dev_err(ctx->dev, "Request firmware failed %d\n", ret); + skl_dsp_disable_core(ctx); + return -EIO; + } + } + + ret = skl_dsp_boot(ctx); if (ret < 0) { - dev_err(ctx->dev, "Request firmware failed %d\n", ret); - skl_dsp_disable_core(ctx); - return -EIO; + dev_err(ctx->dev, "Boot dsp core failed ret: %d", ret); + goto skl_load_base_firmware_failed; + } + + ret = skl_cldma_prepare(ctx); + if (ret < 0) { + dev_err(ctx->dev, "CL dma prepare failed : %d", ret); + goto skl_load_base_firmware_failed; } /* enable Interrupt */ @@ -102,7 +118,7 @@ static int skl_load_base_firmware(struct sst_dsp *ctx) goto skl_load_base_firmware_failed; } - ret = skl_transfer_firmware(ctx, fw->data, fw->size); + ret = skl_transfer_firmware(ctx, ctx->fw->data, ctx->fw->size); if (ret < 0) { dev_err(ctx->dev, "Transfer firmware failed%d\n", ret); goto skl_load_base_firmware_failed; @@ -118,13 +134,12 @@ static int skl_load_base_firmware(struct sst_dsp *ctx) dev_dbg(ctx->dev, "Download firmware successful%d\n", ret); skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING); } - release_firmware(fw); - return 0; skl_load_base_firmware_failed: skl_dsp_disable_core(ctx); - release_firmware(fw); + release_firmware(ctx->fw); + ctx->fw = NULL; return ret; } @@ -172,6 +187,12 @@ static int skl_set_dsp_D3(struct sst_dsp *ctx) } skl_dsp_set_state_locked(ctx, SKL_DSP_RESET); + /* disable Interrupt */ + ctx->cl_dev.ops.cl_cleanup_controller(ctx); + skl_cldma_int_disable(ctx); + skl_ipc_op_int_disable(ctx); + skl_ipc_int_disable(ctx); + return ret; } @@ -235,22 +256,6 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, if (ret) return ret; - skl->boot_complete = false; - init_waitqueue_head(&skl->boot_wait); - - ret = skl_dsp_boot(sst); - if (ret < 0) { - dev_err(skl->dev, "Boot dsp core failed ret: %d", ret); - goto free_ipc; - } - - ret = skl_cldma_prepare(sst); - if (ret < 0) { - dev_err(dev, "CL dma prepare failed : %d", ret); - goto free_ipc; - } - - ret = sst->fw_ops.load_fw(sst); if (ret < 0) { dev_err(dev, "Load base fw failed : %d", ret); @@ -262,7 +267,6 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, return 0; -free_ipc: skl_ipc_free(&skl->ipc); return ret; } diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c new file mode 100644 index 000000000000..a7854c8fc523 --- /dev/null +++ b/sound/soc/intel/skylake/skl-topology.c @@ -0,0 +1,1252 @@ +/* + * skl-topology.c - Implements Platform component ALSA controls/widget + * handlers. + * + * Copyright (C) 2014-2015 Intel Corp + * Author: Jeeja KP <jeeja.kp@intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/slab.h> +#include <linux/types.h> +#include <linux/firmware.h> +#include <sound/soc.h> +#include <sound/soc-topology.h> +#include "skl-sst-dsp.h" +#include "skl-sst-ipc.h" +#include "skl-topology.h" +#include "skl.h" +#include "skl-tplg-interface.h" + +#define SKL_CH_FIXUP_MASK (1 << 0) +#define SKL_RATE_FIXUP_MASK (1 << 1) +#define SKL_FMT_FIXUP_MASK (1 << 2) + +/* + * SKL DSP driver modelling uses only few DAPM widgets so for rest we will + * ignore. This helpers checks if the SKL driver handles this widget type + */ +static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w) +{ + switch (w->id) { + case snd_soc_dapm_dai_link: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: + case snd_soc_dapm_dai_out: + case snd_soc_dapm_switch: + return false; + default: + return true; + } +} + +/* + * Each pipelines needs memory to be allocated. Check if we have free memory + * from available pool. Then only add this to pool + * This is freed when pipe is deleted + * Note: DSP does actual memory management we only keep track for complete + * pool + */ +static bool skl_tplg_alloc_pipe_mem(struct skl *skl, + struct skl_module_cfg *mconfig) +{ + struct skl_sst *ctx = skl->skl_sst; + + if (skl->resource.mem + mconfig->pipe->memory_pages > + skl->resource.max_mem) { + dev_err(ctx->dev, + "%s: module_id %d instance %d\n", __func__, + mconfig->id.module_id, + mconfig->id.instance_id); + dev_err(ctx->dev, + "exceeds ppl memory available %d mem %d\n", + skl->resource.max_mem, skl->resource.mem); + return false; + } + + skl->resource.mem += mconfig->pipe->memory_pages; + return true; +} + +/* + * Pipeline needs needs DSP CPU resources for computation, this is + * quantified in MCPS (Million Clocks Per Second) required for module/pipe + * + * Each pipelines needs mcps to be allocated. Check if we have mcps for this + * pipe. This adds the mcps to driver counter + * This is removed on pipeline delete + */ +static bool skl_tplg_alloc_pipe_mcps(struct skl *skl, + struct skl_module_cfg *mconfig) +{ + struct skl_sst *ctx = skl->skl_sst; + + if (skl->resource.mcps + mconfig->mcps > skl->resource.max_mcps) { + dev_err(ctx->dev, + "%s: module_id %d instance %d\n", __func__, + mconfig->id.module_id, mconfig->id.instance_id); + dev_err(ctx->dev, + "exceeds ppl memory available %d > mem %d\n", + skl->resource.max_mcps, skl->resource.mcps); + return false; + } + + skl->resource.mcps += mconfig->mcps; + return true; +} + +/* + * Free the mcps when tearing down + */ +static void +skl_tplg_free_pipe_mcps(struct skl *skl, struct skl_module_cfg *mconfig) +{ + skl->resource.mcps -= mconfig->mcps; +} + +/* + * Free the memory when tearing down + */ +static void +skl_tplg_free_pipe_mem(struct skl *skl, struct skl_module_cfg *mconfig) +{ + skl->resource.mem -= mconfig->pipe->memory_pages; +} + + +static void skl_dump_mconfig(struct skl_sst *ctx, + struct skl_module_cfg *mcfg) +{ + dev_dbg(ctx->dev, "Dumping config\n"); + dev_dbg(ctx->dev, "Input Format:\n"); + dev_dbg(ctx->dev, "channels = %d\n", mcfg->in_fmt.channels); + dev_dbg(ctx->dev, "s_freq = %d\n", mcfg->in_fmt.s_freq); + dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->in_fmt.ch_cfg); + dev_dbg(ctx->dev, "valid bit depth = %d\n", + mcfg->in_fmt.valid_bit_depth); + dev_dbg(ctx->dev, "Output Format:\n"); + dev_dbg(ctx->dev, "channels = %d\n", mcfg->out_fmt.channels); + dev_dbg(ctx->dev, "s_freq = %d\n", mcfg->out_fmt.s_freq); + dev_dbg(ctx->dev, "valid bit depth = %d\n", + mcfg->out_fmt.valid_bit_depth); + dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->out_fmt.ch_cfg); +} + +static void skl_tplg_update_params(struct skl_module_fmt *fmt, + struct skl_pipe_params *params, int fixup) +{ + if (fixup & SKL_RATE_FIXUP_MASK) + fmt->s_freq = params->s_freq; + if (fixup & SKL_CH_FIXUP_MASK) + fmt->channels = params->ch; + if (fixup & SKL_FMT_FIXUP_MASK) + fmt->valid_bit_depth = params->s_fmt; +} + +/* + * A pipeline may have modules which impact the pcm parameters, like SRC, + * channel converter, format converter. + * We need to calculate the output params by applying the 'fixup' + * Topology will tell driver which type of fixup is to be applied by + * supplying the fixup mask, so based on that we calculate the output + * + * Now In FE the pcm hw_params is source/target format. Same is applicable + * for BE with its hw_params invoked. + * here based on FE, BE pipeline and direction we calculate the input and + * outfix and then apply that for a module + */ +static void skl_tplg_update_params_fixup(struct skl_module_cfg *m_cfg, + struct skl_pipe_params *params, bool is_fe) +{ + int in_fixup, out_fixup; + struct skl_module_fmt *in_fmt, *out_fmt; + + in_fmt = &m_cfg->in_fmt; + out_fmt = &m_cfg->out_fmt; + + if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (is_fe) { + in_fixup = m_cfg->params_fixup; + out_fixup = (~m_cfg->converter) & + m_cfg->params_fixup; + } else { + out_fixup = m_cfg->params_fixup; + in_fixup = (~m_cfg->converter) & + m_cfg->params_fixup; + } + } else { + if (is_fe) { + out_fixup = m_cfg->params_fixup; + in_fixup = (~m_cfg->converter) & + m_cfg->params_fixup; + } else { + in_fixup = m_cfg->params_fixup; + out_fixup = (~m_cfg->converter) & + m_cfg->params_fixup; + } + } + + skl_tplg_update_params(in_fmt, params, in_fixup); + skl_tplg_update_params(out_fmt, params, out_fixup); +} + +/* + * A module needs input and output buffers, which are dependent upon pcm + * params, so once we have calculate params, we need buffer calculation as + * well. + */ +static void skl_tplg_update_buffer_size(struct skl_sst *ctx, + struct skl_module_cfg *mcfg) +{ + int multiplier = 1; + + if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT) + multiplier = 5; + + mcfg->ibs = (mcfg->in_fmt.s_freq / 1000) * + (mcfg->in_fmt.channels) * + (mcfg->in_fmt.bit_depth >> 3) * + multiplier; + + mcfg->obs = (mcfg->out_fmt.s_freq / 1000) * + (mcfg->out_fmt.channels) * + (mcfg->out_fmt.bit_depth >> 3) * + multiplier; +} + +static void skl_tplg_update_module_params(struct snd_soc_dapm_widget *w, + struct skl_sst *ctx) +{ + struct skl_module_cfg *m_cfg = w->priv; + struct skl_pipe_params *params = m_cfg->pipe->p_params; + int p_conn_type = m_cfg->pipe->conn_type; + bool is_fe; + + if (!m_cfg->params_fixup) + return; + + dev_dbg(ctx->dev, "Mconfig for widget=%s BEFORE updation\n", + w->name); + + skl_dump_mconfig(ctx, m_cfg); + + if (p_conn_type == SKL_PIPE_CONN_TYPE_FE) + is_fe = true; + else + is_fe = false; + + skl_tplg_update_params_fixup(m_cfg, params, is_fe); + skl_tplg_update_buffer_size(ctx, m_cfg); + + dev_dbg(ctx->dev, "Mconfig for widget=%s AFTER updation\n", + w->name); + + skl_dump_mconfig(ctx, m_cfg); +} + +/* + * A pipe can have multiple modules, each of them will be a DAPM widget as + * well. While managing a pipeline we need to get the list of all the + * widgets in a pipelines, so this helper - skl_tplg_get_pipe_widget() helps + * to get the SKL type widgets in that pipeline + */ +static int skl_tplg_alloc_pipe_widget(struct device *dev, + struct snd_soc_dapm_widget *w, struct skl_pipe *pipe) +{ + struct skl_module_cfg *src_module = NULL; + struct snd_soc_dapm_path *p = NULL; + struct skl_pipe_module *p_module = NULL; + + p_module = devm_kzalloc(dev, sizeof(*p_module), GFP_KERNEL); + if (!p_module) + return -ENOMEM; + + p_module->w = w; + list_add_tail(&p_module->node, &pipe->w_list); + + snd_soc_dapm_widget_for_each_sink_path(w, p) { + if ((p->sink->priv == NULL) + && (!is_skl_dsp_widget_type(w))) + continue; + + if ((p->sink->priv != NULL) && p->connect + && is_skl_dsp_widget_type(p->sink)) { + + src_module = p->sink->priv; + if (pipe->ppl_id == src_module->pipe->ppl_id) + skl_tplg_alloc_pipe_widget(dev, + p->sink, pipe); + } + } + return 0; +} + +/* + * Inside a pipe instance, we can have various modules. These modules need + * to instantiated in DSP by invoking INIT_MODULE IPC, which is achieved by + * skl_init_module() routine, so invoke that for all modules in a pipeline + */ +static int +skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) +{ + struct skl_pipe_module *w_module; + struct snd_soc_dapm_widget *w; + struct skl_module_cfg *mconfig; + struct skl_sst *ctx = skl->skl_sst; + int ret = 0; + + list_for_each_entry(w_module, &pipe->w_list, node) { + w = w_module->w; + mconfig = w->priv; + + /* check resource available */ + if (!skl_tplg_alloc_pipe_mcps(skl, mconfig)) + return -ENOMEM; + + /* + * apply fix/conversion to module params based on + * FE/BE params + */ + skl_tplg_update_module_params(w, ctx); + ret = skl_init_module(ctx, mconfig, NULL); + if (ret < 0) + return ret; + } + + return 0; +} + +/* + * Mixer module represents a pipeline. So in the Pre-PMU event of mixer we + * need create the pipeline. So we do following: + * - check the resources + * - Create the pipeline + * - Initialize the modules in pipeline + * - finally bind all modules together + */ +static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w, + struct skl *skl) +{ + int ret; + struct skl_module_cfg *mconfig = w->priv; + struct skl_pipe_module *w_module; + struct skl_pipe *s_pipe = mconfig->pipe; + struct skl_module_cfg *src_module = NULL, *dst_module; + struct skl_sst *ctx = skl->skl_sst; + + /* check resource available */ + if (!skl_tplg_alloc_pipe_mcps(skl, mconfig)) + return -EBUSY; + + if (!skl_tplg_alloc_pipe_mem(skl, mconfig)) + return -ENOMEM; + + /* + * Create a list of modules for pipe. + * This list contains modules from source to sink + */ + ret = skl_create_pipeline(ctx, mconfig->pipe); + if (ret < 0) + return ret; + + /* + * we create a w_list of all widgets in that pipe. This list is not + * freed on PMD event as widgets within a pipe are static. This + * saves us cycles to get widgets in pipe every time. + * + * So if we have already initialized all the widgets of a pipeline + * we skip, so check for list_empty and create the list if empty + */ + if (list_empty(&s_pipe->w_list)) { + ret = skl_tplg_alloc_pipe_widget(ctx->dev, w, s_pipe); + if (ret < 0) + return ret; + } + + /* Init all pipe modules from source to sink */ + ret = skl_tplg_init_pipe_modules(skl, s_pipe); + if (ret < 0) + return ret; + + /* Bind modules from source to sink */ + list_for_each_entry(w_module, &s_pipe->w_list, node) { + dst_module = w_module->w->priv; + + if (src_module == NULL) { + src_module = dst_module; + continue; + } + + ret = skl_bind_modules(ctx, src_module, dst_module); + if (ret < 0) + return ret; + + src_module = dst_module; + } + + return 0; +} + +/* + * A PGA represents a module in a pipeline. So in the Pre-PMU event of PGA + * we need to do following: + * - Bind to sink pipeline + * Since the sink pipes can be running and we don't get mixer event on + * connect for already running mixer, we need to find the sink pipes + * here and bind to them. This way dynamic connect works. + * - Start sink pipeline, if not running + * - Then run current pipe + */ +static int skl_tplg_pga_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w, + struct skl *skl) +{ + struct snd_soc_dapm_path *p; + struct skl_dapm_path_list *path_list; + struct snd_soc_dapm_widget *source, *sink; + struct skl_module_cfg *src_mconfig, *sink_mconfig; + struct skl_sst *ctx = skl->skl_sst; + int ret = 0; + + source = w; + src_mconfig = source->priv; + + /* + * find which sink it is connected to, bind with the sink, + * if sink is not started, start sink pipe first, then start + * this pipe + */ + snd_soc_dapm_widget_for_each_source_path(w, p) { + if (!p->connect) + continue; + + dev_dbg(ctx->dev, "%s: src widget=%s\n", __func__, w->name); + dev_dbg(ctx->dev, "%s: sink widget=%s\n", __func__, p->sink->name); + + /* + * here we will check widgets in sink pipelines, so that + * can be any widgets type and we are only interested if + * they are ones used for SKL so check that first + */ + if ((p->sink->priv != NULL) && + is_skl_dsp_widget_type(p->sink)) { + + sink = p->sink; + src_mconfig = source->priv; + sink_mconfig = sink->priv; + + /* Bind source to sink, mixin is always source */ + ret = skl_bind_modules(ctx, src_mconfig, sink_mconfig); + if (ret) + return ret; + + /* Start sinks pipe first */ + if (sink_mconfig->pipe->state != SKL_PIPE_STARTED) { + ret = skl_run_pipe(ctx, sink_mconfig->pipe); + if (ret) + return ret; + } + + path_list = kzalloc( + sizeof(struct skl_dapm_path_list), + GFP_KERNEL); + if (path_list == NULL) + return -ENOMEM; + + /* Add connected path to one global list */ + path_list->dapm_path = p; + list_add_tail(&path_list->node, &skl->dapm_path_list); + break; + } + } + + /* Start source pipe last after starting all sinks */ + ret = skl_run_pipe(ctx, src_mconfig->pipe); + if (ret) + return ret; + + return 0; +} + +/* + * in the Post-PMU event of mixer we need to do following: + * - Check if this pipe is running + * - if not, then + * - bind this pipeline to its source pipeline + * if source pipe is already running, this means it is a dynamic + * connection and we need to bind only to that pipe + * - start this pipeline + */ +static int skl_tplg_mixer_dapm_post_pmu_event(struct snd_soc_dapm_widget *w, + struct skl *skl) +{ + int ret = 0; + struct snd_soc_dapm_path *p; + struct snd_soc_dapm_widget *source, *sink; + struct skl_module_cfg *src_mconfig, *sink_mconfig; + struct skl_sst *ctx = skl->skl_sst; + int src_pipe_started = 0; + + sink = w; + sink_mconfig = sink->priv; + + /* + * If source pipe is already started, that means source is driving + * one more sink before this sink got connected, Since source is + * started, bind this sink to source and start this pipe. + */ + snd_soc_dapm_widget_for_each_sink_path(w, p) { + if (!p->connect) + continue; + + dev_dbg(ctx->dev, "sink widget=%s\n", w->name); + dev_dbg(ctx->dev, "src widget=%s\n", p->source->name); + + /* + * here we will check widgets in sink pipelines, so that + * can be any widgets type and we are only interested if + * they are ones used for SKL so check that first + */ + if ((p->source->priv != NULL) && + is_skl_dsp_widget_type(p->source)) { + source = p->source; + src_mconfig = source->priv; + sink_mconfig = sink->priv; + src_pipe_started = 1; + + /* + * check pipe state, then no need to bind or start + * the pipe + */ + if (src_mconfig->pipe->state != SKL_PIPE_STARTED) + src_pipe_started = 0; + } + } + + if (src_pipe_started) { + ret = skl_bind_modules(ctx, src_mconfig, sink_mconfig); + if (ret) + return ret; + + ret = skl_run_pipe(ctx, sink_mconfig->pipe); + } + + return ret; +} + +/* + * in the Pre-PMD event of mixer we need to do following: + * - Stop the pipe + * - find the source connections and remove that from dapm_path_list + * - unbind with source pipelines if still connected + */ +static int skl_tplg_mixer_dapm_pre_pmd_event(struct snd_soc_dapm_widget *w, + struct skl *skl) +{ + struct snd_soc_dapm_widget *source, *sink; + struct skl_module_cfg *src_mconfig, *sink_mconfig; + int ret = 0, path_found = 0; + struct skl_dapm_path_list *path_list, *tmp_list; + struct skl_sst *ctx = skl->skl_sst; + + sink = w; + sink_mconfig = sink->priv; + + /* Stop the pipe */ + ret = skl_stop_pipe(ctx, sink_mconfig->pipe); + if (ret) + return ret; + + /* + * This list, dapm_path_list handling here does not need any locks + * as we are under dapm lock while handling widget events. + * List can be manipulated safely only under dapm widgets handler + * routines + */ + list_for_each_entry_safe(path_list, tmp_list, + &skl->dapm_path_list, node) { + if (path_list->dapm_path->sink == sink) { + dev_dbg(ctx->dev, "Path found = %s\n", + path_list->dapm_path->name); + source = path_list->dapm_path->source; + src_mconfig = source->priv; + path_found = 1; + + list_del(&path_list->node); + kfree(path_list); + break; + } + } + + /* + * If path_found == 1, that means pmd for source pipe has + * not occurred, source is connected to some other sink. + * so its responsibility of sink to unbind itself from source. + */ + if (path_found) { + ret = skl_stop_pipe(ctx, src_mconfig->pipe); + if (ret < 0) + return ret; + + ret = skl_unbind_modules(ctx, src_mconfig, sink_mconfig); + } + + return ret; +} + +/* + * in the Post-PMD event of mixer we need to do following: + * - Free the mcps used + * - Free the mem used + * - Unbind the modules within the pipeline + * - Delete the pipeline (modules are not required to be explicitly + * deleted, pipeline delete is enough here + */ +static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w, + struct skl *skl) +{ + struct skl_module_cfg *mconfig = w->priv; + struct skl_pipe_module *w_module; + struct skl_module_cfg *src_module = NULL, *dst_module; + struct skl_sst *ctx = skl->skl_sst; + struct skl_pipe *s_pipe = mconfig->pipe; + int ret = 0; + + skl_tplg_free_pipe_mcps(skl, mconfig); + + list_for_each_entry(w_module, &s_pipe->w_list, node) { + dst_module = w_module->w->priv; + + if (src_module == NULL) { + src_module = dst_module; + continue; + } + + ret = skl_unbind_modules(ctx, src_module, dst_module); + if (ret < 0) + return ret; + + src_module = dst_module; + } + + ret = skl_delete_pipe(ctx, mconfig->pipe); + skl_tplg_free_pipe_mem(skl, mconfig); + + return ret; +} + +/* + * in the Post-PMD event of PGA we need to do following: + * - Free the mcps used + * - Stop the pipeline + * - In source pipe is connected, unbind with source pipelines + */ +static int skl_tplg_pga_dapm_post_pmd_event(struct snd_soc_dapm_widget *w, + struct skl *skl) +{ + struct snd_soc_dapm_widget *source, *sink; + struct skl_module_cfg *src_mconfig, *sink_mconfig; + int ret = 0, path_found = 0; + struct skl_dapm_path_list *path_list, *tmp_path_list; + struct skl_sst *ctx = skl->skl_sst; + + source = w; + src_mconfig = source->priv; + + skl_tplg_free_pipe_mcps(skl, src_mconfig); + /* Stop the pipe since this is a mixin module */ + ret = skl_stop_pipe(ctx, src_mconfig->pipe); + if (ret) + return ret; + + list_for_each_entry_safe(path_list, tmp_path_list, &skl->dapm_path_list, node) { + if (path_list->dapm_path->source == source) { + dev_dbg(ctx->dev, "Path found = %s\n", + path_list->dapm_path->name); + sink = path_list->dapm_path->sink; + sink_mconfig = sink->priv; + path_found = 1; + + list_del(&path_list->node); + kfree(path_list); + break; + } + } + + /* + * This is a connector and if path is found that means + * unbind between source and sink has not happened yet + */ + if (path_found) { + ret = skl_stop_pipe(ctx, src_mconfig->pipe); + if (ret < 0) + return ret; + + ret = skl_unbind_modules(ctx, src_mconfig, sink_mconfig); + } + + return ret; +} + +/* + * In modelling, we assume there will be ONLY one mixer in a pipeline. If + * mixer is not required then it is treated as static mixer aka vmixer with + * a hard path to source module + * So we don't need to check if source is started or not as hard path puts + * dependency on each other + */ +static int skl_tplg_vmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct skl *skl = get_skl_ctx(dapm->dev); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return skl_tplg_mixer_dapm_pre_pmu_event(w, skl); + + case SND_SOC_DAPM_POST_PMD: + return skl_tplg_mixer_dapm_post_pmd_event(w, skl); + } + + return 0; +} + +/* + * In modelling, we assume there will be ONLY one mixer in a pipeline. If a + * second one is required that is created as another pipe entity. + * The mixer is responsible for pipe management and represent a pipeline + * instance + */ +static int skl_tplg_mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct skl *skl = get_skl_ctx(dapm->dev); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return skl_tplg_mixer_dapm_pre_pmu_event(w, skl); + + case SND_SOC_DAPM_POST_PMU: + return skl_tplg_mixer_dapm_post_pmu_event(w, skl); + + case SND_SOC_DAPM_PRE_PMD: + return skl_tplg_mixer_dapm_pre_pmd_event(w, skl); + + case SND_SOC_DAPM_POST_PMD: + return skl_tplg_mixer_dapm_post_pmd_event(w, skl); + } + + return 0; +} + +/* + * In modelling, we assumed rest of the modules in pipeline are PGA. But we + * are interested in last PGA (leaf PGA) in a pipeline to disconnect with + * the sink when it is running (two FE to one BE or one FE to two BE) + * scenarios + */ +static int skl_tplg_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) + +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct skl *skl = get_skl_ctx(dapm->dev); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return skl_tplg_pga_dapm_pre_pmu_event(w, skl); + + case SND_SOC_DAPM_POST_PMD: + return skl_tplg_pga_dapm_post_pmd_event(w, skl); + } + + return 0; +} + +/* + * The FE params are passed by hw_params of the DAI. + * On hw_params, the params are stored in Gateway module of the FE and we + * need to calculate the format in DSP module configuration, that + * conversion is done here + */ +int skl_tplg_update_pipe_params(struct device *dev, + struct skl_module_cfg *mconfig, + struct skl_pipe_params *params) +{ + struct skl_pipe *pipe = mconfig->pipe; + struct skl_module_fmt *format = NULL; + + memcpy(pipe->p_params, params, sizeof(*params)); + + if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) + format = &mconfig->in_fmt; + else + format = &mconfig->out_fmt; + + /* set the hw_params */ + format->s_freq = params->s_freq; + format->channels = params->ch; + format->valid_bit_depth = skl_get_bit_depth(params->s_fmt); + + /* + * 16 bit is 16 bit container whereas 24 bit is in 32 bit + * container so update bit depth accordingly + */ + switch (format->valid_bit_depth) { + case SKL_DEPTH_16BIT: + format->bit_depth = format->valid_bit_depth; + break; + + case SKL_DEPTH_24BIT: + format->bit_depth = SKL_DEPTH_32BIT; + break; + + default: + dev_err(dev, "Invalid bit depth %x for pipe\n", + format->valid_bit_depth); + return -EINVAL; + } + + if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mconfig->ibs = (format->s_freq / 1000) * + (format->channels) * + (format->bit_depth >> 3); + } else { + mconfig->obs = (format->s_freq / 1000) * + (format->channels) * + (format->bit_depth >> 3); + } + + return 0; +} + +/* + * Query the module config for the FE DAI + * This is used to find the hw_params set for that DAI and apply to FE + * pipeline + */ +struct skl_module_cfg * +skl_tplg_fe_get_cpr_module(struct snd_soc_dai *dai, int stream) +{ + struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_path *p = NULL; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + w = dai->playback_widget; + snd_soc_dapm_widget_for_each_sink_path(w, p) { + if (p->connect && p->sink->power && + is_skl_dsp_widget_type(p->sink)) + continue; + + if (p->sink->priv) { + dev_dbg(dai->dev, "set params for %s\n", + p->sink->name); + return p->sink->priv; + } + } + } else { + w = dai->capture_widget; + snd_soc_dapm_widget_for_each_source_path(w, p) { + if (p->connect && p->source->power && + is_skl_dsp_widget_type(p->source)) + continue; + + if (p->source->priv) { + dev_dbg(dai->dev, "set params for %s\n", + p->source->name); + return p->source->priv; + } + } + } + + return NULL; +} + +static u8 skl_tplg_be_link_type(int dev_type) +{ + int ret; + + switch (dev_type) { + case SKL_DEVICE_BT: + ret = NHLT_LINK_SSP; + break; + + case SKL_DEVICE_DMIC: + ret = NHLT_LINK_DMIC; + break; + + case SKL_DEVICE_I2S: + ret = NHLT_LINK_SSP; + break; + + case SKL_DEVICE_HDALINK: + ret = NHLT_LINK_HDA; + break; + + default: + ret = NHLT_LINK_INVALID; + break; + } + + return ret; +} + +/* + * Fill the BE gateway parameters + * The BE gateway expects a blob of parameters which are kept in the ACPI + * NHLT blob, so query the blob for interface type (i2s/pdm) and instance. + * The port can have multiple settings so pick based on the PCM + * parameters + */ +static int skl_tplg_be_fill_pipe_params(struct snd_soc_dai *dai, + struct skl_module_cfg *mconfig, + struct skl_pipe_params *params) +{ + struct skl_pipe *pipe = mconfig->pipe; + struct nhlt_specific_cfg *cfg; + struct skl *skl = get_skl_ctx(dai->dev); + int link_type = skl_tplg_be_link_type(mconfig->dev_type); + + memcpy(pipe->p_params, params, sizeof(*params)); + + /* update the blob based on virtual bus_id*/ + cfg = skl_get_ep_blob(skl, mconfig->vbus_id, link_type, + params->s_fmt, params->ch, + params->s_freq, params->stream); + if (cfg) { + mconfig->formats_config.caps_size = cfg->size; + mconfig->formats_config.caps = (u32 *) &cfg->caps; + } else { + dev_err(dai->dev, "Blob NULL for id %x type %d dirn %d\n", + mconfig->vbus_id, link_type, + params->stream); + dev_err(dai->dev, "PCM: ch %d, freq %d, fmt %d\n", + params->ch, params->s_freq, params->s_fmt); + return -EINVAL; + } + + return 0; +} + +static int skl_tplg_be_set_src_pipe_params(struct snd_soc_dai *dai, + struct snd_soc_dapm_widget *w, + struct skl_pipe_params *params) +{ + struct snd_soc_dapm_path *p; + int ret = -EIO; + + snd_soc_dapm_widget_for_each_source_path(w, p) { + if (p->connect && is_skl_dsp_widget_type(p->source) && + p->source->priv) { + + if (!p->source->power) { + ret = skl_tplg_be_fill_pipe_params( + dai, p->source->priv, + params); + if (ret < 0) + return ret; + } else { + return -EBUSY; + } + } else { + ret = skl_tplg_be_set_src_pipe_params( + dai, p->source, params); + if (ret < 0) + return ret; + } + } + + return ret; +} + +static int skl_tplg_be_set_sink_pipe_params(struct snd_soc_dai *dai, + struct snd_soc_dapm_widget *w, struct skl_pipe_params *params) +{ + struct snd_soc_dapm_path *p = NULL; + int ret = -EIO; + + snd_soc_dapm_widget_for_each_sink_path(w, p) { + if (p->connect && is_skl_dsp_widget_type(p->sink) && + p->sink->priv) { + + if (!p->sink->power) { + ret = skl_tplg_be_fill_pipe_params( + dai, p->sink->priv, params); + if (ret < 0) + return ret; + } else { + return -EBUSY; + } + + } else { + ret = skl_tplg_be_set_sink_pipe_params( + dai, p->sink, params); + if (ret < 0) + return ret; + } + } + + return ret; +} + +/* + * BE hw_params can be a source parameters (capture) or sink parameters + * (playback). Based on sink and source we need to either find the source + * list or the sink list and set the pipeline parameters + */ +int skl_tplg_be_update_params(struct snd_soc_dai *dai, + struct skl_pipe_params *params) +{ + struct snd_soc_dapm_widget *w; + + if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { + w = dai->playback_widget; + + return skl_tplg_be_set_src_pipe_params(dai, w, params); + + } else { + w = dai->capture_widget; + + return skl_tplg_be_set_sink_pipe_params(dai, w, params); + } + + return 0; +} + +static const struct snd_soc_tplg_widget_events skl_tplg_widget_ops[] = { + {SKL_MIXER_EVENT, skl_tplg_mixer_event}, + {SKL_VMIXER_EVENT, skl_tplg_vmixer_event}, + {SKL_PGA_EVENT, skl_tplg_pga_event}, +}; + +/* + * The topology binary passes the pin info for a module so initialize the pin + * info passed into module instance + */ +static void skl_fill_module_pin_info(struct skl_dfw_module_pin *dfw_pin, + struct skl_module_pin *m_pin, + bool is_dynamic, int max_pin) +{ + int i; + + for (i = 0; i < max_pin; i++) { + m_pin[i].id.module_id = dfw_pin[i].module_id; + m_pin[i].id.instance_id = dfw_pin[i].instance_id; + m_pin[i].in_use = false; + m_pin[i].is_dynamic = is_dynamic; + } +} + +/* + * Add pipeline from topology binary into driver pipeline list + * + * If already added we return that instance + * Otherwise we create a new instance and add into driver list + */ +static struct skl_pipe *skl_tplg_add_pipe(struct device *dev, + struct skl *skl, struct skl_dfw_pipe *dfw_pipe) +{ + struct skl_pipeline *ppl; + struct skl_pipe *pipe; + struct skl_pipe_params *params; + + list_for_each_entry(ppl, &skl->ppl_list, node) { + if (ppl->pipe->ppl_id == dfw_pipe->pipe_id) + return ppl->pipe; + } + + ppl = devm_kzalloc(dev, sizeof(*ppl), GFP_KERNEL); + if (!ppl) + return NULL; + + pipe = devm_kzalloc(dev, sizeof(*pipe), GFP_KERNEL); + if (!pipe) + return NULL; + + params = devm_kzalloc(dev, sizeof(*params), GFP_KERNEL); + if (!params) + return NULL; + + pipe->ppl_id = dfw_pipe->pipe_id; + pipe->memory_pages = dfw_pipe->memory_pages; + pipe->pipe_priority = dfw_pipe->pipe_priority; + pipe->conn_type = dfw_pipe->conn_type; + pipe->state = SKL_PIPE_INVALID; + pipe->p_params = params; + INIT_LIST_HEAD(&pipe->w_list); + + ppl->pipe = pipe; + list_add(&ppl->node, &skl->ppl_list); + + return ppl->pipe; +} + +/* + * Topology core widget load callback + * + * This is used to save the private data for each widget which gives + * information to the driver about module and pipeline parameters which DSP + * FW expects like ids, resource values, formats etc + */ +static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, + struct snd_soc_dapm_widget *w, + struct snd_soc_tplg_dapm_widget *tplg_w) +{ + int ret; + struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); + struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl_module_cfg *mconfig; + struct skl_pipe *pipe; + struct skl_dfw_module *dfw_config = + (struct skl_dfw_module *)tplg_w->priv.data; + + if (!tplg_w->priv.size) + goto bind_event; + + mconfig = devm_kzalloc(bus->dev, sizeof(*mconfig), GFP_KERNEL); + + if (!mconfig) + return -ENOMEM; + + w->priv = mconfig; + mconfig->id.module_id = dfw_config->module_id; + mconfig->id.instance_id = dfw_config->instance_id; + mconfig->mcps = dfw_config->max_mcps; + mconfig->ibs = dfw_config->ibs; + mconfig->obs = dfw_config->obs; + mconfig->core_id = dfw_config->core_id; + mconfig->max_in_queue = dfw_config->max_in_queue; + mconfig->max_out_queue = dfw_config->max_out_queue; + mconfig->is_loadable = dfw_config->is_loadable; + mconfig->in_fmt.channels = dfw_config->in_fmt.channels; + mconfig->in_fmt.s_freq = dfw_config->in_fmt.freq; + mconfig->in_fmt.bit_depth = dfw_config->in_fmt.bit_depth; + mconfig->in_fmt.valid_bit_depth = + dfw_config->in_fmt.valid_bit_depth; + mconfig->in_fmt.ch_cfg = dfw_config->in_fmt.ch_cfg; + mconfig->out_fmt.channels = dfw_config->out_fmt.channels; + mconfig->out_fmt.s_freq = dfw_config->out_fmt.freq; + mconfig->out_fmt.bit_depth = dfw_config->out_fmt.bit_depth; + mconfig->out_fmt.valid_bit_depth = + dfw_config->out_fmt.valid_bit_depth; + mconfig->out_fmt.ch_cfg = dfw_config->out_fmt.ch_cfg; + mconfig->params_fixup = dfw_config->params_fixup; + mconfig->converter = dfw_config->converter; + mconfig->m_type = dfw_config->module_type; + mconfig->vbus_id = dfw_config->vbus_id; + + pipe = skl_tplg_add_pipe(bus->dev, skl, &dfw_config->pipe); + if (pipe) + mconfig->pipe = pipe; + + mconfig->dev_type = dfw_config->dev_type; + mconfig->hw_conn_type = dfw_config->hw_conn_type; + mconfig->time_slot = dfw_config->time_slot; + mconfig->formats_config.caps_size = dfw_config->caps.caps_size; + + mconfig->m_in_pin = devm_kzalloc(bus->dev, + (mconfig->max_in_queue) * + sizeof(*mconfig->m_in_pin), + GFP_KERNEL); + if (!mconfig->m_in_pin) + return -ENOMEM; + + mconfig->m_out_pin = devm_kzalloc(bus->dev, (mconfig->max_out_queue) * + sizeof(*mconfig->m_out_pin), + GFP_KERNEL); + if (!mconfig->m_out_pin) + return -ENOMEM; + + skl_fill_module_pin_info(dfw_config->in_pin, mconfig->m_in_pin, + dfw_config->is_dynamic_in_pin, + mconfig->max_in_queue); + + skl_fill_module_pin_info(dfw_config->out_pin, mconfig->m_out_pin, + dfw_config->is_dynamic_out_pin, + mconfig->max_out_queue); + + + if (mconfig->formats_config.caps_size == 0) + goto bind_event; + + mconfig->formats_config.caps = (u32 *)devm_kzalloc(bus->dev, + mconfig->formats_config.caps_size, GFP_KERNEL); + + if (mconfig->formats_config.caps == NULL) + return -ENOMEM; + + memcpy(mconfig->formats_config.caps, dfw_config->caps.caps, + dfw_config->caps.caps_size); + +bind_event: + if (tplg_w->event_type == 0) { + dev_dbg(bus->dev, "ASoC: No event handler required\n"); + return 0; + } + + ret = snd_soc_tplg_widget_bind_event(w, skl_tplg_widget_ops, + ARRAY_SIZE(skl_tplg_widget_ops), + tplg_w->event_type); + + if (ret) { + dev_err(bus->dev, "%s: No matching event handlers found for %d\n", + __func__, tplg_w->event_type); + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_tplg_ops skl_tplg_ops = { + .widget_load = skl_tplg_widget_load, +}; + +/* This will be read from topology manifest, currently defined here */ +#define SKL_MAX_MCPS 30000000 +#define SKL_FW_MAX_MEM 1000000 + +/* + * SKL topology init routine + */ +int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus) +{ + int ret; + const struct firmware *fw; + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = ebus_to_skl(ebus); + + ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); + if (ret < 0) { + dev_err(bus->dev, "tplg fw %s load failed with %d\n", + "dfw_sst.bin", ret); + return ret; + } + + /* + * The complete tplg for SKL is loaded as index 0, we don't use + * any other index + */ + ret = snd_soc_tplg_component_load(&platform->component, + &skl_tplg_ops, fw, 0); + if (ret < 0) { + dev_err(bus->dev, "tplg component load failed%d\n", ret); + return -EINVAL; + } + + skl->resource.max_mcps = SKL_MAX_MCPS; + skl->resource.max_mem = SKL_FW_MAX_MEM; + + return 0; +} diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 8c7767baa94f..76053a8de41c 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -129,6 +129,11 @@ struct skl_src_module_cfg { enum skl_s_freq src_cfg; } __packed; +struct notification_mask { + u32 notify; + u32 enable; +} __packed; + struct skl_up_down_mixer_cfg { struct skl_base_cfg base_cfg; enum skl_ch_cfg out_ch_cfg; @@ -153,8 +158,7 @@ enum skl_dma_type { union skl_ssp_dma_node { u8 val; struct { - u8 dual_mono:1; - u8 time_slot:3; + u8 time_slot_index:4; u8 i2s_instance:4; } dma_node; }; @@ -263,6 +267,34 @@ struct skl_module_cfg { struct skl_specific_cfg formats_config; }; +struct skl_pipeline { + struct skl_pipe *pipe; + struct list_head node; +}; + +struct skl_dapm_path_list { + struct snd_soc_dapm_path *dapm_path; + struct list_head node; +}; + +static inline struct skl *get_skl_ctx(struct device *dev) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + + return ebus_to_skl(ebus); +} + +int skl_tplg_be_update_params(struct snd_soc_dai *dai, + struct skl_pipe_params *params); +void skl_tplg_set_be_dmic_config(struct snd_soc_dai *dai, + struct skl_pipe_params *params, int stream); +int skl_tplg_init(struct snd_soc_platform *platform, + struct hdac_ext_bus *ebus); +struct skl_module_cfg *skl_tplg_fe_get_cpr_module( + struct snd_soc_dai *dai, int stream); +int skl_tplg_update_pipe_params(struct device *dev, + struct skl_module_cfg *mconfig, struct skl_pipe_params *params); + int skl_create_pipeline(struct skl_sst *ctx, struct skl_pipe *pipe); int skl_run_pipe(struct skl_sst *ctx, struct skl_pipe *pipe); diff --git a/sound/soc/intel/skylake/skl-tplg-interface.h b/sound/soc/intel/skylake/skl-tplg-interface.h index a50689825bca..2bc396d54cbe 100644 --- a/sound/soc/intel/skylake/skl-tplg-interface.h +++ b/sound/soc/intel/skylake/skl-tplg-interface.h @@ -19,6 +19,29 @@ #ifndef __HDA_TPLG_INTERFACE_H__ #define __HDA_TPLG_INTERFACE_H__ +/* + * Default types range from 0~12. type can range from 0 to 0xff + * SST types start at higher to avoid any overlapping in future + */ +#define SOC_CONTROL_TYPE_HDA_SST_ALGO_PARAMS 0x100 +#define SOC_CONTROL_TYPE_HDA_SST_MUX 0x101 +#define SOC_CONTROL_TYPE_HDA_SST_MIX 0x101 +#define SOC_CONTROL_TYPE_HDA_SST_BYTE 0x103 + +#define HDA_SST_CFG_MAX 900 /* size of copier cfg*/ +#define MAX_IN_QUEUE 8 +#define MAX_OUT_QUEUE 8 + +/* Event types goes here */ +/* Reserve event type 0 for no event handlers */ +enum skl_event_types { + SKL_EVENT_NONE = 0, + SKL_MIXER_EVENT, + SKL_MUX_EVENT, + SKL_VMIXER_EVENT, + SKL_PGA_EVENT +}; + /** * enum skl_ch_cfg - channel configuration * @@ -83,6 +106,66 @@ enum skl_dev_type { SKL_DEVICE_I2S = 0x2, SKL_DEVICE_SLIMBUS = 0x3, SKL_DEVICE_HDALINK = 0x4, + SKL_DEVICE_HDAHOST = 0x5, SKL_DEVICE_NONE }; + +struct skl_dfw_module_pin { + u16 module_id; + u16 instance_id; +} __packed; + +struct skl_dfw_module_fmt { + u32 channels; + u32 freq; + u32 bit_depth; + u32 valid_bit_depth; + u32 ch_cfg; +} __packed; + +struct skl_dfw_module_caps { + u32 caps_size; + u32 caps[HDA_SST_CFG_MAX]; +}; + +struct skl_dfw_pipe { + u8 pipe_id; + u8 pipe_priority; + u16 conn_type; + u32 memory_pages; +} __packed; + +struct skl_dfw_module { + u16 module_id; + u16 instance_id; + u32 max_mcps; + u8 core_id; + u8 max_in_queue; + u8 max_out_queue; + u8 is_loadable; + u8 conn_type; + u8 dev_type; + u8 hw_conn_type; + u8 time_slot; + u32 obs; + u32 ibs; + u32 params_fixup; + u32 converter; + u32 module_type; + u32 vbus_id; + u8 is_dynamic_in_pin; + u8 is_dynamic_out_pin; + struct skl_dfw_pipe pipe; + struct skl_dfw_module_fmt in_fmt; + struct skl_dfw_module_fmt out_fmt; + struct skl_dfw_module_pin in_pin[MAX_IN_QUEUE]; + struct skl_dfw_module_pin out_pin[MAX_OUT_QUEUE]; + struct skl_dfw_module_caps caps; +} __packed; + +struct skl_dfw_algo_data { + u32 max; + char *params; +} __packed; + #endif diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 348d094e81d6..5319529aedf7 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -166,12 +166,20 @@ static int skl_runtime_suspend(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct hdac_ext_bus *ebus = pci_get_drvdata(pci); struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = ebus_to_skl(ebus); + int ret; dev_dbg(bus->dev, "in %s\n", __func__); /* enable controller wake up event */ snd_hdac_chip_updatew(bus, WAKEEN, 0, STATESTS_INT_MASK); + snd_hdac_ext_bus_link_power_down_all(ebus); + + ret = skl_suspend_dsp(skl); + if (ret < 0) + return ret; + snd_hdac_bus_stop_chip(bus); snd_hdac_bus_enter_link_reset(bus); @@ -183,7 +191,7 @@ static int skl_runtime_resume(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct hdac_ext_bus *ebus = pci_get_drvdata(pci); struct hdac_bus *bus = ebus_to_hbus(ebus); - struct skl *hda = ebus_to_skl(ebus); + struct skl *skl = ebus_to_skl(ebus); int status; dev_dbg(bus->dev, "in %s\n", __func__); @@ -191,12 +199,12 @@ static int skl_runtime_resume(struct device *dev) /* Read STATESTS before controller reset */ status = snd_hdac_chip_readw(bus, STATESTS); - skl_init_pci(hda); + skl_init_pci(skl); snd_hdac_bus_init_chip(bus, true); /* disable controller Wake Up event */ snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, 0); - return 0; + return skl_resume_dsp(skl); } #endif /* CONFIG_PM */ @@ -453,21 +461,28 @@ static int skl_probe(struct pci_dev *pci, if (err < 0) goto out_free; + skl->nhlt = skl_nhlt_init(bus->dev); + + if (skl->nhlt == NULL) + goto out_free; + pci_set_drvdata(skl->pci, ebus); /* check if dsp is there */ if (ebus->ppcap) { - /* TODO register with dsp IPC */ - dev_dbg(bus->dev, "Register dsp\n"); + err = skl_init_dsp(skl); + if (err < 0) { + dev_dbg(bus->dev, "error failed to register dsp\n"); + goto out_free; + } } - if (ebus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(ebus); /* create device for soc dmic */ err = skl_dmic_device_register(skl); if (err < 0) - goto out_free; + goto out_dsp_free; /* register platform dai and controls */ err = skl_platform_register(bus->dev); @@ -491,6 +506,8 @@ out_unregister: skl_platform_unregister(bus->dev); out_dmic_free: skl_dmic_device_unregister(skl); +out_dsp_free: + skl_free_dsp(skl); out_free: skl->init_failed = 1; skl_free(ebus); @@ -507,6 +524,7 @@ static void skl_remove(struct pci_dev *pci) pm_runtime_get_noresume(&pci->dev); pci_dev_put(pci); skl_platform_unregister(&pci->dev); + skl_free_dsp(skl); skl_dmic_device_unregister(skl); skl_free(ebus); dev_set_drvdata(&pci->dev, NULL); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index f7fdbb02947f..dd2e79ae45a8 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -48,6 +48,13 @@ #define AZX_REG_VS_SDXEFIFOS_XBASE 0x1094 #define AZX_REG_VS_SDXEFIFOS_XINTERVAL 0x20 +struct skl_dsp_resource { + u32 max_mcps; + u32 max_mem; + u32 mcps; + u32 mem; +}; + struct skl { struct hdac_ext_bus ebus; struct pci_dev *pci; @@ -55,8 +62,12 @@ struct skl { unsigned int init_failed:1; /* delayed init failed */ struct platform_device *dmic_dev; - void __iomem *nhlt; /* nhlt ptr */ + void *nhlt; /* nhlt ptr */ struct skl_sst *skl_sst; /* sst skl ctx */ + + struct skl_dsp_resource resource; + struct list_head ppl_list; + struct list_head dapm_path_list; }; #define skl_to_ebus(s) (&(s)->ebus) @@ -72,8 +83,8 @@ struct skl_dma_params { int skl_platform_unregister(struct device *dev); int skl_platform_register(struct device *dev); -void __iomem *skl_nhlt_init(struct device *dev); -void skl_nhlt_free(void __iomem *addr); +void *skl_nhlt_init(struct device *dev); +void skl_nhlt_free(void *addr); struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance, u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn); diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index b05fb1c1a848..794a3499e567 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -485,6 +485,7 @@ static const struct of_device_id jz4740_of_matches[] = { { .compatible = "ingenic,jz4780-i2s", .data = (void *)JZ_I2S_JZ4780 }, { /* sentinel */ } }; +MODULE_DEVICE_TABLE(of, jz4740_of_matches); #endif static int jz4740_i2s_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c index de7563bdc5c2..e0304d544f26 100644 --- a/sound/soc/kirkwood/armada-370-db.c +++ b/sound/soc/kirkwood/armada-370-db.c @@ -130,6 +130,7 @@ static const struct of_device_id a370db_dt_ids[] = { { .compatible = "marvell,a370db-audio" }, { }, }; +MODULE_DEVICE_TABLE(of, a370db_dt_ids); static struct platform_driver a370db_driver = { .driver = { diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c index 684e8a78bed0..71a1a35047ba 100644 --- a/sound/soc/mediatek/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173-max98090.c @@ -179,21 +179,13 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) } card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); return ret; } -static int mt8173_max98090_dev_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - static const struct of_device_id mt8173_max98090_dt_match[] = { { .compatible = "mediatek,mt8173-max98090", }, { } @@ -209,7 +201,6 @@ static struct platform_driver mt8173_max98090_driver = { #endif }, .probe = mt8173_max98090_dev_probe, - .remove = mt8173_max98090_dev_remove, }; module_platform_driver(mt8173_max98090_driver); diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index 86cf9752f18a..50ba538eccb3 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -246,21 +246,13 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) card->dev = &pdev->dev; platform_set_drvdata(pdev, card); - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); return ret; } -static int mt8173_rt5650_rt5676_dev_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - static const struct of_device_id mt8173_rt5650_rt5676_dt_match[] = { { .compatible = "mediatek,mt8173-rt5650-rt5676", }, { } @@ -276,7 +268,6 @@ static struct platform_driver mt8173_rt5650_rt5676_driver = { #endif }, .probe = mt8173_rt5650_rt5676_dev_probe, - .remove = mt8173_rt5650_rt5676_dev_remove, }; module_platform_driver(mt8173_rt5650_rt5676_driver); diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 6e6fce6a14ba..2b23ffbac6b1 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -142,7 +142,7 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev) card->dev = &pdev->dev; platform_set_drvdata(pdev, card); - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); @@ -154,12 +154,8 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev) static int mxs_sgtl5000_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - mxs_saif_put_mclk(0); - snd_soc_unregister_card(card); - return 0; } diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index dcb5336b5698..190f868e78b2 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -99,8 +99,7 @@ static int n810_startup(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); + snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2); n810_ext_control(&rtd->card->dapm); return clk_prepare_enable(sys_clkout2); diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 3bebfb1d3a6f..5e21f08579d8 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -107,8 +107,7 @@ static int rx51_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); + snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2); rx51_ext_control(&card->dapm); return 0; @@ -297,7 +296,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) dev_err(card->dev, "Failed to add TPA6130A2 controls\n"); return err; } - snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); + snd_soc_limit_volume(card, "TPA6130A2 Headphone Playback Volume", 42); err = omap_mcbsp_st_add_controls(rtd, 2); if (err < 0) { diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 2b26318bc200..6147e86e9b0f 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -116,26 +116,19 @@ static int brownstone_probe(struct platform_device *pdev) int ret; brownstone.dev = &pdev->dev; - ret = snd_soc_register_card(&brownstone); + ret = devm_snd_soc_register_card(&pdev->dev, &brownstone); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int brownstone_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&brownstone); - return 0; -} - static struct platform_driver mmp_driver = { .driver = { .name = "brownstone-audio", .pm = &snd_soc_pm_ops, }, .probe = brownstone_probe, - .remove = brownstone_remove, }; module_platform_driver(mmp_driver); diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 3580d10c9f28..c97dc13d3608 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -295,28 +295,19 @@ static int corgi_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int corgi_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - static struct platform_driver corgi_driver = { .driver = { .name = "corgi-audio", .pm = &snd_soc_pm_ops, }, .probe = corgi_probe, - .remove = corgi_remove, }; module_platform_driver(corgi_driver); diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index d72e124a3676..1de876529aa1 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -138,7 +138,7 @@ static int e740_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -149,10 +149,7 @@ static int e740_probe(struct platform_device *pdev) static int e740_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios)); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 48f2d7c2e68c..b7eb7cd5df7d 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -120,7 +120,7 @@ static int e750_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -131,10 +131,7 @@ static int e750_probe(struct platform_device *pdev) static int e750_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios)); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 45d4bd46fff6..41bf71466a7b 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -119,7 +119,7 @@ static int e800_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -130,10 +130,7 @@ static int e800_probe(struct platform_device *pdev) static int e800_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios)); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 9f8be7cd567e..ecbf2873b7ff 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -193,7 +193,7 @@ static int hx4700_audio_probe(struct platform_device *pdev) return ret; snd_soc_card_hx4700.dev = &pdev->dev; - ret = snd_soc_register_card(&snd_soc_card_hx4700); + ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_hx4700); if (ret) gpio_free_array(hx4700_audio_gpios, ARRAY_SIZE(hx4700_audio_gpios)); @@ -203,8 +203,6 @@ static int hx4700_audio_probe(struct platform_device *pdev) static int hx4700_audio_remove(struct platform_device *pdev) { - snd_soc_unregister_card(&snd_soc_card_hx4700); - gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0); gpio_set_value(GPIO107_HX4700_SPK_nSD, 0); diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 29fabbfd21f1..9d0e40771ef5 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -72,28 +72,19 @@ static int imote2_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int imote2_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - static struct platform_driver imote2_driver = { .driver = { .name = "imote2-audio", .pm = &snd_soc_pm_ops, }, .probe = imote2_probe, - .remove = imote2_remove, }; module_platform_driver(imote2_driver); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index a9615a574546..29bc60e85e92 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -181,7 +181,7 @@ static int mioa701_wm9713_probe(struct platform_device *pdev) return -ENODEV; mioa701.dev = &pdev->dev; - rc = snd_soc_register_card(&mioa701); + rc = devm_snd_soc_register_card(&pdev->dev, &mioa701); if (!rc) dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" "lead to overheating and possible destruction of your device." @@ -189,17 +189,8 @@ static int mioa701_wm9713_probe(struct platform_device *pdev) return rc; } -static int mioa701_wm9713_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - static struct platform_driver mioa701_wm9713_driver = { .probe = mioa701_wm9713_probe, - .remove = mioa701_wm9713_remove, .driver = { .name = "mioa701-wm9713", .pm = &snd_soc_pm_ops, diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index c20bbc042425..4e74d9573f03 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -140,22 +140,15 @@ static int palm27x_asoc_probe(struct platform_device *pdev) palm27x_asoc.dev = &pdev->dev; - ret = snd_soc_register_card(&palm27x_asoc); + ret = devm_snd_soc_register_card(&pdev->dev, &palm27x_asoc); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int palm27x_asoc_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&palm27x_asoc); - return 0; -} - static struct platform_driver palm27x_wm9712_driver = { .probe = palm27x_asoc_probe, - .remove = palm27x_asoc_remove, .driver = { .name = "palm27x-asoc", .pm = &snd_soc_pm_ops, diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 80b457ac522a..84d0e2e50808 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -267,28 +267,19 @@ static int poodle_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int poodle_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - static struct platform_driver poodle_driver = { .driver = { .name = "poodle-audio", .pm = &snd_soc_pm_ops, }, .probe = poodle_probe, - .remove = poodle_remove, }; module_platform_driver(poodle_driver); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 3da485ec1de7..da03fad1b9cd 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -809,6 +809,7 @@ static const struct of_device_id pxa_ssp_of_ids[] = { { .compatible = "mrvl,pxa-ssp-dai" }, {} }; +MODULE_DEVICE_TABLE(of, pxa_ssp_of_ids); #endif static int asoc_ssp_probe(struct platform_device *pdev) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9e4b04e0fbd1..f3de615aacd7 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> #include <sound/core.h> #include <sound/ac97_codec.h> @@ -49,7 +50,11 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11; +static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 11, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -57,7 +62,11 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, }; -static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12; +static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 12, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -65,7 +74,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10; +static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 10, +}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, @@ -73,7 +85,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9; +static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 9, +}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, @@ -81,7 +96,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8; +static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 8, +}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, @@ -89,9 +107,8 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; -static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -105,9 +122,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, return 0; } -static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int pxa2xx_ac97_aux_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -121,9 +137,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, return 0; } -static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int pxa2xx_ac97_mic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -139,15 +154,15 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_48000) static const struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = { - .hw_params = pxa2xx_ac97_hw_params, + .startup = pxa2xx_ac97_hifi_startup, }; static const struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = { - .hw_params = pxa2xx_ac97_hw_aux_params, + .startup = pxa2xx_ac97_aux_startup, }; static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { - .hw_params = pxa2xx_ac97_hw_mic_params, + .startup = pxa2xx_ac97_mic_startup, }; /* diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 6b4e40036910..0389cf7b4b1e 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -319,6 +319,9 @@ static int pxa2xx_i2s_probe(struct snd_soc_dai *dai) /* Along with FIFO servicing */ SAIMR &= ~(SAIMR_RFS | SAIMR_TFS); + snd_soc_dai_init_dma_data(dai, &pxa2xx_i2s_pcm_stereo_out, + &pxa2xx_i2s_pcm_stereo_in); + return 0; } diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 831ee37d2e3e..9f390398d518 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -15,8 +15,6 @@ #include <linux/dmaengine.h> #include <linux/of.h> -#include <mach/dma.h> - #include <sound/core.h> #include <sound/soc.h> #include <sound/pxa2xx-lib.h> @@ -27,11 +25,8 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_dmaengine_dai_dma_data *dma; - int ret; dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); @@ -40,40 +35,13 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, if (!dma) return 0; - /* this may get called several times by oss emulation - * with different params */ - if (prtd->params == NULL) { - prtd->params = dma; - ret = pxa_request_dma("name", DMA_PRIO_LOW, - pxa2xx_pcm_dma_irq, substream); - if (ret < 0) - return ret; - prtd->dma_ch = ret; - } else if (prtd->params != dma) { - pxa_free_dma(prtd->dma_ch); - prtd->params = dma; - ret = pxa_request_dma("name", DMA_PRIO_LOW, - pxa2xx_pcm_dma_irq, substream); - if (ret < 0) - return ret; - prtd->dma_ch = ret; - } - return __pxa2xx_pcm_hw_params(substream, params); } static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { - struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; - __pxa2xx_pcm_hw_free(substream); - if (prtd->dma_ch >= 0) { - pxa_free_dma(prtd->dma_ch); - prtd->dma_ch = -1; - prtd->params = NULL; - } - return 0; } @@ -132,6 +100,7 @@ static const struct of_device_id snd_soc_pxa_audio_match[] = { { .compatible = "mrvl,pxa-pcm-audio" }, { } }; +MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match); #endif static struct platform_driver pxa_pcm_driver = { diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 461123ad5ff2..b00222620fd0 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -305,7 +305,7 @@ static int spitz_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -322,9 +322,6 @@ err1: static int spitz_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); gpio_free(spitz_mic_gpio); return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index f59f566551ef..49518dd642aa 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -233,7 +233,7 @@ static int tosa_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -244,10 +244,7 @@ static int tosa_probe(struct platform_device *pdev) static int tosa_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - gpio_free(TOSA_GPIO_L_MUTE); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index 1753c7d9e760..65c20f779177 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -128,7 +128,7 @@ static int ttc_dkb_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -136,22 +136,12 @@ static int ttc_dkb_probe(struct platform_device *pdev) return ret; } -static int ttc_dkb_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static struct platform_driver ttc_dkb_driver = { .driver = { .name = "ttc-dkb-audio", .pm = &snd_soc_pm_ops, }, .probe = ttc_dkb_probe, - .remove = ttc_dkb_remove, }; module_platform_driver(ttc_dkb_driver); diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 97bc2023f08a..e5101e0d2d37 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -438,7 +438,8 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) if (IS_ERR(drvdata->mi2s_bit_clk[dai_id])) { dev_err(&pdev->dev, "%s() error getting mi2s-bit-clk: %ld\n", - __func__, PTR_ERR(drvdata->mi2s_bit_clk[i])); + __func__, + PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } } diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index 58bae8e2cf5f..f1e0c703e0d2 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -15,9 +15,17 @@ config SND_SOC_ROCKCHIP_I2S Rockchip I2S device. The device supports upto maximum of 8 channels each for play and record. +config SND_SOC_ROCKCHIP_SPDIF + tristate "Rockchip SPDIF Device Driver" + depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for SPDIF driver for + Rockchip SPDIF transceiver device. + config SND_SOC_ROCKCHIP_MAX98090 tristate "ASoC support for Rockchip boards using a MAX98090 codec" - depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP select SND_SOC_ROCKCHIP_I2S select SND_SOC_MAX98090 select SND_SOC_TS3A227E @@ -27,7 +35,7 @@ config SND_SOC_ROCKCHIP_MAX98090 config SND_SOC_ROCKCHIP_RT5645 tristate "ASoC support for Rockchip boards using a RT5645/RT5650 codec" - depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP select SND_SOC_ROCKCHIP_I2S select SND_SOC_RT5645 help diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 1bc1dc3c729a..c0bf560125f3 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,7 +1,9 @@ # ROCKCHIP Platform Support -snd-soc-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-spdif-objs := rockchip_spdif.o -obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index b93610212e3d..58ee64594f07 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -226,6 +226,7 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct rk_i2s_dev *i2s = to_info(dai); + struct snd_soc_pcm_runtime *rtd = substream->private_data; unsigned int val = 0; switch (params_format(params)) { @@ -245,13 +246,46 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - regmap_update_bits(i2s->regmap, I2S_TXCR, I2S_TXCR_VDW_MASK, val); - regmap_update_bits(i2s->regmap, I2S_RXCR, I2S_RXCR_VDW_MASK, val); + switch (params_channels(params)) { + case 8: + val |= I2S_CHN_8; + break; + case 6: + val |= I2S_CHN_6; + break; + case 4: + val |= I2S_CHN_4; + break; + case 2: + val |= I2S_CHN_2; + break; + default: + dev_err(i2s->dev, "invalid channel: %d\n", + params_channels(params)); + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + regmap_update_bits(i2s->regmap, I2S_RXCR, + I2S_RXCR_VDW_MASK | I2S_RXCR_CSR_MASK, + val); + else + regmap_update_bits(i2s->regmap, I2S_TXCR, + I2S_TXCR_VDW_MASK | I2S_TXCR_CSR_MASK, + val); + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK, I2S_DMACR_TDL(16)); regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK, I2S_DMACR_RDL(16)); + val = I2S_CKR_TRCM_TXRX; + if (dai->driver->symmetric_rates || rtd->dai_link->symmetric_rates) + val = I2S_CKR_TRCM_TXSHARE; + + regmap_update_bits(i2s->regmap, I2S_CKR, + I2S_CKR_TRCM_MASK, + val); return 0; } @@ -415,10 +449,12 @@ static const struct regmap_config rockchip_i2s_regmap_config = { static int rockchip_i2s_probe(struct platform_device *pdev) { + struct device_node *node = pdev->dev.of_node; struct rk_i2s_dev *i2s; struct resource *res; void __iomem *regs; int ret; + int val; i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); if (!i2s) { @@ -475,6 +511,14 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_pm_disable; } + /* refine capture channels */ + if (!of_property_read_u32(node, "rockchip,capture-channels", &val)) { + if (val >= 2 && val <= 8) + rockchip_i2s_dai.capture.channels_max = val; + else + rockchip_i2s_dai.capture.channels_max = 2; + } + ret = devm_snd_soc_register_component(&pdev->dev, &rockchip_i2s_component, &rockchip_i2s_dai, 1); diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h index 93f456f518a9..dc6e2c74d088 100644 --- a/sound/soc/rockchip/rockchip_i2s.h +++ b/sound/soc/rockchip/rockchip_i2s.h @@ -49,6 +49,9 @@ * RXCR * receive operation control register */ +#define I2S_RXCR_CSR_SHIFT 15 +#define I2S_RXCR_CSR(x) (x << I2S_RXCR_CSR_SHIFT) +#define I2S_RXCR_CSR_MASK (3 << I2S_RXCR_CSR_SHIFT) #define I2S_RXCR_HWT BIT(14) #define I2S_RXCR_SJM_SHIFT 12 #define I2S_RXCR_SJM_R (0 << I2S_RXCR_SJM_SHIFT) @@ -75,6 +78,12 @@ * CKR * clock generation register */ +#define I2S_CKR_TRCM_SHIFT 28 +#define I2S_CKR_TRCM(x) (x << I2S_CKR_TRCM_SHIFT) +#define I2S_CKR_TRCM_TXRX (0 << I2S_CKR_TRCM_SHIFT) +#define I2S_CKR_TRCM_TXSHARE (1 << I2S_CKR_TRCM_SHIFT) +#define I2S_CKR_TRCM_RXSHARE (2 << I2S_CKR_TRCM_SHIFT) +#define I2S_CKR_TRCM_MASK (3 << I2S_CKR_TRCM_SHIFT) #define I2S_CKR_MSS_SHIFT 27 #define I2S_CKR_MSS_MASTER (0 << I2S_CKR_MSS_SHIFT) #define I2S_CKR_MSS_SLAVE (1 << I2S_CKR_MSS_SHIFT) @@ -207,6 +216,13 @@ enum { ROCKCHIP_DIV_BCLK, }; +/* channel select */ +#define I2S_CSR_SHIFT 15 +#define I2S_CHN_2 (0 << I2S_CSR_SHIFT) +#define I2S_CHN_4 (1 << I2S_CSR_SHIFT) +#define I2S_CHN_6 (2 << I2S_CSR_SHIFT) +#define I2S_CHN_8 (3 << I2S_CSR_SHIFT) + /* I2S REGS */ #define I2S_TXCR (0x0000) #define I2S_RXCR (0x0004) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c new file mode 100644 index 000000000000..a38a3029062c --- /dev/null +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -0,0 +1,405 @@ +/* sound/soc/rockchip/rk_spdif.c + * + * ALSA SoC Audio Layer - Rockchip I2S Controller driver + * + * Copyright (c) 2014 Rockchip Electronics Co. Ltd. + * Author: Jianqun <jay.xu@rock-chips.com> + * Copyright (c) 2015 Collabora Ltd. + * Author: Sjoerd Simons <sjoerd.simons@collabora.co.uk> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/of_gpio.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> +#include <linux/mfd/syscon.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> + +#include "rockchip_spdif.h" + +enum rk_spdif_type { + RK_SPDIF_RK3066, + RK_SPDIF_RK3188, + RK_SPDIF_RK3288, +}; + +#define RK3288_GRF_SOC_CON2 0x24c + +struct rk_spdif_dev { + struct device *dev; + + struct clk *mclk; + struct clk *hclk; + + struct snd_dmaengine_dai_dma_data playback_dma_data; + + struct regmap *regmap; +}; + +static const struct of_device_id rk_spdif_match[] = { + { .compatible = "rockchip,rk3066-spdif", + .data = (void *) RK_SPDIF_RK3066 }, + { .compatible = "rockchip,rk3188-spdif", + .data = (void *) RK_SPDIF_RK3188 }, + { .compatible = "rockchip,rk3288-spdif", + .data = (void *) RK_SPDIF_RK3288 }, + {}, +}; +MODULE_DEVICE_TABLE(of, rk_spdif_match); + +static int rk_spdif_runtime_suspend(struct device *dev) +{ + struct rk_spdif_dev *spdif = dev_get_drvdata(dev); + + clk_disable_unprepare(spdif->mclk); + clk_disable_unprepare(spdif->hclk); + + return 0; +} + +static int rk_spdif_runtime_resume(struct device *dev) +{ + struct rk_spdif_dev *spdif = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(spdif->mclk); + if (ret) { + dev_err(spdif->dev, "mclk clock enable failed %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(spdif->hclk); + if (ret) { + dev_err(spdif->dev, "hclk clock enable failed %d\n", ret); + return ret; + } + + return 0; +} + +static int rk_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai); + unsigned int val = SPDIF_CFGR_HALFWORD_ENABLE; + int srate, mclk; + int ret; + + srate = params_rate(params); + switch (srate) { + case 32000: + case 48000: + case 96000: + mclk = 96000 * 128; /* 12288000 hz */ + break; + case 44100: + mclk = 44100 * 256; /* 11289600 hz */ + break; + case 192000: + mclk = 192000 * 128; /* 24576000 hz */ + break; + default: + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val |= SPDIF_CFGR_VDW_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val |= SPDIF_CFGR_VDW_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val |= SPDIF_CFGR_VDW_24; + break; + default: + return -EINVAL; + } + + /* Set clock and calculate divider */ + ret = clk_set_rate(spdif->mclk, mclk); + if (ret != 0) { + dev_err(spdif->dev, "Failed to set module clock rate: %d\n", + ret); + return ret; + } + + val |= SPDIF_CFGR_CLK_DIV(mclk/(srate * 256)); + ret = regmap_update_bits(spdif->regmap, SPDIF_CFGR, + SPDIF_CFGR_CLK_DIV_MASK | SPDIF_CFGR_HALFWORD_ENABLE | + SDPIF_CFGR_VDW_MASK, + val); + + return ret; +} + +static int rk_spdif_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai); + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR, + SPDIF_DMACR_TDE_ENABLE, + SPDIF_DMACR_TDE_ENABLE); + + if (ret != 0) + return ret; + + ret = regmap_update_bits(spdif->regmap, SPDIF_XFER, + SPDIF_XFER_TXS_START, + SPDIF_XFER_TXS_START); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR, + SPDIF_DMACR_TDE_ENABLE, + SPDIF_DMACR_TDE_DISABLE); + + if (ret != 0) + return ret; + + ret = regmap_update_bits(spdif->regmap, SPDIF_XFER, + SPDIF_XFER_TXS_START, + SPDIF_XFER_TXS_STOP); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int rk_spdif_dai_probe(struct snd_soc_dai *dai) +{ + struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &spdif->playback_dma_data; + + return 0; +} + +static const struct snd_soc_dai_ops rk_spdif_dai_ops = { + .hw_params = rk_spdif_hw_params, + .trigger = rk_spdif_trigger, +}; + +static struct snd_soc_dai_driver rk_spdif_dai = { + .probe = rk_spdif_dai_probe, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &rk_spdif_dai_ops, +}; + +static const struct snd_soc_component_driver rk_spdif_component = { + .name = "rockchip-spdif", +}; + +static bool rk_spdif_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SPDIF_CFGR: + case SPDIF_DMACR: + case SPDIF_INTCR: + case SPDIF_XFER: + case SPDIF_SMPDR: + return true; + default: + return false; + } +} + +static bool rk_spdif_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SPDIF_CFGR: + case SPDIF_SDBLR: + case SPDIF_INTCR: + case SPDIF_INTSR: + case SPDIF_XFER: + return true; + default: + return false; + } +} + +static bool rk_spdif_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SPDIF_INTSR: + case SPDIF_SDBLR: + return true; + default: + return false; + } +} + +static const struct regmap_config rk_spdif_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SPDIF_SMPDR, + .writeable_reg = rk_spdif_wr_reg, + .readable_reg = rk_spdif_rd_reg, + .volatile_reg = rk_spdif_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int rk_spdif_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct rk_spdif_dev *spdif; + const struct of_device_id *match; + struct resource *res; + void __iomem *regs; + int ret; + + match = of_match_node(rk_spdif_match, np); + if ((int) match->data == RK_SPDIF_RK3288) { + struct regmap *grf; + + grf = syscon_regmap_lookup_by_phandle(np, "rockchip,grf"); + if (IS_ERR(grf)) { + dev_err(&pdev->dev, + "rockchip_spdif missing 'rockchip,grf' \n"); + return PTR_ERR(grf); + } + + /* Select the 8 channel SPDIF solution on RK3288 as + * the 2 channel one does not appear to work + */ + regmap_write(grf, RK3288_GRF_SOC_CON2, BIT(1) << 16); + } + + spdif = devm_kzalloc(&pdev->dev, sizeof(*spdif), GFP_KERNEL); + if (!spdif) + return -ENOMEM; + + spdif->hclk = devm_clk_get(&pdev->dev, "hclk"); + if (IS_ERR(spdif->hclk)) { + dev_err(&pdev->dev, "Can't retrieve rk_spdif bus clock\n"); + return PTR_ERR(spdif->hclk); + } + ret = clk_prepare_enable(spdif->hclk); + if (ret) { + dev_err(spdif->dev, "hclock enable failed %d\n", ret); + return ret; + } + + spdif->mclk = devm_clk_get(&pdev->dev, "mclk"); + if (IS_ERR(spdif->mclk)) { + dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n"); + return PTR_ERR(spdif->mclk); + } + + ret = clk_prepare_enable(spdif->mclk); + if (ret) { + dev_err(spdif->dev, "clock enable failed %d\n", ret); + return ret; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs, + &rk_spdif_regmap_config); + if (IS_ERR(spdif->regmap)) { + dev_err(&pdev->dev, + "Failed to initialise managed register map\n"); + return PTR_ERR(spdif->regmap); + } + + spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR; + spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + spdif->playback_dma_data.maxburst = 4; + + spdif->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, spdif); + + pm_runtime_set_active(&pdev->dev); + pm_runtime_enable(&pdev->dev); + pm_request_idle(&pdev->dev); + + ret = devm_snd_soc_register_component(&pdev->dev, + &rk_spdif_component, + &rk_spdif_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI\n"); + goto err_pm_runtime; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM\n"); + goto err_pm_runtime; + } + + return 0; + +err_pm_runtime: + pm_runtime_disable(&pdev->dev); + + return ret; +} + +static int rk_spdif_remove(struct platform_device *pdev) +{ + struct rk_spdif_dev *spdif = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + rk_spdif_runtime_suspend(&pdev->dev); + + clk_disable_unprepare(spdif->mclk); + clk_disable_unprepare(spdif->hclk); + + return 0; +} + +static const struct dev_pm_ops rk_spdif_pm_ops = { + SET_RUNTIME_PM_OPS(rk_spdif_runtime_suspend, rk_spdif_runtime_resume, + NULL) +}; + +static struct platform_driver rk_spdif_driver = { + .probe = rk_spdif_probe, + .remove = rk_spdif_remove, + .driver = { + .name = "rockchip-spdif", + .of_match_table = of_match_ptr(rk_spdif_match), + .pm = &rk_spdif_pm_ops, + }, +}; +module_platform_driver(rk_spdif_driver); + +MODULE_ALIAS("platform:rockchip-spdif"); +MODULE_DESCRIPTION("ROCKCHIP SPDIF transceiver Interface"); +MODULE_AUTHOR("Sjoerd Simons <sjoerd.simons@collabora.co.uk>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h new file mode 100644 index 000000000000..07f86a21046a --- /dev/null +++ b/sound/soc/rockchip/rockchip_spdif.h @@ -0,0 +1,63 @@ +/* + * ALSA SoC Audio Layer - Rockchip SPDIF transceiver driver + * + * Copyright (c) 2015 Collabora Ltd. + * Author: Sjoerd Simons <sjoerd.simons@collabora.co.uk> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ROCKCHIP_SPDIF_H +#define _ROCKCHIP_SPDIF_H + +/* + * CFGR + * transfer configuration register +*/ +#define SPDIF_CFGR_CLK_DIV_SHIFT (16) +#define SPDIF_CFGR_CLK_DIV_MASK (0xff << SPDIF_CFGR_CLK_DIV_SHIFT) +#define SPDIF_CFGR_CLK_DIV(x) (x << SPDIF_CFGR_CLK_DIV_SHIFT) + +#define SPDIF_CFGR_HALFWORD_SHIFT 2 +#define SPDIF_CFGR_HALFWORD_DISABLE (0 << SPDIF_CFGR_HALFWORD_SHIFT) +#define SPDIF_CFGR_HALFWORD_ENABLE (1 << SPDIF_CFGR_HALFWORD_SHIFT) + +#define SPDIF_CFGR_VDW_SHIFT 0 +#define SPDIF_CFGR_VDW(x) (x << SPDIF_CFGR_VDW_SHIFT) +#define SDPIF_CFGR_VDW_MASK (0xf << SPDIF_CFGR_VDW_SHIFT) + +#define SPDIF_CFGR_VDW_16 SPDIF_CFGR_VDW(0x00) +#define SPDIF_CFGR_VDW_20 SPDIF_CFGR_VDW(0x01) +#define SPDIF_CFGR_VDW_24 SPDIF_CFGR_VDW(0x10) + +/* + * DMACR + * DMA control register +*/ +#define SPDIF_DMACR_TDE_SHIFT 5 +#define SPDIF_DMACR_TDE_DISABLE (0 << SPDIF_DMACR_TDE_SHIFT) +#define SPDIF_DMACR_TDE_ENABLE (1 << SPDIF_DMACR_TDE_SHIFT) + +#define SPDIF_DMACR_TDL_SHIFT 0 +#define SPDIF_DMACR_TDL(x) ((x) << SPDIF_DMACR_TDL_SHIFT) +#define SPDIF_DMACR_TDL_MASK (0x1f << SDPIF_DMACR_TDL_SHIFT) + +/* + * XFER + * Transfer control register +*/ +#define SPDIF_XFER_TXS_SHIFT 0 +#define SPDIF_XFER_TXS_STOP (0 << SPDIF_XFER_TXS_SHIFT) +#define SPDIF_XFER_TXS_START (1 << SPDIF_XFER_TXS_SHIFT) + +#define SPDIF_CFGR (0x0000) +#define SPDIF_SDBLR (0x0004) +#define SPDIF_DMACR (0x0008) +#define SPDIF_INTCR (0x000c) +#define SPDIF_INTSR (0x0010) +#define SPDIF_XFER (0x0018) +#define SPDIF_SMPDR (0x0020) + +#endif /* _ROCKCHIP_SPDIF_H */ diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index c72e9fb26658..5f5825faeb2a 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -26,16 +26,15 @@ #include <mach/gpio-samsung.h> #include "s3c24xx-i2s.h" -static unsigned int rates[] = { +static const unsigned int rates[] = { 11025, 22050, 44100, }; -static struct snd_pcm_hw_constraint_list hw_rates = { +static const struct snd_pcm_hw_constraint_list hw_rates = { .count = ARRAY_SIZE(rates), .list = rates, - .mask = 0, }; static struct snd_soc_jack hp_jack; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 35e37c457f1f..fa096abe9e75 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -38,16 +38,15 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, static int rx1950_spk_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -static unsigned int rates[] = { +static const unsigned int rates[] = { 16000, 44100, 48000, }; -static struct snd_pcm_hw_constraint_list hw_rates = { +static const struct snd_pcm_hw_constraint_list hw_rates = { .count = ARRAY_SIZE(rates), .list = rates, - .mask = 0, }; static struct snd_soc_jack hp_jack; diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 07114b0b0dc1..206d1edab07c 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,10 +37,11 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" depends on DMA_OF + depends on COMMON_CLK select SND_SIMPLE_CARD select REGMAP_MMIO help - This option enables R-Car SUR/SCU/SSIU/SSI sound support + This option enables R-Car SRU/SCU/SSIU/SSI sound support config SND_SOC_RSRC_CARD tristate "Renesas Sampling Rate Convert Sound Card" diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index fefc881dbac2..2a5b3a293cd2 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -7,7 +7,7 @@ * License. See the file "COPYING" in the main directory of this archive * for more details. */ -#include <linux/sh_clk.h> +#include <linux/clk-provider.h> #include "rsnd.h" #define CLKA 0 @@ -16,12 +16,26 @@ #define CLKI 3 #define CLKMAX 4 +#define CLKOUT 0 +#define CLKOUT1 1 +#define CLKOUT2 2 +#define CLKOUT3 3 +#define CLKOUTMAX 4 + +#define BRRx_MASK(x) (0x3FF & x) + +static struct rsnd_mod_ops adg_ops = { + .name = "adg", +}; + struct rsnd_adg { struct clk *clk[CLKMAX]; + struct clk *clkout[CLKOUTMAX]; + struct clk_onecell_data onecell; + struct rsnd_mod mod; - int rbga_rate_for_441khz_div_6; /* RBGA */ - int rbgb_rate_for_48khz_div_6; /* RBGB */ - u32 ckr; + int rbga_rate_for_441khz; /* RBGA */ + int rbgb_rate_for_48khz; /* RBGB */ }; #define for_each_rsnd_clk(pos, adg, i) \ @@ -29,17 +43,36 @@ struct rsnd_adg { (i < CLKMAX) && \ ((pos) = adg->clk[i]); \ i++) +#define for_each_rsnd_clkout(pos, adg, i) \ + for (i = 0; \ + (i < CLKOUTMAX) && \ + ((pos) = adg->clkout[i]); \ + i++) #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg) +static u32 rsnd_adg_calculate_rbgx(unsigned long div) +{ + int i, ratio; + + if (!div) + return 0; + + for (i = 3; i >= 0; i--) { + ratio = 2 << (i * 2); + if (0 == (div % ratio)) + return (u32)((i << 8) | ((div / ratio) - 1)); + } + + return ~0; +} static u32 rsnd_adg_ssi_ws_timing_gen2(struct rsnd_dai_stream *io) { struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io); - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); int id = rsnd_mod_id(mod); int ws = id; - if (rsnd_ssi_is_pin_sharing(rsnd_ssi_mod_get(priv, id))) { + if (rsnd_ssi_is_pin_sharing(io)) { switch (id) { case 1: case 2: @@ -60,6 +93,9 @@ static u32 rsnd_adg_ssi_ws_timing_gen2(struct rsnd_dai_stream *io) int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_adg *adg = rsnd_priv_to_adg(priv); + struct rsnd_mod *adg_mod = rsnd_mod_get(adg); int id = rsnd_mod_id(mod); int shift = (id % 2) ? 16 : 0; u32 mask, val; @@ -69,21 +105,26 @@ int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *mod, val = val << shift; mask = 0xffff << shift; - rsnd_mod_bset(mod, CMDOUT_TIMSEL, mask, val); + rsnd_mod_bset(adg_mod, CMDOUT_TIMSEL, mask, val); return 0; } -static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *mod, +static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *src_mod, struct rsnd_dai_stream *io, u32 timsel) { + struct rsnd_priv *priv = rsnd_mod_to_priv(src_mod); + struct rsnd_adg *adg = rsnd_priv_to_adg(priv); + struct rsnd_mod *adg_mod = rsnd_mod_get(adg); int is_play = rsnd_io_is_play(io); - int id = rsnd_mod_id(mod); + int id = rsnd_mod_id(src_mod); int shift = (id % 2) ? 16 : 0; u32 mask, ws; u32 in, out; + rsnd_mod_confirm_src(src_mod); + ws = rsnd_adg_ssi_ws_timing_gen2(io); in = (is_play) ? timsel : ws; @@ -95,37 +136,38 @@ static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *mod, switch (id / 2) { case 0: - rsnd_mod_bset(mod, SRCIN_TIMSEL0, mask, in); - rsnd_mod_bset(mod, SRCOUT_TIMSEL0, mask, out); + rsnd_mod_bset(adg_mod, SRCIN_TIMSEL0, mask, in); + rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL0, mask, out); break; case 1: - rsnd_mod_bset(mod, SRCIN_TIMSEL1, mask, in); - rsnd_mod_bset(mod, SRCOUT_TIMSEL1, mask, out); + rsnd_mod_bset(adg_mod, SRCIN_TIMSEL1, mask, in); + rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL1, mask, out); break; case 2: - rsnd_mod_bset(mod, SRCIN_TIMSEL2, mask, in); - rsnd_mod_bset(mod, SRCOUT_TIMSEL2, mask, out); + rsnd_mod_bset(adg_mod, SRCIN_TIMSEL2, mask, in); + rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL2, mask, out); break; case 3: - rsnd_mod_bset(mod, SRCIN_TIMSEL3, mask, in); - rsnd_mod_bset(mod, SRCOUT_TIMSEL3, mask, out); + rsnd_mod_bset(adg_mod, SRCIN_TIMSEL3, mask, in); + rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL3, mask, out); break; case 4: - rsnd_mod_bset(mod, SRCIN_TIMSEL4, mask, in); - rsnd_mod_bset(mod, SRCOUT_TIMSEL4, mask, out); + rsnd_mod_bset(adg_mod, SRCIN_TIMSEL4, mask, in); + rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL4, mask, out); break; } return 0; } -int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod, +int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *src_mod, struct rsnd_dai_stream *io, unsigned int src_rate, unsigned int dst_rate) { - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_priv *priv = rsnd_mod_to_priv(src_mod); struct rsnd_adg *adg = rsnd_priv_to_adg(priv); + struct rsnd_mod *adg_mod = rsnd_mod_get(adg); struct device *dev = rsnd_priv_to_dev(priv); int idx, sel, div, step, ret; u32 val, en; @@ -134,10 +176,12 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod, clk_get_rate(adg->clk[CLKA]), /* 0000: CLKA */ clk_get_rate(adg->clk[CLKB]), /* 0001: CLKB */ clk_get_rate(adg->clk[CLKC]), /* 0010: CLKC */ - adg->rbga_rate_for_441khz_div_6,/* 0011: RBGA */ - adg->rbgb_rate_for_48khz_div_6, /* 0100: RBGB */ + adg->rbga_rate_for_441khz, /* 0011: RBGA */ + adg->rbgb_rate_for_48khz, /* 0100: RBGB */ }; + rsnd_mod_confirm_src(src_mod); + min = ~0; val = 0; en = 0; @@ -175,25 +219,27 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod, return -EIO; } - ret = rsnd_adg_set_src_timsel_gen2(mod, io, val); + ret = rsnd_adg_set_src_timsel_gen2(src_mod, io, val); if (ret < 0) { dev_err(dev, "timsel error\n"); return ret; } - rsnd_mod_bset(mod, DIV_EN, en, en); + rsnd_mod_bset(adg_mod, DIV_EN, en, en); dev_dbg(dev, "convert rate %d <-> %d\n", src_rate, dst_rate); return 0; } -int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod, +int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *src_mod, struct rsnd_dai_stream *io) { u32 val = rsnd_adg_ssi_ws_timing_gen2(io); - return rsnd_adg_set_src_timsel_gen2(mod, io, val); + rsnd_mod_confirm_src(src_mod); + + return rsnd_adg_set_src_timsel_gen2(src_mod, io, val); } int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv, @@ -202,6 +248,7 @@ int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv, unsigned int dst_rate) { struct rsnd_adg *adg = rsnd_priv_to_adg(priv); + struct rsnd_mod *adg_mod = rsnd_mod_get(adg); struct device *dev = rsnd_priv_to_dev(priv); int idx, sel, div, shift; u32 mask, val; @@ -211,8 +258,8 @@ int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv, clk_get_rate(adg->clk[CLKB]), /* 001: CLKB */ clk_get_rate(adg->clk[CLKC]), /* 010: CLKC */ 0, /* 011: MLBCLK (not used) */ - adg->rbga_rate_for_441khz_div_6,/* 100: RBGA */ - adg->rbgb_rate_for_48khz_div_6, /* 101: RBGB */ + adg->rbga_rate_for_441khz, /* 100: RBGA */ + adg->rbgb_rate_for_48khz, /* 101: RBGB */ }; /* find div (= 1/128, 1/256, 1/512, 1/1024, 1/2048 */ @@ -238,13 +285,13 @@ find_rate: switch (id / 4) { case 0: - rsnd_mod_bset(mod, AUDIO_CLK_SEL3, mask, val); + rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL3, mask, val); break; case 1: - rsnd_mod_bset(mod, AUDIO_CLK_SEL4, mask, val); + rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL4, mask, val); break; case 2: - rsnd_mod_bset(mod, AUDIO_CLK_SEL5, mask, val); + rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL5, mask, val); break; } @@ -257,12 +304,17 @@ find_rate: return 0; } -static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val) +static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) { - int id = rsnd_mod_id(mod); + struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod); + struct rsnd_adg *adg = rsnd_priv_to_adg(priv); + struct rsnd_mod *adg_mod = rsnd_mod_get(adg); + int id = rsnd_mod_id(ssi_mod); int shift = (id % 4) * 8; u32 mask = 0xFF << shift; + rsnd_mod_confirm_ssi(ssi_mod); + val = val << shift; /* @@ -274,13 +326,13 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val) switch (id / 4) { case 0: - rsnd_mod_bset(mod, AUDIO_CLK_SEL0, mask, val); + rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL0, mask, val); break; case 1: - rsnd_mod_bset(mod, AUDIO_CLK_SEL1, mask, val); + rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL1, mask, val); break; case 2: - rsnd_mod_bset(mod, AUDIO_CLK_SEL2, mask, val); + rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL2, mask, val); break; } } @@ -326,14 +378,14 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate) } /* - * find 1/6 clock from BRGA/BRGB + * find divided clock from BRGA/BRGB */ - if (rate == adg->rbga_rate_for_441khz_div_6) { + if (rate == adg->rbga_rate_for_441khz) { data = 0x10; goto found_clock; } - if (rate == adg->rbgb_rate_for_48khz_div_6) { + if (rate == adg->rbgb_rate_for_48khz) { data = 0x20; goto found_clock; } @@ -342,29 +394,60 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate) found_clock: - /* see rsnd_adg_ssi_clk_init() */ - rsnd_mod_bset(mod, SSICKR, 0x00FF0000, adg->ckr); - rsnd_mod_write(mod, BRRA, 0x00000002); /* 1/6 */ - rsnd_mod_write(mod, BRRB, 0x00000002); /* 1/6 */ - /* * This "mod" = "ssi" here. * we can get "ssi id" from mod */ rsnd_adg_set_ssi_clk(mod, data); - dev_dbg(dev, "ADG: ssi%d selects clk%d = %d", - rsnd_mod_id(mod), i, rate); + dev_dbg(dev, "ADG: %s[%d] selects 0x%x for %d\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), + data, rate); return 0; } -static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) +static void rsnd_adg_get_clkin(struct rsnd_priv *priv, + struct rsnd_adg *adg) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; + static const char * const clk_name[] = { + [CLKA] = "clk_a", + [CLKB] = "clk_b", + [CLKC] = "clk_c", + [CLKI] = "clk_i", + }; + int i; + + for (i = 0; i < CLKMAX; i++) { + clk = devm_clk_get(dev, clk_name[i]); + adg->clk[i] = IS_ERR(clk) ? NULL : clk; + } + + for_each_rsnd_clk(clk, adg, i) + dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk)); +} + +static void rsnd_adg_get_clkout(struct rsnd_priv *priv, + struct rsnd_adg *adg) { struct clk *clk; - unsigned long rate; - u32 ckr; + struct rsnd_mod *adg_mod = rsnd_mod_get(adg); + struct device *dev = rsnd_priv_to_dev(priv); + struct device_node *np = dev->of_node; + u32 ckr, rbgx, rbga, rbgb; + u32 rate, req_rate, div; + uint32_t count = 0; + unsigned long req_48kHz_rate, req_441kHz_rate; int i; + const char *parent_clk_name = NULL; + static const char * const clkout_name[] = { + [CLKOUT] = "audio_clkout", + [CLKOUT1] = "audio_clkout1", + [CLKOUT2] = "audio_clkout2", + [CLKOUT3] = "audio_clkout3", + }; int brg_table[] = { [CLKA] = 0x0, [CLKB] = 0x1, @@ -372,19 +455,34 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) [CLKI] = 0x2, }; + of_property_read_u32(np, "#clock-cells", &count); + + /* + * ADG supports BRRA/BRRB output only + * this means all clkout0/1/2/3 will be same rate + */ + of_property_read_u32(np, "clock-frequency", &req_rate); + req_48kHz_rate = 0; + req_441kHz_rate = 0; + if (0 == (req_rate % 44100)) + req_441kHz_rate = req_rate; + if (0 == (req_rate % 48000)) + req_48kHz_rate = req_rate; + /* * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC * have 44.1kHz or 48kHz base clocks for now. * * SSI itself can divide parent clock by 1/1 - 1/16 - * So, BRGA outputs 44.1kHz base parent clock 1/32, - * and, BRGB outputs 48.0kHz base parent clock 1/32 here. * see * rsnd_adg_ssi_clk_try_start() + * rsnd_ssi_master_clk_start() */ ckr = 0; - adg->rbga_rate_for_441khz_div_6 = 0; - adg->rbgb_rate_for_48khz_div_6 = 0; + rbga = 2; /* default 1/6 */ + rbgb = 2; /* default 1/6 */ + adg->rbga_rate_for_441khz = 0; + adg->rbgb_rate_for_48khz = 0; for_each_rsnd_clk(clk, adg, i) { rate = clk_get_rate(clk); @@ -392,19 +490,86 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) continue; /* RBGA */ - if (!adg->rbga_rate_for_441khz_div_6 && (0 == rate % 44100)) { - adg->rbga_rate_for_441khz_div_6 = rate / 6; - ckr |= brg_table[i] << 20; + if (!adg->rbga_rate_for_441khz && (0 == rate % 44100)) { + div = 6; + if (req_441kHz_rate) + div = rate / req_441kHz_rate; + rbgx = rsnd_adg_calculate_rbgx(div); + if (BRRx_MASK(rbgx) == rbgx) { + rbga = rbgx; + adg->rbga_rate_for_441khz = rate / div; + ckr |= brg_table[i] << 20; + if (req_441kHz_rate) + parent_clk_name = __clk_get_name(clk); + } } /* RBGB */ - if (!adg->rbgb_rate_for_48khz_div_6 && (0 == rate % 48000)) { - adg->rbgb_rate_for_48khz_div_6 = rate / 6; - ckr |= brg_table[i] << 16; + if (!adg->rbgb_rate_for_48khz && (0 == rate % 48000)) { + div = 6; + if (req_48kHz_rate) + div = rate / req_48kHz_rate; + rbgx = rsnd_adg_calculate_rbgx(div); + if (BRRx_MASK(rbgx) == rbgx) { + rbgb = rbgx; + adg->rbgb_rate_for_48khz = rate / div; + ckr |= brg_table[i] << 16; + if (req_48kHz_rate) { + parent_clk_name = __clk_get_name(clk); + ckr |= 0x80000000; + } + } } } - adg->ckr = ckr; + /* + * ADG supports BRRA/BRRB output only. + * this means all clkout0/1/2/3 will be * same rate + */ + + /* + * for clkout + */ + if (!count) { + clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], + parent_clk_name, + (parent_clk_name) ? + 0 : CLK_IS_ROOT, req_rate); + if (!IS_ERR(clk)) { + adg->clkout[CLKOUT] = clk; + of_clk_add_provider(np, of_clk_src_simple_get, clk); + } + } + /* + * for clkout0/1/2/3 + */ + else { + for (i = 0; i < CLKOUTMAX; i++) { + clk = clk_register_fixed_rate(dev, clkout_name[i], + parent_clk_name, + (parent_clk_name) ? + 0 : CLK_IS_ROOT, + req_rate); + if (!IS_ERR(clk)) { + adg->onecell.clks = adg->clkout; + adg->onecell.clk_num = CLKOUTMAX; + + adg->clkout[i] = clk; + + of_clk_add_provider(np, of_clk_src_onecell_get, + &adg->onecell); + } + } + } + + rsnd_mod_bset(adg_mod, SSICKR, 0x00FF0000, ckr); + rsnd_mod_write(adg_mod, BRRA, rbga); + rsnd_mod_write(adg_mod, BRRB, rbgb); + + for_each_rsnd_clkout(clk, adg, i) + dev_dbg(dev, "clkout %d : %p : %ld\n", i, clk, clk_get_rate(clk)); + dev_dbg(dev, "SSICKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", + ckr, rbga, rbgb); } int rsnd_adg_probe(struct platform_device *pdev, @@ -413,8 +578,6 @@ int rsnd_adg_probe(struct platform_device *pdev, { struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); - struct clk *clk; - int i; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); if (!adg) { @@ -422,15 +585,16 @@ int rsnd_adg_probe(struct platform_device *pdev, return -ENOMEM; } - adg->clk[CLKA] = devm_clk_get(dev, "clk_a"); - adg->clk[CLKB] = devm_clk_get(dev, "clk_b"); - adg->clk[CLKC] = devm_clk_get(dev, "clk_c"); - adg->clk[CLKI] = devm_clk_get(dev, "clk_i"); - - for_each_rsnd_clk(clk, adg, i) - dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk)); + /* + * ADG is special module. + * Use ADG mod without rsnd_mod_init() to make debug easy + * for rsnd_write/rsnd_read + */ + adg->mod.ops = &adg_ops; + adg->mod.priv = priv; - rsnd_adg_ssi_clk_init(priv, adg); + rsnd_adg_get_clkin(priv, adg); + rsnd_adg_get_clkout(priv, adg); priv->adg = adg; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f3feed5ce9b6..deed48ef28b8 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -110,6 +110,7 @@ static const struct rsnd_of_data rsnd_of_data_gen2 = { static const struct of_device_id rsnd_of_match[] = { { .compatible = "renesas,rcar_sound-gen1", .data = &rsnd_of_data_gen1 }, { .compatible = "renesas,rcar_sound-gen2", .data = &rsnd_of_data_gen2 }, + { .compatible = "renesas,rcar_sound-gen3", .data = &rsnd_of_data_gen2 }, /* gen2 compatible */ {}, }; MODULE_DEVICE_TABLE(of, rsnd_of_match); @@ -126,6 +127,17 @@ MODULE_DEVICE_TABLE(of, rsnd_of_match); #define rsnd_info_id(priv, io, name) \ ((io)->info->name - priv->info->name##_info) +void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type) +{ + if (mod->type != type) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_warn(dev, "%s[%d] is not your expected module\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + } +} + /* * rsnd_mod functions */ @@ -288,7 +300,7 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) /* * rsnd_dai functions */ -#define __rsnd_mod_call(mod, io, func, param...) \ +#define rsnd_mod_call(mod, io, func, param...) \ ({ \ struct rsnd_priv *priv = rsnd_mod_to_priv(mod); \ struct device *dev = rsnd_priv_to_dev(priv); \ @@ -296,24 +308,17 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) u8 val = (mod->status >> __rsnd_mod_shift_##func) & 0xF; \ u8 add = ((val + __rsnd_mod_add_##func) & 0xF); \ int ret = 0; \ - int called = 0; \ - if (val == __rsnd_mod_call_##func) { \ - called = 1; \ - ret = (mod)->ops->func(mod, io, param); \ - } \ + int call = (val == __rsnd_mod_call_##func) && (mod)->ops->func; \ mod->status = (mod->status & ~mask) + \ (add << __rsnd_mod_shift_##func); \ - dev_dbg(dev, "%s[%d] 0x%08x %s\n", \ - rsnd_mod_name(mod), rsnd_mod_id(mod), mod->status, \ - called ? #func : ""); \ + dev_dbg(dev, "%s[%d]\t0x%08x %s\n", \ + rsnd_mod_name(mod), rsnd_mod_id(mod), \ + mod->status, call ? #func : ""); \ + if (call) \ + ret = (mod)->ops->func(mod, io, param); \ ret; \ }) -#define rsnd_mod_call(mod, io, func, param...) \ - (!(mod) ? -ENODEV : \ - !((mod)->ops->func) ? 0 : \ - __rsnd_mod_call(mod, io, func, param)) - #define rsnd_dai_call(fn, io, param...) \ ({ \ struct rsnd_mod *mod; \ @@ -322,9 +327,7 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) mod = (io)->mod[i]; \ if (!mod) \ continue; \ - ret = rsnd_mod_call(mod, io, fn, param); \ - if (ret < 0) \ - break; \ + ret |= rsnd_mod_call(mod, io, fn, param); \ } \ ret; \ }) @@ -490,16 +493,10 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, break; case SNDRV_PCM_TRIGGER_STOP: ret = rsnd_dai_call(stop, io, priv); - if (ret < 0) - goto dai_trigger_end; - ret = rsnd_dai_call(quit, io, priv); - if (ret < 0) - goto dai_trigger_end; + ret |= rsnd_dai_call(quit, io, priv); - ret = rsnd_platform_call(priv, dai, stop, ssi_id); - if (ret < 0) - goto dai_trigger_end; + ret |= rsnd_platform_call(priv, dai, stop, ssi_id); rsnd_dai_stream_quit(io); break; @@ -1224,20 +1221,11 @@ static int rsnd_probe(struct platform_device *pdev) }; int ret, i; - info = NULL; - of_data = NULL; - if (of_id) { - info = devm_kzalloc(&pdev->dev, - sizeof(struct rcar_snd_info), GFP_KERNEL); - of_data = of_id->data; - } else { - info = pdev->dev.platform_data; - } - - if (!info) { - dev_err(dev, "driver needs R-Car sound information\n"); - return -ENODEV; - } + info = devm_kzalloc(&pdev->dev, sizeof(struct rcar_snd_info), + GFP_KERNEL); + if (!info) + return -ENOMEM; + of_data = of_id->data; /* * init priv data diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 05498bba5874..3cb214ab848b 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -35,7 +35,7 @@ static int rsnd_ctu_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_hw_start(mod); + rsnd_mod_power_on(mod); rsnd_ctu_initialize_lock(mod); @@ -50,7 +50,7 @@ static int rsnd_ctu_quit(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_hw_stop(mod); + rsnd_mod_power_off(mod); return 0; } @@ -66,7 +66,7 @@ struct rsnd_mod *rsnd_ctu_mod_get(struct rsnd_priv *priv, int id) if (WARN_ON(id < 0 || id >= rsnd_ctu_nr(priv))) id = 0; - return &((struct rsnd_ctu *)(priv->ctu) + id)->mod; + return rsnd_mod_get((struct rsnd_ctu *)(priv->ctu) + id); } static void rsnd_of_parse_ctu(struct platform_device *pdev, @@ -118,10 +118,8 @@ int rsnd_ctu_probe(struct platform_device *pdev, int i, nr, ret; /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) { - dev_warn(dev, "CTU is not supported on Gen1\n"); - return -EINVAL; - } + if (rsnd_is_gen1(priv)) + return 0; rsnd_of_parse_ctu(pdev, of_data, priv); @@ -150,7 +148,7 @@ int rsnd_ctu_probe(struct platform_device *pdev, ctu->info = &info->ctu_info[i]; - ret = rsnd_mod_init(priv, &ctu->mod, &rsnd_ctu_ops, + ret = rsnd_mod_init(priv, rsnd_mod_get(ctu), &rsnd_ctu_ops, clk, RSND_MOD_CTU, i); if (ret) return ret; @@ -166,6 +164,6 @@ void rsnd_ctu_remove(struct platform_device *pdev, int i; for_each_rsnd_ctu(ctu, priv, i) { - rsnd_mod_quit(&ctu->mod); + rsnd_mod_quit(rsnd_mod_get(ctu)); } } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index bfbb8a5e93bd..5d084d040961 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -470,7 +470,7 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io, dev_err(dev, "DVC is selected without SRC\n"); /* use SSIU or SSI ? */ - if (is_ssi && rsnd_ssi_use_busif(io, mod)) + if (is_ssi && rsnd_ssi_use_busif(io)) is_ssi++; return (is_from) ? diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 57796387d482..58f690900e6d 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -153,7 +153,7 @@ static int rsnd_dvc_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_hw_start(mod); + rsnd_mod_power_on(mod); rsnd_dvc_soft_reset(mod); @@ -175,7 +175,7 @@ static int rsnd_dvc_quit(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_hw_stop(mod); + rsnd_mod_power_off(mod); return 0; } @@ -282,7 +282,7 @@ struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id) if (WARN_ON(id < 0 || id >= rsnd_dvc_nr(priv))) id = 0; - return &((struct rsnd_dvc *)(priv->dvc) + id)->mod; + return rsnd_mod_get((struct rsnd_dvc *)(priv->dvc) + id); } static void rsnd_of_parse_dvc(struct platform_device *pdev, @@ -333,10 +333,8 @@ int rsnd_dvc_probe(struct platform_device *pdev, int i, nr, ret; /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) { - dev_warn(dev, "CMD is not supported on Gen1\n"); - return -EINVAL; - } + if (rsnd_is_gen1(priv)) + return 0; rsnd_of_parse_dvc(pdev, of_data, priv); @@ -361,7 +359,7 @@ int rsnd_dvc_probe(struct platform_device *pdev, dvc->info = &info->dvc_info[i]; - ret = rsnd_mod_init(priv, &dvc->mod, &rsnd_dvc_ops, + ret = rsnd_mod_init(priv, rsnd_mod_get(dvc), &rsnd_dvc_ops, clk, RSND_MOD_DVC, i); if (ret) return ret; @@ -377,6 +375,6 @@ void rsnd_dvc_remove(struct platform_device *pdev, int i; for_each_rsnd_dvc(dvc, priv, i) { - rsnd_mod_quit(&dvc->mod); + rsnd_mod_quit(rsnd_mod_get(dvc)); } } diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index f04d17bc6e3d..76da7620904c 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -22,13 +22,15 @@ #include "rsnd.h" struct rsnd_gen { - void __iomem *base[RSND_BASE_MAX]; - struct rsnd_gen_ops *ops; + /* RSND_BASE_MAX base */ + void __iomem *base[RSND_BASE_MAX]; + phys_addr_t res[RSND_BASE_MAX]; struct regmap *regmap[RSND_BASE_MAX]; + + /* RSND_REG_MAX base */ struct regmap_field *regs[RSND_REG_MAX]; - phys_addr_t res[RSND_REG_MAX]; }; #define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) @@ -79,11 +81,11 @@ u32 rsnd_read(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return 0; + regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); + dev_dbg(dev, "r %s[%d] - %4d : %08x\n", rsnd_mod_name(mod), rsnd_mod_id(mod), reg, val); - regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); - return val; } @@ -182,6 +184,7 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, if (IS_ERR(regmap)) return PTR_ERR(regmap); + /* RSND_BASE_MAX base */ gen->base[reg_id] = base; gen->regmap[reg_id] = regmap; gen->res[reg_id] = res->start; @@ -198,6 +201,7 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, if (IS_ERR(regs)) return PTR_ERR(regs); + /* RSND_REG_MAX base */ gen->regs[conf[i].idx] = regs; } diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 0d5c102db6f5..953dd0be9b60 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -58,7 +58,7 @@ static int rsnd_mix_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_hw_start(mod); + rsnd_mod_power_on(mod); rsnd_mix_soft_reset(mod); @@ -83,7 +83,7 @@ static int rsnd_mix_quit(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_hw_stop(mod); + rsnd_mod_power_off(mod); return 0; } @@ -99,7 +99,7 @@ struct rsnd_mod *rsnd_mix_mod_get(struct rsnd_priv *priv, int id) if (WARN_ON(id < 0 || id >= rsnd_mix_nr(priv))) id = 0; - return &((struct rsnd_mix *)(priv->mix) + id)->mod; + return rsnd_mod_get((struct rsnd_mix *)(priv->mix) + id); } static void rsnd_of_parse_mix(struct platform_device *pdev, @@ -151,10 +151,8 @@ int rsnd_mix_probe(struct platform_device *pdev, int i, nr, ret; /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) { - dev_warn(dev, "MIX is not supported on Gen1\n"); - return -EINVAL; - } + if (rsnd_is_gen1(priv)) + return 0; rsnd_of_parse_mix(pdev, of_data, priv); @@ -179,7 +177,7 @@ int rsnd_mix_probe(struct platform_device *pdev, mix->info = &info->mix_info[i]; - ret = rsnd_mod_init(priv, &mix->mod, &rsnd_mix_ops, + ret = rsnd_mod_init(priv, rsnd_mod_get(mix), &rsnd_mix_ops, clk, RSND_MOD_MIX, i); if (ret) return ret; @@ -195,6 +193,6 @@ void rsnd_mix_remove(struct platform_device *pdev, int i; for_each_rsnd_mix(mix, priv, i) { - rsnd_mod_quit(&mix->mod); + rsnd_mod_quit(rsnd_mod_get(mix)); } } diff --git a/sound/soc/sh/rcar/rcar_snd.h b/sound/soc/sh/rcar/rcar_snd.h new file mode 100644 index 000000000000..d8e33d38da43 --- /dev/null +++ b/sound/soc/sh/rcar/rcar_snd.h @@ -0,0 +1,117 @@ +/* + * Renesas R-Car SRU/SCU/SSIU/SSI support + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef RCAR_SND_H +#define RCAR_SND_H + + +#define RSND_GEN1_SRU 0 +#define RSND_GEN1_ADG 1 +#define RSND_GEN1_SSI 2 + +#define RSND_GEN2_SCU 0 +#define RSND_GEN2_ADG 1 +#define RSND_GEN2_SSIU 2 +#define RSND_GEN2_SSI 3 + +#define RSND_BASE_MAX 4 + +/* + * flags + * + * 0xAB000000 + * + * A : clock sharing settings + * B : SSI direction + */ +#define RSND_SSI_CLK_PIN_SHARE (1 << 31) +#define RSND_SSI_NO_BUSIF (1 << 30) /* SSI+DMA without BUSIF */ + +#define RSND_SSI(_dma_id, _irq, _flags) \ +{ .dma_id = _dma_id, .irq = _irq, .flags = _flags } +#define RSND_SSI_UNUSED \ +{ .dma_id = -1, .irq = -1, .flags = 0 } + +struct rsnd_ssi_platform_info { + int dma_id; + int irq; + u32 flags; +}; + +#define RSND_SRC(rate, _dma_id) \ +{ .convert_rate = rate, .dma_id = _dma_id, } +#define RSND_SRC_UNUSED \ +{ .convert_rate = 0, .dma_id = -1, } + +struct rsnd_src_platform_info { + u32 convert_rate; /* sampling rate convert */ + int dma_id; /* for Gen2 SCU */ + int irq; +}; + +/* + * flags + */ +struct rsnd_ctu_platform_info { + u32 flags; +}; + +struct rsnd_mix_platform_info { + u32 flags; +}; + +struct rsnd_dvc_platform_info { + u32 flags; +}; + +struct rsnd_dai_path_info { + struct rsnd_ssi_platform_info *ssi; + struct rsnd_src_platform_info *src; + struct rsnd_ctu_platform_info *ctu; + struct rsnd_mix_platform_info *mix; + struct rsnd_dvc_platform_info *dvc; +}; + +struct rsnd_dai_platform_info { + struct rsnd_dai_path_info playback; + struct rsnd_dai_path_info capture; +}; + +/* + * flags + * + * 0x0000000A + * + * A : generation + */ +#define RSND_GEN_MASK (0xF << 0) +#define RSND_GEN1 (1 << 0) /* fixme */ +#define RSND_GEN2 (2 << 0) /* fixme */ + +struct rcar_snd_info { + u32 flags; + struct rsnd_ssi_platform_info *ssi_info; + int ssi_info_nr; + struct rsnd_src_platform_info *src_info; + int src_info_nr; + struct rsnd_ctu_platform_info *ctu_info; + int ctu_info_nr; + struct rsnd_mix_platform_info *mix_info; + int mix_info_nr; + struct rsnd_dvc_platform_info *dvc_info; + int dvc_info_nr; + struct rsnd_dai_platform_info *dai_info; + int dai_info_nr; + int (*start)(int id); + int (*stop)(int id); +}; + +#endif diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 7a0e52b4640a..085329878525 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -21,10 +21,11 @@ #include <linux/of_irq.h> #include <linux/sh_dma.h> #include <linux/workqueue.h> -#include <sound/rcar_snd.h> #include <sound/soc.h> #include <sound/pcm_params.h> +#include "rcar_snd.h" + /* * pseudo register * @@ -214,6 +215,7 @@ struct rsnd_dma { }; #define rsnd_dma_to_dmaen(dma) (&(dma)->dma.en) #define rsnd_dma_to_dmapp(dma) (&(dma)->dma.pp) +#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma) void rsnd_dma_start(struct rsnd_dai_stream *io, struct rsnd_dma *dma); void rsnd_dma_stop(struct rsnd_dai_stream *io, struct rsnd_dma *dma); @@ -225,8 +227,6 @@ int rsnd_dma_probe(struct platform_device *pdev, struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, struct rsnd_mod *mod, char *name); -#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma) - /* * R-Car sound mod */ @@ -330,8 +330,9 @@ struct rsnd_mod { #define rsnd_mod_to_priv(mod) ((mod)->priv) #define rsnd_mod_to_dma(mod) (&(mod)->dma) #define rsnd_mod_id(mod) ((mod) ? (mod)->id : -1) -#define rsnd_mod_hw_start(mod) clk_enable((mod)->clk) -#define rsnd_mod_hw_stop(mod) clk_disable((mod)->clk) +#define rsnd_mod_power_on(mod) clk_enable((mod)->clk) +#define rsnd_mod_power_off(mod) clk_disable((mod)->clk) +#define rsnd_mod_get(ip) (&(ip)->mod) int rsnd_mod_init(struct rsnd_priv *priv, struct rsnd_mod *mod, @@ -571,9 +572,12 @@ int rsnd_ssi_probe(struct platform_device *pdev, void rsnd_ssi_remove(struct platform_device *pdev, struct rsnd_priv *priv); struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); -int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod); int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); -int rsnd_ssi_use_busif(struct rsnd_dai_stream *io, struct rsnd_mod *mod); +int rsnd_ssi_use_busif(struct rsnd_dai_stream *io); + +#define rsnd_ssi_is_pin_sharing(io) \ + __rsnd_ssi_is_pin_sharing(rsnd_io_to_mod_ssi(io)) +int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod); /* * R-Car SRC @@ -627,4 +631,15 @@ void rsnd_dvc_remove(struct platform_device *pdev, struct rsnd_priv *priv); struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id); +#ifdef DEBUG +void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type); +#define rsnd_mod_confirm_ssi(mssi) rsnd_mod_make_sure(mssi, RSND_MOD_SSI) +#define rsnd_mod_confirm_src(msrc) rsnd_mod_make_sure(msrc, RSND_MOD_SRC) +#define rsnd_mod_confirm_dvc(mdvc) rsnd_mod_make_sure(mdvc, RSND_MOD_DVC) +#else +#define rsnd_mod_confirm_ssi(mssi) +#define rsnd_mod_confirm_src(msrc) +#define rsnd_mod_confirm_dvc(mdvc) +#endif + #endif diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 89a18e102feb..261b50217c48 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -159,7 +159,7 @@ int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, /* * SSI_MODE1 */ - if (rsnd_ssi_is_pin_sharing(ssi_mod)) { + if (rsnd_ssi_is_pin_sharing(io)) { int shift = -1; switch (ssi_id) { case 1: @@ -352,7 +352,7 @@ static int rsnd_src_init(struct rsnd_mod *mod, { struct rsnd_src *src = rsnd_mod_to_src(mod); - rsnd_mod_hw_start(mod); + rsnd_mod_power_on(mod); rsnd_src_soft_reset(mod); @@ -373,7 +373,7 @@ static int rsnd_src_quit(struct rsnd_mod *mod, struct rsnd_src *src = rsnd_mod_to_src(mod); struct device *dev = rsnd_priv_to_dev(priv); - rsnd_mod_hw_stop(mod); + rsnd_mod_power_off(mod); if (src->err) dev_warn(dev, "%s[%d] under/over flow err = %d\n", @@ -918,11 +918,10 @@ static void rsnd_src_reconvert_update(struct rsnd_dai_stream *io, rsnd_mod_write(mod, SRC_IFSVR, fsrate); } -static int rsnd_src_pcm_new(struct rsnd_mod *mod, +static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd) { - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct rsnd_dai *rdai = rsnd_io_to_rdai(io); struct rsnd_src *src = rsnd_mod_to_src(mod); int ret; @@ -932,12 +931,6 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, */ /* - * Gen1 is not supported - */ - if (rsnd_is_gen1(priv)) - return 0; - - /* * SRC sync convert needs clock master */ if (!rsnd_rdai_is_clk_master(rdai)) @@ -975,7 +968,7 @@ static struct rsnd_mod_ops rsnd_src_gen2_ops = { .start = rsnd_src_start_gen2, .stop = rsnd_src_stop_gen2, .hw_params = rsnd_src_hw_params, - .pcm_new = rsnd_src_pcm_new, + .pcm_new = rsnd_src_pcm_new_gen2, }; struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id) @@ -983,7 +976,7 @@ struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id) if (WARN_ON(id < 0 || id >= rsnd_src_nr(priv))) id = 0; - return &((struct rsnd_src *)(priv->src) + id)->mod; + return rsnd_mod_get((struct rsnd_src *)(priv->src) + id); } static void rsnd_of_parse_src(struct platform_device *pdev, @@ -1043,8 +1036,10 @@ int rsnd_src_probe(struct platform_device *pdev, int i, nr, ret; ops = NULL; - if (rsnd_is_gen1(priv)) + if (rsnd_is_gen1(priv)) { ops = &rsnd_src_gen1_ops; + dev_warn(dev, "Gen1 support will be removed soon\n"); + } if (rsnd_is_gen2(priv)) ops = &rsnd_src_gen2_ops; if (!ops) { @@ -1078,7 +1073,7 @@ int rsnd_src_probe(struct platform_device *pdev, src->info = &info->src_info[i]; - ret = rsnd_mod_init(priv, &src->mod, ops, clk, RSND_MOD_SRC, i); + ret = rsnd_mod_init(priv, rsnd_mod_get(src), ops, clk, RSND_MOD_SRC, i); if (ret) return ret; } @@ -1093,6 +1088,6 @@ void rsnd_src_remove(struct platform_device *pdev, int i; for_each_rsnd_src(src, priv, i) { - rsnd_mod_quit(&src->mod); + rsnd_mod_quit(rsnd_mod_get(src)); } } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index d45b9a7e324e..1427ec21bd7e 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -79,7 +79,6 @@ struct rsnd_ssi { #define rsnd_ssi_nr(priv) ((priv)->ssi_nr) #define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod) -#define rsnd_dma_to_ssi(dma) rsnd_mod_to_ssi(rsnd_dma_to_mod(dma)) #define rsnd_ssi_pio_available(ssi) ((ssi)->info->irq > 0) #define rsnd_ssi_parent(ssi) ((ssi)->parent) #define rsnd_ssi_mode_flags(p) ((p)->info->flags) @@ -87,8 +86,9 @@ struct rsnd_ssi { #define rsnd_ssi_of_node(priv) \ of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, "rcar_sound,ssi") -int rsnd_ssi_use_busif(struct rsnd_dai_stream *io, struct rsnd_mod *mod) +int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) { + struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); int use_busif = 0; @@ -128,10 +128,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, struct rsnd_priv *priv = rsnd_io_to_priv(io); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct device *dev = rsnd_priv_to_dev(priv); - int i, j, ret; - int adg_clk_div_table[] = { - 1, 6, /* see adg.c */ - }; + struct rsnd_mod *mod = rsnd_mod_get(ssi); + int j, ret; int ssi_clk_mul_table[] = { 1, 2, 4, 8, 16, 6, 12, }; @@ -141,28 +139,25 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, /* * Find best clock, and try to start ADG */ - for (i = 0; i < ARRAY_SIZE(adg_clk_div_table); i++) { - for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) { - - /* - * this driver is assuming that - * system word is 64fs (= 2 x 32bit) - * see rsnd_ssi_init() - */ - main_rate = rate / adg_clk_div_table[i] - * 32 * 2 * ssi_clk_mul_table[j]; - - ret = rsnd_adg_ssi_clk_try_start(&ssi->mod, main_rate); - if (0 == ret) { - ssi->cr_clk = FORCE | SWL_32 | - SCKD | SWSD | CKDV(j); - - dev_dbg(dev, "%s[%d] outputs %u Hz\n", - rsnd_mod_name(&ssi->mod), - rsnd_mod_id(&ssi->mod), rate); - - return 0; - } + for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) { + + /* + * this driver is assuming that + * system word is 64fs (= 2 x 32bit) + * see rsnd_ssi_init() + */ + main_rate = rate * 32 * 2 * ssi_clk_mul_table[j]; + + ret = rsnd_adg_ssi_clk_try_start(mod, main_rate); + if (0 == ret) { + ssi->cr_clk = FORCE | SWL_32 | + SCKD | SWSD | CKDV(j); + + dev_dbg(dev, "%s[%d] outputs %u Hz\n", + rsnd_mod_name(mod), + rsnd_mod_id(mod), rate); + + return 0; } } @@ -172,8 +167,10 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, static void rsnd_ssi_master_clk_stop(struct rsnd_ssi *ssi) { + struct rsnd_mod *mod = rsnd_mod_get(ssi); + ssi->cr_clk = 0; - rsnd_adg_ssi_clk_stop(&ssi->mod); + rsnd_adg_ssi_clk_stop(mod); } static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, @@ -182,11 +179,12 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, struct rsnd_priv *priv = rsnd_io_to_priv(io); struct rsnd_dai *rdai = rsnd_io_to_rdai(io); struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_mod *mod = rsnd_mod_get(ssi); u32 cr_mode; u32 cr; if (0 == ssi->usrcnt) { - rsnd_mod_hw_start(&ssi->mod); + rsnd_mod_power_on(mod); if (rsnd_rdai_is_clk_master(rdai)) { struct rsnd_ssi *ssi_parent = rsnd_ssi_parent(ssi); @@ -198,7 +196,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, } } - if (rsnd_ssi_is_dma_mode(&ssi->mod)) { + if (rsnd_ssi_is_dma_mode(mod)) { cr_mode = UIEN | OIEN | /* over/under run */ DMEN; /* DMA : enable DMA */ } else { @@ -210,24 +208,25 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, cr_mode | EN; - rsnd_mod_write(&ssi->mod, SSICR, cr); + rsnd_mod_write(mod, SSICR, cr); /* enable WS continue */ if (rsnd_rdai_is_clk_master(rdai)) - rsnd_mod_write(&ssi->mod, SSIWSR, CONT); + rsnd_mod_write(mod, SSIWSR, CONT); /* clear error status */ - rsnd_mod_write(&ssi->mod, SSISR, 0); + rsnd_mod_write(mod, SSISR, 0); ssi->usrcnt++; dev_dbg(dev, "%s[%d] hw started\n", - rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod)); + rsnd_mod_name(mod), rsnd_mod_id(mod)); } static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi) { - struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod); + struct rsnd_mod *mod = rsnd_mod_get(ssi); + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct rsnd_dai *rdai = rsnd_io_to_rdai(io); struct device *dev = rsnd_priv_to_dev(priv); u32 cr; @@ -247,15 +246,15 @@ static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi) cr = ssi->cr_own | ssi->cr_clk; - rsnd_mod_write(&ssi->mod, SSICR, cr | EN); - rsnd_ssi_status_check(&ssi->mod, DIRQ); + rsnd_mod_write(mod, SSICR, cr | EN); + rsnd_ssi_status_check(mod, DIRQ); /* * disable SSI, * and, wait idle state */ - rsnd_mod_write(&ssi->mod, SSICR, cr); /* disabled all */ - rsnd_ssi_status_check(&ssi->mod, IIRQ); + rsnd_mod_write(mod, SSICR, cr); /* disabled all */ + rsnd_ssi_status_check(mod, IIRQ); if (rsnd_rdai_is_clk_master(rdai)) { struct rsnd_ssi *ssi_parent = rsnd_ssi_parent(ssi); @@ -266,13 +265,13 @@ static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi) rsnd_ssi_master_clk_stop(ssi); } - rsnd_mod_hw_stop(&ssi->mod); + rsnd_mod_power_off(mod); ssi->chan = 0; } dev_dbg(dev, "%s[%d] hw stopped\n", - rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod)); + rsnd_mod_name(mod), rsnd_mod_id(mod)); } /* @@ -371,7 +370,7 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod, /* It will be removed on rsnd_ssi_hw_stop */ ssi->chan = chan; if (ssi_parent) - return rsnd_ssi_hw_params(&ssi_parent->mod, io, + return rsnd_ssi_hw_params(rsnd_mod_get(ssi_parent), io, substream, params); return 0; @@ -379,12 +378,14 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod, static void rsnd_ssi_record_error(struct rsnd_ssi *ssi, u32 status) { + struct rsnd_mod *mod = rsnd_mod_get(ssi); + /* under/over flow error */ if (status & (UIRQ | OIRQ)) { ssi->err++; /* clear error status */ - rsnd_mod_write(&ssi->mod, SSISR, 0); + rsnd_mod_write(mod, SSISR, 0); } } @@ -394,7 +395,7 @@ static int rsnd_ssi_start(struct rsnd_mod *mod, { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - rsnd_src_ssiu_start(mod, io, rsnd_ssi_use_busif(io, mod)); + rsnd_src_ssiu_start(mod, io, rsnd_ssi_use_busif(io)); rsnd_ssi_hw_start(ssi, io); @@ -554,7 +555,7 @@ static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, rsnd_dma_quit(io, rsnd_mod_to_dma(mod)); /* PIO will request IRQ again */ - devm_free_irq(dev, irq, ssi); + devm_free_irq(dev, irq, mod); return 0; } @@ -613,7 +614,7 @@ static struct dma_chan *rsnd_ssi_dma_req(struct rsnd_dai_stream *io, int is_play = rsnd_io_is_play(io); char *name; - if (rsnd_ssi_use_busif(io, mod)) + if (rsnd_ssi_use_busif(io)) name = is_play ? "rxu" : "txu"; else name = is_play ? "rx" : "tx"; @@ -656,10 +657,10 @@ struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id) if (WARN_ON(id < 0 || id >= rsnd_ssi_nr(priv))) id = 0; - return &((struct rsnd_ssi *)(priv->ssi) + id)->mod; + return rsnd_mod_get((struct rsnd_ssi *)(priv->ssi) + id); } -int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod) +int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); @@ -668,10 +669,12 @@ int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod) static void rsnd_ssi_parent_setup(struct rsnd_priv *priv, struct rsnd_ssi *ssi) { - if (!rsnd_ssi_is_pin_sharing(&ssi->mod)) + struct rsnd_mod *mod = rsnd_mod_get(ssi); + + if (!__rsnd_ssi_is_pin_sharing(mod)) return; - switch (rsnd_mod_id(&ssi->mod)) { + switch (rsnd_mod_id(mod)) { case 1: case 2: ssi->parent = rsnd_mod_to_ssi(rsnd_ssi_mod_get(priv, 0)); @@ -697,9 +700,6 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, struct device *dev = &pdev->dev; int nr, i; - if (!of_data) - return; - node = rsnd_ssi_of_node(priv); if (!node) return; @@ -794,7 +794,8 @@ int rsnd_ssi_probe(struct platform_device *pdev, else if (rsnd_ssi_pio_available(ssi)) ops = &rsnd_ssi_pio_ops; - ret = rsnd_mod_init(priv, &ssi->mod, ops, clk, RSND_MOD_SSI, i); + ret = rsnd_mod_init(priv, rsnd_mod_get(ssi), ops, clk, + RSND_MOD_SSI, i); if (ret) return ret; @@ -811,6 +812,6 @@ void rsnd_ssi_remove(struct platform_device *pdev, int i; for_each_rsnd_ssi(ssi, priv, i) { - rsnd_mod_quit(&ssi->mod); + rsnd_mod_quit(rsnd_mod_get(ssi)); } } diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index abb0d956231c..76b2ab8c2b4a 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -738,7 +738,7 @@ static int siu_probe(struct platform_device *pdev) struct siu_info *info; int ret; - info = kmalloc(sizeof(*info), GFP_KERNEL); + info = devm_kmalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; siu_i2s_data = info; @@ -746,7 +746,7 @@ static int siu_probe(struct platform_device *pdev) ret = request_firmware(&fw_entry, "siu_spb.bin", &pdev->dev); if (ret) - goto ereqfw; + return ret; /* * Loaded firmware is "const" - read only, but we have to modify it in @@ -757,89 +757,52 @@ static int siu_probe(struct platform_device *pdev) release_firmware(fw_entry); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - ret = -ENODEV; - goto egetres; - } + if (!res) + return -ENODEV; - region = request_mem_region(res->start, resource_size(res), - pdev->name); + region = devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name); if (!region) { dev_err(&pdev->dev, "SIU region already claimed\n"); - ret = -EBUSY; - goto ereqmemreg; + return -EBUSY; } - ret = -ENOMEM; - info->pram = ioremap(res->start, PRAM_SIZE); + info->pram = devm_ioremap(&pdev->dev, res->start, PRAM_SIZE); if (!info->pram) - goto emappram; - info->xram = ioremap(res->start + XRAM_OFFSET, XRAM_SIZE); + return -ENOMEM; + info->xram = devm_ioremap(&pdev->dev, res->start + XRAM_OFFSET, + XRAM_SIZE); if (!info->xram) - goto emapxram; - info->yram = ioremap(res->start + YRAM_OFFSET, YRAM_SIZE); + return -ENOMEM; + info->yram = devm_ioremap(&pdev->dev, res->start + YRAM_OFFSET, + YRAM_SIZE); if (!info->yram) - goto emapyram; - info->reg = ioremap(res->start + REG_OFFSET, resource_size(res) - - REG_OFFSET); + return -ENOMEM; + info->reg = devm_ioremap(&pdev->dev, res->start + REG_OFFSET, + resource_size(res) - REG_OFFSET); if (!info->reg) - goto emapreg; + return -ENOMEM; dev_set_drvdata(&pdev->dev, info); /* register using ARRAY version so we can keep dai name */ - ret = snd_soc_register_component(&pdev->dev, &siu_i2s_component, - &siu_i2s_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &siu_i2s_component, + &siu_i2s_dai, 1); if (ret < 0) - goto edaiinit; + return ret; - ret = snd_soc_register_platform(&pdev->dev, &siu_platform); + ret = devm_snd_soc_register_platform(&pdev->dev, &siu_platform); if (ret < 0) - goto esocregp; + return ret; pm_runtime_enable(&pdev->dev); - return ret; - -esocregp: - snd_soc_unregister_component(&pdev->dev); -edaiinit: - iounmap(info->reg); -emapreg: - iounmap(info->yram); -emapyram: - iounmap(info->xram); -emapxram: - iounmap(info->pram); -emappram: - release_mem_region(res->start, resource_size(res)); -ereqmemreg: -egetres: -ereqfw: - kfree(info); - - return ret; + return 0; } static int siu_remove(struct platform_device *pdev) { - struct siu_info *info = dev_get_drvdata(&pdev->dev); - struct resource *res; - pm_runtime_disable(&pdev->dev); - - snd_soc_unregister_platform(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - - iounmap(info->reg); - iounmap(info->yram); - iounmap(info->xram); - iounmap(info->pram); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res) - release_mem_region(res->start, resource_size(res)); - kfree(info); - return 0; } diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 025c38fbe3c0..12a9820feac1 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -612,8 +612,15 @@ static struct snd_compr_ops soc_compr_dyn_ops = { .get_codec_caps = soc_compr_get_codec_caps }; -/* create a new compress */ -int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +/** + * snd_soc_new_compress - create a new compress. + * + * @rtd: The runtime for which we will create compress + * @num: the device index number (zero based - shared with normal PCMs) + * + * Return: 0 for success, else error. + */ +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; @@ -703,3 +710,4 @@ compr_err: kfree(compr); return ret; } +EXPORT_SYMBOL_GPL(snd_soc_new_compress); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6173d15236c3..24b096066a07 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1370,9 +1370,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) soc_dpcm_debugfs_add(rtd); #endif - if (cpu_dai->driver->compress_dai) { + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = soc_new_compress(rtd, num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -3291,13 +3291,38 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); +static int snd_soc_of_get_slot_mask(struct device_node *np, + const char *prop_name, + unsigned int *mask) +{ + u32 val; + const __be32 *of_slot_mask = of_get_property(np, prop_name, &val); + int i; + + if (!of_slot_mask) + return 0; + val /= sizeof(u32); + for (i = 0; i < val; i++) + if (be32_to_cpup(&of_slot_mask[i])) + *mask |= (1 << i); + + return val; +} + int snd_soc_of_parse_tdm_slot(struct device_node *np, + unsigned int *tx_mask, + unsigned int *rx_mask, unsigned int *slots, unsigned int *slot_width) { u32 val; int ret; + if (tx_mask) + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask", tx_mask); + if (rx_mask) + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask", rx_mask); + if (of_property_read_bool(np, "dai-tdm-slot-num")) { ret = of_property_read_u32(np, "dai-tdm-slot-num", &val); if (ret) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ff8bda471b25..016eba10b1ec 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -509,6 +509,18 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, } /** + * snd_soc_dapm_kcontrol_widget() - Returns the widget associated to a + * kcontrol + * @kcontrol: The kcontrol + */ +struct snd_soc_dapm_widget *snd_soc_dapm_kcontrol_widget( + struct snd_kcontrol *kcontrol) +{ + return dapm_kcontrol_get_wlist(kcontrol)->widgets[0]; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_widget); + +/** * snd_soc_dapm_kcontrol_dapm() - Returns the dapm context associated to a * kcontrol * @kcontrol: The kcontrol @@ -779,7 +791,7 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, * Determine if a kcontrol is shared. If it is, look it up. If it isn't, * create it. Either way, add the widget into the control's widget list */ -static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, +static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w, int kci) { struct snd_soc_dapm_context *dapm = w->dapm; @@ -810,6 +822,7 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, switch (w->id) { case snd_soc_dapm_switch: case snd_soc_dapm_mixer: + case snd_soc_dapm_pga: wname_in_long_name = true; kcname_in_long_name = true; break; @@ -899,7 +912,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) continue; if (!w->kcontrols[i]) { - ret = dapm_create_or_share_mixmux_kcontrol(w, i); + ret = dapm_create_or_share_kcontrol(w, i); if (ret < 0) return ret; } @@ -952,7 +965,7 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return -EINVAL; } - ret = dapm_create_or_share_mixmux_kcontrol(w, 0); + ret = dapm_create_or_share_kcontrol(w, 0); if (ret < 0) return ret; @@ -967,9 +980,13 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) /* create new dapm volume control */ static int dapm_new_pga(struct snd_soc_dapm_widget *w) { - if (w->num_kcontrols) - dev_err(w->dapm->dev, - "ASoC: PGA controls not supported: '%s'\n", w->name); + int i, ret; + + for (i = 0; i < w->num_kcontrols; i++) { + ret = dapm_create_or_share_kcontrol(w, i); + if (ret < 0) + return ret; + } return 0; } @@ -3473,11 +3490,29 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: substream.stream = SNDRV_PCM_STREAM_CAPTURE; + if (source->driver->ops && source->driver->ops->startup) { + ret = source->driver->ops->startup(&substream, source); + if (ret < 0) { + dev_err(source->dev, + "ASoC: startup() failed: %d\n", ret); + goto out; + } + source->active++; + } ret = soc_dai_hw_params(&substream, params, source); if (ret < 0) goto out; substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + if (sink->driver->ops && sink->driver->ops->startup) { + ret = sink->driver->ops->startup(&substream, sink); + if (ret < 0) { + dev_err(sink->dev, + "ASoC: startup() failed: %d\n", ret); + goto out; + } + sink->active++; + } ret = soc_dai_hw_params(&substream, params, sink); if (ret < 0) goto out; @@ -3497,6 +3532,18 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret); ret = 0; + + source->active--; + if (source->driver->ops && source->driver->ops->shutdown) { + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + source->driver->ops->shutdown(&substream, source); + } + + sink->active--; + if (sink->driver->ops && sink->driver->ops->shutdown) { + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + sink->driver->ops->shutdown(&substream, sink); + } break; default: diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 05977ae1ff2a..ecd38e52285a 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -588,16 +588,16 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); /** * snd_soc_limit_volume - Set new limit to an existing volume control. * - * @codec: where to look for the control + * @card: where to look for the control * @name: Name of the control * @max: new maximum limit * * Return 0 for success, else error. */ -int snd_soc_limit_volume(struct snd_soc_codec *codec, +int snd_soc_limit_volume(struct snd_soc_card *card, const char *name, int max) { - struct snd_card *card = codec->component.card->snd_card; + struct snd_card *snd_card = card->snd_card; struct snd_kcontrol *kctl; struct soc_mixer_control *mc; int found = 0; @@ -607,7 +607,7 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, if (unlikely(!name || max <= 0)) return -EINVAL; - list_for_each_entry(kctl, &card->controls, list) { + list_for_each_entry(kctl, &snd_card->controls, list) { if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { found = 1; break; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 70e4b9d8bdcd..c86dc96e8986 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -34,6 +34,24 @@ #define DPCM_MAX_BE_USERS 8 +/* + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream + * + * Returns true if the DAI supports the indicated stream type. + */ +static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) +{ + struct snd_soc_pcm_stream *codec_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &dai->driver->playback; + else + codec_stream = &dai->driver->capture; + + /* If the codec specifies any rate at all, it supports the stream. */ + return codec_stream->rates; +} + /** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated @@ -182,9 +200,9 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %dHz rate\n", soc_dai->rate); - ret = snd_pcm_hw_constraint_minmax(substream->runtime, + ret = snd_pcm_hw_constraint_single(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, - soc_dai->rate, soc_dai->rate); + soc_dai->rate); if (ret < 0) { dev_err(soc_dai->dev, "ASoC: Unable to apply rate constraint: %d\n", @@ -198,9 +216,8 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %d channel(s)\n", soc_dai->channels); - ret = snd_pcm_hw_constraint_minmax(substream->runtime, + ret = snd_pcm_hw_constraint_single(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, - soc_dai->channels, soc_dai->channels); if (ret < 0) { dev_err(soc_dai->dev, @@ -215,9 +232,8 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %d sample bits\n", soc_dai->sample_bits); - ret = snd_pcm_hw_constraint_minmax(substream->runtime, + ret = snd_pcm_hw_constraint_single(substream->runtime, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - soc_dai->sample_bits, soc_dai->sample_bits); if (ret < 0) { dev_err(soc_dai->dev, @@ -371,6 +387,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* first calculate min/max only for CODECs in the DAI link */ for (i = 0; i < rtd->num_codecs; i++) { + + /* + * Skip CODECs which don't support the current stream type. + * Otherwise, since the rate, channel, and format values will + * zero in that case, we would have no usable settings left, + * causing the resulting setup to fail. + * At least one CODEC should match, otherwise we should have + * bailed out on a higher level, since there would be no + * CODEC to support the transfer direction in that case. + */ + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], + substream->stream)) + continue; + codec_dai_drv = rtd->codec_dais[i]->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; @@ -827,6 +857,23 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; struct snd_pcm_hw_params codec_params; + /* + * Skip CODECs which don't support the current stream type, + * the idea being that if a CODEC is not used for the currently + * set up transfer direction, it should not need to be + * configured, especially since the configuration used might + * not even be supported by that CODEC. There may be cases + * however where a CODEC needs to be set up although it is + * actually not being used for the transfer, e.g. if a + * capture-only CODEC is acting as an LRCLK and/or BCLK master + * for the DAI link including a playback-only CODEC. + * If this becomes necessary, we will have to augment the + * machine driver setup with information on how to act, so + * we can do the right thing here. + */ + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + continue; + /* copy params for each codec */ codec_params = *params; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 69d01cd925ce..8d7ec80af51b 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1558,7 +1558,7 @@ static int soc_tplg_pcm_dai_elems_load(struct soc_tplg *tplg, pcm_dai = (struct snd_soc_tplg_pcm_dai *)tplg->pos; if (soc_tplg_check_elem_count(tplg, - sizeof(struct snd_soc_tplg_pcm_dai), count, + sizeof(struct snd_soc_tplg_pcm), count, hdr->payload_size, "PCM DAI")) { dev_err(tplg->dev, "ASoC: invalid count %d for PCM DAI elems\n", count); @@ -1566,7 +1566,7 @@ static int soc_tplg_pcm_dai_elems_load(struct soc_tplg *tplg, } dev_dbg(tplg->dev, "ASoC: adding %d PCM DAIs\n", count); - tplg->pos += sizeof(struct snd_soc_tplg_pcm_dai) * count; + tplg->pos += sizeof(struct snd_soc_tplg_pcm) * count; dobj = kzalloc(sizeof(struct snd_soc_dobj), GFP_KERNEL); if (dobj == NULL) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig new file mode 100644 index 000000000000..84c72ec6ad73 --- /dev/null +++ b/sound/soc/sunxi/Kconfig @@ -0,0 +1,11 @@ +menu "Allwinner SoC Audio support" + +config SND_SUN4I_CODEC + tristate "Allwinner A10 Codec Support" + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Select Y or M to add support for the Codec embedded in the Allwinner + A10 and affiliated SoCs. + +endmenu diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile new file mode 100644 index 000000000000..ea8a08c881d6 --- /dev/null +++ b/sound/soc/sunxi/Makefile @@ -0,0 +1,2 @@ +obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o + diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c new file mode 100644 index 000000000000..bcbf4da168b6 --- /dev/null +++ b/sound/soc/sunxi/sun4i-codec.c @@ -0,0 +1,712 @@ +/* + * Copyright 2014 Emilio López <emilio@elopez.com.ar> + * Copyright 2014 Jon Smirl <jonsmirl@gmail.com> + * Copyright 2015 Maxime Ripard <maxime.ripard@free-electrons.com> + * + * Based on the Allwinner SDK driver, released under the GPL. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/init.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_address.h> +#include <linux/clk.h> +#include <linux/regmap.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/initval.h> +#include <sound/dmaengine_pcm.h> + +/* Codec DAC register offsets and bit fields */ +#define SUN4I_CODEC_DAC_DPC (0x00) +#define SUN4I_CODEC_DAC_DPC_EN_DA (31) +#define SUN4I_CODEC_DAC_DPC_DVOL (12) +#define SUN4I_CODEC_DAC_FIFOC (0x04) +#define SUN4I_CODEC_DAC_FIFOC_DAC_FS (29) +#define SUN4I_CODEC_DAC_FIFOC_FIR_VERSION (28) +#define SUN4I_CODEC_DAC_FIFOC_SEND_LASAT (26) +#define SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE (24) +#define SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT (21) +#define SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL (8) +#define SUN4I_CODEC_DAC_FIFOC_MONO_EN (6) +#define SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS (5) +#define SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN (4) +#define SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH (0) +#define SUN4I_CODEC_DAC_FIFOS (0x08) +#define SUN4I_CODEC_DAC_TXDATA (0x0c) +#define SUN4I_CODEC_DAC_ACTL (0x10) +#define SUN4I_CODEC_DAC_ACTL_DACAENR (31) +#define SUN4I_CODEC_DAC_ACTL_DACAENL (30) +#define SUN4I_CODEC_DAC_ACTL_MIXEN (29) +#define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15) +#define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14) +#define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13) +#define SUN4I_CODEC_DAC_ACTL_DACPAS (8) +#define SUN4I_CODEC_DAC_ACTL_MIXPAS (7) +#define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6) +#define SUN4I_CODEC_DAC_ACTL_PA_VOL (0) +#define SUN4I_CODEC_DAC_TUNE (0x14) +#define SUN4I_CODEC_DAC_DEBUG (0x18) + +/* Codec ADC register offsets and bit fields */ +#define SUN4I_CODEC_ADC_FIFOC (0x1c) +#define SUN4I_CODEC_ADC_FIFOC_EN_AD (28) +#define SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE (24) +#define SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL (8) +#define SUN4I_CODEC_ADC_FIFOC_MONO_EN (7) +#define SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS (6) +#define SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN (4) +#define SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH (0) +#define SUN4I_CODEC_ADC_FIFOS (0x20) +#define SUN4I_CODEC_ADC_RXDATA (0x24) +#define SUN4I_CODEC_ADC_ACTL (0x28) +#define SUN4I_CODEC_ADC_ACTL_ADC_R_EN (31) +#define SUN4I_CODEC_ADC_ACTL_ADC_L_EN (30) +#define SUN4I_CODEC_ADC_ACTL_PREG1EN (29) +#define SUN4I_CODEC_ADC_ACTL_PREG2EN (28) +#define SUN4I_CODEC_ADC_ACTL_VMICEN (27) +#define SUN4I_CODEC_ADC_ACTL_VADCG (20) +#define SUN4I_CODEC_ADC_ACTL_ADCIS (17) +#define SUN4I_CODEC_ADC_ACTL_PA_EN (4) +#define SUN4I_CODEC_ADC_ACTL_DDE (3) +#define SUN4I_CODEC_ADC_DEBUG (0x2c) + +/* Other various ADC registers */ +#define SUN4I_CODEC_DAC_TXCNT (0x30) +#define SUN4I_CODEC_ADC_RXCNT (0x34) +#define SUN4I_CODEC_AC_SYS_VERI (0x38) +#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c) + +struct sun4i_codec { + struct device *dev; + struct regmap *regmap; + struct clk *clk_apb; + struct clk *clk_module; + + struct snd_dmaengine_dai_dma_data playback_dma_data; +}; + +static void sun4i_codec_start_playback(struct sun4i_codec *scodec) +{ + /* + * FIXME: according to the BSP, we might need to drive a PA + * GPIO high here on some boards + */ + + /* Flush TX FIFO */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH), + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH)); + + /* Enable DAC DRQ */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN), + BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN)); +} + +static void sun4i_codec_stop_playback(struct sun4i_codec *scodec) +{ + /* + * FIXME: according to the BSP, we might need to drive a PA + * GPIO low here on some boards + */ + + /* Disable DAC DRQ */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN), + 0); +} + +static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENOTSUPP; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sun4i_codec_start_playback(scodec); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sun4i_codec_stop_playback(scodec); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int sun4i_codec_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + u32 val; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENOTSUPP; + + /* Flush the TX FIFO */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH), + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH)); + + /* Set TX FIFO Empty Trigger Level */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + 0x3f << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL, + 0xf << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL); + + if (substream->runtime->rate > 32000) + /* Use 64 bits FIR filter */ + val = 0; + else + /* Use 32 bits FIR filter */ + val = BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION); + + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION), + val); + + /* Send zeros when we have an underrun */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_SEND_LASAT), + 0); + + return 0; +} + +static unsigned long sun4i_codec_get_mod_freq(struct snd_pcm_hw_params *params) +{ + unsigned int rate = params_rate(params); + + switch (rate) { + case 176400: + case 88200: + case 44100: + case 33075: + case 22050: + case 14700: + case 11025: + case 7350: + return 22579200; + + case 192000: + case 96000: + case 48000: + case 32000: + case 24000: + case 16000: + case 12000: + case 8000: + return 24576000; + + default: + return 0; + } +} + +static int sun4i_codec_get_hw_rate(struct snd_pcm_hw_params *params) +{ + unsigned int rate = params_rate(params); + + switch (rate) { + case 192000: + case 176400: + return 6; + + case 96000: + case 88200: + return 7; + + case 48000: + case 44100: + return 0; + + case 32000: + case 33075: + return 1; + + case 24000: + case 22050: + return 2; + + case 16000: + case 14700: + return 3; + + case 12000: + case 11025: + return 4; + + case 8000: + case 7350: + return 5; + + default: + return -EINVAL; + } +} + +static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + unsigned long clk_freq; + int ret, hwrate; + u32 val; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENOTSUPP; + + clk_freq = sun4i_codec_get_mod_freq(params); + if (!clk_freq) + return -EINVAL; + + ret = clk_set_rate(scodec->clk_module, clk_freq); + if (ret) + return ret; + + hwrate = sun4i_codec_get_hw_rate(params); + if (hwrate < 0) + return hwrate; + + /* Set DAC sample rate */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + 7 << SUN4I_CODEC_DAC_FIFOC_DAC_FS, + hwrate << SUN4I_CODEC_DAC_FIFOC_DAC_FS); + + /* Set the number of channels we want to use */ + if (params_channels(params) == 1) + val = BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN); + else + val = 0; + + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN), + val); + + /* Set the number of sample bits to either 16 or 24 bits */ + if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min == 32) { + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS), + BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS)); + + /* Set TX FIFO mode to padding the LSBs with 0 */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE), + 0); + + scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + } else { + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS), + 0); + + /* Set TX FIFO mode to repeat the MSB */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE), + BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE)); + + scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + } + + return 0; +} + +static int sun4i_codec_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + + /* + * Stop issuing DRQ when we have room for less than 16 samples + * in our TX FIFO + */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT, + 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT); + + return clk_prepare_enable(scodec->clk_module); +} + +static void sun4i_codec_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + + clk_disable_unprepare(scodec->clk_module); +} + +static const struct snd_soc_dai_ops sun4i_codec_dai_ops = { + .startup = sun4i_codec_startup, + .shutdown = sun4i_codec_shutdown, + .trigger = sun4i_codec_trigger, + .hw_params = sun4i_codec_hw_params, + .prepare = sun4i_codec_prepare, +}; + +static struct snd_soc_dai_driver sun4i_codec_dai = { + .name = "Codec", + .ops = &sun4i_codec_dai_ops, + .playback = { + .stream_name = "Codec Playback", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 192000, + .rates = SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .sig_bits = 24, + }, +}; + +/*** Codec ***/ +static const struct snd_kcontrol_new sun4i_codec_pa_mute = + SOC_DAPM_SINGLE("Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0); + +static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1); + +static const struct snd_kcontrol_new sun4i_codec_widgets[] = { + SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, + sun4i_codec_pa_volume_scale), +}; + +static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0), +}; + +static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { + SOC_DAPM_SINGLE("Right DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0), + SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0), +}; + +static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_DACPAS, 1, 0), + SOC_DAPM_SINGLE("Mixer Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIXPAS, 1, 0), +}; + +static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { + /* Digital parts of the DACs */ + SND_SOC_DAPM_SUPPLY("DAC", SUN4I_CODEC_DAC_DPC, + SUN4I_CODEC_DAC_DPC_EN_DA, 0, + NULL, 0), + + /* Analog parts of the DACs */ + SND_SOC_DAPM_DAC("Left DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_DACAENL, 0), + SND_SOC_DAPM_DAC("Right DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_DACAENR, 0), + + /* Mixers */ + SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + sun4i_codec_left_mixer_controls, + ARRAY_SIZE(sun4i_codec_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + sun4i_codec_right_mixer_controls, + ARRAY_SIZE(sun4i_codec_right_mixer_controls)), + + /* Global Mixer Enable */ + SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0), + + /* Pre-Amplifier */ + SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PA_EN, 0, + sun4i_codec_pa_mixer_controls, + ARRAY_SIZE(sun4i_codec_pa_mixer_controls)), + SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0, + &sun4i_codec_pa_mute), + + SND_SOC_DAPM_OUTPUT("HP Right"), + SND_SOC_DAPM_OUTPUT("HP Left"), +}; + +static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { + /* Left DAC Routes */ + { "Left DAC", NULL, "DAC" }, + + /* Right DAC Routes */ + { "Right DAC", NULL, "DAC" }, + + /* Right Mixer Routes */ + { "Right Mixer", NULL, "Mixer Enable" }, + { "Right Mixer", "Left DAC Playback Switch", "Left DAC" }, + { "Right Mixer", "Right DAC Playback Switch", "Right DAC" }, + + /* Left Mixer Routes */ + { "Left Mixer", NULL, "Mixer Enable" }, + { "Left Mixer", "Left DAC Playback Switch", "Left DAC" }, + + /* Pre-Amplifier Mixer Routes */ + { "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" }, + { "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" }, + { "Pre-Amplifier", "DAC Playback Switch", "Left DAC" }, + { "Pre-Amplifier", "DAC Playback Switch", "Right DAC" }, + + /* PA -> HP path */ + { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, + { "HP Right", NULL, "Pre-Amplifier Mute" }, + { "HP Left", NULL, "Pre-Amplifier Mute" }, +}; + +static struct snd_soc_codec_driver sun4i_codec_codec = { + .controls = sun4i_codec_widgets, + .num_controls = ARRAY_SIZE(sun4i_codec_widgets), + .dapm_widgets = sun4i_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_dapm_widgets), + .dapm_routes = sun4i_codec_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sun4i_codec_dapm_routes), +}; + +static const struct snd_soc_component_driver sun4i_codec_component = { + .name = "sun4i-codec", +}; + +#define SUN4I_CODEC_RATES SNDRV_PCM_RATE_8000_192000 +#define SUN4I_CODEC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static int sun4i_codec_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_card *card = snd_soc_dai_get_drvdata(dai); + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card); + + snd_soc_dai_init_dma_data(dai, &scodec->playback_dma_data, + NULL); + + return 0; +} + +static struct snd_soc_dai_driver dummy_cpu_dai = { + .name = "sun4i-codec-cpu-dai", + .probe = sun4i_codec_dai_probe, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SUN4I_CODEC_RATES, + .formats = SUN4I_CODEC_FORMATS, + .sig_bits = 24, + }, +}; + +static const struct regmap_config sun4i_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL, +}; + +static const struct of_device_id sun4i_codec_of_match[] = { + { .compatible = "allwinner,sun4i-a10-codec" }, + { .compatible = "allwinner,sun7i-a20-codec" }, + {} +}; +MODULE_DEVICE_TABLE(of, sun4i_codec_of_match); + +static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev, + int *num_links) +{ + struct snd_soc_dai_link *link = devm_kzalloc(dev, sizeof(*link), + GFP_KERNEL); + if (!link) + return NULL; + + link->name = "cdc"; + link->stream_name = "CDC PCM"; + link->codec_dai_name = "Codec"; + link->cpu_dai_name = dev_name(dev); + link->codec_name = dev_name(dev); + link->platform_name = dev_name(dev); + link->dai_fmt = SND_SOC_DAIFMT_I2S; + + *num_links = 1; + + return link; +}; + +static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) +{ + struct snd_soc_card *card; + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return NULL; + + card->dai_link = sun4i_codec_create_link(dev, &card->num_links); + if (!card->dai_link) + return NULL; + + card->dev = dev; + card->name = "sun4i-codec"; + + return card; +}; + +static int sun4i_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card; + struct sun4i_codec *scodec; + struct resource *res; + void __iomem *base; + int ret; + + scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL); + if (!scodec) + return -ENOMEM; + + scodec->dev = &pdev->dev; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) { + dev_err(&pdev->dev, "Failed to map the registers\n"); + return PTR_ERR(base); + } + + scodec->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun4i_codec_regmap_config); + if (IS_ERR(scodec->regmap)) { + dev_err(&pdev->dev, "Failed to create our regmap\n"); + return PTR_ERR(scodec->regmap); + } + + /* Get the clocks from the DT */ + scodec->clk_apb = devm_clk_get(&pdev->dev, "apb"); + if (IS_ERR(scodec->clk_apb)) { + dev_err(&pdev->dev, "Failed to get the APB clock\n"); + return PTR_ERR(scodec->clk_apb); + } + + scodec->clk_module = devm_clk_get(&pdev->dev, "codec"); + if (IS_ERR(scodec->clk_module)) { + dev_err(&pdev->dev, "Failed to get the module clock\n"); + return PTR_ERR(scodec->clk_module); + } + + /* Enable the bus clock */ + if (clk_prepare_enable(scodec->clk_apb)) { + dev_err(&pdev->dev, "Failed to enable the APB clock\n"); + return -EINVAL; + } + + /* DMA configuration for TX FIFO */ + scodec->playback_dma_data.addr = res->start + SUN4I_CODEC_DAC_TXDATA; + scodec->playback_dma_data.maxburst = 4; + scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + + ret = snd_soc_register_codec(&pdev->dev, &sun4i_codec_codec, + &sun4i_codec_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Failed to register our codec\n"); + goto err_clk_disable; + } + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun4i_codec_component, + &dummy_cpu_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Failed to register our DAI\n"); + goto err_unregister_codec; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "Failed to register against DMAEngine\n"); + goto err_unregister_codec; + } + + card = sun4i_codec_create_card(&pdev->dev); + if (!card) { + dev_err(&pdev->dev, "Failed to create our card\n"); + goto err_unregister_codec; + } + + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, scodec); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "Failed to register our card\n"); + goto err_unregister_codec; + } + + return 0; + +err_unregister_codec: + snd_soc_unregister_codec(&pdev->dev); +err_clk_disable: + clk_disable_unprepare(scodec->clk_apb); + return ret; +} + +static int sun4i_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + snd_soc_unregister_codec(&pdev->dev); + clk_disable_unprepare(scodec->clk_apb); + + return 0; +} + +static struct platform_driver sun4i_codec_driver = { + .driver = { + .name = "sun4i-codec", + .of_match_table = sun4i_codec_of_match, + }, + .probe = sun4i_codec_probe, + .remove = sun4i_codec_remove, +}; +module_platform_driver(sun4i_codec_driver); + +MODULE_DESCRIPTION("Allwinner A10 codec driver"); +MODULE_AUTHOR("Emilio López <emilio@elopez.com.ar>"); +MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>"); +MODULE_AUTHOR("Maxime Ripard <maxime.ripard@free-electrons.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 4e0c0e502ade..ba9fc099cf67 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -152,6 +152,7 @@ static const struct of_device_id snd_soc_mop500_match[] = { { .compatible = "stericsson,snd-soc-mop500", }, {}, }; +MODULE_DEVICE_TABLE(of, snd_soc_mop500_match); static struct platform_driver snd_soc_mop500_driver = { .driver = { diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index f5df08ded770..6d5698b25bd4 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -522,9 +522,9 @@ static int ux500_msp_dai_hw_params(struct snd_pcm_substream *substream, slots_active = hweight32(mask); dev_dbg(dai->dev, "TDM-slots active: %d", slots_active); - snd_pcm_hw_constraint_minmax(runtime, + snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, - slots_active, slots_active); + slots_active); break; default: @@ -843,6 +843,7 @@ static const struct of_device_id ux500_msp_i2s_match[] = { { .compatible = "stericsson,ux500-msp-i2s", }, {}, }; +MODULE_DEVICE_TABLE(of, ux500_msp_i2s_match); static struct platform_driver msp_i2s_driver = { .driver = { diff --git a/sound/usb/card.h b/sound/usb/card.h index ef580b43f1e3..71778ca4b26a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -122,6 +122,7 @@ struct snd_usb_substream { unsigned int buffer_periods; /* current periods per buffer */ unsigned int altset_idx; /* USB data format: index of alternate setting */ unsigned int txfr_quirk:1; /* allow sub-frame alignment */ + unsigned int tx_length_quirk:1; /* add length specifier to transfers */ unsigned int fmt_type; /* USB audio format type (1-3) */ unsigned int pkt_offset_adj; /* Bytes to drop from beginning of packets (for non-compliant devices) */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e6f71894ecdc..7b1cb365ffab 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -183,13 +183,53 @@ static void retire_inbound_urb(struct snd_usb_endpoint *ep, ep->retire_data_urb(ep->data_subs, urb); } +static void prepare_silent_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *ctx) +{ + struct urb *urb = ctx->urb; + unsigned int offs = 0; + unsigned int extra = 0; + __le32 packet_length; + int i; + + /* For tx_length_quirk, put packet length at start of packet */ + if (ep->chip->tx_length_quirk) + extra = sizeof(packet_length); + + for (i = 0; i < ctx->packets; ++i) { + unsigned int offset; + unsigned int length; + int counts; + + if (ctx->packet_size[i]) + counts = ctx->packet_size[i]; + else + counts = snd_usb_endpoint_next_packet_size(ep); + + length = counts * ep->stride; /* number of silent bytes */ + offset = offs * ep->stride + extra * i; + urb->iso_frame_desc[i].offset = offset; + urb->iso_frame_desc[i].length = length + extra; + if (extra) { + packet_length = cpu_to_le32(length); + memcpy(urb->transfer_buffer + offset, + &packet_length, sizeof(packet_length)); + } + memset(urb->transfer_buffer + offset + extra, + ep->silence_value, length); + offs += counts; + } + + urb->number_of_packets = ctx->packets; + urb->transfer_buffer_length = offs * ep->stride + ctx->packets * extra; +} + /* * Prepare a PLAYBACK urb for submission to the bus. */ static void prepare_outbound_urb(struct snd_usb_endpoint *ep, struct snd_urb_ctx *ctx) { - int i; struct urb *urb = ctx->urb; unsigned char *cp = urb->transfer_buffer; @@ -201,24 +241,7 @@ static void prepare_outbound_urb(struct snd_usb_endpoint *ep, ep->prepare_data_urb(ep->data_subs, urb); } else { /* no data provider, so send silence */ - unsigned int offs = 0; - for (i = 0; i < ctx->packets; ++i) { - int counts; - - if (ctx->packet_size[i]) - counts = ctx->packet_size[i]; - else - counts = snd_usb_endpoint_next_packet_size(ep); - - urb->iso_frame_desc[i].offset = offs * ep->stride; - urb->iso_frame_desc[i].length = counts * ep->stride; - offs += counts; - } - - urb->number_of_packets = ctx->packets; - urb->transfer_buffer_length = offs * ep->stride; - memset(urb->transfer_buffer, ep->silence_value, - offs * ep->stride); + prepare_silent_urb(ep, ctx); } break; @@ -594,6 +617,8 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, unsigned int max_packs_per_period, urbs_per_period, urb_packs; unsigned int max_urbs, i; int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels; + int tx_length_quirk = (ep->chip->tx_length_quirk && + usb_pipeout(ep->pipe)); if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) { /* @@ -610,13 +635,34 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, /* assume max. frequency is 25% higher than nominal */ ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - ep->datainterval); + /* Round up freqmax to nearest integer in order to calculate maximum + * packet size, which must represent a whole number of frames. + * This is accomplished by adding 0x0.ffff before converting the + * Q16.16 format into integer. + * In order to accurately calculate the maximum packet size when + * the data interval is more than 1 (i.e. ep->datainterval > 0), + * multiply by the data interval prior to rounding. For instance, + * a freqmax of 41 kHz will result in a max packet size of 6 (5.125) + * frames with a data interval of 1, but 11 (10.25) frames with a + * data interval of 2. + * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the + * maximum datainterval value of 3, at USB full speed, higher for + * USB high speed, noting that ep->freqmax is in units of + * frames per packet in Q16.16 format.) + */ + maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * + (frame_bits >> 3); + if (tx_length_quirk) + maxsize += sizeof(__le32); /* Space for length descriptor */ /* but wMaxPacketSize might reduce this */ if (ep->maxpacksize && ep->maxpacksize < maxsize) { /* whatever fits into a max. size packet */ - maxsize = ep->maxpacksize; - ep->freqmax = (maxsize / (frame_bits >> 3)) + unsigned int data_maxsize = maxsize = ep->maxpacksize; + + if (tx_length_quirk) + /* Need to remove the length descriptor to calc freq */ + data_maxsize -= sizeof(__le32); + ep->freqmax = (data_maxsize / (frame_bits >> 3)) << (16 - ep->datainterval); } diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 417ebb11cf48..7661616f3636 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1903,11 +1903,14 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi *umidi) hostif = &intf->altsetting[1]; intfd = get_iface_desc(hostif); + /* If either or both of the endpoints support interrupt transfer, + * then use the alternate setting + */ if (intfd->bNumEndpoints != 2 || - (get_endpoint(hostif, 0)->bmAttributes & - USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK || - (get_endpoint(hostif, 1)->bmAttributes & - USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT) + !((get_endpoint(hostif, 0)->bmAttributes & + USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT || + (get_endpoint(hostif, 1)->bmAttributes & + USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)) return; dev_dbg(&umidi->dev->dev, "switching to altsetting %d with int ep\n", diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index d3608c0a29f3..fe91184ce832 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -338,7 +338,7 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); struct usb_mixer_interface *mixer = list->mixer; int index = kcontrol->private_value & 0xff; - int value = ucontrol->value.integer.value[0]; + unsigned int value = ucontrol->value.integer.value[0]; int old_value = kcontrol->private_value >> 8; int err; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index cdac5179db3f..9245f52d43bd 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1383,6 +1383,56 @@ static inline void fill_playback_urb_dsd_dop(struct snd_usb_substream *subs, subs->hwptr_done++; } } + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; +} + +static void copy_to_urb(struct snd_usb_substream *subs, struct urb *urb, + int offset, int stride, unsigned int bytes) +{ + struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + + if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { + /* err, the transferred area goes over buffer boundary. */ + unsigned int bytes1 = + runtime->buffer_size * stride - subs->hwptr_done; + memcpy(urb->transfer_buffer + offset, + runtime->dma_area + subs->hwptr_done, bytes1); + memcpy(urb->transfer_buffer + offset + bytes1, + runtime->dma_area, bytes - bytes1); + } else { + memcpy(urb->transfer_buffer + offset, + runtime->dma_area + subs->hwptr_done, bytes); + } + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; +} + +static unsigned int copy_to_urb_quirk(struct snd_usb_substream *subs, + struct urb *urb, int stride, + unsigned int bytes) +{ + __le32 packet_length; + int i; + + /* Put __le32 length descriptor at start of each packet. */ + for (i = 0; i < urb->number_of_packets; i++) { + unsigned int length = urb->iso_frame_desc[i].length; + unsigned int offset = urb->iso_frame_desc[i].offset; + + packet_length = cpu_to_le32(length); + offset += i * sizeof(packet_length); + urb->iso_frame_desc[i].offset = offset; + urb->iso_frame_desc[i].length += sizeof(packet_length); + memcpy(urb->transfer_buffer + offset, + &packet_length, sizeof(packet_length)); + copy_to_urb(subs, urb, offset + sizeof(packet_length), + stride, length); + } + /* Adjust transfer size accordingly. */ + bytes += urb->number_of_packets * sizeof(packet_length); + return bytes; } static void prepare_playback_urb(struct snd_usb_substream *subs, @@ -1460,27 +1510,17 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, } subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; } else { /* usual PCM */ - if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { - /* err, the transferred area goes over buffer boundary. */ - unsigned int bytes1 = - runtime->buffer_size * stride - subs->hwptr_done; - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes1); - memcpy(urb->transfer_buffer + bytes1, - runtime->dma_area, bytes - bytes1); - } else { - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes); - } - - subs->hwptr_done += bytes; + if (!subs->tx_length_quirk) + copy_to_urb(subs, urb, 0, stride, bytes); + else + bytes = copy_to_urb_quirk(subs, urb, stride, bytes); + /* bytes is now amount of outgoing data */ } - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - /* update delay with exact number of samples queued */ runtime->delay = subs->last_delay; runtime->delay += frames; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e4756651a52c..1a1e2e4df35e 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2664,6 +2664,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE(0x1235, 0x000a), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Novation", */ + /* .product_name = "Nocturn", */ + .ifnum = 0, + .type = QUIRK_MIDI_RAW_BYTES + } +}, +{ USB_DEVICE(0x1235, 0x000e), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "Novation", */ @@ -3182,10 +3191,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), { /* * ZOOM R16/24 in audio interface mode. - * Mixer descriptors are garbage, further quirks will be needed - * to make any of it functional, thus disabled for now. - * Playback stream appears to start and run fine but no sound - * is produced, so also disabled for now. + * Playback requires an extra four byte LE length indicator + * at the start of each isochronous packet. This quirk is + * enabled in create_standard_audio_quirk(). */ USB_DEVICE(0x1686, 0x00dd), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { @@ -3193,14 +3201,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { - /* Mixer */ - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE, - }, - { /* Playback */ .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, }, { /* Capture */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 00ebc0ca008e..4897ea171194 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -115,6 +115,9 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, struct usb_interface_descriptor *altsd; int err; + if (chip->usb_id == USB_ID(0x1686, 0x00dd)) /* Zoom R16/24 */ + chip->tx_length_quirk = 1; + alts = &iface->altsetting[0]; altsd = get_iface_desc(alts); err = snd_usb_parse_audio_interface(chip, altsd->bInterfaceNumber); diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 970086015cde..8ee14f2365e7 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -92,6 +92,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->direction = stream; subs->dev = as->chip->dev; subs->txfr_quirk = as->chip->txfr_quirk; + subs->tx_length_quirk = as->chip->tx_length_quirk; subs->speed = snd_usb_get_speed(subs->dev); subs->pkt_offset_adj = 0; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 33a176437e2e..15a12715bd05 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -43,6 +43,7 @@ struct snd_usb_audio { atomic_t usage_count; wait_queue_head_t shutdown_wait; unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ + unsigned int tx_length_quirk:1; /* Put length specifier in transfers */ int num_interfaces; int num_suspended_intf; |