diff options
Diffstat (limited to 'sound')
87 files changed, 3675 insertions, 586 deletions
diff --git a/sound/core/info.c b/sound/core/info.c index 051d55b05521..9f404e965ea2 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -684,7 +684,7 @@ int snd_info_card_free(struct snd_card *card) * snd_info_get_line - read one line from the procfs buffer * @buffer: the procfs buffer * @line: the buffer to store - * @len: the max. buffer size - 1 + * @len: the max. buffer size * * Reads one line from the buffer and stores the string. * @@ -704,7 +704,7 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) buffer->stop = 1; if (c == '\n') break; - if (len) { + if (len > 1) { len--; *line++ = c; } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9acc77eae487..0032278567ad 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1782,14 +1782,16 @@ static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream, { struct snd_pcm_hw_params *params = arg; snd_pcm_format_t format; - int channels, width; + int channels; + ssize_t frame_size; params->fifo_size = substream->runtime->hw.fifo_size; if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_FIFO_IN_FRAMES)) { format = params_format(params); channels = params_channels(params); - width = snd_pcm_format_physical_width(format); - params->fifo_size /= width * channels; + frame_size = snd_pcm_format_size(format, channels); + if (frame_size > 0) + params->fifo_size /= (unsigned)frame_size; } return 0; } diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 4560ca0e5651..2c6fd80e0bd1 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -142,11 +142,11 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { }, [SNDRV_PCM_FORMAT_DSD_U8] = { .width = 8, .phys = 8, .le = 1, .signd = 0, - .silence = {}, + .silence = { 0x69 }, }, [SNDRV_PCM_FORMAT_DSD_U16_LE] = { .width = 16, .phys = 16, .le = 1, .signd = 0, - .silence = {}, + .silence = { 0x69, 0x69 }, }, /* FIXME: the following three formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index f96bf4c7c232..95fc2eaf11dc 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -507,7 +507,16 @@ static void amdtp_pull_midi(struct amdtp_stream *s, static void update_pcm_pointers(struct amdtp_stream *s, struct snd_pcm_substream *pcm, unsigned int frames) -{ unsigned int ptr; +{ + 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) diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index d8ee7b0e9386..4823c08196ac 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -125,6 +125,7 @@ struct amdtp_stream { 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]; diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index a9a30c0161f1..e3a04d69c853 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -567,10 +567,14 @@ static int dice_hw_params(struct snd_pcm_substream *substream, return err; /* - * At rates above 96 kHz, pretend that the stream runs at half the - * actual sample rate with twice the number of channels; two samples - * of a channel are stored consecutively in the packet. Requires - * blocking mode and PCM buffer size should be aligned to SYT_INTERVAL. + * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in + * one data block of AMDTP packet. Thus sampling transfer frequency is + * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are + * transferred on AMDTP packets at 96 kHz. Two successive samples of a + * channel are stored consecutively in the packet. This quirk is called + * as 'Dual Wire'. + * For this quirk, blocking mode is required and PCM buffer size should + * be aligned to SYT_INTERVAL. */ channels = params_channels(hw_params); if (rate_index > 4) { @@ -579,18 +583,25 @@ static int dice_hw_params(struct snd_pcm_substream *substream, return err; } - for (i = 0; i < channels; i++) { - dice->stream.pcm_positions[i * 2] = i; - dice->stream.pcm_positions[i * 2 + 1] = i + channels; - } - rate /= 2; channels *= 2; + dice->stream.double_pcm_frames = true; + } else { + dice->stream.double_pcm_frames = false; } mode = rate_index_to_mode(rate_index); amdtp_stream_set_parameters(&dice->stream, rate, channels, dice->rx_midi_ports[mode]); + if (rate_index > 4) { + channels /= 2; + + for (i = 0; i < channels; i++) { + dice->stream.pcm_positions[i] = i * 2; + dice->stream.pcm_positions[i + channels] = i * 2 + 1; + } + } + amdtp_stream_set_pcm_format(&dice->stream, params_format(hw_params)); diff --git a/sound/pci/ctxfi/ct20k1reg.h b/sound/pci/ctxfi/ct20k1reg.h index f2e34e3f27ee..5851249f11d9 100644 --- a/sound/pci/ctxfi/ct20k1reg.h +++ b/sound/pci/ctxfi/ct20k1reg.h @@ -7,7 +7,7 @@ */ #ifndef CT20K1REG_H -#define CT20k1REG_H +#define CT20K1REG_H /* 20k1 registers */ #define DSPXRAM_START 0x000000 @@ -632,5 +632,3 @@ #define I2SD_R 0x19L #endif /* CT20K1REG_H */ - - diff --git a/sound/pci/hda/ca0132_regs.h b/sound/pci/hda/ca0132_regs.h index 07e760937d3c..8371274aa811 100644 --- a/sound/pci/hda/ca0132_regs.h +++ b/sound/pci/hda/ca0132_regs.h @@ -20,7 +20,7 @@ */ #ifndef __CA0132_REGS_H -#define __CA0312_REGS_H +#define __CA0132_REGS_H #define DSP_CHIP_OFFSET 0x100000 #define DSP_DBGCNTL_MODULE_OFFSET 0xE30 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6f2fa838b635..47ccb8f44adb 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -217,6 +217,7 @@ enum { CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, CXT_FIXUP_GPIO1, + CXT_FIXUP_ASPIRE_DMIC, CXT_FIXUP_THINKPAD_ACPI, CXT_FIXUP_OLPC_XO, CXT_FIXUP_CAP_MIX_AMP, @@ -664,6 +665,12 @@ static const struct hda_fixup cxt_fixups[] = { { } }, }, + [CXT_FIXUP_ASPIRE_DMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_stereo_dmic, + .chained = true, + .chain_id = CXT_FIXUP_GPIO1, + }, [CXT_FIXUP_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, .v.func = hda_fixup_thinkpad_acpi, @@ -744,7 +751,7 @@ static const struct hda_model_fixup cxt5051_fixup_models[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), @@ -770,6 +777,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_PINCFG_LENOVO_TP410, .name = "tp410" }, { .id = CXT_FIXUP_THINKPAD_ACPI, .name = "thinkpad" }, { .id = CXT_PINCFG_LEMOTE_A1004, .name = "lemote-a1004" }, + { .id = CXT_PINCFG_LEMOTE_A1205, .name = "lemote-a1205" }, { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" }, {} }; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 36badba2dcec..99d7d7fecaad 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -50,6 +50,8 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec)) #define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) +#define is_cherryview(codec) ((codec)->vendor_id == 0x80862883) +#define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec)) struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -1459,7 +1461,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (is_haswell_plus(codec) || is_valleyview(codec)) + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1598,7 +1600,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) * and this can make HW reset converter selection on a pin. */ if (eld->eld_valid && !old_eld_valid && per_pin->setup) { - if (is_haswell_plus(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || + is_valleyview_plus(codec)) { intel_verify_pin_cvt_connect(codec, per_pin); intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx); @@ -1779,7 +1782,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, bool non_pcm; int pinctl; - if (is_haswell_plus(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { /* Verify pin:cvt selections to avoid silent audio after S3. * After S3, the audio driver restores pin:cvt selections * but this can happen before gfx is ready and such selection @@ -2330,9 +2333,8 @@ static int patch_generic_hdmi(struct hda_codec *codec) intel_haswell_fixup_enable_dp12(codec); } - if (is_haswell(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) codec->depop_delay = 0; - } if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6b38ec3c6e57..1ba22fb527c2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -181,6 +181,8 @@ static void alc_fix_pll(struct hda_codec *codec) spec->pll_coef_idx); val = snd_hda_codec_read(codec, spec->pll_nid, 0, AC_VERB_GET_PROC_COEF, 0); + if (val == -1) + return; snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX, spec->pll_coef_idx); snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF, @@ -326,6 +328,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0885: case 0x10ec0887: /*case 0x10ec0889:*/ /* this causes an SPDIF problem */ + case 0x10ec0900: alc889_coef_init(codec); break; case 0x10ec0888: @@ -2348,6 +2351,7 @@ static int patch_alc882(struct hda_codec *codec) switch (codec->vendor_id) { case 0x10ec0882: case 0x10ec0885: + case 0x10ec0900: break; default: /* ALC883 and variants */ @@ -2806,6 +2810,8 @@ static void alc286_shutup(struct hda_codec *codec) static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { int val = alc_read_coef_idx(codec, 0x04); + if (val == -1) + return; if (power_up) val |= 1 << 11; else @@ -3264,6 +3270,15 @@ static int alc269_resume(struct hda_codec *codec) snd_hda_codec_resume_cache(codec); alc_inv_dmic_sync(codec, true); hda_call_check_power_status(codec, 0x01); + + /* on some machine, the BIOS will clear the codec gpio data when enter + * suspend, and won't restore the data after resume, so we restore it + * in the driver. + */ + if (spec->gpio_led) + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_led); + if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); @@ -4395,6 +4410,7 @@ enum { ALC292_FIXUP_TPT440_DOCK, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, + ALC282_FIXUP_ASPIRE_V5_PINS, }; static const struct hda_fixup alc269_fixups[] = { @@ -4842,6 +4858,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained_before = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC282_FIXUP_ASPIRE_V5_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x90a60130 }, + { 0x14, 0x90170110 }, + { 0x17, 0x40000008 }, + { 0x18, 0x411111f0 }, + { 0x19, 0x411111f0 }, + { 0x1a, 0x411111f0 }, + { 0x1b, 0x411111f0 }, + { 0x1d, 0x40f89b2d }, + { 0x1e, 0x411111f0 }, + { 0x21, 0x0321101f }, + { }, + }, + }, }; @@ -4853,6 +4885,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), + SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), @@ -5311,27 +5344,30 @@ static void alc269_fill_coef(struct hda_codec *codec) if ((alc_get_coef0(codec) & 0x00ff) == 0x017) { val = alc_read_coef_idx(codec, 0x04); /* Power up output pin */ - alc_write_coef_idx(codec, 0x04, val | (1<<11)); + if (val != -1) + alc_write_coef_idx(codec, 0x04, val | (1<<11)); } if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { val = alc_read_coef_idx(codec, 0xd); - if ((val & 0x0c00) >> 10 != 0x1) { + if (val != -1 && (val & 0x0c00) >> 10 != 0x1) { /* Capless ramp up clock control */ alc_write_coef_idx(codec, 0xd, val | (1<<10)); } val = alc_read_coef_idx(codec, 0x17); - if ((val & 0x01c0) >> 6 != 0x4) { + if (val != -1 && (val & 0x01c0) >> 6 != 0x4) { /* Class D power on reset */ alc_write_coef_idx(codec, 0x17, val | (1<<7)); } } val = alc_read_coef_idx(codec, 0xd); /* Class D */ - alc_write_coef_idx(codec, 0xd, val | (1<<14)); + if (val != -1) + alc_write_coef_idx(codec, 0xd, val | (1<<14)); val = alc_read_coef_idx(codec, 0x4); /* HP */ - alc_write_coef_idx(codec, 0x4, val | (1<<11)); + if (val != -1) + alc_write_coef_idx(codec, 0x4, val | (1<<11)); } /* diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ea823e1100da..98cd1908c039 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -566,8 +566,8 @@ static void stac_init_power_map(struct hda_codec *codec) if (snd_hda_jack_tbl_get(codec, nid)) continue; if (def_conf == AC_JACK_PORT_COMPLEX && - !(spec->vref_mute_led_nid == nid || - is_jack_detectable(codec, nid))) { + spec->vref_mute_led_nid != nid && + is_jack_detectable(codec, nid)) { snd_hda_jack_detect_enable_callback(codec, nid, STAC_PWR_EVENT, jack_update_power); @@ -4276,11 +4276,18 @@ static int stac_parse_auto_config(struct hda_codec *codec) return err; } - stac_init_power_map(codec); - return 0; } +static int stac_build_controls(struct hda_codec *codec) +{ + int err = snd_hda_gen_build_controls(codec); + + if (err < 0) + return err; + stac_init_power_map(codec); + return 0; +} static int stac_init(struct hda_codec *codec) { @@ -4392,7 +4399,7 @@ static int stac_suspend(struct hda_codec *codec) #endif /* CONFIG_PM */ static const struct hda_codec_ops stac_patch_ops = { - .build_controls = snd_hda_gen_build_controls, + .build_controls = stac_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = stac_init, .free = stac_free, diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e25ed..7678122f8fe0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC + select SND_SOC_CS35L32 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -56,7 +57,10 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C + select SND_SOC_DMIC select SND_SOC_BT_SCO + select SND_SOC_ES8328_SPI if SPI_MASTER + select SND_SOC_ES8328_I2C if I2C select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -323,6 +327,10 @@ config SND_SOC_ALC5632 config SND_SOC_CQ0093VC tristate +config SND_SOC_CS35L32 + tristate "Cirrus Logic CS35L32 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate @@ -405,6 +413,17 @@ config SND_SOC_DMIC config SND_SOC_HDMI_CODEC tristate "HDMI stub CODEC" +config SND_SOC_ES8328 + tristate "Everest Semi ES8328 CODEC" + +config SND_SOC_ES8328_I2C + tristate + select SND_SOC_ES8328 + +config SND_SOC_ES8328_SPI + tristate + select SND_SOC_ES8328 + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 20afe0f0c5be..afba944657bc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -32,6 +32,7 @@ snd-soc-ak4671-objs := ak4671.o snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o +snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -49,6 +50,9 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o +snd-soc-es8328-objs := es8328.o +snd-soc-es8328-i2c-objs := es8328-i2c.o +snd-soc-es8328-spi-objs := es8328-spi.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -203,6 +207,7 @@ obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o +obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o @@ -220,6 +225,9 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o +obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o +obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 1fb4402bf72d..fd43827bb856 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -56,8 +56,7 @@ #define GPIO31_DIR_OUTPUT 0x40 /* Macrocell register definitions */ -#define AB8500_CTRL3_REG 0x0200 -#define AB8500_GPIO_DIR4_REG 0x1013 +#define AB8500_GPIO_DIR4_REG 0x13 /* Bank AB8500_MISC */ /* Nr of FIR/IIR-coeff banks in ANC-block */ #define AB8500_NR_OF_ANC_COEFF_BANKS 2 @@ -126,6 +125,8 @@ struct ab8500_codec_drvdata_dbg { /* Private data for AB8500 device-driver */ struct ab8500_codec_drvdata { + struct regmap *regmap; + /* Sidetone */ long *sid_fir_values; enum sid_state sid_status; @@ -166,49 +167,35 @@ static inline const char *amic_type_str(enum amic_type type) */ /* Read a register from the audio-bank of AB8500 */ -static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, - unsigned int reg) +static int ab8500_codec_read_reg(void *context, unsigned int reg, + unsigned int *value) { + struct device *dev = context; int status; - unsigned int value = 0; u8 value8; - status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO, - reg, &value8); - if (status < 0) { - dev_err(codec->dev, - "%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - } else { - dev_dbg(codec->dev, - "%s: Read 0x%02x from register 0x%02x:0x%02x\n", - __func__, value8, (u8)AB8500_AUDIO, (u8)reg); - value = (unsigned int)value8; - } + status = abx500_get_register_interruptible(dev, AB8500_AUDIO, + reg, &value8); + *value = (unsigned int)value8; - return value; + return status; } /* Write to a register in the audio-bank of AB8500 */ -static int ab8500_codec_write_reg(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int ab8500_codec_write_reg(void *context, unsigned int reg, + unsigned int value) { - int status; - - status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO, - reg, value); - if (status < 0) - dev_err(codec->dev, - "%s: ERROR: Register (%02x:%02x) write failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - else - dev_dbg(codec->dev, - "%s: Wrote 0x%02x into register %02x:%02x\n", - __func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg); + struct device *dev = context; - return status; + return abx500_set_register_interruptible(dev, AB8500_AUDIO, + reg, value); } +static const struct regmap_config ab8500_codec_regmap = { + .reg_read = ab8500_codec_read_reg, + .reg_write = ab8500_codec_write_reg, +}; + /* * Controls - DAPM */ @@ -1968,16 +1955,16 @@ static int ab8500_audio_setup_mics(struct snd_soc_codec *codec, dev_dbg(codec->dev, "%s: Enter.\n", __func__); /* Set DMic-clocks to outputs */ - status = abx500_get_register_interruptible(codec->dev, (u8)AB8500_MISC, - (u8)AB8500_GPIO_DIR4_REG, + status = abx500_get_register_interruptible(codec->dev, AB8500_MISC, + AB8500_GPIO_DIR4_REG, &value8); if (status < 0) return status; value = value8 | GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT | GPIO31_DIR_OUTPUT; status = abx500_set_register_interruptible(codec->dev, - (u8)AB8500_MISC, - (u8)AB8500_GPIO_DIR4_REG, + AB8500_MISC, + AB8500_GPIO_DIR4_REG, value); if (status < 0) return status; @@ -2565,9 +2552,6 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver ab8500_codec_driver = { .probe = ab8500_codec_probe, - .read = ab8500_codec_read_reg, - .write = ab8500_codec_write_reg, - .reg_word_size = sizeof(u8), .controls = ab8500_ctrls, .num_controls = ARRAY_SIZE(ab8500_ctrls), .dapm_widgets = ab8500_dapm_widgets, @@ -2592,6 +2576,15 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) drvdata->anc_status = ANC_UNCONFIGURED; dev_set_drvdata(&pdev->dev, drvdata); + drvdata->regmap = devm_regmap_init(&pdev->dev, NULL, &pdev->dev, + &ab8500_codec_regmap); + if (IS_ERR(drvdata->regmap)) { + status = PTR_ERR(drvdata->regmap); + dev_err(&pdev->dev, "%s: Failed to allocate regmap: %d\n", + __func__, status); + return status; + } + dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__); status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver, ab8500_codec_dai, diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index e889e1b84192..bd9b1839c8b0 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -69,19 +69,6 @@ static struct snd_soc_dai_driver ac97_dai = { .ops = &ac97_dai_ops, }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return soc_ac97_ops->read(codec->ac97, reg); -} - -static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) -{ - soc_ac97_ops->write(codec->ac97, reg, val); - return 0; -} - static int ac97_soc_probe(struct snd_soc_codec *codec) { struct snd_ac97_bus *ac97_bus; @@ -122,8 +109,6 @@ static int ac97_soc_resume(struct snd_soc_codec *codec) #endif static struct snd_soc_codec_driver soc_codec_dev_ac97 = { - .write = ac97_write, - .read = ac97_read, .probe = ac97_soc_probe, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1ff7d4d027e9..7c784ad3e8b2 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1448,29 +1448,10 @@ static int adau1373_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int adau1373_remove(struct snd_soc_codec *codec) -{ - adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int adau1373_suspend(struct snd_soc_codec *codec) -{ - struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); - int ret; - - ret = adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); - regcache_cache_only(adau1373->regmap, true); - - return ret; -} - static int adau1373_resume(struct snd_soc_codec *codec) { struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(adau1373->regmap, false); - adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); regcache_sync(adau1373->regmap); return 0; @@ -1501,8 +1482,6 @@ static const struct regmap_config adau1373_regmap_config = { static struct snd_soc_codec_driver adau1373_codec_driver = { .probe = adau1373_probe, - .remove = adau1373_remove, - .suspend = adau1373_suspend, .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, .idle_bias_off = true, diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 848cab839553..5518ebd6947c 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -714,9 +714,9 @@ static int adau1761_codec_probe(struct snd_soc_codec *codec) static const struct snd_soc_codec_driver adau1761_codec_driver = { .probe = adau1761_codec_probe, - .suspend = adau17x1_suspend, .resume = adau17x1_resume, .set_bias_level = adau1761_set_bias_level, + .suspend_bias_off = true, .controls = adau1761_controls, .num_controls = ARRAY_SIZE(adau1761_controls), diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c index 045a61413840..e9fc00fb13dd 100644 --- a/sound/soc/codecs/adau1781.c +++ b/sound/soc/codecs/adau1781.c @@ -446,9 +446,9 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec) static const struct snd_soc_codec_driver adau1781_codec_driver = { .probe = adau1781_codec_probe, - .suspend = adau17x1_suspend, .resume = adau17x1_resume, .set_bias_level = adau1781_set_bias_level, + .suspend_bias_off = true, .controls = adau1781_controls, .num_controls = ARRAY_SIZE(adau1781_controls), diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 0b659704e60c..3e16c1c64115 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -815,13 +815,6 @@ int adau17x1_add_routes(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(adau17x1_add_routes); -int adau17x1_suspend(struct snd_soc_codec *codec) -{ - codec->driver->set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} -EXPORT_SYMBOL_GPL(adau17x1_suspend); - int adau17x1_resume(struct snd_soc_codec *codec) { struct adau *adau = snd_soc_codec_get_drvdata(codec); @@ -829,7 +822,6 @@ int adau17x1_resume(struct snd_soc_codec *codec) if (adau->switch_mode) adau->switch_mode(codec->dev); - codec->driver->set_bias_level(codec, SND_SOC_BIAS_STANDBY); regcache_sync(adau->regmap); return 0; diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 3ffabaf4c7a8..e4a557fd7155 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -52,7 +52,6 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); bool adau17x1_volatile_register(struct device *dev, unsigned int reg); -int adau17x1_suspend(struct snd_soc_codec *codec); int adau17x1_resume(struct snd_soc_codec *codec); extern const struct snd_soc_dai_ops adau17x1_dai_ops; diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index c43b93fdf0df..ce3cdca9fc62 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -812,42 +812,23 @@ static int adav80x_probe(struct snd_soc_codec *codec) /* Disable DAC zero flag */ regmap_write(adav80x->regmap, ADAV80X_DAC_CTRL3, 0x6); - return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -} - -static int adav80x_suspend(struct snd_soc_codec *codec) -{ - struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - int ret; - - ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); - regcache_cache_only(adav80x->regmap, true); - - return ret; + return 0; } static int adav80x_resume(struct snd_soc_codec *codec) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(adav80x->regmap, false); - adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); regcache_sync(adav80x->regmap); return 0; } -static int adav80x_remove(struct snd_soc_codec *codec) -{ - return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); -} - static struct snd_soc_codec_driver adav80x_codec_driver = { .probe = adav80x_probe, - .remove = adav80x_remove, - .suspend = adav80x_suspend, .resume = adav80x_resume, .set_bias_level = adav80x_set_bias_level, + .suspend_bias_off = true, .set_pll = adav80x_set_pll, .set_sysclk = adav80x_set_sysclk, diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index bd41ee4da078..2c71f16bd661 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1278,6 +1278,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; + wl = snd_pcm_format_width(params_format(params)); + if (tdm_slots) { arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", tdm_slots, tdm_width); @@ -1285,6 +1287,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, channels = tdm_slots; } else { bclk_target = snd_soc_params_to_bclk(params); + tdm_width = wl; } if (chan_limit && chan_limit < channels) { @@ -1319,8 +1322,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", rates[bclk], rates[bclk] / lrclk); - wl = snd_pcm_format_width(params_format(params)); - frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + frame = wl << ARIZONA_AIF1TX_WL_SHIFT | tdm_width; reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame); diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c new file mode 100644 index 000000000000..c125925da92e --- /dev/null +++ b/sound/soc/codecs/cs35l32.c @@ -0,0 +1,631 @@ +/* + * cs35l32.c -- CS35L32 ALSA SoC audio driver + * + * Copyright 2014 CirrusLogic, Inc. + * + * Author: Brian Austin <brian.austin@cirrus.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/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.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 <dt-bindings/sound/cs35l32.h> + +#include "cs35l32.h" + +#define CS35L32_NUM_SUPPLIES 2 +static const char *const cs35l32_supply_names[CS35L32_NUM_SUPPLIES] = { + "VA", + "VP", +}; + +struct cs35l32_private { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct regulator_bulk_data supplies[CS35L32_NUM_SUPPLIES]; + struct cs35l32_platform_data pdata; + struct gpio_desc *reset_gpio; +}; + +static const struct reg_default cs35l32_reg_defaults[] = { + + { 0x06, 0x04 }, /* Power Ctl 1 */ + { 0x07, 0xE8 }, /* Power Ctl 2 */ + { 0x08, 0x40 }, /* Clock Ctl */ + { 0x09, 0x20 }, /* Low Battery Threshold */ + { 0x0A, 0x00 }, /* Voltage Monitor [RO] */ + { 0x0B, 0x40 }, /* Conv Peak Curr Protection CTL */ + { 0x0C, 0x07 }, /* IMON Scaling */ + { 0x0D, 0x03 }, /* Audio/LED Pwr Manager */ + { 0x0F, 0x20 }, /* Serial Port Control */ + { 0x10, 0x14 }, /* Class D Amp CTL */ + { 0x11, 0x00 }, /* Protection Release CTL */ + { 0x12, 0xFF }, /* Interrupt Mask 1 */ + { 0x13, 0xFF }, /* Interrupt Mask 2 */ + { 0x14, 0xFF }, /* Interrupt Mask 3 */ + { 0x19, 0x00 }, /* LED Flash Mode Current */ + { 0x1A, 0x00 }, /* LED Movie Mode Current */ + { 0x1B, 0x20 }, /* LED Flash Timer */ + { 0x1C, 0x00 }, /* LED Flash Inhibit Current */ +}; + +static bool cs35l32_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L32_DEVID_AB: + case CS35L32_DEVID_CD: + case CS35L32_DEVID_E: + case CS35L32_FAB_ID: + case CS35L32_REV_ID: + case CS35L32_PWRCTL1: + case CS35L32_PWRCTL2: + case CS35L32_CLK_CTL: + case CS35L32_BATT_THRESHOLD: + case CS35L32_VMON: + case CS35L32_BST_CPCP_CTL: + case CS35L32_IMON_SCALING: + case CS35L32_AUDIO_LED_MNGR: + case CS35L32_ADSP_CTL: + case CS35L32_CLASSD_CTL: + case CS35L32_PROTECT_CTL: + case CS35L32_INT_MASK_1: + case CS35L32_INT_MASK_2: + case CS35L32_INT_MASK_3: + case CS35L32_INT_STATUS_1: + case CS35L32_INT_STATUS_2: + case CS35L32_INT_STATUS_3: + case CS35L32_LED_STATUS: + case CS35L32_FLASH_MODE: + case CS35L32_MOVIE_MODE: + case CS35L32_FLASH_TIMER: + case CS35L32_FLASH_INHIBIT: + return true; + default: + return false; + } +} + +static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L32_DEVID_AB: + case CS35L32_DEVID_CD: + case CS35L32_DEVID_E: + case CS35L32_FAB_ID: + case CS35L32_REV_ID: + case CS35L32_INT_STATUS_1: + case CS35L32_INT_STATUS_2: + case CS35L32_INT_STATUS_3: + case CS35L32_LED_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l32_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L32_INT_STATUS_1: + case CS35L32_INT_STATUS_2: + case CS35L32_INT_STATUS_3: + case CS35L32_LED_STATUS: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 300, 0); + +static const struct snd_kcontrol_new imon_ctl = + SOC_DAPM_SINGLE("Switch", CS35L32_PWRCTL2, 6, 1, 1); + +static const struct snd_kcontrol_new vmon_ctl = + SOC_DAPM_SINGLE("Switch", CS35L32_PWRCTL2, 7, 1, 1); + +static const struct snd_kcontrol_new vpmon_ctl = + SOC_DAPM_SINGLE("Switch", CS35L32_PWRCTL2, 5, 1, 1); + +static const struct snd_kcontrol_new cs35l32_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", CS35L32_CLASSD_CTL, + 3, 0x04, 1, classd_ctl_tlv), + SOC_SINGLE("Zero Cross Switch", CS35L32_CLASSD_CTL, 2, 1, 0), + SOC_SINGLE("Gain Manager Switch", CS35L32_AUDIO_LED_MNGR, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget cs35l32_dapm_widgets[] = { + + SND_SOC_DAPM_SUPPLY("BOOST", CS35L32_PWRCTL1, 2, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker", CS35L32_PWRCTL1, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("SDOUT", NULL, 0, CS35L32_PWRCTL2, 3, 1), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("ISENSE"), + SND_SOC_DAPM_INPUT("VSENSE"), + + SND_SOC_DAPM_SWITCH("VMON ADC", CS35L32_PWRCTL2, 7, 1, &vmon_ctl), + SND_SOC_DAPM_SWITCH("IMON ADC", CS35L32_PWRCTL2, 6, 1, &imon_ctl), + SND_SOC_DAPM_SWITCH("VPMON ADC", CS35L32_PWRCTL2, 5, 1, &vpmon_ctl), +}; + +static const struct snd_soc_dapm_route cs35l32_audio_map[] = { + + {"Speaker", NULL, "BOOST"}, + + {"VMON ADC", NULL, "VSENSE"}, + {"IMON ADC", NULL, "ISENSE"}, + {"VPMON ADC", NULL, "VP"}, + + {"SDOUT", "Switch", "VMON ADC"}, + {"SDOUT", "Switch", "IMON ADC"}, + {"SDOUT", "Switch", "VPMON ADC"}, + + {"Capture", NULL, "SDOUT"}, +}; + +static int cs35l32_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, CS35L32_ADSP_CTL, + CS35L32_ADSP_MASTER_MASK, + CS35L32_ADSP_MASTER_MASK); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, CS35L32_ADSP_CTL, + CS35L32_ADSP_MASTER_MASK, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs35l32_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + + return snd_soc_update_bits(codec, CS35L32_PWRCTL2, + CS35L32_SDOUT_3ST, tristate << 3); +} + +static const struct snd_soc_dai_ops cs35l32_ops = { + .set_fmt = cs35l32_set_dai_fmt, + .set_tristate = cs35l32_set_tristate, +}; + +static struct snd_soc_dai_driver cs35l32_dai[] = { + { + .name = "cs35l32-monitor", + .id = 0, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS35L32_RATES, + .formats = CS35L32_FORMATS, + }, + .ops = &cs35l32_ops, + .symmetric_rates = 1, + } +}; + +static int cs35l32_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + unsigned int val; + + switch (freq) { + case 6000000: + val = CS35L32_MCLK_RATIO; + break; + case 12000000: + val = CS35L32_MCLK_DIV2_MASK | CS35L32_MCLK_RATIO; + break; + case 6144000: + val = 0; + break; + case 12288000: + val = CS35L32_MCLK_DIV2_MASK; + break; + default: + return -EINVAL; + } + + return snd_soc_update_bits(codec, CS35L32_CLK_CTL, + CS35L32_MCLK_DIV2_MASK | CS35L32_MCLK_RATIO_MASK, val); +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l32 = { + .set_sysclk = cs35l32_codec_set_sysclk, + + .dapm_widgets = cs35l32_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l32_dapm_widgets), + .dapm_routes = cs35l32_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l32_audio_map), + + .controls = cs35l32_snd_controls, + .num_controls = ARRAY_SIZE(cs35l32_snd_controls), +}; + +/* Current and threshold powerup sequence Pg37 in datasheet */ +static const struct reg_default cs35l32_monitor_patch[] = { + + { 0x00, 0x99 }, + { 0x48, 0x17 }, + { 0x49, 0x56 }, + { 0x43, 0x01 }, + { 0x3B, 0x62 }, + { 0x3C, 0x80 }, + { 0x00, 0x00 }, +}; + +static struct regmap_config cs35l32_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L32_MAX_REGISTER, + .reg_defaults = cs35l32_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs35l32_reg_defaults), + .volatile_reg = cs35l32_volatile_register, + .readable_reg = cs35l32_readable_register, + .precious_reg = cs35l32_precious_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs35l32_handle_of_data(struct i2c_client *i2c_client, + struct cs35l32_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + unsigned int val; + + if (of_property_read_u32(np, "cirrus,sdout-share", &val) >= 0) + pdata->sdout_share = val; + + of_property_read_u32(np, "cirrus,boost-manager", &val); + switch (val) { + case CS35L32_BOOST_MGR_AUTO: + case CS35L32_BOOST_MGR_AUTO_AUDIO: + case CS35L32_BOOST_MGR_BYPASS: + case CS35L32_BOOST_MGR_FIXED: + pdata->boost_mng = val; + break; + default: + dev_err(&i2c_client->dev, + "Wrong cirrus,boost-manager DT value %d\n", val); + pdata->boost_mng = CS35L32_BOOST_MGR_BYPASS; + } + + of_property_read_u32(np, "cirrus,sdout-datacfg", &val); + switch (val) { + case CS35L32_DATA_CFG_LR_VP: + case CS35L32_DATA_CFG_LR_STAT: + case CS35L32_DATA_CFG_LR: + case CS35L32_DATA_CFG_LR_VPSTAT: + pdata->sdout_datacfg = val; + break; + default: + dev_err(&i2c_client->dev, + "Wrong cirrus,sdout-datacfg DT value %d\n", val); + pdata->sdout_datacfg = CS35L32_DATA_CFG_LR; + } + + of_property_read_u32(np, "cirrus,battery-threshold", &val); + switch (val) { + case CS35L32_BATT_THRESH_3_1V: + case CS35L32_BATT_THRESH_3_2V: + case CS35L32_BATT_THRESH_3_3V: + case CS35L32_BATT_THRESH_3_4V: + pdata->batt_thresh = val; + break; + default: + dev_err(&i2c_client->dev, + "Wrong cirrus,battery-threshold DT value %d\n", val); + pdata->batt_thresh = CS35L32_BATT_THRESH_3_3V; + } + + of_property_read_u32(np, "cirrus,battery-recovery", &val); + switch (val) { + case CS35L32_BATT_RECOV_3_1V: + case CS35L32_BATT_RECOV_3_2V: + case CS35L32_BATT_RECOV_3_3V: + case CS35L32_BATT_RECOV_3_4V: + case CS35L32_BATT_RECOV_3_5V: + case CS35L32_BATT_RECOV_3_6V: + pdata->batt_recov = val; + break; + default: + dev_err(&i2c_client->dev, + "Wrong cirrus,battery-recovery DT value %d\n", val); + pdata->batt_recov = CS35L32_BATT_RECOV_3_4V; + } + + return 0; +} + +static int cs35l32_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l32_private *cs35l32; + struct cs35l32_platform_data *pdata = + dev_get_platdata(&i2c_client->dev); + int ret, i; + unsigned int devid = 0; + unsigned int reg; + + + cs35l32 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l32_private), + GFP_KERNEL); + if (!cs35l32) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs35l32); + + cs35l32->regmap = devm_regmap_init_i2c(i2c_client, &cs35l32_regmap); + if (IS_ERR(cs35l32->regmap)) { + ret = PTR_ERR(cs35l32->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + if (pdata) { + cs35l32->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l32_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + ret = cs35l32_handle_of_data(i2c_client, + &cs35l32->pdata); + if (ret != 0) + return ret; + } + } + + for (i = 0; i < ARRAY_SIZE(cs35l32->supplies); i++) + cs35l32->supplies[i].supply = cs35l32_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c_client->dev, + ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + /* Reset the Device */ + cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev, + "reset-gpios"); + if (IS_ERR(cs35l32->reset_gpio)) { + ret = PTR_ERR(cs35l32->reset_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + cs35l32->reset_gpio = NULL; + } else { + ret = gpiod_direction_output(cs35l32->reset_gpio, 0); + if (ret) + return ret; + gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); + } + + /* initialize codec */ + ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, ®); + devid = (reg & 0xFF) << 12; + + ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_CD, ®); + devid |= (reg & 0xFF) << 4; + + ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_E, ®); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L32_CHIP_ID) { + ret = -ENODEV; + dev_err(&i2c_client->dev, + "CS35L32 Device ID (%X). Expected %X\n", + devid, CS35L32_CHIP_ID); + return ret; + } + + ret = regmap_read(cs35l32->regmap, CS35L32_REV_ID, ®); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + return ret; + } + + ret = regmap_register_patch(cs35l32->regmap, cs35l32_monitor_patch, + ARRAY_SIZE(cs35l32_monitor_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to apply errata patch\n"); + return ret; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L32, Revision: %02X\n", reg & 0xFF); + + /* Setup VBOOST Management */ + if (cs35l32->pdata.boost_mng) + regmap_update_bits(cs35l32->regmap, CS35L32_AUDIO_LED_MNGR, + CS35L32_BOOST_MASK, + cs35l32->pdata.boost_mng); + + /* Setup ADSP Format Config */ + if (cs35l32->pdata.sdout_share) + regmap_update_bits(cs35l32->regmap, CS35L32_ADSP_CTL, + CS35L32_ADSP_SHARE_MASK, + cs35l32->pdata.sdout_share << 3); + + /* Setup ADSP Data Configuration */ + if (cs35l32->pdata.sdout_datacfg) + regmap_update_bits(cs35l32->regmap, CS35L32_ADSP_CTL, + CS35L32_ADSP_DATACFG_MASK, + cs35l32->pdata.sdout_datacfg << 4); + + /* Setup Low Battery Recovery */ + if (cs35l32->pdata.batt_recov) + regmap_update_bits(cs35l32->regmap, CS35L32_BATT_THRESHOLD, + CS35L32_BATT_REC_MASK, + cs35l32->pdata.batt_recov << 1); + + /* Setup Low Battery Threshold */ + if (cs35l32->pdata.batt_thresh) + regmap_update_bits(cs35l32->regmap, CS35L32_BATT_THRESHOLD, + CS35L32_BATT_THRESH_MASK, + cs35l32->pdata.batt_thresh << 4); + + /* Power down the AMP */ + regmap_update_bits(cs35l32->regmap, CS35L32_PWRCTL1, CS35L32_PDN_AMP, + CS35L32_PDN_AMP); + + /* Clear MCLK Error Bit since we don't have the clock yet */ + ret = regmap_read(cs35l32->regmap, CS35L32_INT_STATUS_1, ®); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l32, cs35l32_dai, + ARRAY_SIZE(cs35l32_dai)); + if (ret < 0) + goto err_disable; + + return 0; + +err_disable: + regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + return ret; +} + +static int cs35l32_i2c_remove(struct i2c_client *i2c_client) +{ + struct cs35l32_private *cs35l32 = i2c_get_clientdata(i2c_client); + + snd_soc_unregister_codec(&i2c_client->dev); + + /* Hold down reset */ + if (cs35l32->reset_gpio) + gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int cs35l32_runtime_suspend(struct device *dev) +{ + struct cs35l32_private *cs35l32 = dev_get_drvdata(dev); + + regcache_cache_only(cs35l32->regmap, true); + regcache_mark_dirty(cs35l32->regmap); + + /* Hold down reset */ + if (cs35l32->reset_gpio) + gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); + + /* remove power */ + regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + + return 0; +} + +static int cs35l32_runtime_resume(struct device *dev) +{ + struct cs35l32_private *cs35l32 = dev_get_drvdata(dev); + int ret; + + /* Enable power */ + ret = regulator_bulk_enable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (cs35l32->reset_gpio) + gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); + + regcache_cache_only(cs35l32->regmap, false); + regcache_sync(cs35l32->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs35l32_runtime_pm = { + SET_RUNTIME_PM_OPS(cs35l32_runtime_suspend, cs35l32_runtime_resume, + NULL) +}; + +static const struct of_device_id cs35l32_of_match[] = { + { .compatible = "cirrus,cs35l32", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l32_of_match); + + +static const struct i2c_device_id cs35l32_id[] = { + {"cs35l32", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l32_id); + +static struct i2c_driver cs35l32_i2c_driver = { + .driver = { + .name = "cs35l32", + .owner = THIS_MODULE, + .pm = &cs35l32_runtime_pm, + .of_match_table = cs35l32_of_match, + }, + .id_table = cs35l32_id, + .probe = cs35l32_i2c_probe, + .remove = cs35l32_i2c_remove, +}; + +module_i2c_driver(cs35l32_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L32 driver"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin@cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l32.h b/sound/soc/codecs/cs35l32.h new file mode 100644 index 000000000000..31ab804a22bc --- /dev/null +++ b/sound/soc/codecs/cs35l32.h @@ -0,0 +1,93 @@ +/* + * cs35l32.h -- CS35L32 ALSA SoC audio driver + * + * Copyright 2014 CirrusLogic, Inc. + * + * Author: Brian Austin <brian.austin@cirrus.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 __CS35L32_H__ +#define __CS35L32_H__ + +struct cs35l32_platform_data { + /* Low Battery Threshold */ + unsigned int batt_thresh; + /* Low Battery Recovery */ + unsigned int batt_recov; + /* LED Current Management*/ + unsigned int led_mng; + /* Audio Gain w/ LED */ + unsigned int audiogain_mng; + /* Boost Management */ + unsigned int boost_mng; + /* Data CFG for DUAL device */ + unsigned int sdout_datacfg; + /* SDOUT Sharing */ + unsigned int sdout_share; +}; + +#define CS35L32_CHIP_ID 0x00035A32 +#define CS35L32_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L32_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L32_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L32_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L32_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L32_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L32_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L32_CLK_CTL 0x08 /* Clock Ctl */ +#define CS35L32_BATT_THRESHOLD 0x09 /* Low Battery Threshold */ +#define CS35L32_VMON 0x0A /* Voltage Monitor [RO] */ +#define CS35L32_BST_CPCP_CTL 0x0B /* Conv Peak Curr Protection CTL */ +#define CS35L32_IMON_SCALING 0x0C /* IMON Scaling */ +#define CS35L32_AUDIO_LED_MNGR 0x0D /* Audio/LED Pwr Manager */ +#define CS35L32_ADSP_CTL 0x0F /* Serial Port Control */ +#define CS35L32_CLASSD_CTL 0x10 /* Class D Amp CTL */ +#define CS35L32_PROTECT_CTL 0x11 /* Protection Release CTL */ +#define CS35L32_INT_MASK_1 0x12 /* Interrupt Mask 1 */ +#define CS35L32_INT_MASK_2 0x13 /* Interrupt Mask 2 */ +#define CS35L32_INT_MASK_3 0x14 /* Interrupt Mask 3 */ +#define CS35L32_INT_STATUS_1 0x15 /* Interrupt Status 1 [RO] */ +#define CS35L32_INT_STATUS_2 0x16 /* Interrupt Status 2 [RO] */ +#define CS35L32_INT_STATUS_3 0x17 /* Interrupt Status 3 [RO] */ +#define CS35L32_LED_STATUS 0x18 /* LED Lighting Status [RO] */ +#define CS35L32_FLASH_MODE 0x19 /* LED Flash Mode Current */ +#define CS35L32_MOVIE_MODE 0x1A /* LED Movie Mode Current */ +#define CS35L32_FLASH_TIMER 0x1B /* LED Flash Timer */ +#define CS35L32_FLASH_INHIBIT 0x1C /* LED Flash Inhibit Current */ +#define CS35L32_MAX_REGISTER 0x1C + +#define CS35L32_MCLK_DIV2 0x01 +#define CS35L32_MCLK_RATIO 0x01 +#define CS35L32_MCLKDIS 0x80 +#define CS35L32_PDN_ALL 0x01 +#define CS35L32_PDN_AMP 0x80 +#define CS35L32_PDN_BOOST 0x04 +#define CS35L32_PDN_IMON 0x40 +#define CS35L32_PDN_VMON 0x80 +#define CS35L32_PDN_VPMON 0x20 +#define CS35L32_PDN_ADSP 0x08 + +#define CS35L32_MCLK_DIV2_MASK 0x40 +#define CS35L32_MCLK_RATIO_MASK 0x01 +#define CS35L32_MCLK_MASK 0x41 +#define CS35L32_ADSP_MASTER_MASK 0x40 +#define CS35L32_BOOST_MASK 0x03 +#define CS35L32_GAIN_MGR_MASK 0x08 +#define CS35L32_ADSP_SHARE_MASK 0x08 +#define CS35L32_ADSP_DATACFG_MASK 0x30 +#define CS35L32_SDOUT_3ST 0x80 +#define CS35L32_BATT_REC_MASK 0x0E +#define CS35L32_BATT_THRESH_MASK 0x30 + +#define CS35L32_RATES (SNDRV_PCM_RATE_48000) +#define CS35L32_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + + +#endif diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index a20b30ca52c0..4fdd47d700e3 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -77,6 +77,7 @@ static bool cs4265_readable_register(struct device *dev, unsigned int reg) case CS4265_INT_MASK: case CS4265_STATUS_MODE_MSB: case CS4265_STATUS_MODE_LSB: + case CS4265_CHIP_ID: return true; default: return false; @@ -282,10 +283,10 @@ static const struct cs4265_clk_para clk_map_table[] = { /*64k*/ {8192000, 64000, 1, 0}, - {1228800, 64000, 1, 1}, - {1693440, 64000, 1, 2}, - {2457600, 64000, 1, 3}, - {3276800, 64000, 1, 4}, + {12288000, 64000, 1, 1}, + {16934400, 64000, 1, 2}, + {24576000, 64000, 1, 3}, + {32768000, 64000, 1, 4}, /* 88.2k */ {11289600, 88200, 1, 0}, @@ -435,10 +436,10 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params)); if (index >= 0) { snd_soc_update_bits(codec, CS4265_ADC_CTL, - CS4265_ADC_FM, clk_map_table[index].fm_mode); + CS4265_ADC_FM, clk_map_table[index].fm_mode << 6); snd_soc_update_bits(codec, CS4265_MCLK_FREQ, CS4265_MCLK_FREQ_MASK, - clk_map_table[index].mclkdiv); + clk_map_table[index].mclkdiv << 4); } else { dev_err(codec->dev, "can't get correct mclk\n"); @@ -458,12 +459,12 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, if (params_width(params) == 16) { snd_soc_update_bits(codec, CS4265_DAC_CTL, CS4265_DAC_CTL_DIF, (1 << 5)); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 7)); } else { snd_soc_update_bits(codec, CS4265_DAC_CTL, CS4265_DAC_CTL_DIF, (3 << 5)); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 7)); } break; @@ -472,7 +473,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, CS4265_DAC_CTL_DIF, 0); snd_soc_update_bits(codec, CS4265_ADC_CTL, CS4265_ADC_DIF, 0); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 6)); break; diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 969167d8b71e..da4f758cd12a 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -176,9 +176,9 @@ static bool cs42l52_volatile_register(struct device *dev, unsigned int reg) case CS42L52_BATT_LEVEL: case CS42L52_SPK_STATUS: case CS42L52_CHARGE_PUMP: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index c766a5a9ce80..bb74dd17fa26 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -171,9 +171,9 @@ static bool cs42l56_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case CS42L56_INT_STATUS: - return 1; + return true; default: - return 0; + return false; } } @@ -1175,11 +1175,8 @@ static int cs42l56_probe(struct snd_soc_codec *codec) static int cs42l56_remove(struct snd_soc_codec *codec) { - struct cs42l56_private *cs42l56 = snd_soc_codec_get_drvdata(codec); - cs42l56_free_beep(codec); cs42l56_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(cs42l56->supplies), cs42l56->supplies); return 0; } diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 2fae31cb0067..fa15fa1c0516 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -217,7 +217,7 @@ static void da732x_set_charge_pump(struct snd_soc_codec *codec, int state) snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA723X_CP_DIS); break; default: - pr_err(KERN_ERR "Wrong charge pump state\n"); + pr_err("Wrong charge pump state\n"); break; } } diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h index 1dceafeec415..f586cbd30b77 100644 --- a/sound/soc/codecs/da732x.h +++ b/sound/soc/codecs/da732x.h @@ -11,7 +11,7 @@ */ #ifndef __DA732X_H_ -#define __DA732X_H +#define __DA732X_H_ #include <sound/soc.h> diff --git a/sound/soc/codecs/es8328-i2c.c b/sound/soc/codecs/es8328-i2c.c new file mode 100644 index 000000000000..aae410d122ee --- /dev/null +++ b/sound/soc/codecs/es8328-i2c.c @@ -0,0 +1,60 @@ +/* + * es8328-i2c.c -- ES8328 ALSA SoC I2C Audio driver + * + * Copyright 2014 Sutajio Ko-Usagi PTE LTD + * + * Author: Sean Cross <xobs@kosagi.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/i2c.h> +#include <linux/regmap.h> + +#include <sound/soc.h> + +#include "es8328.h" + +static const struct i2c_device_id es8328_id[] = { + { "everest,es8328", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, es8328_id); + +static const struct of_device_id es8328_of_match[] = { + { .compatible = "everest,es8328", }, + { } +}; +MODULE_DEVICE_TABLE(of, es8328_of_match); + +static int es8328_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + return es8328_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &es8328_regmap_config)); +} + +static int es8328_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static struct i2c_driver es8328_i2c_driver = { + .driver = { + .name = "es8328", + .of_match_table = es8328_of_match, + }, + .probe = es8328_i2c_probe, + .remove = es8328_i2c_remove, + .id_table = es8328_id, +}; + +module_i2c_driver(es8328_i2c_driver); + +MODULE_DESCRIPTION("ASoC ES8328 audio CODEC I2C driver"); +MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8328-spi.c b/sound/soc/codecs/es8328-spi.c new file mode 100644 index 000000000000..8fbd935e1c76 --- /dev/null +++ b/sound/soc/codecs/es8328-spi.c @@ -0,0 +1,49 @@ +/* + * es8328.c -- ES8328 ALSA SoC SPI Audio driver + * + * Copyright 2014 Sutajio Ko-Usagi PTE LTD + * + * Author: Sean Cross <xobs@kosagi.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/regmap.h> +#include <linux/spi/spi.h> +#include <sound/soc.h> +#include "es8328.h" + +static const struct of_device_id es8328_of_match[] = { + { .compatible = "everest,es8328", }, + { } +}; +MODULE_DEVICE_TABLE(of, es8328_of_match); + +static int es8328_spi_probe(struct spi_device *spi) +{ + return es8328_probe(&spi->dev, + devm_regmap_init_spi(spi, &es8328_regmap_config)); +} + +static int es8328_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver es8328_spi_driver = { + .driver = { + .name = "es8328", + .of_match_table = es8328_of_match, + }, + .probe = es8328_spi_probe, + .remove = es8328_spi_remove, +}; + +module_spi_driver(es8328_spi_driver); +MODULE_DESCRIPTION("ASoC ES8328 audio CODEC SPI driver"); +MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c new file mode 100644 index 000000000000..f27325155ace --- /dev/null +++ b/sound/soc/codecs/es8328.c @@ -0,0 +1,756 @@ +/* + * es8328.c -- ES8328 ALSA SoC Audio driver + * + * Copyright 2014 Sutajio Ko-Usagi PTE LTD + * + * Author: Sean Cross <xobs@kosagi.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/clk.h> +#include <linux/delay.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/regulator/consumer.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "es8328.h" + +#define ES8328_SYSCLK_RATE_1X 11289600 +#define ES8328_SYSCLK_RATE_2X 22579200 + +/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */ +static struct { + int rate; + u8 ratio; +} mclk_ratios[] = { + { 8000, 9 }, + {11025, 7 }, + {22050, 4 }, + {44100, 2 }, +}; + +/* regulator supplies for sgtl5000, VDDD is an optional external supply */ +enum sgtl5000_regulator_supplies { + DVDD, + AVDD, + PVDD, + HPVDD, + ES8328_SUPPLY_NUM +}; + +/* vddd is optional supply */ +static const char * const supply_names[ES8328_SUPPLY_NUM] = { + "DVDD", + "AVDD", + "PVDD", + "HPVDD", +}; + +#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_11025) +#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +struct es8328_priv { + struct regmap *regmap; + struct clk *clk; + int playback_fs; + bool deemph; + struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM]; +}; + +/* + * ES8328 Controls + */ + +static const char * const adcpol_txt[] = {"Normal", "L Invert", "R Invert", + "L + R Invert"}; +static SOC_ENUM_SINGLE_DECL(adcpol, + ES8328_ADCCONTROL6, 6, adcpol_txt); + +static const DECLARE_TLV_DB_SCALE(play_tlv, -3000, 100, 0); +static const DECLARE_TLV_DB_SCALE(dac_adc_tlv, -9600, 50, 0); +static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); +static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0); + +static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; + +static int es8328_set_deemph(struct snd_soc_codec *codec) +{ + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int val, i, best; + + /* + * If we're using deemphasis select the nearest available sample + * rate. + */ + if (es8328->deemph) { + best = 1; + for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i] - es8328->playback_fs) < + abs(deemph_settings[best] - es8328->playback_fs)) + best = i; + } + + val = best << 1; + } else { + val = 0; + } + + dev_dbg(codec->dev, "Set deemphasis %d\n", val); + + return snd_soc_update_bits(codec, ES8328_DACCONTROL6, 0x6, val); +} + +static int es8328_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = es8328->deemph; + return 0; +} + +static int es8328_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int deemph = ucontrol->value.enumerated.item[0]; + int ret; + + if (deemph > 1) + return -EINVAL; + + ret = es8328_set_deemph(codec); + if (ret < 0) + return ret; + + es8328->deemph = deemph; + + return 0; +} + + + +static const struct snd_kcontrol_new es8328_snd_controls[] = { + SOC_DOUBLE_R_TLV("Capture Digital Volume", + ES8328_ADCCONTROL8, ES8328_ADCCONTROL9, + 0, 0xc0, 1, dac_adc_tlv), + SOC_SINGLE("Capture ZC Switch", ES8328_ADCCONTROL7, 6, 1, 0), + + SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0, + es8328_get_deemph, es8328_put_deemph), + + SOC_ENUM("Capture Polarity", adcpol), + + SOC_SINGLE_TLV("Left Mixer Left Bypass Volume", + ES8328_DACCONTROL17, 3, 7, 1, bypass_tlv), + SOC_SINGLE_TLV("Left Mixer Right Bypass Volume", + ES8328_DACCONTROL19, 3, 7, 1, bypass_tlv), + SOC_SINGLE_TLV("Right Mixer Left Bypass Volume", + ES8328_DACCONTROL18, 3, 7, 1, bypass_tlv), + SOC_SINGLE_TLV("Right Mixer Right Bypass Volume", + ES8328_DACCONTROL20, 3, 7, 1, bypass_tlv), + + SOC_DOUBLE_R_TLV("PCM Volume", + ES8328_LDACVOL, ES8328_RDACVOL, + 0, ES8328_DACVOL_MAX, 1, dac_adc_tlv), + + SOC_DOUBLE_R_TLV("Output 1 Playback Volume", + ES8328_LOUT1VOL, ES8328_ROUT1VOL, + 0, ES8328_OUT1VOL_MAX, 0, play_tlv), + + SOC_DOUBLE_R_TLV("Output 2 Playback Volume", + ES8328_LOUT2VOL, ES8328_ROUT2VOL, + 0, ES8328_OUT2VOL_MAX, 0, play_tlv), + + SOC_DOUBLE_TLV("Mic PGA Volume", ES8328_ADCCONTROL1, + 4, 0, 8, 0, mic_tlv), +}; + +/* + * DAPM Controls + */ + +static const char * const es8328_line_texts[] = { + "Line 1", "Line 2", "PGA", "Differential"}; + +static const struct soc_enum es8328_lline_enum = + SOC_ENUM_SINGLE(ES8328_DACCONTROL16, 3, + ARRAY_SIZE(es8328_line_texts), + es8328_line_texts); +static const struct snd_kcontrol_new es8328_left_line_controls = + SOC_DAPM_ENUM("Route", es8328_lline_enum); + +static const struct soc_enum es8328_rline_enum = + SOC_ENUM_SINGLE(ES8328_DACCONTROL16, 0, + ARRAY_SIZE(es8328_line_texts), + es8328_line_texts); +static const struct snd_kcontrol_new es8328_right_line_controls = + SOC_DAPM_ENUM("Route", es8328_lline_enum); + +/* Left Mixer */ +static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0), +}; + +/* Right Mixer */ +static const struct snd_kcontrol_new es8328_right_mixer_controls[] = { + SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0), +}; + +static const char * const es8328_pga_sel[] = { + "Line 1", "Line 2", "Line 3", "Differential"}; + +/* Left PGA Mux */ +static const struct soc_enum es8328_lpga_enum = + SOC_ENUM_SINGLE(ES8328_ADCCONTROL2, 6, + ARRAY_SIZE(es8328_pga_sel), + es8328_pga_sel); +static const struct snd_kcontrol_new es8328_left_pga_controls = + SOC_DAPM_ENUM("Route", es8328_lpga_enum); + +/* Right PGA Mux */ +static const struct soc_enum es8328_rpga_enum = + SOC_ENUM_SINGLE(ES8328_ADCCONTROL2, 4, + ARRAY_SIZE(es8328_pga_sel), + es8328_pga_sel); +static const struct snd_kcontrol_new es8328_right_pga_controls = + SOC_DAPM_ENUM("Route", es8328_rpga_enum); + +/* Differential Mux */ +static const char * const es8328_diff_sel[] = {"Line 1", "Line 2"}; +static SOC_ENUM_SINGLE_DECL(diffmux, + ES8328_ADCCONTROL3, 7, es8328_diff_sel); +static const struct snd_kcontrol_new es8328_diffmux_controls = + SOC_DAPM_ENUM("Route", diffmux); + +/* Mono ADC Mux */ +static const char * const es8328_mono_mux[] = {"Stereo", "Mono (Left)", + "Mono (Right)", "Digital Mono"}; +static SOC_ENUM_SINGLE_DECL(monomux, + ES8328_ADCCONTROL3, 3, es8328_mono_mux); +static const struct snd_kcontrol_new es8328_monomux_controls = + SOC_DAPM_ENUM("Route", monomux); + +static const struct snd_soc_dapm_widget es8328_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &es8328_diffmux_controls), + SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, + &es8328_monomux_controls), + SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, + &es8328_monomux_controls), + + SND_SOC_DAPM_MUX("Left PGA Mux", ES8328_ADCPOWER, + ES8328_ADCPOWER_AINL_OFF, 1, + &es8328_left_pga_controls), + SND_SOC_DAPM_MUX("Right PGA Mux", ES8328_ADCPOWER, + ES8328_ADCPOWER_AINR_OFF, 1, + &es8328_right_pga_controls), + + SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, + &es8328_left_line_controls), + SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, + &es8328_right_line_controls), + + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ES8328_ADCPOWER, + ES8328_ADCPOWER_ADCR_OFF, 1), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ES8328_ADCPOWER, + ES8328_ADCPOWER_ADCL_OFF, 1), + + SND_SOC_DAPM_SUPPLY("Mic Bias", ES8328_ADCPOWER, + ES8328_ADCPOWER_MIC_BIAS_OFF, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias Gen", ES8328_ADCPOWER, + ES8328_ADCPOWER_ADC_BIAS_GEN_OFF, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DAC STM", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_DACSTM_RESET, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC STM", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_ADCSTM_RESET, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DAC DIG", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_DACDIG_OFF, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC DIG", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_ADCDIG_OFF, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DAC DLL", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_DACDLL_OFF, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC DLL", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_ADCDLL_OFF, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_ADCVREF_OFF, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_DACVREF_OFF, 1, NULL, 0), + + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", ES8328_DACPOWER, + ES8328_DACPOWER_RDAC_OFF, 1), + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", ES8328_DACPOWER, + ES8328_DACPOWER_LDAC_OFF, 1), + + SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + &es8328_left_mixer_controls[0], + ARRAY_SIZE(es8328_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + &es8328_right_mixer_controls[0], + ARRAY_SIZE(es8328_right_mixer_controls)), + + SND_SOC_DAPM_PGA("Right Out 2", ES8328_DACPOWER, + ES8328_DACPOWER_ROUT2_ON, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 2", ES8328_DACPOWER, + ES8328_DACPOWER_LOUT2_ON, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Out 1", ES8328_DACPOWER, + ES8328_DACPOWER_ROUT1_ON, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 1", ES8328_DACPOWER, + ES8328_DACPOWER_LOUT1_ON, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + + SND_SOC_DAPM_INPUT("LINPUT1"), + SND_SOC_DAPM_INPUT("LINPUT2"), + SND_SOC_DAPM_INPUT("RINPUT1"), + SND_SOC_DAPM_INPUT("RINPUT2"), +}; + +static const struct snd_soc_dapm_route es8328_dapm_routes[] = { + + { "Left Line Mux", "Line 1", "LINPUT1" }, + { "Left Line Mux", "Line 2", "LINPUT2" }, + { "Left Line Mux", "PGA", "Left PGA Mux" }, + { "Left Line Mux", "Differential", "Differential Mux" }, + + { "Right Line Mux", "Line 1", "RINPUT1" }, + { "Right Line Mux", "Line 2", "RINPUT2" }, + { "Right Line Mux", "PGA", "Right PGA Mux" }, + { "Right Line Mux", "Differential", "Differential Mux" }, + + { "Left PGA Mux", "Line 1", "LINPUT1" }, + { "Left PGA Mux", "Line 2", "LINPUT2" }, + { "Left PGA Mux", "Differential", "Differential Mux" }, + + { "Right PGA Mux", "Line 1", "RINPUT1" }, + { "Right PGA Mux", "Line 2", "RINPUT2" }, + { "Right PGA Mux", "Differential", "Differential Mux" }, + + { "Differential Mux", "Line 1", "LINPUT1" }, + { "Differential Mux", "Line 1", "RINPUT1" }, + { "Differential Mux", "Line 2", "LINPUT2" }, + { "Differential Mux", "Line 2", "RINPUT2" }, + + { "Left ADC Mux", "Stereo", "Left PGA Mux" }, + { "Left ADC Mux", "Mono (Left)", "Left PGA Mux" }, + { "Left ADC Mux", "Digital Mono", "Left PGA Mux" }, + + { "Right ADC Mux", "Stereo", "Right PGA Mux" }, + { "Right ADC Mux", "Mono (Right)", "Right PGA Mux" }, + { "Right ADC Mux", "Digital Mono", "Right PGA Mux" }, + + { "Left ADC", NULL, "Left ADC Mux" }, + { "Right ADC", NULL, "Right ADC Mux" }, + + { "ADC DIG", NULL, "ADC STM" }, + { "ADC DIG", NULL, "ADC Vref" }, + { "ADC DIG", NULL, "ADC DLL" }, + + { "Left ADC", NULL, "ADC DIG" }, + { "Right ADC", NULL, "ADC DIG" }, + + { "Mic Bias", NULL, "Mic Bias Gen" }, + + { "Left Line Mux", "Line 1", "LINPUT1" }, + { "Left Line Mux", "Line 2", "LINPUT2" }, + { "Left Line Mux", "PGA", "Left PGA Mux" }, + { "Left Line Mux", "Differential", "Differential Mux" }, + + { "Right Line Mux", "Line 1", "RINPUT1" }, + { "Right Line Mux", "Line 2", "RINPUT2" }, + { "Right Line Mux", "PGA", "Right PGA Mux" }, + { "Right Line Mux", "Differential", "Differential Mux" }, + + { "Left Out 1", NULL, "Left DAC" }, + { "Right Out 1", NULL, "Right DAC" }, + { "Left Out 2", NULL, "Left DAC" }, + { "Right Out 2", NULL, "Right DAC" }, + + { "Left Mixer", "Playback Switch", "Left DAC" }, + { "Left Mixer", "Left Bypass Switch", "Left Line Mux" }, + { "Left Mixer", "Right Playback Switch", "Right DAC" }, + { "Left Mixer", "Right Bypass Switch", "Right Line Mux" }, + + { "Right Mixer", "Left Playback Switch", "Left DAC" }, + { "Right Mixer", "Left Bypass Switch", "Left Line Mux" }, + { "Right Mixer", "Playback Switch", "Right DAC" }, + { "Right Mixer", "Right Bypass Switch", "Right Line Mux" }, + + { "DAC DIG", NULL, "DAC STM" }, + { "DAC DIG", NULL, "DAC Vref" }, + { "DAC DIG", NULL, "DAC DLL" }, + + { "Left DAC", NULL, "DAC DIG" }, + { "Right DAC", NULL, "DAC DIG" }, + + { "Left Out 1", NULL, "Left Mixer" }, + { "LOUT1", NULL, "Left Out 1" }, + { "Right Out 1", NULL, "Right Mixer" }, + { "ROUT1", NULL, "Right Out 1" }, + + { "Left Out 2", NULL, "Left Mixer" }, + { "LOUT2", NULL, "Left Out 2" }, + { "Right Out 2", NULL, "Right Mixer" }, + { "ROUT2", NULL, "Right Out 2" }, +}; + +static int es8328_mute(struct snd_soc_dai *dai, int mute) +{ + return snd_soc_update_bits(dai->codec, ES8328_DACCONTROL3, + ES8328_DACCONTROL3_DACMUTE, + mute ? ES8328_DACCONTROL3_DACMUTE : 0); +} + +static int es8328_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 es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int clk_rate; + int i; + int reg; + u8 ratio; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = ES8328_DACCONTROL2; + else + reg = ES8328_ADCCONTROL5; + + clk_rate = clk_get_rate(es8328->clk); + + if ((clk_rate != ES8328_SYSCLK_RATE_1X) && + (clk_rate != ES8328_SYSCLK_RATE_2X)) { + dev_err(codec->dev, + "%s: clock is running at %d Hz, not %d or %d Hz\n", + __func__, clk_rate, + ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X); + return -EINVAL; + } + + /* find master mode MCLK to sampling frequency ratio */ + ratio = mclk_ratios[0].rate; + for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++) + if (params_rate(params) <= mclk_ratios[i].rate) + ratio = mclk_ratios[i].ratio; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + es8328->playback_fs = params_rate(params); + es8328_set_deemph(codec); + } + + return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio); +} + +static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int clk_rate; + u8 mode = ES8328_DACCONTROL1_DACWL_16; + + /* set master/slave audio interface */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM) + return -EINVAL; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + mode |= ES8328_DACCONTROL1_DACFORMAT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + + snd_soc_write(codec, ES8328_DACCONTROL1, mode); + snd_soc_write(codec, ES8328_ADCCONTROL4, mode); + + /* Master serial port mode, with BCLK generated automatically */ + clk_rate = clk_get_rate(es8328->clk); + if (clk_rate == ES8328_SYSCLK_RATE_1X) + snd_soc_write(codec, ES8328_MASTERMODE, + ES8328_MASTERMODE_MSC); + else + snd_soc_write(codec, ES8328_MASTERMODE, + ES8328_MASTERMODE_MCLKDIV2 | + ES8328_MASTERMODE_MSC); + + return 0; +} + +static int es8328_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VREF, VMID=2x50k, digital enabled */ + snd_soc_write(codec, ES8328_CHIPPOWER, 0); + snd_soc_update_bits(codec, ES8328_CONTROL1, + ES8328_CONTROL1_VMIDSEL_MASK | + ES8328_CONTROL1_ENREF, + ES8328_CONTROL1_VMIDSEL_50k | + ES8328_CONTROL1_ENREF); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, ES8328_CONTROL1, + ES8328_CONTROL1_VMIDSEL_MASK | + ES8328_CONTROL1_ENREF, + ES8328_CONTROL1_VMIDSEL_5k | + ES8328_CONTROL1_ENREF); + + /* Charge caps */ + msleep(100); + } + + snd_soc_write(codec, ES8328_CONTROL2, + ES8328_CONTROL2_OVERCURRENT_ON | + ES8328_CONTROL2_THERMAL_SHUTDOWN_ON); + + /* VREF, VMID=2*500k, digital stopped */ + snd_soc_update_bits(codec, ES8328_CONTROL1, + ES8328_CONTROL1_VMIDSEL_MASK | + ES8328_CONTROL1_ENREF, + ES8328_CONTROL1_VMIDSEL_500k | + ES8328_CONTROL1_ENREF); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ES8328_CONTROL1, + ES8328_CONTROL1_VMIDSEL_MASK | + ES8328_CONTROL1_ENREF, + 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops es8328_dai_ops = { + .hw_params = es8328_hw_params, + .digital_mute = es8328_mute, + .set_fmt = es8328_set_dai_fmt, +}; + +static struct snd_soc_dai_driver es8328_dai = { + .name = "es8328-hifi-analog", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ES8328_RATES, + .formats = ES8328_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ES8328_RATES, + .formats = ES8328_FORMATS, + }, + .ops = &es8328_dai_ops, +}; + +static int es8328_suspend(struct snd_soc_codec *codec) +{ + struct es8328_priv *es8328; + int ret; + + es8328 = snd_soc_codec_get_drvdata(codec); + + clk_disable_unprepare(es8328->clk); + + ret = regulator_bulk_disable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + if (ret) { + dev_err(codec->dev, "unable to disable regulators\n"); + return ret; + } + return 0; +} + +static int es8328_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + struct es8328_priv *es8328; + int ret; + + es8328 = snd_soc_codec_get_drvdata(codec); + + ret = clk_prepare_enable(es8328->clk); + if (ret) { + dev_err(codec->dev, "unable to enable clock\n"); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + if (ret) { + dev_err(codec->dev, "unable to enable regulators\n"); + return ret; + } + + regcache_mark_dirty(regmap); + ret = regcache_sync(regmap); + if (ret) { + dev_err(codec->dev, "unable to sync regcache\n"); + return ret; + } + + return 0; +} + +static int es8328_codec_probe(struct snd_soc_codec *codec) +{ + struct es8328_priv *es8328; + int ret; + + es8328 = snd_soc_codec_get_drvdata(codec); + + ret = regulator_bulk_enable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + if (ret) { + dev_err(codec->dev, "unable to enable regulators\n"); + return ret; + } + + /* Setup clocks */ + es8328->clk = devm_clk_get(codec->dev, NULL); + if (IS_ERR(es8328->clk)) { + dev_err(codec->dev, "codec clock missing or invalid\n"); + ret = PTR_ERR(es8328->clk); + goto clk_fail; + } + + ret = clk_prepare_enable(es8328->clk); + if (ret) { + dev_err(codec->dev, "unable to prepare codec clk\n"); + goto clk_fail; + } + + return 0; + +clk_fail: + regulator_bulk_disable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + return ret; +} + +static int es8328_remove(struct snd_soc_codec *codec) +{ + struct es8328_priv *es8328; + + es8328 = snd_soc_codec_get_drvdata(codec); + + if (es8328->clk) + clk_disable_unprepare(es8328->clk); + + regulator_bulk_disable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + + return 0; +} + +const struct regmap_config es8328_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ES8328_REG_MAX, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(es8328_regmap_config); + +static struct snd_soc_codec_driver es8328_codec_driver = { + .probe = es8328_codec_probe, + .suspend = es8328_suspend, + .resume = es8328_resume, + .remove = es8328_remove, + .set_bias_level = es8328_set_bias_level, + .suspend_bias_off = true, + + .controls = es8328_snd_controls, + .num_controls = ARRAY_SIZE(es8328_snd_controls), + .dapm_widgets = es8328_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8328_dapm_widgets), + .dapm_routes = es8328_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8328_dapm_routes), +}; + +int es8328_probe(struct device *dev, struct regmap *regmap) +{ + struct es8328_priv *es8328; + int ret; + int i; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + es8328 = devm_kzalloc(dev, sizeof(*es8328), GFP_KERNEL); + if (es8328 == NULL) + return -ENOMEM; + + es8328->regmap = regmap; + + for (i = 0; i < ARRAY_SIZE(es8328->supplies); i++) + es8328->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(es8328->supplies), + es8328->supplies); + if (ret) { + dev_err(dev, "unable to get regulators\n"); + return ret; + } + + dev_set_drvdata(dev, es8328); + + return snd_soc_register_codec(dev, + &es8328_codec_driver, &es8328_dai, 1); +} +EXPORT_SYMBOL_GPL(es8328_probe); + +MODULE_DESCRIPTION("ASoC ES8328 driver"); +MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h new file mode 100644 index 000000000000..cb36afe10c0e --- /dev/null +++ b/sound/soc/codecs/es8328.h @@ -0,0 +1,314 @@ +/* + * es8328.h -- ES8328 ALSA SoC Audio driver + */ + +#ifndef _ES8328_H +#define _ES8328_H + +#include <linux/regmap.h> + +struct device; + +extern const struct regmap_config es8328_regmap_config; +int es8328_probe(struct device *dev, struct regmap *regmap); + +#define ES8328_DACLVOL 46 +#define ES8328_DACRVOL 47 +#define ES8328_DACCTL 28 +#define ES8328_RATEMASK (0x1f << 0) + +#define ES8328_CONTROL1 0x00 +#define ES8328_CONTROL1_VMIDSEL_OFF (0 << 0) +#define ES8328_CONTROL1_VMIDSEL_50k (1 << 0) +#define ES8328_CONTROL1_VMIDSEL_500k (2 << 0) +#define ES8328_CONTROL1_VMIDSEL_5k (3 << 0) +#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0) +#define ES8328_CONTROL1_ENREF (1 << 2) +#define ES8328_CONTROL1_SEQEN (1 << 3) +#define ES8328_CONTROL1_SAMEFS (1 << 4) +#define ES8328_CONTROL1_DACMCLK_ADC (0 << 5) +#define ES8328_CONTROL1_DACMCLK_DAC (1 << 5) +#define ES8328_CONTROL1_LRCM (1 << 6) +#define ES8328_CONTROL1_SCP_RESET (1 << 7) + +#define ES8328_CONTROL2 0x01 +#define ES8328_CONTROL2_VREF_BUF_OFF (1 << 0) +#define ES8328_CONTROL2_VREF_LOWPOWER (1 << 1) +#define ES8328_CONTROL2_IBIASGEN_OFF (1 << 2) +#define ES8328_CONTROL2_ANALOG_OFF (1 << 3) +#define ES8328_CONTROL2_VREF_BUF_LOWPOWER (1 << 4) +#define ES8328_CONTROL2_VCM_MOD_LOWPOWER (1 << 5) +#define ES8328_CONTROL2_OVERCURRENT_ON (1 << 6) +#define ES8328_CONTROL2_THERMAL_SHUTDOWN_ON (1 << 7) + +#define ES8328_CHIPPOWER 0x02 +#define ES8328_CHIPPOWER_DACVREF_OFF 0 +#define ES8328_CHIPPOWER_ADCVREF_OFF 1 +#define ES8328_CHIPPOWER_DACDLL_OFF 2 +#define ES8328_CHIPPOWER_ADCDLL_OFF 3 +#define ES8328_CHIPPOWER_DACSTM_RESET 4 +#define ES8328_CHIPPOWER_ADCSTM_RESET 5 +#define ES8328_CHIPPOWER_DACDIG_OFF 6 +#define ES8328_CHIPPOWER_ADCDIG_OFF 7 + +#define ES8328_ADCPOWER 0x03 +#define ES8328_ADCPOWER_INT1_LOWPOWER 0 +#define ES8328_ADCPOWER_FLASH_ADC_LOWPOWER 1 +#define ES8328_ADCPOWER_ADC_BIAS_GEN_OFF 2 +#define ES8328_ADCPOWER_MIC_BIAS_OFF 3 +#define ES8328_ADCPOWER_ADCR_OFF 4 +#define ES8328_ADCPOWER_ADCL_OFF 5 +#define ES8328_ADCPOWER_AINR_OFF 6 +#define ES8328_ADCPOWER_AINL_OFF 7 + +#define ES8328_DACPOWER 0x04 +#define ES8328_DACPOWER_OUT3_ON 0 +#define ES8328_DACPOWER_MONO_ON 1 +#define ES8328_DACPOWER_ROUT2_ON 2 +#define ES8328_DACPOWER_LOUT2_ON 3 +#define ES8328_DACPOWER_ROUT1_ON 4 +#define ES8328_DACPOWER_LOUT1_ON 5 +#define ES8328_DACPOWER_RDAC_OFF 6 +#define ES8328_DACPOWER_LDAC_OFF 7 + +#define ES8328_CHIPLOPOW1 0x05 +#define ES8328_CHIPLOPOW2 0x06 +#define ES8328_ANAVOLMANAG 0x07 + +#define ES8328_MASTERMODE 0x08 +#define ES8328_MASTERMODE_BCLKDIV (0 << 0) +#define ES8328_MASTERMODE_BCLK_INV (1 << 5) +#define ES8328_MASTERMODE_MCLKDIV2 (1 << 6) +#define ES8328_MASTERMODE_MSC (1 << 7) + +#define ES8328_ADCCONTROL1 0x09 +#define ES8328_ADCCONTROL2 0x0a +#define ES8328_ADCCONTROL3 0x0b +#define ES8328_ADCCONTROL4 0x0c +#define ES8328_ADCCONTROL5 0x0d +#define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0) + +#define ES8328_ADCCONTROL6 0x0e + +#define ES8328_ADCCONTROL7 0x0f +#define ES8328_ADCCONTROL7_ADC_MUTE (1 << 2) +#define ES8328_ADCCONTROL7_ADC_LER (1 << 3) +#define ES8328_ADCCONTROL7_ADC_ZERO_CROSS (1 << 4) +#define ES8328_ADCCONTROL7_ADC_SOFT_RAMP (1 << 5) +#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_4 (0 << 6) +#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_8 (1 << 6) +#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_16 (2 << 6) +#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_32 (3 << 6) + +#define ES8328_ADCCONTROL8 0x10 +#define ES8328_ADCCONTROL9 0x11 +#define ES8328_ADCCONTROL10 0x12 +#define ES8328_ADCCONTROL11 0x13 +#define ES8328_ADCCONTROL12 0x14 +#define ES8328_ADCCONTROL13 0x15 +#define ES8328_ADCCONTROL14 0x16 + +#define ES8328_DACCONTROL1 0x17 +#define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1) +#define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1) +#define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1) +#define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1) +#define ES8328_DACCONTROL1_DACWL_24 (0 << 3) +#define ES8328_DACCONTROL1_DACWL_20 (1 << 3) +#define ES8328_DACCONTROL1_DACWL_18 (2 << 3) +#define ES8328_DACCONTROL1_DACWL_16 (3 << 3) +#define ES8328_DACCONTROL1_DACWL_32 (4 << 3) +#define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6) +#define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6) +#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6) +#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK1 (1 << 6) +#define ES8328_DACCONTROL1_LRSWAP (1 << 7) + +#define ES8328_DACCONTROL2 0x18 +#define ES8328_DACCONTROL2_RATEMASK (0x1f << 0) +#define ES8328_DACCONTROL2_DOUBLESPEED (1 << 5) + +#define ES8328_DACCONTROL3 0x19 +#define ES8328_DACCONTROL3_AUTOMUTE (1 << 2) +#define ES8328_DACCONTROL3_DACMUTE (1 << 2) +#define ES8328_DACCONTROL3_LEFTGAINVOL (1 << 3) +#define ES8328_DACCONTROL3_DACZEROCROSS (1 << 4) +#define ES8328_DACCONTROL3_DACSOFTRAMP (1 << 5) +#define ES8328_DACCONTROL3_DACRAMPRATE (3 << 6) + +#define ES8328_LDACVOL 0x1a +#define ES8328_LDACVOL_MASK (0 << 0) +#define ES8328_LDACVOL_MAX (0xc0) + +#define ES8328_RDACVOL 0x1b +#define ES8328_RDACVOL_MASK (0 << 0) +#define ES8328_RDACVOL_MAX (0xc0) + +#define ES8328_DACVOL_MAX (0xc0) + +#define ES8328_DACCONTROL4 0x1a +#define ES8328_DACCONTROL5 0x1b + +#define ES8328_DACCONTROL6 0x1c +#define ES8328_DACCONTROL6_CLICKFREE (1 << 3) +#define ES8328_DACCONTROL6_DAC_INVR (1 << 4) +#define ES8328_DACCONTROL6_DAC_INVL (1 << 5) +#define ES8328_DACCONTROL6_DEEMPH_OFF (0 << 6) +#define ES8328_DACCONTROL6_DEEMPH_32k (1 << 6) +#define ES8328_DACCONTROL6_DEEMPH_44_1k (2 << 6) +#define ES8328_DACCONTROL6_DEEMPH_48k (3 << 6) + +#define ES8328_DACCONTROL7 0x1d +#define ES8328_DACCONTROL7_VPP_SCALE_3p5 (0 << 0) +#define ES8328_DACCONTROL7_VPP_SCALE_4p0 (1 << 0) +#define ES8328_DACCONTROL7_VPP_SCALE_3p0 (2 << 0) +#define ES8328_DACCONTROL7_VPP_SCALE_2p5 (3 << 0) +#define ES8328_DACCONTROL7_SHELVING_STRENGTH (1 << 2) /* In eights */ +#define ES8328_DACCONTROL7_MONO (1 << 5) +#define ES8328_DACCONTROL7_ZEROR (1 << 6) +#define ES8328_DACCONTROL7_ZEROL (1 << 7) + +/* Shelving filter */ +#define ES8328_DACCONTROL8 0x1e +#define ES8328_DACCONTROL9 0x1f +#define ES8328_DACCONTROL10 0x20 +#define ES8328_DACCONTROL11 0x21 +#define ES8328_DACCONTROL12 0x22 +#define ES8328_DACCONTROL13 0x23 +#define ES8328_DACCONTROL14 0x24 +#define ES8328_DACCONTROL15 0x25 + +#define ES8328_DACCONTROL16 0x26 +#define ES8328_DACCONTROL16_RMIXSEL_RIN1 (0 << 0) +#define ES8328_DACCONTROL16_RMIXSEL_RIN2 (1 << 0) +#define ES8328_DACCONTROL16_RMIXSEL_RIN3 (2 << 0) +#define ES8328_DACCONTROL16_RMIXSEL_RADC (3 << 0) +#define ES8328_DACCONTROL16_LMIXSEL_LIN1 (0 << 3) +#define ES8328_DACCONTROL16_LMIXSEL_LIN2 (1 << 3) +#define ES8328_DACCONTROL16_LMIXSEL_LIN3 (2 << 3) +#define ES8328_DACCONTROL16_LMIXSEL_LADC (3 << 3) + +#define ES8328_DACCONTROL17 0x27 +#define ES8328_DACCONTROL17_LI2LOVOL (7 << 3) +#define ES8328_DACCONTROL17_LI2LO (1 << 6) +#define ES8328_DACCONTROL17_LD2LO (1 << 7) + +#define ES8328_DACCONTROL18 0x28 +#define ES8328_DACCONTROL18_RI2LOVOL (7 << 3) +#define ES8328_DACCONTROL18_RI2LO (1 << 6) +#define ES8328_DACCONTROL18_RD2LO (1 << 7) + +#define ES8328_DACCONTROL19 0x29 +#define ES8328_DACCONTROL19_LI2ROVOL (7 << 3) +#define ES8328_DACCONTROL19_LI2RO (1 << 6) +#define ES8328_DACCONTROL19_LD2RO (1 << 7) + +#define ES8328_DACCONTROL20 0x2a +#define ES8328_DACCONTROL20_RI2ROVOL (7 << 3) +#define ES8328_DACCONTROL20_RI2RO (1 << 6) +#define ES8328_DACCONTROL20_RD2RO (1 << 7) + +#define ES8328_DACCONTROL21 0x2b +#define ES8328_DACCONTROL21_LI2MOVOL (7 << 3) +#define ES8328_DACCONTROL21_LI2MO (1 << 6) +#define ES8328_DACCONTROL21_LD2MO (1 << 7) + +#define ES8328_DACCONTROL22 0x2c +#define ES8328_DACCONTROL22_RI2MOVOL (7 << 3) +#define ES8328_DACCONTROL22_RI2MO (1 << 6) +#define ES8328_DACCONTROL22_RD2MO (1 << 7) + +#define ES8328_DACCONTROL23 0x2d +#define ES8328_DACCONTROL23_MOUTINV (1 << 1) +#define ES8328_DACCONTROL23_HPSWPOL (1 << 2) +#define ES8328_DACCONTROL23_HPSWEN (1 << 3) +#define ES8328_DACCONTROL23_VROI_1p5k (0 << 4) +#define ES8328_DACCONTROL23_VROI_40k (1 << 4) +#define ES8328_DACCONTROL23_OUT3_VREF (0 << 5) +#define ES8328_DACCONTROL23_OUT3_ROUT1 (1 << 5) +#define ES8328_DACCONTROL23_OUT3_MONOOUT (2 << 5) +#define ES8328_DACCONTROL23_OUT3_RIGHT_MIXER (3 << 5) +#define ES8328_DACCONTROL23_ROUT2INV (1 << 7) + +/* LOUT1 Amplifier */ +#define ES8328_LOUT1VOL 0x2e +#define ES8328_LOUT1VOL_MASK (0 << 5) +#define ES8328_LOUT1VOL_MAX (0x24) + +/* ROUT1 Amplifier */ +#define ES8328_ROUT1VOL 0x2f +#define ES8328_ROUT1VOL_MASK (0 << 5) +#define ES8328_ROUT1VOL_MAX (0x24) + +#define ES8328_OUT1VOL_MAX (0x24) + +/* LOUT2 Amplifier */ +#define ES8328_LOUT2VOL 0x30 +#define ES8328_LOUT2VOL_MASK (0 << 5) +#define ES8328_LOUT2VOL_MAX (0x24) + +/* ROUT2 Amplifier */ +#define ES8328_ROUT2VOL 0x31 +#define ES8328_ROUT2VOL_MASK (0 << 5) +#define ES8328_ROUT2VOL_MAX (0x24) + +#define ES8328_OUT2VOL_MAX (0x24) + +/* Mono Out Amplifier */ +#define ES8328_MONOOUTVOL 0x32 +#define ES8328_MONOOUTVOL_MASK (0 << 5) +#define ES8328_MONOOUTVOL_MAX (0x24) + +#define ES8328_DACCONTROL29 0x33 +#define ES8328_DACCONTROL30 0x34 + +#define ES8328_SYSCLK 0 + +#define ES8328_REG_MAX 0x35 + +#define ES8328_PLL1 0 +#define ES8328_PLL2 1 + +/* clock inputs */ +#define ES8328_MCLK 0 +#define ES8328_PCMCLK 1 + +/* clock divider id's */ +#define ES8328_PCMDIV 0 +#define ES8328_BCLKDIV 1 +#define ES8328_VXCLKDIV 2 + +/* PCM clock dividers */ +#define ES8328_PCM_DIV_1 (0 << 6) +#define ES8328_PCM_DIV_3 (2 << 6) +#define ES8328_PCM_DIV_5_5 (3 << 6) +#define ES8328_PCM_DIV_2 (4 << 6) +#define ES8328_PCM_DIV_4 (5 << 6) +#define ES8328_PCM_DIV_6 (6 << 6) +#define ES8328_PCM_DIV_8 (7 << 6) + +/* BCLK clock dividers */ +#define ES8328_BCLK_DIV_1 (0 << 7) +#define ES8328_BCLK_DIV_2 (1 << 7) +#define ES8328_BCLK_DIV_4 (2 << 7) +#define ES8328_BCLK_DIV_8 (3 << 7) + +/* VXCLK clock dividers */ +#define ES8328_VXCLK_DIV_1 (0 << 6) +#define ES8328_VXCLK_DIV_2 (1 << 6) +#define ES8328_VXCLK_DIV_4 (2 << 6) +#define ES8328_VXCLK_DIV_8 (3 << 6) +#define ES8328_VXCLK_DIV_16 (4 << 6) + +#define ES8328_DAI_HIFI 0 +#define ES8328_DAI_VOICE 1 + +#define ES8328_1536FS 1536 +#define ES8328_1024FS 1024 +#define ES8328_768FS 768 +#define ES8328_512FS 512 +#define ES8328_384FS 384 +#define ES8328_256FS 256 +#define ES8328_128FS 128 + +#endif diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 275b3f72f3f4..c1ae5764983f 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1395,18 +1395,6 @@ static struct snd_soc_dai_driver lm49453_dai[] = { }, }; -static int lm49453_suspend(struct snd_soc_codec *codec) -{ - lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int lm49453_resume(struct snd_soc_codec *codec) -{ - lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - /* power down chip */ static int lm49453_remove(struct snd_soc_codec *codec) { @@ -1416,8 +1404,6 @@ static int lm49453_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { .remove = lm49453_remove, - .suspend = lm49453_suspend, - .resume = lm49453_resume, .set_bias_level = lm49453_set_bias_level, .controls = lm49453_snd_controls, .num_controls = ARRAY_SIZE(lm49453_snd_controls), diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 163ec3855fd4..0c8aefab404c 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -259,13 +259,13 @@ static const struct soc_enum pcm512x_veds = pcm512x_ramp_step_text); static const struct snd_kcontrol_new pcm512x_controls[] = { -SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2, +SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2, PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL, PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), -SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, +SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, PCM512x_RQMR_SHIFT, 1, 1), SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e4f6102efc1a..b86b426f159d 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -51,7 +51,7 @@ static struct reg_default rt286_index_def[] = { { 0x04, 0xaf01 }, { 0x08, 0x000d }, { 0x09, 0xd810 }, - { 0x0a, 0x0060 }, + { 0x0a, 0x0120 }, { 0x0b, 0x0000 }, { 0x0d, 0x2800 }, { 0x0f, 0x0000 }, @@ -60,7 +60,7 @@ static struct reg_default rt286_index_def[] = { { 0x33, 0x0208 }, { 0x49, 0x0004 }, { 0x4f, 0x50e9 }, - { 0x50, 0x2c00 }, + { 0x50, 0x2000 }, { 0x63, 0x2902 }, { 0x67, 0x1111 }, { 0x68, 0x1016 }, @@ -104,7 +104,6 @@ static const struct reg_default rt286_reg[] = { { 0x02170700, 0x00000000 }, { 0x02270100, 0x00000000 }, { 0x02370100, 0x00000000 }, - { 0x02040000, 0x00004002 }, { 0x01870700, 0x00000020 }, { 0x00830000, 0x000000c3 }, { 0x00930000, 0x000000c3 }, @@ -192,7 +191,6 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) /*handle index registers*/ if (reg <= 0xff) { rt286_hw_write(client, RT286_COEF_INDEX, reg); - reg = RT286_PROC_COEF; for (i = 0; i < INDEX_CACHE_SIZE; i++) { if (reg == rt286->index_cache[i].reg) { rt286->index_cache[i].def = value; @@ -200,6 +198,7 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) } } + reg = RT286_PROC_COEF; } data[0] = (reg >> 24) & 0xff; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 2fdcbb8e8a2a..c3f2decd643c 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2089,6 +2089,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = { static const struct regmap_config rt5640_regmap = { .reg_bits = 8, .val_bits = 16, + .use_single_rw = true, .max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) * RT5640_PR_SPACING), diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 67f14556462f..dc978ad59fc7 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -19,6 +19,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -1700,14 +1701,19 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_INPUT("Haptic Generator"), - SND_SOC_DAPM_PGA("DMIC1", RT5677_DMIC_CTRL1, RT5677_DMIC_1_EN_SFT, 0, - NULL, 0), - SND_SOC_DAPM_PGA("DMIC2", RT5677_DMIC_CTRL1, RT5677_DMIC_2_EN_SFT, 0, - NULL, 0), - SND_SOC_DAPM_PGA("DMIC3", RT5677_DMIC_CTRL1, RT5677_DMIC_3_EN_SFT, 0, - NULL, 0), - SND_SOC_DAPM_PGA("DMIC4", RT5677_DMIC_CTRL2, RT5677_DMIC_4_EN_SFT, 0, - NULL, 0), + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC1 power", RT5677_DMIC_CTRL1, + RT5677_DMIC_1_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 power", RT5677_DMIC_CTRL1, + RT5677_DMIC_2_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC3 power", RT5677_DMIC_CTRL1, + RT5677_DMIC_3_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC4 power", RT5677_DMIC_CTRL2, + RT5677_DMIC_4_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, set_dmic_clk, SND_SOC_DAPM_PRE_PMU), @@ -2130,15 +2136,22 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DMIC L4", NULL, "DMIC CLK" }, { "DMIC R4", NULL, "DMIC CLK" }, + { "DMIC L1", NULL, "DMIC1 power" }, + { "DMIC R1", NULL, "DMIC1 power" }, + { "DMIC L3", NULL, "DMIC3 power" }, + { "DMIC R3", NULL, "DMIC3 power" }, + { "DMIC L4", NULL, "DMIC4 power" }, + { "DMIC R4", NULL, "DMIC4 power" }, + { "BST1", NULL, "IN1P" }, { "BST1", NULL, "IN1N" }, { "BST2", NULL, "IN2P" }, { "BST2", NULL, "IN2N" }, - { "IN1P", NULL, "micbias1" }, - { "IN1N", NULL, "micbias1" }, - { "IN2P", NULL, "micbias1" }, - { "IN2N", NULL, "micbias1" }, + { "IN1P", NULL, "MICBIAS1" }, + { "IN1N", NULL, "MICBIAS1" }, + { "IN2P", NULL, "MICBIAS1" }, + { "IN2N", NULL, "MICBIAS1" }, { "ADC 1", NULL, "BST1" }, { "ADC 1", NULL, "ADC 1 power" }, @@ -2793,6 +2806,16 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "PDM2R", NULL, "PDM2 R Mux" }, }; +static const struct snd_soc_dapm_route rt5677_dmic2_clk_1[] = { + { "DMIC L2", NULL, "DMIC1 power" }, + { "DMIC R2", NULL, "DMIC1 power" }, +}; + +static const struct snd_soc_dapm_route rt5677_dmic2_clk_2[] = { + { "DMIC L2", NULL, "DMIC2 power" }, + { "DMIC R2", NULL, "DMIC2 power" }, +}; + static int rt5677_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -3138,12 +3161,148 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, return 0; } +#ifdef CONFIG_GPIOLIB +static inline struct rt5677_priv *gpio_to_rt5677(struct gpio_chip *chip) +{ + return container_of(chip, struct rt5677_priv, gpio_chip); +} + +static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + + switch (offset) { + case RT5677_GPIO1 ... RT5677_GPIO5: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + 0x1 << (offset * 3 + 1), !!value << (offset * 3 + 1)); + break; + + case RT5677_GPIO6: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3, + RT5677_GPIO6_OUT_MASK, !!value << RT5677_GPIO6_OUT_SFT); + break; + + default: + break; + } +} + +static int rt5677_gpio_direction_out(struct gpio_chip *chip, + unsigned offset, int value) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + + switch (offset) { + case RT5677_GPIO1 ... RT5677_GPIO5: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + 0x3 << (offset * 3 + 1), + (0x2 | !!value) << (offset * 3 + 1)); + break; + + case RT5677_GPIO6: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3, + RT5677_GPIO6_DIR_MASK | RT5677_GPIO6_OUT_MASK, + RT5677_GPIO6_DIR_OUT | !!value << RT5677_GPIO6_OUT_SFT); + break; + + default: + break; + } + + return 0; +} + +static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + int value, ret; + + ret = regmap_read(rt5677->regmap, RT5677_GPIO_ST, &value); + if (ret < 0) + return ret; + + return (value & (0x1 << offset)) >> offset; +} + +static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + + switch (offset) { + case RT5677_GPIO1 ... RT5677_GPIO5: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + 0x1 << (offset * 3 + 2), 0x0); + break; + + case RT5677_GPIO6: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3, + RT5677_GPIO6_DIR_MASK, RT5677_GPIO6_DIR_IN); + break; + + default: + break; + } + + return 0; +} + +static struct gpio_chip rt5677_template_chip = { + .label = "rt5677", + .owner = THIS_MODULE, + .direction_output = rt5677_gpio_direction_out, + .set = rt5677_gpio_set, + .direction_input = rt5677_gpio_direction_in, + .get = rt5677_gpio_get, + .can_sleep = 1, +}; + +static void rt5677_init_gpio(struct i2c_client *i2c) +{ + struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); + int ret; + + rt5677->gpio_chip = rt5677_template_chip; + rt5677->gpio_chip.ngpio = RT5677_GPIO_NUM; + rt5677->gpio_chip.dev = &i2c->dev; + rt5677->gpio_chip.base = -1; + + ret = gpiochip_add(&rt5677->gpio_chip); + if (ret != 0) + dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret); +} + +static void rt5677_free_gpio(struct i2c_client *i2c) +{ + struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); + + gpiochip_remove(&rt5677->gpio_chip); +} +#else +static void rt5677_init_gpio(struct i2c_client *i2c) +{ +} + +static void rt5677_free_gpio(struct i2c_client *i2c) +{ +} +#endif + static int rt5677_probe(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); rt5677->codec = codec; + if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) { + snd_soc_dapm_add_routes(&codec->dapm, + rt5677_dmic2_clk_2, + ARRAY_SIZE(rt5677_dmic2_clk_2)); + } else { /*use dmic1 clock by default*/ + snd_soc_dapm_add_routes(&codec->dapm, + rt5677_dmic2_clk_1, + ARRAY_SIZE(rt5677_dmic2_clk_1)); + } + rt5677_set_bias_level(codec, SND_SOC_BIAS_OFF); regmap_write(rt5677->regmap, RT5677_DIG_MISC, 0x0020); @@ -3381,6 +3540,17 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5677->regmap, RT5677_IN1, RT5677_IN_DF2, RT5677_IN_DF2); + if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) { + regmap_update_bits(rt5677->regmap, RT5677_GEN_CTRL2, + RT5677_GPIO5_FUNC_MASK, + RT5677_GPIO5_FUNC_DMIC); + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + RT5677_GPIO5_DIR_MASK, + RT5677_GPIO5_DIR_OUT); + } + + rt5677_init_gpio(i2c); + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, rt5677_dai, ARRAY_SIZE(rt5677_dai)); } @@ -3388,6 +3558,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, static int rt5677_i2c_remove(struct i2c_client *i2c) { snd_soc_unregister_codec(&i2c->dev); + rt5677_free_gpio(i2c); return 0; } diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 863393e62096..b61b72cfcbd7 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1287,16 +1287,16 @@ #define RT5677_PLL1_PD_SFT 8 #define RT5677_PLL1_PD_1 (0x0 << 8) #define RT5677_PLL1_PD_2 (0x1 << 8) -#define RT5671_DAC_OSR_MASK (0x3 << 6) -#define RT5671_DAC_OSR_SFT 6 -#define RT5671_DAC_OSR_128 (0x0 << 6) -#define RT5671_DAC_OSR_64 (0x1 << 6) -#define RT5671_DAC_OSR_32 (0x2 << 6) -#define RT5671_ADC_OSR_MASK (0x3 << 4) -#define RT5671_ADC_OSR_SFT 4 -#define RT5671_ADC_OSR_128 (0x0 << 4) -#define RT5671_ADC_OSR_64 (0x1 << 4) -#define RT5671_ADC_OSR_32 (0x2 << 4) +#define RT5677_DAC_OSR_MASK (0x3 << 6) +#define RT5677_DAC_OSR_SFT 6 +#define RT5677_DAC_OSR_128 (0x0 << 6) +#define RT5677_DAC_OSR_64 (0x1 << 6) +#define RT5677_DAC_OSR_32 (0x2 << 6) +#define RT5677_ADC_OSR_MASK (0x3 << 4) +#define RT5677_ADC_OSR_SFT 4 +#define RT5677_ADC_OSR_128 (0x0 << 4) +#define RT5677_ADC_OSR_64 (0x1 << 4) +#define RT5677_ADC_OSR_32 (0x2 << 4) /* Global Clock Control 2 (0x81) */ #define RT5677_PLL2_PR_SRC_MASK (0x1 << 15) @@ -1312,18 +1312,18 @@ #define RT5677_PLL2_SRC_BCLK4 (0x4 << 12) #define RT5677_PLL2_SRC_RCCLK (0x5 << 12) #define RT5677_PLL2_SRC_SLIM (0x6 << 12) -#define RT5671_DSP_ASRC_O_SRC (0x3 << 10) -#define RT5671_DSP_ASRC_O_SRC_SFT 10 -#define RT5671_DSP_ASRC_O_MCLK (0x0 << 10) -#define RT5671_DSP_ASRC_O_PLL1 (0x1 << 10) -#define RT5671_DSP_ASRC_O_SLIM (0x2 << 10) -#define RT5671_DSP_ASRC_O_RCCLK (0x3 << 10) -#define RT5671_DSP_ASRC_I_SRC (0x3 << 8) -#define RT5671_DSP_ASRC_I_SRC_SFT 8 -#define RT5671_DSP_ASRC_I_MCLK (0x0 << 8) -#define RT5671_DSP_ASRC_I_PLL1 (0x1 << 8) -#define RT5671_DSP_ASRC_I_SLIM (0x2 << 8) -#define RT5671_DSP_ASRC_I_RCCLK (0x3 << 8) +#define RT5677_DSP_ASRC_O_SRC (0x3 << 10) +#define RT5677_DSP_ASRC_O_SRC_SFT 10 +#define RT5677_DSP_ASRC_O_MCLK (0x0 << 10) +#define RT5677_DSP_ASRC_O_PLL1 (0x1 << 10) +#define RT5677_DSP_ASRC_O_SLIM (0x2 << 10) +#define RT5677_DSP_ASRC_O_RCCLK (0x3 << 10) +#define RT5677_DSP_ASRC_I_SRC (0x3 << 8) +#define RT5677_DSP_ASRC_I_SRC_SFT 8 +#define RT5677_DSP_ASRC_I_MCLK (0x0 << 8) +#define RT5677_DSP_ASRC_I_PLL1 (0x1 << 8) +#define RT5677_DSP_ASRC_I_SLIM (0x2 << 8) +#define RT5677_DSP_ASRC_I_RCCLK (0x3 << 8) #define RT5677_DSP_CLK_SRC_MASK (0x1 << 7) #define RT5677_DSP_CLK_SRC_SFT 7 #define RT5677_DSP_CLK_SRC_PLL2 (0x0 << 7) @@ -1363,6 +1363,110 @@ #define RT5677_SEL_SRC_IB01 (0x1 << 0) #define RT5677_SEL_SRC_IB01_SFT 0 +/* GPIO status (0xbf) */ +#define RT5677_GPIO6_STATUS_MASK (0x1 << 5) +#define RT5677_GPIO6_STATUS_SFT 5 +#define RT5677_GPIO5_STATUS_MASK (0x1 << 4) +#define RT5677_GPIO5_STATUS_SFT 4 +#define RT5677_GPIO4_STATUS_MASK (0x1 << 3) +#define RT5677_GPIO4_STATUS_SFT 3 +#define RT5677_GPIO3_STATUS_MASK (0x1 << 2) +#define RT5677_GPIO3_STATUS_SFT 2 +#define RT5677_GPIO2_STATUS_MASK (0x1 << 1) +#define RT5677_GPIO2_STATUS_SFT 1 +#define RT5677_GPIO1_STATUS_MASK (0x1 << 0) +#define RT5677_GPIO1_STATUS_SFT 0 + +/* GPIO Control 1 (0xc0) */ +#define RT5677_GPIO1_PIN_MASK (0x1 << 15) +#define RT5677_GPIO1_PIN_SFT 15 +#define RT5677_GPIO1_PIN_GPIO1 (0x0 << 15) +#define RT5677_GPIO1_PIN_IRQ (0x1 << 15) +#define RT5677_IPTV_MODE_MASK (0x1 << 14) +#define RT5677_IPTV_MODE_SFT 14 +#define RT5677_IPTV_MODE_GPIO (0x0 << 14) +#define RT5677_IPTV_MODE_IPTV (0x1 << 14) +#define RT5677_FUNC_MODE_MASK (0x1 << 13) +#define RT5677_FUNC_MODE_SFT 13 +#define RT5677_FUNC_MODE_DMIC_GPIO (0x0 << 13) +#define RT5677_FUNC_MODE_JTAG (0x1 << 13) + +/* GPIO Control 2 (0xc1) */ +#define RT5677_GPIO5_DIR_MASK (0x1 << 14) +#define RT5677_GPIO5_DIR_SFT 14 +#define RT5677_GPIO5_DIR_IN (0x0 << 14) +#define RT5677_GPIO5_DIR_OUT (0x1 << 14) +#define RT5677_GPIO5_OUT_MASK (0x1 << 13) +#define RT5677_GPIO5_OUT_SFT 13 +#define RT5677_GPIO5_OUT_LO (0x0 << 13) +#define RT5677_GPIO5_OUT_HI (0x1 << 13) +#define RT5677_GPIO5_P_MASK (0x1 << 12) +#define RT5677_GPIO5_P_SFT 12 +#define RT5677_GPIO5_P_NOR (0x0 << 12) +#define RT5677_GPIO5_P_INV (0x1 << 12) +#define RT5677_GPIO4_DIR_MASK (0x1 << 11) +#define RT5677_GPIO4_DIR_SFT 11 +#define RT5677_GPIO4_DIR_IN (0x0 << 11) +#define RT5677_GPIO4_DIR_OUT (0x1 << 11) +#define RT5677_GPIO4_OUT_MASK (0x1 << 10) +#define RT5677_GPIO4_OUT_SFT 10 +#define RT5677_GPIO4_OUT_LO (0x0 << 10) +#define RT5677_GPIO4_OUT_HI (0x1 << 10) +#define RT5677_GPIO4_P_MASK (0x1 << 9) +#define RT5677_GPIO4_P_SFT 9 +#define RT5677_GPIO4_P_NOR (0x0 << 9) +#define RT5677_GPIO4_P_INV (0x1 << 9) +#define RT5677_GPIO3_DIR_MASK (0x1 << 8) +#define RT5677_GPIO3_DIR_SFT 8 +#define RT5677_GPIO3_DIR_IN (0x0 << 8) +#define RT5677_GPIO3_DIR_OUT (0x1 << 8) +#define RT5677_GPIO3_OUT_MASK (0x1 << 7) +#define RT5677_GPIO3_OUT_SFT 7 +#define RT5677_GPIO3_OUT_LO (0x0 << 7) +#define RT5677_GPIO3_OUT_HI (0x1 << 7) +#define RT5677_GPIO3_P_MASK (0x1 << 6) +#define RT5677_GPIO3_P_SFT 6 +#define RT5677_GPIO3_P_NOR (0x0 << 6) +#define RT5677_GPIO3_P_INV (0x1 << 6) +#define RT5677_GPIO2_DIR_MASK (0x1 << 5) +#define RT5677_GPIO2_DIR_SFT 5 +#define RT5677_GPIO2_DIR_IN (0x0 << 5) +#define RT5677_GPIO2_DIR_OUT (0x1 << 5) +#define RT5677_GPIO2_OUT_MASK (0x1 << 4) +#define RT5677_GPIO2_OUT_SFT 4 +#define RT5677_GPIO2_OUT_LO (0x0 << 4) +#define RT5677_GPIO2_OUT_HI (0x1 << 4) +#define RT5677_GPIO2_P_MASK (0x1 << 3) +#define RT5677_GPIO2_P_SFT 3 +#define RT5677_GPIO2_P_NOR (0x0 << 3) +#define RT5677_GPIO2_P_INV (0x1 << 3) +#define RT5677_GPIO1_DIR_MASK (0x1 << 2) +#define RT5677_GPIO1_DIR_SFT 2 +#define RT5677_GPIO1_DIR_IN (0x0 << 2) +#define RT5677_GPIO1_DIR_OUT (0x1 << 2) +#define RT5677_GPIO1_OUT_MASK (0x1 << 1) +#define RT5677_GPIO1_OUT_SFT 1 +#define RT5677_GPIO1_OUT_LO (0x0 << 1) +#define RT5677_GPIO1_OUT_HI (0x1 << 1) +#define RT5677_GPIO1_P_MASK (0x1 << 0) +#define RT5677_GPIO1_P_SFT 0 +#define RT5677_GPIO1_P_NOR (0x0 << 0) +#define RT5677_GPIO1_P_INV (0x1 << 0) + +/* GPIO Control 3 (0xc2) */ +#define RT5677_GPIO6_DIR_MASK (0x1 << 2) +#define RT5677_GPIO6_DIR_SFT 2 +#define RT5677_GPIO6_DIR_IN (0x0 << 2) +#define RT5677_GPIO6_DIR_OUT (0x1 << 2) +#define RT5677_GPIO6_OUT_MASK (0x1 << 1) +#define RT5677_GPIO6_OUT_SFT 1 +#define RT5677_GPIO6_OUT_LO (0x0 << 1) +#define RT5677_GPIO6_OUT_HI (0x1 << 1) +#define RT5677_GPIO6_P_MASK (0x1 << 0) +#define RT5677_GPIO6_P_SFT 0 +#define RT5677_GPIO6_P_NOR (0x0 << 0) +#define RT5677_GPIO6_P_INV (0x1 << 0) + /* Virtual DSP Mixer Control (0xf7 0xf8 0xf9) */ #define RT5677_DSP_IB_01_H (0x1 << 15) #define RT5677_DSP_IB_01_H_SFT 15 @@ -1393,6 +1497,11 @@ #define RT5677_DSP_IB_9_L (0x1 << 1) #define RT5677_DSP_IB_9_L_SFT 1 +/* General Control2 (0xfc)*/ +#define RT5677_GPIO5_FUNC_MASK (0x1 << 9) +#define RT5677_GPIO5_FUNC_GPIO (0x0 << 9) +#define RT5677_GPIO5_FUNC_DMIC (0x1 << 9) + /* System Clock Source */ enum { RT5677_SCLK_S_MCLK, @@ -1418,6 +1527,16 @@ enum { RT5677_AIFS, }; +enum { + RT5677_GPIO1, + RT5677_GPIO2, + RT5677_GPIO3, + RT5677_GPIO4, + RT5677_GPIO5, + RT5677_GPIO6, + RT5677_GPIO_NUM, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1431,6 +1550,9 @@ struct rt5677_priv { int pll_src; int pll_in; int pll_out; +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif }; #endif /* __RT5677_H__ */ diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index e8680bea5f86..67ea55adb307 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -646,17 +646,6 @@ static struct snd_soc_dai_driver ssm2518_dai = { .ops = &ssm2518_dai_ops, }; -static int ssm2518_probe(struct snd_soc_codec *codec) -{ - return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); -} - -static int ssm2518_remove(struct snd_soc_codec *codec) -{ - ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { @@ -727,8 +716,6 @@ static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, } static struct snd_soc_codec_driver ssm2518_codec_driver = { - .probe = ssm2518_probe, - .remove = ssm2518_remove, .set_bias_level = ssm2518_set_bias_level, .set_sysclk = ssm2518_set_sysclk, .idle_bias_off = true, diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 484b3bbe8624..527de0463548 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -502,18 +502,11 @@ static struct snd_soc_dai_driver ssm2602_dai = { .symmetric_samplebits = 1, }; -static int ssm2602_suspend(struct snd_soc_codec *codec) -{ - ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int ssm2602_resume(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); regcache_sync(ssm2602->regmap); - ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -586,27 +579,14 @@ static int ssm260x_codec_probe(struct snd_soc_codec *codec) break; } - if (ret) - return ret; - - ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -/* remove everything here */ -static int ssm2602_remove(struct snd_soc_codec *codec) -{ - ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; + return ret; } static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .probe = ssm260x_codec_probe, - .remove = ssm2602_remove, - .suspend = ssm2602_suspend, .resume = ssm2602_resume, .set_bias_level = ssm2602_set_bias_level, + .suspend_bias_off = true, .controls = ssm260x_snd_controls, .num_controls = ARRAY_SIZE(ssm260x_snd_controls), @@ -647,7 +627,7 @@ int ssm2602_probe(struct device *dev, enum ssm2602_type type, return -ENOMEM; dev_set_drvdata(dev, ssm2602); - ssm2602->type = SSM2602; + ssm2602->type = type; ssm2602->regmap = regmap; return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602, diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 9aa1323fb2ab..89c748dd3d6e 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -4,7 +4,7 @@ * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver * * Copyright (C) 2012 ST Microelectronics - * Rajeev Kumar <rajeev-dlh.kumar@st.com> + * Rajeev Kumar <rajeevkumar.linux@gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -426,5 +426,5 @@ static struct i2c_driver sta529_i2c_driver = { module_i2c_driver(sta529_i2c_driver); MODULE_DESCRIPTION("ASoC STA529 codec driver"); -MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 0f64c7890eed..aea9e1ff9126 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -189,46 +189,57 @@ static const struct aic31xx_rate_divs aic31xx_divs[] = { /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */ /* 8k rate */ {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2}, + {12000000, 8000, 1, 8, 1920, 128, 32, 3, 128, 32, 3}, {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2}, {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2}, /* 11.025k rate */ {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2}, + {12000000, 11025, 1, 8, 4672, 128, 24, 3, 128, 24, 3}, {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2}, {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2}, /* 16k rate */ {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2}, + {12000000, 16000, 1, 8, 1920, 128, 16, 3, 128, 16, 3}, {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2}, {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2}, /* 22.05k rate */ {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2}, + {12000000, 22050, 1, 8, 4672, 128, 12, 3, 128, 12, 3}, {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2}, {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2}, /* 32k rate */ {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2}, + {12000000, 32000, 1, 8, 1920, 128, 8, 3, 128, 8, 3}, {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2}, {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2}, /* 44.1k rate */ {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2}, + {12000000, 44100, 1, 8, 4672, 128, 6, 3, 128, 6, 3}, {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2}, {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2}, /* 48k rate */ {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2}, + {12000000, 48000, 1, 7, 6800, 96, 5, 4, 96, 5, 4}, {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2}, {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2}, /* 88.2k rate */ {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2}, + {12000000, 88200, 1, 8, 4672, 64, 6, 3, 64, 6, 3}, {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2}, {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2}, /* 96k rate */ {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2}, + {12000000, 96000, 1, 7, 6800, 48, 5, 4, 48, 5, 4}, {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2}, {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2}, /* 176.4k rate */ {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2}, + {12000000, 176400, 1, 8, 4672, 32, 6, 3, 32, 6, 3}, {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2}, {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2}, /* 192k rate */ {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2}, + {12000000, 192000, 1, 7, 6800, 24, 5, 4, 24, 5, 4}, {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2}, {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2}, }; @@ -680,7 +691,9 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, struct snd_pcm_hw_params *params) { struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int bclk_score = snd_soc_params_to_frame_size(params); int bclk_n = 0; + int match = -1; int i; /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */ @@ -691,15 +704,37 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { if (aic31xx_divs[i].rate == params_rate(params) && - aic31xx_divs[i].mclk == aic31xx->sysclk) - break; + aic31xx_divs[i].mclk == aic31xx->sysclk) { + int s = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) % + snd_soc_params_to_frame_size(params); + int bn = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) / + snd_soc_params_to_frame_size(params); + if (s < bclk_score && bn > 0) { + match = i; + bclk_n = bn; + bclk_score = s; + } + } } - if (i == ARRAY_SIZE(aic31xx_divs)) { - dev_err(codec->dev, "%s: Sampling rate %u not supported\n", + if (match == -1) { + dev_err(codec->dev, + "%s: Sample rate (%u) and format not supported\n", __func__, params_rate(params)); + /* See bellow for details how fix this. */ return -EINVAL; } + if (bclk_score != 0) { + dev_warn(codec->dev, "Can not produce exact bitclock"); + /* This is fine if using dsp format, but if using i2s + there may be trouble. To fix the issue edit the + aic31xx_divs table for your mclk and sample + rate. Details can be found from: + http://www.ti.com/lit/ds/symlink/tlv320aic3100.pdf + Section: 5.6 CLOCK Generation and PLL + */ + } + i = match; /* PLL configuration */ snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, @@ -729,14 +764,6 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr); /* Bit clock divider configuration. */ - bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) - / snd_soc_params_to_frame_size(params); - if (bclk_n == 0) { - dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n", - __func__); - return -EINVAL; - } - snd_soc_update_bits(codec, AIC31XX_BCLKN, AIC31XX_PLL_MASK, bclk_n); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64f179ee9834..f2c416d16b6c 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1222,20 +1222,6 @@ static struct snd_soc_dai_driver aic3x_dai = { .symmetric_rates = 1, }; -static int aic3x_suspend(struct snd_soc_codec *codec) -{ - aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int aic3x_resume(struct snd_soc_codec *codec) -{ - aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static void aic3x_mono_init(struct snd_soc_codec *codec) { /* DAC to Mono Line Out default volume and route to Output mixer */ @@ -1429,8 +1415,6 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .idle_bias_off = true, .probe = aic3x_probe, .remove = aic3x_remove, - .suspend = aic3x_suspend, - .resume = aic3x_resume, .controls = aic3x_snd_controls, .num_controls = ARRAY_SIZE(aic3x_snd_controls), .dapm_widgets = aic3x_dapm_widgets, diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 7bb0d36d4c54..a01ad629ed61 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2319,11 +2319,8 @@ static void wm5100_init_gpio(struct i2c_client *i2c) static void wm5100_free_gpio(struct i2c_client *i2c) { struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c); - int ret; - ret = gpiochip_remove(&wm5100->gpio_chip); - if (ret != 0) - dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm5100->gpio_chip); } #else static void wm5100_init_gpio(struct i2c_client *i2c) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3dfdcc4197fa..628ec774cf22 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -212,7 +212,7 @@ static void wm8350_pga_work(struct work_struct *work) { struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out1 = &wm8350_data->out1, *out2 = &wm8350_data->out2; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e54e097f4fcb..21ca3a94fc96 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1433,7 +1433,7 @@ static void wm8753_work(struct work_struct *work) struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); wm8753_set_bias_level(codec, dapm->bias_level); } diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 0ea01dfcb6e1..3addc5fe5cb2 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -518,23 +518,6 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, return 0; } -#ifdef CONFIG_PM -static int wm8804_suspend(struct snd_soc_codec *codec) -{ - wm8804_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8804_resume(struct snd_soc_codec *codec) -{ - wm8804_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8804_suspend NULL -#define wm8804_resume NULL -#endif - static int wm8804_remove(struct snd_soc_codec *codec) { struct wm8804_priv *wm8804; @@ -671,8 +654,6 @@ static struct snd_soc_dai_driver wm8804_dai = { static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .probe = wm8804_probe, .remove = wm8804_remove, - .suspend = wm8804_suspend, - .resume = wm8804_resume, .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index aa0984864e76..c038b3e04398 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1877,11 +1877,7 @@ static void wm8903_init_gpio(struct wm8903_priv *wm8903) static void wm8903_free_gpio(struct wm8903_priv *wm8903) { - int ret; - - ret = gpiochip_remove(&wm8903->gpio_chip); - if (ret != 0) - dev_err(wm8903->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm8903->gpio_chip); } #else static void wm8903_init_gpio(struct wm8903_priv *wm8903) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 1098ae32f1f9..9077411e62ce 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3398,11 +3398,8 @@ static void wm8962_init_gpio(struct snd_soc_codec *codec) static void wm8962_free_gpio(struct snd_soc_codec *codec) { struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - int ret; - ret = gpiochip_remove(&wm8962->gpio_chip); - if (ret != 0) - dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm8962->gpio_chip); } #else static void wm8962_init_gpio(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 0499cd4cfb71..39ddb9b8834c 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -615,7 +615,7 @@ static void wm8971_work(struct work_struct *work) struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); wm8971_set_bias_level(codec, codec->dapm.bias_level); } diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index cae4ac5a5730..1288edeb8c7d 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1998,23 +1998,6 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, return 0; } -#ifdef CONFIG_PM -static int wm8995_suspend(struct snd_soc_codec *codec) -{ - wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8995_resume(struct snd_soc_codec *codec) -{ - wm8995_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8995_suspend NULL -#define wm8995_resume NULL -#endif - static int wm8995_remove(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; @@ -2220,8 +2203,6 @@ static struct snd_soc_dai_driver wm8995_dai[] = { static struct snd_soc_codec_driver soc_codec_dev_wm8995 = { .probe = wm8995_probe, .remove = wm8995_remove, - .suspend = wm8995_suspend, - .resume = wm8995_resume, .set_bias_level = wm8995_set_bias_level, .idle_bias_off = true, }; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index f16ff4f56923..b1dcc11c1b23 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2216,11 +2216,7 @@ static void wm8996_init_gpio(struct wm8996_priv *wm8996) static void wm8996_free_gpio(struct wm8996_priv *wm8996) { - int ret; - - ret = gpiochip_remove(&wm8996->gpio_chip); - if (ret != 0) - dev_err(wm8996->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm8996->gpio_chip); } #else static void wm8996_init_gpio(struct wm8996_priv *wm8996) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index d69510c53239..8e948c63f3d9 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -63,7 +63,8 @@ config SND_DM365_AIC3X_CODEC Say Y if you want to add support for AIC3101 audio codec config SND_DM365_VOICE_CODEC - bool "Voice Codec - CQ93VC" + tristate "Voice Codec - CQ93VC" + depends on SND_DAVINCI_SOC select MFD_DAVINCI_VOICECODEC select SND_DAVINCI_SOC_VCIF select SND_SOC_CQ0093VC diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c28508da34cf..0eed9b1b24e1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -42,14 +42,26 @@ #define MCASP_MAX_AFIFO_DEPTH 64 +static u32 context_regs[] = { + DAVINCI_MCASP_TXFMCTL_REG, + DAVINCI_MCASP_RXFMCTL_REG, + DAVINCI_MCASP_TXFMT_REG, + DAVINCI_MCASP_RXFMT_REG, + DAVINCI_MCASP_ACLKXCTL_REG, + DAVINCI_MCASP_ACLKRCTL_REG, + DAVINCI_MCASP_AHCLKXCTL_REG, + DAVINCI_MCASP_AHCLKRCTL_REG, + DAVINCI_MCASP_PDIR_REG, + DAVINCI_MCASP_RXMASK_REG, + DAVINCI_MCASP_TXMASK_REG, + DAVINCI_MCASP_RXTDM_REG, + DAVINCI_MCASP_TXTDM_REG, +}; + struct davinci_mcasp_context { - u32 txfmtctl; - u32 rxfmtctl; - u32 txfmt; - u32 rxfmt; - u32 aclkxctl; - u32 aclkrctl; - u32 pdir; + u32 config_regs[ARRAY_SIZE(context_regs)]; + u32 afifo_regs[2]; /* for read/write fifo control registers */ + u32 *xrsr_regs; /* for serializer configuration */ }; struct davinci_mcasp { @@ -403,7 +415,8 @@ out: return ret; } -static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, + int div, bool explicit) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); @@ -420,7 +433,8 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div ACLKXDIV(div - 1), ACLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRDIV(div - 1), ACLKRDIV_MASK); - mcasp->bclk_div = div; + if (explicit) + mcasp->bclk_div = div; break; case 2: /* BCLK/LRCLK ratio */ @@ -434,6 +448,12 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div return 0; } +static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, + int div) +{ + return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1); +} + static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { @@ -459,8 +479,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7; - u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * For captured data we should not rotate, inversion and masking is + * enoguh to get the data to the right position: + * Format data from bus after reverse (XRBUF) + * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| + * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| + */ + u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() @@ -738,7 +767,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, "Inaccurate BCLK: %u Hz / %u != %u Hz\n", mcasp->sysclk_freq, div, bclk_freq); } - davinci_mcasp_set_clkdiv(cpu_dai, 1, div); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } ret = mcasp_common_hw_param(mcasp, substream->stream, @@ -857,14 +886,24 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; + int i; - context->txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG); - context->rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG); - context->txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG); - context->rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG); - context->aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG); - context->aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG); - context->pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG); + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); + + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + context->afifo_regs[0] = mcasp_get_reg(mcasp, reg); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + context->afifo_regs[1] = mcasp_get_reg(mcasp, reg); + } + + for (i = 0; i < mcasp->num_serializer; i++) + context->xrsr_regs[i] = mcasp_get_reg(mcasp, + DAVINCI_MCASP_XRSRCTL_REG(i)); return 0; } @@ -873,14 +912,24 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; + int i; + + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); + + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[0]); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[1]); + } - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, context->txfmtctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, context->rxfmtctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, context->txfmt); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, context->rxfmt); - mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, context->aclkxctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, context->aclkrctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, context->pdir); + for (i = 0; i < mcasp->num_serializer; i++) + mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + context->xrsr_regs[i]); return 0; } @@ -1199,6 +1248,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->op_mode = pdata->op_mode; mcasp->tdm_slots = pdata->tdm_slots; mcasp->num_serializer = pdata->num_serializer; +#ifdef CONFIG_PM_SLEEP + mcasp->context.xrsr_regs = devm_kzalloc(&pdev->dev, + sizeof(u32) * mcasp->num_serializer, + GFP_KERNEL); +#endif mcasp->serial_dir = pdata->serial_dir; mcasp->version = pdata->version; mcasp->txnumevt = pdata->txnumevt; diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/davinci/edma-pcm.c index 605e643133db..59e588abe54b 100644 --- a/sound/soc/davinci/edma-pcm.c +++ b/sound/soc/davinci/edma-pcm.c @@ -25,6 +25,8 @@ #include <sound/dmaengine_pcm.h> #include <linux/edma.h> +#include "edma-pcm.h" + static const struct snd_pcm_hardware edma_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 25c31f1655f6..e961388e6e9c 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -4,7 +4,7 @@ * sound/soc/dwc/designware_i2s.c * * Copyright (C) 2010 ST Microelectronics - * Rajeev Kumar <rajeev-dlh.kumar@st.com> + * Rajeev Kumar <rajeevkumar.linux@gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -455,7 +455,7 @@ static struct platform_driver dw_i2s_driver = { module_platform_driver(dw_i2s_driver); -MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>"); MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:designware_i2s"); diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index f54a8fc99291..081e406b3713 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -49,7 +49,6 @@ config SND_SOC_FSL_ESAI tristate "Enhanced Serial Audio Interface (ESAI) module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_FSL_UTILS help Say Y if you want to add Enhanced Synchronous Audio Interface (ESAI) support for the Freescale CPUs. @@ -241,6 +240,18 @@ config SND_SOC_IMX_WM8962 Say Y if you want to add support for SoC audio on an i.MX board with a wm8962 codec. +config SND_SOC_IMX_ES8328 + tristate "SoC Audio support for i.MX boards with the ES8328 codec" + depends on OF && (I2C || SPI) + select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ES8328_SPI if SPI_MASTER + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + help + Say Y if you want to add support for the ES8328 audio codec connected + via SSI/I2S over either SPI or I2C. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C @@ -269,6 +280,20 @@ config SND_SOC_IMX_MC13783 select SND_SOC_MC13783 select SND_SOC_IMX_PCM_DMA +config SND_SOC_FSL_ASOC_CARD + tristate "Generic ASoC Sound Card with ASRC support" + depends on OF && I2C + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_ESAI + select SND_SOC_FSL_SAI + select SND_SOC_FSL_SSI + help + ALSA SoC Audio support with ASRC feature for Freescale SoCs that have + ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 + and SGTL5000. + Say Y if you want to add support for Freescale Generic ASoC Sound Card. + endif # SND_IMX_SOC endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 9ff59267eac9..d28dc25c9375 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-y := fsl_ssi.o @@ -19,6 +20,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o @@ -50,6 +52,7 @@ snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o +snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o @@ -59,6 +62,7 @@ obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o +obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c new file mode 100644 index 000000000000..007c772f3cef --- /dev/null +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -0,0 +1,574 @@ +/* + * Freescale Generic ASoC Sound Card driver with ASRC + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <nicoleotsuka@gmail.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "fsl_esai.h" +#include "fsl_sai.h" +#include "imx-audmux.h" + +#include "../codecs/sgtl5000.h" +#include "../codecs/wm8962.h" + +#define RX 0 +#define TX 1 + +/* Default DAI format without Master and Slave flag */ +#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF) + +/** + * CODEC private data + * + * @mclk_freq: Clock rate of MCLK + * @mclk_id: MCLK (or main clock) id for set_sysclk() + * @fll_id: FLL (or secordary clock) id for set_sysclk() + * @pll_id: PLL id for set_pll() + */ +struct codec_priv { + unsigned long mclk_freq; + u32 mclk_id; + u32 fll_id; + u32 pll_id; +}; + +/** + * CPU private data + * + * @sysclk_freq[2]: SYSCLK rates for set_sysclk() + * @sysclk_dir[2]: SYSCLK directions for set_sysclk() + * @sysclk_id[2]: SYSCLK ids for set_sysclk() + * + * Note: [1] for tx and [0] for rx + */ +struct cpu_priv { + unsigned long sysclk_freq[2]; + u32 sysclk_dir[2]; + u32 sysclk_id[2]; +}; + +/** + * Freescale Generic ASOC card private data + * + * @dai_link[3]: DAI link structure including normal one and DPCM link + * @pdev: platform device pointer + * @codec_priv: CODEC private data + * @cpu_priv: CPU private data + * @card: ASoC card structure + * @sample_rate: Current sample rate + * @sample_format: Current sample format + * @asrc_rate: ASRC sample rate used by Back-Ends + * @asrc_format: ASRC sample format used by Back-Ends + * @dai_fmt: DAI format between CPU and CODEC + * @name: Card name + */ + +struct fsl_asoc_card_priv { + struct snd_soc_dai_link dai_link[3]; + struct platform_device *pdev; + struct codec_priv codec_priv; + struct cpu_priv cpu_priv; + struct snd_soc_card card; + u32 sample_rate; + u32 sample_format; + u32 asrc_rate; + u32 asrc_format; + u32 dai_fmt; + char name[32]; +}; + +/** + * This dapm route map exsits for DPCM link only. + * The other routes shall go through Device Tree. + */ +static const struct snd_soc_dapm_route audio_map[] = { + {"CPU-Playback", NULL, "ASRC-Playback"}, + {"Playback", NULL, "CPU-Playback"}, + {"ASRC-Capture", NULL, "CPU-Capture"}, + {"CPU-Capture", NULL, "Capture"}, +}; + +/* Add all possible widgets into here without being redundant */ +static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Line Out Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct cpu_priv *cpu_priv = &priv->cpu_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->sample_rate = params_rate(params); + priv->sample_format = params_format(params); + + if (priv->card.set_bias_level) + return 0; + + /* Specific configurations of DAIs starts from here */ + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx], + cpu_priv->sysclk_freq[tx], + cpu_priv->sysclk_dir[tx]); + if (ret) { + dev_err(dev, "failed to set sysclk for cpu dai\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops fsl_asoc_card_ops = { + .hw_params = fsl_asoc_card_hw_params, +}; + +static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct snd_interval *rate; + struct snd_mask *mask; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + rate->max = rate->min = priv->asrc_rate; + + mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_none(mask); + snd_mask_set(mask, priv->asrc_format); + + return 0; +} + +static struct snd_soc_dai_link fsl_asoc_card_dai[] = { + /* Default ASoC DAI Link*/ + { + .name = "HiFi", + .stream_name = "HiFi", + .ops = &fsl_asoc_card_ops, + }, + /* DPCM Link between Front-End and Back-End (Optional) */ + { + .name = "HiFi-ASRC-FE", + .stream_name = "HiFi-ASRC-FE", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .dpcm_capture = 1, + .dynamic = 1, + }, + { + .name = "HiFi-ASRC-BE", + .stream_name = "HiFi-ASRC-BE", + .platform_name = "snd-soc-dummy", + .be_hw_params_fixup = be_hw_params_fixup, + .ops = &fsl_asoc_card_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = card->dev; + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level != SND_SOC_BIAS_STANDBY) + break; + + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, + codec_priv->mclk_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level != SND_SOC_BIAS_PREPARE) + break; + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, + codec_priv->mclk_freq, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); + if (ret) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + return 0; +} + +static int fsl_asoc_card_audmux_init(struct device_node *np, + struct fsl_asoc_card_priv *priv) +{ + struct device *dev = &priv->pdev->dev; + u32 int_ptcr = 0, ext_ptcr = 0; + int int_port, ext_port; + int ret; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the AUDMUX API expects it starts at 0. + */ + int_port--; + ext_port--; + + /* + * Use asynchronous mode (6 wires) for all cases. + * If only 4 wires are needed, just set SSI into + * synchronous mode and enable 4 PADs in IOMUX. + */ + switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR; + break; + case SND_SOC_DAIFMT_CBS_CFM: + int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + break; + default: + return -EINVAL; + } + + /* 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; + } + + ret = imx_audmux_v2_configure_port(int_port, int_ptcr, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + 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; + } + + ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } + + return 0; +} + +static int fsl_asoc_card_late_probe(struct snd_soc_card *card) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = card->dev; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, + codec_priv->mclk_freq, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set sysclk in %s\n", __func__); + return ret; + } + + return 0; +} + +static int fsl_asoc_card_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np, *asrc_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *asrc_pdev = NULL; + struct platform_device *cpu_pdev; + struct fsl_asoc_card_priv *priv; + struct i2c_client *codec_dev; + struct clk *codec_clk; + u32 width; + int ret; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + cpu_np = of_parse_phandle(np, "audio-cpu", 0); + /* 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"); + ret = -EINVAL; + goto fail; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find CPU DAI device\n"); + ret = -EINVAL; + 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; + } + + 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); + } + + /* Default sample rate and format, will be updated in hw_params() */ + priv->sample_rate = 44100; + priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; + + /* Assign a default DAI format, and allow each card to overwrite it */ + priv->dai_fmt = DAI_FMT_BASE; + + /* Diversify the card configurations */ + if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { + priv->card.set_bias_level = NULL; + priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; + priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { + priv->codec_priv.mclk_id = SGTL5000_SYSCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { + priv->card.set_bias_level = fsl_asoc_card_set_bias_level; + priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; + priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; + priv->codec_priv.pll_id = WM8962_FLL; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else { + dev_err(&pdev->dev, "unknown Device Tree compatible\n"); + return -EINVAL; + } + + /* Common settings for corresponding Freescale CPU DAI driver */ + if (strstr(cpu_np->name, "ssi")) { + /* Only SSI needs to configure AUDMUX */ + ret = fsl_asoc_card_audmux_init(np, priv); + if (ret) { + dev_err(&pdev->dev, "failed to init audmux\n"); + goto asrc_fail; + } + } else if (strstr(cpu_np->name, "esai")) { + priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; + priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; + } else if (strstr(cpu_np->name, "sai")) { + priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; + priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; + } + + sprintf(priv->name, "%s-audio", codec_dev->name); + + /* Initialize sound card */ + priv->pdev = pdev; + priv->card.dev = &pdev->dev; + priv->card.name = priv->name; + priv->card.dai_link = priv->dai_link; + priv->card.dapm_routes = audio_map; + priv->card.late_probe = fsl_asoc_card_late_probe; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); + priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; + priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); + + memcpy(priv->dai_link, fsl_asoc_card_dai, + sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + + /* 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_dev->name; + priv->dai_link[0].platform_of_node = cpu_np; + priv->dai_link[0].dai_fmt = priv->dai_fmt; + priv->card.num_links = 1; + + if (asrc_pdev) { + /* DPCM DAI Links only if ASRC exsits */ + priv->dai_link[1].cpu_of_node = asrc_np; + priv->dai_link[1].platform_of_node = asrc_np; + priv->dai_link[2].codec_dai_name = codec_dev->name; + priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].cpu_of_node = cpu_np; + priv->dai_link[2].dai_fmt = priv->dai_fmt; + priv->card.num_links = 3; + + ret = of_property_read_u32(asrc_np, "fsl,asrc-rate", + &priv->asrc_rate); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + ret = -EINVAL; + goto asrc_fail; + } + + ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + ret = -EINVAL; + goto asrc_fail; + } + + if (width == 24) + priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE; + else + priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE; + } + + /* Finish card registering */ + platform_set_drvdata(pdev, priv); + snd_soc_card_set_drvdata(&priv->card, priv); + + ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + +asrc_fail: + of_node_put(asrc_np); +fail: + of_node_put(codec_np); + of_node_put(cpu_np); + + return ret; +} + +static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-cs42888", }, + { .compatible = "fsl,imx-audio-sgtl5000", }, + { .compatible = "fsl,imx-audio-wm8962", }, + {} +}; + +static struct platform_driver fsl_asoc_card_driver = { + .probe = fsl_asoc_card_probe, + .driver = { + .name = "fsl-asoc-card", + .pm = &snd_soc_pm_ops, + .of_match_table = fsl_asoc_card_dt_ids, + }, +}; +module_platform_driver(fsl_asoc_card_driver); + +MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC"); +MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>"); +MODULE_ALIAS("platform:fsl-asoc-card"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 822110420b71..3b145313f93e 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -684,7 +684,7 @@ static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg) } } -static struct regmap_config fsl_asrc_regmap_config = { +static const struct regmap_config fsl_asrc_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -802,10 +802,6 @@ static int fsl_asrc_probe(struct platform_device *pdev) asrc_priv->paddr = res->start; - /* Register regmap and let it prepare core clock */ - if (of_property_read_bool(np, "big-endian")) - fsl_asrc_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; - asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs, &fsl_asrc_regmap_config); if (IS_ERR(asrc_priv->regmap)) { diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 72d154e7dd03..8bcdfda09d7a 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -18,7 +18,6 @@ #include "fsl_esai.h" #include "imx-pcm.h" -#include "fsl_utils.h" #define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000 #define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ @@ -38,6 +37,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot + * @slots: number of slots * @hck_rate: clock rate of desired HCKx clock * @sck_rate: clock rate of desired SCKx clock * @hck_dir: the direction of HCKx pads @@ -56,6 +56,7 @@ struct fsl_esai { struct clk *fsysclk; u32 fifo_depth; u32 slot_width; + u32 slots; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; @@ -363,6 +364,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); esai_priv->slot_width = slot_width; + esai_priv->slots = slots; return 0; } @@ -510,10 +512,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 width = snd_pcm_format_width(params_format(params)); u32 channels = params_channels(params); + u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); u32 bclk, mask, val; int ret; - bclk = params_rate(params) * esai_priv->slot_width * 2; + bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots; ret = fsl_esai_set_bclk(dai, tx, bclk); if (ret) @@ -530,7 +533,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK | (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK); val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) | - (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels)); + (tx ? ESAI_xFCR_TE(pins) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(pins)); regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); @@ -565,6 +568,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u8 i, channels = substream->runtime->channels; + u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -579,7 +583,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, - tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels)); + tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins)); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: @@ -607,7 +611,6 @@ static struct snd_soc_dai_ops fsl_esai_dai_ops = { .hw_params = fsl_esai_hw_params, .set_sysclk = fsl_esai_set_dai_sysclk, .set_fmt = fsl_esai_set_dai_fmt, - .xlate_tdm_slot_mask = fsl_asoc_xlate_tdm_slot_mask, .set_tdm_slot = fsl_esai_set_dai_tdm_slot, }; @@ -707,7 +710,7 @@ static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg) } } -static struct regmap_config fsl_esai_regmap_config = { +static const struct regmap_config fsl_esai_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -733,9 +736,6 @@ static int fsl_esai_probe(struct platform_device *pdev) esai_priv->pdev = pdev; strcpy(esai_priv->name, np->name); - if (of_property_read_bool(np, "big-endian")) - fsl_esai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; - /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); @@ -783,6 +783,9 @@ static int fsl_esai_probe(struct platform_device *pdev) /* Set a default slot size */ esai_priv->slot_width = 32; + /* Set a default slot number */ + esai_priv->slots = 2; + /* Set a default master/slave state */ esai_priv->slave_mode = true; diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 75e14033e8d8..91a550f4a10d 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -130,8 +130,8 @@ #define ESAI_xFCR_RE_WIDTH 4 #define ESAI_xFCR_TE_MASK (((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) #define ESAI_xFCR_RE_MASK (((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) -#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK) -#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK) +#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - x)) & ESAI_xFCR_TE_MASK) +#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - x)) & ESAI_xFCR_RE_MASK) #define ESAI_xFCR_xFR_SHIFT 1 #define ESAI_xFCR_xFR_MASK (1 << ESAI_xFCR_xFR_SHIFT) #define ESAI_xFCR_xFR (1 << ESAI_xFCR_xFR_SHIFT) @@ -272,8 +272,8 @@ #define ESAI_xCR_RE_WIDTH 4 #define ESAI_xCR_TE_MASK (((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) #define ESAI_xCR_RE_MASK (((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) -#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK) -#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK) +#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - x)) & ESAI_xCR_TE_MASK) +#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - x)) & ESAI_xCR_RE_MASK) /* * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8 diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index faa049797897..7eeb1dd8ce27 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -175,7 +175,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, bool tx = fsl_dir == FSL_FMT_TRANSMITTER; u32 val_cr2 = 0, val_cr4 = 0; - if (!sai->big_endian_data) + if (!sai->is_lsb_first) val_cr4 |= FSL_SAI_CR4_MF; /* DAI mode */ @@ -304,7 +304,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); - if (sai->big_endian_data) + if (sai->is_lsb_first) val_cr5 |= FSL_SAI_CR5_FBT(0); else val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); @@ -330,13 +330,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, u32 xcsr, count = 100; /* - * The transmitter bit clock and frame sync are to be - * used by both the transmitter and receiver. + * Asynchronous mode: Clear SYNC for both Tx and Rx. + * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx. + * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx. */ - regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, - ~FSL_SAI_CR2_SYNC); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, - FSL_SAI_CR2_SYNC); + sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0); /* * It is recommended that the transmitter is the last enabled @@ -437,8 +437,13 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_TX * 2); regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, @@ -539,7 +544,7 @@ static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg) } } -static struct regmap_config fsl_sai_regmap_config = { +static const struct regmap_config fsl_sai_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -568,11 +573,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) sai->sai_on_imx = true; - sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); - if (sai->big_endian_regs) - fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; - - sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, res); @@ -621,6 +622,33 @@ static int fsl_sai_probe(struct platform_device *pdev) return ret; } + /* Sync Tx with Rx as default by following old DT binding */ + sai->synchronous[RX] = true; + sai->synchronous[TX] = false; + fsl_sai_dai.symmetric_rates = 1; + fsl_sai_dai.symmetric_channels = 1; + fsl_sai_dai.symmetric_samplebits = 1; + + if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) && + of_find_property(np, "fsl,sai-asynchronous", NULL)) { + /* error out if both synchronous and asynchronous are present */ + dev_err(&pdev->dev, "invalid binding for synchronous mode\n"); + return -EINVAL; + } + + if (of_find_property(np, "fsl,sai-synchronous-rx", NULL)) { + /* Sync Rx with Tx */ + sai->synchronous[RX] = false; + sai->synchronous[TX] = true; + } else if (of_find_property(np, "fsl,sai-asynchronous", NULL)) { + /* Discard all settings for asynchronous mode */ + sai->synchronous[RX] = false; + sai->synchronous[TX] = false; + fsl_sai_dai.symmetric_rates = 0; + fsl_sai_dai.symmetric_channels = 0; + fsl_sai_dai.symmetric_samplebits = 0; + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0e6c9f595d75..34667209b607 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -48,6 +48,7 @@ /* SAI Transmit/Recieve Control Register */ #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FR BIT(25) +#define FSL_SAI_CSR_SR BIT(24) #define FSL_SAI_CSR_xF_SHIFT 16 #define FSL_SAI_CSR_xF_W_SHIFT 18 #define FSL_SAI_CSR_xF_MASK (0x1f << FSL_SAI_CSR_xF_SHIFT) @@ -131,13 +132,16 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; - bool big_endian_regs; - bool big_endian_data; + bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; + bool synchronous[2]; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; +#define TX 1 +#define RX 0 + #endif /* __FSL_SAI_H */ diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 70acfe4a9bd5..ae4e408810ec 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1040,7 +1040,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) } } -static struct regmap_config fsl_spdif_regmap_config = { +static const struct regmap_config fsl_spdif_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -1184,9 +1184,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); spdif_priv->cpu_dai_drv.name = spdif_priv->name; - if (of_property_read_bool(np, "big-endian")) - fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; - /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb5776a39b..e6955170dc42 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -169,6 +169,7 @@ struct fsl_ssi_private { u8 i2s_mode; bool use_dma; bool use_dual_fifo; + bool has_ipg_clk_name; unsigned int fifo_depth; struct fsl_ssi_rxtx_reg_val rxtx_reg_val; @@ -259,6 +260,11 @@ static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) SND_SOC_DAIFMT_CBS_CFS; } +static bool fsl_ssi_is_i2s_cbm_cfs(struct fsl_ssi_private *ssi_private) +{ + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) == + SND_SOC_DAIFMT_CBM_CFS; +} /** * fsl_ssi_isr: SSI interrupt handler * @@ -525,6 +531,11 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + int ret; + + ret = clk_prepare_enable(ssi_private->clk); + if (ret) + return ret; /* When using dual fifo mode, it is safer to ensure an even period * size. If appearing to an odd number while DMA always starts its @@ -539,6 +550,21 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, } /** + * fsl_ssi_shutdown: shutdown the SSI + * + */ +static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + + clk_disable_unprepare(ssi_private->clk); + +} + +/** * fsl_ssi_set_bclk - configure Digital Audio Interface bit clock * * Note: This function can be only called when using SSI as DAI master @@ -705,6 +731,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } } + if (!fsl_ssi_is_ac97(ssi_private)) { + u8 i2smode; + /* + * Switch to normal net mode in order to have a frame sync + * signal every 32 bits instead of 16 bits + */ + if (fsl_ssi_is_i2s_cbm_cfs(ssi_private) && sample_size == 16) + i2smode = CCSR_SSI_SCR_I2S_MODE_NORMAL | + CCSR_SSI_SCR_NET; + else + i2smode = ssi_private->i2s_mode; + + regmap_update_bits(regs, CCSR_SSI_SCR, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : i2smode); + } + /* * FIXME: The documentation says that SxCCR[WL] should not be * modified while the SSI is enabled. The only time this can @@ -724,11 +767,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(regs, CCSR_SSI_SRCCR, CCSR_SSI_SxCCR_WL_MASK, wl); - if (!fsl_ssi_is_ac97(ssi_private)) - regmap_update_bits(regs, CCSR_SSI_SCR, - CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, - channels == 1 ? 0 : ssi_private->i2s_mode); - return 0; } @@ -748,8 +786,9 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, return 0; } -static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, - unsigned int fmt) +static int _fsl_ssi_set_dai_fmt(struct device *dev, + struct fsl_ssi_private *ssi_private, + unsigned int fmt) { struct regmap *regs = ssi_private->regs; u32 strcr = 0, stcr, srcr, scr, mask; @@ -758,7 +797,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->dai_fmt = fmt; if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) { - dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n"); + dev_err(dev, "baudclk is missing which is necessary for master mode\n"); return -EINVAL; } @@ -780,6 +819,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFS: ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER; regmap_update_bits(regs, CCSR_SSI_STCCR, @@ -853,6 +893,11 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, case SND_SOC_DAIFMT_CBM_CFM: scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; + case SND_SOC_DAIFMT_CBM_CFS: + strcr &= ~CCSR_SSI_STCR_TXDIR; + strcr |= CCSR_SSI_STCR_TFDIR; + scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; + break; default: return -EINVAL; } @@ -913,7 +958,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); - return _fsl_ssi_set_dai_fmt(ssi_private, fmt); + return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt); } /** @@ -1020,6 +1065,7 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, + .shutdown = fsl_ssi_shutdown, .hw_params = fsl_ssi_hw_params, .hw_free = fsl_ssi_hw_free, .set_fmt = fsl_ssi_set_dai_fmt, @@ -1145,17 +1191,22 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, u32 dmas[4]; int ret; - ssi_private->clk = devm_clk_get(&pdev->dev, NULL); + if (ssi_private->has_ipg_clk_name) + ssi_private->clk = devm_clk_get(&pdev->dev, "ipg"); + else + ssi_private->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi_private->clk)) { ret = PTR_ERR(ssi_private->clk); dev_err(&pdev->dev, "could not get clock: %d\n", ret); return ret; } - ret = clk_prepare_enable(ssi_private->clk); - if (ret) { - dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret); - return ret; + if (!ssi_private->has_ipg_clk_name) { + ret = clk_prepare_enable(ssi_private->clk); + if (ret) { + dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret); + return ret; + } } /* For those SLAVE implementations, we ingore non-baudclk cases @@ -1213,8 +1264,9 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, return 0; error_pcm: - clk_disable_unprepare(ssi_private->clk); + if (!ssi_private->has_ipg_clk_name) + clk_disable_unprepare(ssi_private->clk); return ret; } @@ -1223,7 +1275,8 @@ static void fsl_ssi_imx_clean(struct platform_device *pdev, { if (!ssi_private->use_dma) imx_pcm_fiq_exit(pdev); - clk_disable_unprepare(ssi_private->clk); + if (!ssi_private->has_ipg_clk_name) + clk_disable_unprepare(ssi_private->clk); } static int fsl_ssi_probe(struct platform_device *pdev) @@ -1262,9 +1315,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (sprop) { if (!strcmp(sprop, "ac97-slave")) ssi_private->dai_fmt = SND_SOC_DAIFMT_AC97; - else if (!strcmp(sprop, "i2s-slave")) - ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_CBM_CFM; } ssi_private->use_dma = !of_property_read_bool(np, @@ -1298,8 +1348,16 @@ static int fsl_ssi_probe(struct platform_device *pdev) return -ENOMEM; } - ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, + ret = of_property_match_string(np, "clock-names", "ipg"); + if (ret < 0) { + ssi_private->has_ipg_clk_name = false; + ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, &fsl_ssi_regconfig); + } else { + ssi_private->has_ipg_clk_name = true; + ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev, + "ipg", iomem, &fsl_ssi_regconfig); + } if (IS_ERR(ssi_private->regs)) { dev_err(&pdev->dev, "Failed to init register map\n"); return PTR_ERR(ssi_private->regs); @@ -1387,7 +1445,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) done: if (ssi_private->dai_fmt) - _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt); + _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, + ssi_private->dai_fmt); return 0; diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c new file mode 100644 index 000000000000..653e66d150c8 --- /dev/null +++ b/sound/soc/fsl/imx-es8328.c @@ -0,0 +1,232 @@ +/* + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/gpio.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 +#define MUX_PORT_MAX 7 + +struct imx_es8328_data { + struct device *dev; + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + int jack_gpio; +}; + +static struct snd_soc_jack_gpio headset_jack_gpios[] = { + { + .gpio = -1, + .name = "headset-gpio", + .report = SND_JACK_HEADSET, + .invert = 0, + .debounce_time = 200, + }, +}; + +static struct snd_soc_jack headset_jack; + +static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct imx_es8328_data *data = container_of(rtd->card, + struct imx_es8328_data, card); + int ret = 0; + + /* Headphone jack detection */ + if (gpio_is_valid(data->jack_gpio)) { + ret = snd_soc_jack_new(rtd->codec, "Headphone", + SND_JACK_HEADPHONE | SND_JACK_BTN_0, + &headset_jack); + if (ret) + return ret; + + headset_jack_gpios[0].gpio = data->jack_gpio; + ret = snd_soc_jack_add_gpios(&headset_jack, + ARRAY_SIZE(headset_jack_gpios), + headset_jack_gpios); + } + + return ret; +} + +static const struct snd_soc_dapm_widget imx_es8328_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_REGULATOR_SUPPLY("audio-amp", 1, 0), +}; + +static int imx_es8328_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_es8328_data *data; + u32 int_port, ext_port; + int ret; + struct device *dev = &pdev->dev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(dev, "mux-int-port missing or invalid\n"); + goto fail; + } + if (int_port > MUX_PORT_MAX || int_port == 0) { + dev_err(dev, "mux-int-port: hardware only has %d mux ports\n", + MUX_PORT_MAX); + goto fail; + } + + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(dev, "mux-ext-port missing or invalid\n"); + goto fail; + } + if (ext_port > MUX_PORT_MAX || ext_port == 0) { + dev_err(dev, "mux-ext-port: hardware only has %d mux ports\n", + MUX_PORT_MAX); + goto fail; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } + ret = imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->dev = dev; + + data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0); + + data->dai.name = "hifi"; + data->dai.stream_name = "hifi"; + data->dai.codec_dai_name = "es8328-hifi-analog"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_of_node = ssi_np; + data->dai.platform_of_node = ssi_np; + data->dai.init = &imx_es8328_dai_init; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = dev; + data->card.dapm_widgets = imx_es8328_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_es8328_dapm_widgets); + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) { + dev_err(dev, "Unable to parse card name\n"); + goto fail; + } + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) { + dev_err(dev, "Unable to parse routing: %d\n", ret); + goto fail; + } + data->card.num_links = 1; + data->card.owner = THIS_MODULE; + data->card.dai_link = &data->dai; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(dev, "Unable to register: %d\n", ret); + goto fail; + } + + platform_set_drvdata(pdev, data); +fail: + of_node_put(ssi_np); + of_node_put(codec_np); + + return ret; +} + +static int imx_es8328_remove(struct platform_device *pdev) +{ + struct imx_es8328_data *data = platform_get_drvdata(pdev); + + snd_soc_jack_free_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios), + headset_jack_gpios); + + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_es8328_dt_ids[] = { + { .compatible = "fsl,imx-audio-es8328", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_es8328_dt_ids); + +static struct platform_driver imx_es8328_driver = { + .driver = { + .name = "imx-es8328", + .of_match_table = imx_es8328_dt_ids, + }, + .probe = imx_es8328_probe, + .remove = imx_es8328_remove, +}; +module_platform_driver(imx_es8328_driver); + +MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>"); +MODULE_DESCRIPTION("Kosagi i.MX6 ES8328 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-audio-es8328"); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 159e517fa09a..cef7776b712c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -481,12 +481,19 @@ static int asoc_simple_card_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&priv->snd_card, priv); ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); + if (ret >= 0) + return ret; err: asoc_simple_card_unref(pdev); return ret; } +static int asoc_simple_card_remove(struct platform_device *pdev) +{ + return asoc_simple_card_unref(pdev); +} + static const struct of_device_id asoc_simple_of_match[] = { { .compatible = "simple-audio-card", }, {}, @@ -500,6 +507,7 @@ static struct platform_driver asoc_simple_card = { .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, + .remove = asoc_simple_card_remove, }; module_platform_driver(asoc_simple_card); diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 42edc6f4fc4a..03d0a166b635 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -246,8 +246,8 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { }; static struct sst_acpi_mach baytrail_machines[] = { - { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" }, - { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" }, + { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, + { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, {} }; diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index 67673a2c0f41..b4ad98c43e5c 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -817,7 +817,7 @@ static struct sst_dsp_device byt_dev = { .ops = &sst_byt_ops, }; -int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata) +int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata) { struct sst_byt *byt = pdata->dsp; @@ -826,14 +826,6 @@ int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata) sst_byt_drop_all(byt); dev_dbg(byt->dev, "dsp in reset\n"); - return 0; -} -EXPORT_SYMBOL_GPL(sst_byt_dsp_suspend_noirq); - -int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata) -{ - struct sst_byt *byt = pdata->dsp; - dev_dbg(byt->dev, "free all blocks and unload fw\n"); sst_fw_unload(byt->fw); diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h index 06a4d202689b..8faff6dcf25d 100644 --- a/sound/soc/intel/sst-baytrail-ipc.h +++ b/sound/soc/intel/sst-baytrail-ipc.h @@ -66,7 +66,6 @@ int sst_byt_get_dsp_position(struct sst_byt *byt, int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata); void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata); struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt); -int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata); int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata); int sst_byt_dsp_boot(struct device *dev, struct sst_pdata *pdata); int sst_byt_dsp_wait_for_ready(struct device *dev, struct sst_pdata *pdata); diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 599401c0c655..eab1c7d85187 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -59,6 +59,9 @@ struct sst_byt_priv_data { /* DAI data */ struct sst_byt_pcm_data pcm[BYT_PCM_COUNT]; + + /* flag indicating is stream context restore needed after suspend */ + bool restore_stream; }; /* this may get called several times by oss emulation */ @@ -184,7 +187,10 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) sst_byt_stream_start(byt, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_RESUME: - schedule_work(&pcm_data->work); + if (pdata->restore_stream == true) + schedule_work(&pcm_data->work); + else + sst_byt_stream_resume(byt, pcm_data->stream); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: sst_byt_stream_resume(byt, pcm_data->stream); @@ -193,6 +199,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) sst_byt_stream_stop(byt, pcm_data->stream); break; case SNDRV_PCM_TRIGGER_SUSPEND: + pdata->restore_stream = false; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sst_byt_stream_pause(byt, pcm_data->stream); break; @@ -404,26 +411,10 @@ static const struct snd_soc_component_driver byt_dai_component = { }; #ifdef CONFIG_PM -static int sst_byt_pcm_dev_suspend_noirq(struct device *dev) -{ - struct sst_pdata *sst_pdata = dev_get_platdata(dev); - int ret; - - dev_dbg(dev, "suspending noirq\n"); - - /* at this point all streams will be stopped and context saved */ - ret = sst_byt_dsp_suspend_noirq(dev, sst_pdata); - if (ret < 0) { - dev_err(dev, "failed to suspend %d\n", ret); - return ret; - } - - return ret; -} - static int sst_byt_pcm_dev_suspend_late(struct device *dev) { struct sst_pdata *sst_pdata = dev_get_platdata(dev); + struct sst_byt_priv_data *priv_data = dev_get_drvdata(dev); int ret; dev_dbg(dev, "suspending late\n"); @@ -434,34 +425,30 @@ static int sst_byt_pcm_dev_suspend_late(struct device *dev) return ret; } + priv_data->restore_stream = true; + return ret; } static int sst_byt_pcm_dev_resume_early(struct device *dev) { struct sst_pdata *sst_pdata = dev_get_platdata(dev); + int ret; dev_dbg(dev, "resume early\n"); /* load fw and boot DSP */ - return sst_byt_dsp_boot(dev, sst_pdata); -} - -static int sst_byt_pcm_dev_resume(struct device *dev) -{ - struct sst_pdata *sst_pdata = dev_get_platdata(dev); - - dev_dbg(dev, "resume\n"); + ret = sst_byt_dsp_boot(dev, sst_pdata); + if (ret) + return ret; /* wait for FW to finish booting */ return sst_byt_dsp_wait_for_ready(dev, sst_pdata); } static const struct dev_pm_ops sst_byt_pm_ops = { - .suspend_noirq = sst_byt_pcm_dev_suspend_noirq, .suspend_late = sst_byt_pcm_dev_suspend_late, .resume_early = sst_byt_pcm_dev_resume_early, - .resume = sst_byt_pcm_dev_resume, }; #define SST_BYT_PM_OPS (&sst_byt_pm_ops) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index f8a6adc2d81c..4336d1831485 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -260,7 +260,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { .stream_name = "TWL4030 Voice", .cpu_dai_name = "omap-mcbsp.3", .codec_dai_name = "twl4030-voice", - .platform_name = "omap-mcbsp.2", + .platform_name = "omap-mcbsp.3", .codec_name = "twl4030-codec", .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM, diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 943922c79f78..b10ae8074461 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -168,7 +168,7 @@ static int rx51_spk_event(struct snd_soc_dapm_widget *w, static int rx51_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - struct snd_soc_codec *codec = w->dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); if (SND_SOC_DAPM_EVENT_ON(event)) tpa6130a2_stereo_enable(codec, 1); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 0109f6c2334e..a8e097433074 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -765,9 +765,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai) SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .startup = pxa_ssp_startup, diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 8d8e4b59049f..fb9e05c9f471 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -165,13 +165,14 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; - mask = I2S_CKR_MSS_SLAVE; + mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = I2S_CKR_MSS_SLAVE; + /* Set source clock in Master mode */ + val = I2S_CKR_MSS_MASTER; break; case SND_SOC_DAIFMT_CBM_CFM: - val = I2S_CKR_MSS_MASTER; + val = I2S_CKR_MSS_SLAVE; break; default: return -EINVAL; @@ -361,6 +362,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) case I2S_XFER: case I2S_CLR: case I2S_RXDR: + case I2S_FIFOLR: + case I2S_INTSR: return true; default: return false; @@ -370,8 +373,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { - case I2S_FIFOLR: case I2S_INTSR: + case I2S_CLR: return true; default: return false; @@ -381,8 +384,6 @@ static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg) { switch (reg) { - case I2S_FIFOLR: - return true; default: return false; } diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 03eec22f0f46..9d513473b300 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -462,7 +462,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (dir == SND_SOC_CLOCK_IN) rfs = 0; - if ((rfs && other->rfs && (other->rfs != rfs)) || + if ((rfs && other && other->rfs && (other->rfs != rfs)) || (any_active(i2s) && (((dir == SND_SOC_CLOCK_IN) && !(mod & MOD_CDCLKCON)) || @@ -762,7 +762,8 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, } else { u32 mod = readl(i2s->addr + I2SMOD); i2s->cdclk_out = !(mod & MOD_CDCLKCON); - other->cdclk_out = i2s->cdclk_out; + if (other) + other->cdclk_out = i2s->cdclk_out; } /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 3fdf3be7b99a..f95e7ab135e8 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -247,7 +247,7 @@ rsnd_gen2_dma_addr(struct rsnd_priv *priv, }; /* it shouldn't happen */ - if (use_dvc & !use_src) + if (use_dvc && !use_src) dev_err(dev, "DVC is selected without SRC\n"); /* use SSIU or SSI ? */ diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 27c06acce205..cecfab3cc948 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -101,10 +101,12 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].runtime = fe_substream->runtime; - if (dpcm_path_get(fe, stream, &list) <= 0) { + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) + goto fe_err; + else if (ret == 0) dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); - } /* calculate valid and active FE <-> BE dpcms */ dpcm_process_paths(fe, stream, &list, 1); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1b422c5c36c8..3d8cff629a18 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -637,7 +637,7 @@ int snd_soc_suspend(struct device *dev) list_for_each_entry(codec, &card->codec_dev_list, card_list) { /* If there are paths active then the CODEC will be held with * bias _ON and should not be suspended. */ - if (!codec->suspended && codec->driver->suspend) { + if (!codec->suspended) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: /* @@ -651,8 +651,10 @@ int snd_soc_suspend(struct device *dev) "ASoC: idle_bias_off CODEC on over suspend\n"); break; } + case SND_SOC_BIAS_OFF: - codec->driver->suspend(codec); + if (codec->driver->suspend) + codec->driver->suspend(codec); codec->suspended = 1; codec->cache_sync = 1; if (codec->component.regmap) @@ -726,11 +728,12 @@ static void soc_resume_deferred(struct work_struct *work) * left with bias OFF or STANDBY and suspended so we must now * resume. Otherwise the suspend was suppressed. */ - if (codec->driver->resume && codec->suspended) { + if (codec->suspended) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: case SND_SOC_BIAS_OFF: - codec->driver->resume(codec); + if (codec->driver->resume) + codec->driver->resume(codec); codec->suspended = 0; break; default: @@ -1087,7 +1090,6 @@ static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - struct snd_soc_component *dai_component, *component2; struct snd_soc_dai *dai; int ret; @@ -1114,44 +1116,12 @@ static int soc_probe_component(struct snd_soc_card *card, } } - /* - * This is rather ugly, but certain platforms expect that the DAPM - * widgets for the DAIs for components with the same parent device are - * created in the platforms DAPM context. Until that is fixed we need to - * keep this. - */ - if (component->steal_sibling_dai_widgets) { - dai_component = NULL; - list_for_each_entry(component2, &component_list, list) { - if (component == component2) - continue; - - if (component2->dev == component->dev && - !list_empty(&component2->dai_list)) { - dai_component = component2; - break; - } - } - } else { - dai_component = component; - list_for_each_entry(component2, &component_list, list) { - if (component2->dev == component->dev && - component2->steal_sibling_dai_widgets) { - dai_component = NULL; - break; - } - } - } - - if (dai_component) { - list_for_each_entry(dai, &dai_component->dai_list, list) { - snd_soc_dapm_new_dai_widgets(dapm, dai); - if (ret != 0) { - dev_err(component->dev, - "Failed to create DAI widgets %d\n", - ret); - goto err_probe; - } + list_for_each_entry(dai, &component->dai_list, list) { + ret = snd_soc_dapm_new_dai_widgets(dapm, dai); + if (ret != 0) { + dev_err(component->dev, + "Failed to create DAI widgets %d\n", ret); + goto err_probe; } } @@ -1209,7 +1179,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, device_initialize(rtd->dev); rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; - rtd->dev->init_name = name; + dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); @@ -1989,19 +1959,14 @@ static struct platform_driver soc_driver = { int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num) { - mutex_lock(&codec->mutex); - codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); - if (codec->ac97 == NULL) { - mutex_unlock(&codec->mutex); + if (codec->ac97 == NULL) return -ENOMEM; - } codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL); if (codec->ac97->bus == NULL) { kfree(codec->ac97); codec->ac97 = NULL; - mutex_unlock(&codec->mutex); return -ENOMEM; } @@ -2014,7 +1979,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, */ codec->ac97_created = 1; - mutex_unlock(&codec->mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); @@ -2184,7 +2148,6 @@ EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops_of_reset); */ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) { - mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS soc_unregister_ac97_codec(codec); #endif @@ -2192,7 +2155,6 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) kfree(codec->ac97); codec->ac97 = NULL; codec->ac97_created = 0; - mutex_unlock(&codec->mutex); } EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); @@ -2909,9 +2871,10 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int val, val_mask; int ret; - val = ((ucontrol->value.integer.value[0] + min) & mask); if (invert) - val = max - val; + val = (max - ucontrol->value.integer.value[0]) & mask; + else + val = ((ucontrol->value.integer.value[0] + min) & mask); val_mask = mask << shift; val = val << shift; @@ -2920,9 +2883,10 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return ret; if (snd_soc_volsw_is_stereo(mc)) { - val = ((ucontrol->value.integer.value[1] + min) & mask); if (invert) - val = max - val; + val = (max - ucontrol->value.integer.value[1]) & mask; + else + val = ((ucontrol->value.integer.value[1] + min) & mask); val_mask = mask << shift; val = val << shift; @@ -2967,8 +2931,9 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, if (invert) ucontrol->value.integer.value[0] = max - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[0] - min; + else + ucontrol->value.integer.value[0] = + ucontrol->value.integer.value[0] - min; if (snd_soc_volsw_is_stereo(mc)) { ret = snd_soc_component_read(component, rreg, &val); @@ -2979,8 +2944,9 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, if (invert) ucontrol->value.integer.value[1] = max - ucontrol->value.integer.value[1]; - ucontrol->value.integer.value[1] = - ucontrol->value.integer.value[1] - min; + else + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; } return 0; @@ -3085,7 +3051,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, unsigned int val, mask; void *data; - if (!component->regmap) + if (!component->regmap || !params->num_regs) return -EINVAL; len = params->num_regs * component->val_bytes; @@ -3810,8 +3776,11 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); */ int snd_soc_unregister_card(struct snd_soc_card *card) { - if (card->instantiated) + if (card->instantiated) { + card->instantiated = false; + snd_soc_dapm_shutdown(card); soc_cleanup_card_resources(card); + } dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); return 0; @@ -4164,19 +4133,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->dev = dev; platform->driver = platform_drv; - if (platform_drv->controls) { - platform->component.controls = platform_drv->controls; - platform->component.num_controls = platform_drv->num_controls; - } - if (platform_drv->dapm_widgets) { - platform->component.dapm_widgets = platform_drv->dapm_widgets; - platform->component.num_dapm_widgets = platform_drv->num_dapm_widgets; - platform->component.steal_sibling_dai_widgets = true; - } - if (platform_drv->dapm_routes) { - platform->component.dapm_routes = platform_drv->dapm_routes; - platform->component.num_dapm_routes = platform_drv->num_dapm_routes; - } if (platform_drv->probe) platform->component.probe = snd_soc_platform_drv_probe; @@ -4397,8 +4353,8 @@ int snd_soc_register_codec(struct device *dev, if (codec_drv->read) codec->component.read = snd_soc_codec_drv_read; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; - codec->dapm.codec = codec; codec->dapm.idle_bias_off = codec_drv->idle_bias_off; + codec->dapm.suspend_bias_off = codec_drv->suspend_bias_off; if (codec_drv->seq_notifier) codec->dapm.seq_notifier = codec_drv->seq_notifier; if (codec_drv->set_bias_level) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8348352dc2c6..2c456a376ade 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -326,12 +326,13 @@ static struct list_head *dapm_kcontrol_get_path_list( list_for_each_entry(path, dapm_kcontrol_get_path_list(kcontrol), \ list_kcontrol) -static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) +unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); return data->value; } +EXPORT_SYMBOL_GPL(dapm_kcontrol_get_value); static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, unsigned int value) @@ -1683,6 +1684,22 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, } } +static bool dapm_idle_bias_off(struct snd_soc_dapm_context *dapm) +{ + if (dapm->idle_bias_off) + return true; + + switch (snd_power_get_state(dapm->card->snd_card)) { + case SNDRV_CTL_POWER_D3hot: + case SNDRV_CTL_POWER_D3cold: + return dapm->suspend_bias_off; + default: + break; + } + + return false; +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- @@ -1706,7 +1723,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { - if (d->idle_bias_off) + if (dapm_idle_bias_off(d)) d->target_bias_level = SND_SOC_BIAS_OFF; else d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1772,7 +1789,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) if (d->target_bias_level > bias) bias = d->target_bias_level; list_for_each_entry(d, &card->dapm_list, list) - if (!d->idle_bias_off) + if (!dapm_idle_bias_off(d)) d->target_bias_level = bias; trace_snd_soc_dapm_walk_done(card); @@ -2860,12 +2877,14 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - int ret = 0; - if (e->reg != SND_SOC_NOPM) - ret = soc_dapm_read(dapm, e->reg, ®_val); - else + if (e->reg != SND_SOC_NOPM) { + int ret = soc_dapm_read(dapm, e->reg, ®_val); + if (ret) + return ret; + } else { reg_val = dapm_kcontrol_get_value(kcontrol); + } val = (reg_val >> e->shift_l) & e->mask; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); @@ -2875,7 +2894,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, ucontrol->value.enumerated.item[1] = val; } - return ret; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); @@ -3107,7 +3126,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } w->dapm = dapm; - w->codec = dapm->codec; + if (dapm->component) + w->codec = dapm->component->codec; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 731fdb5b5f9b..642c86240752 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2352,7 +2352,11 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); fe->dpcm[stream].runtime = fe_substream->runtime; - if (dpcm_path_get(fe, stream, &list) <= 0) { + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) { + mutex_unlock(&fe->card->mutex); + return ret; + } else if (ret == 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 0e5a8f35d0ad..a7dc3c56f44d 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -4,7 +4,7 @@ * sound/soc/spear/spear_pcm.c * * Copyright (C) 2012 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar@st.com> + * Rajeev Kumar<rajeevkumar.linux@gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -50,6 +50,6 @@ int devm_spear_pcm_platform_register(struct device *dev, } EXPORT_SYMBOL_GPL(devm_spear_pcm_platform_register); -MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>"); MODULE_DESCRIPTION("SPEAr PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 9577121ce971..ca8037634100 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -21,7 +21,7 @@ */ #ifndef __TEGRA_ASOC_UTILS_H__ -#define __TEGRA_ASOC_UTILS_H_ +#define __TEGRA_ASOC_UTILS_H__ struct clk; struct device; diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index f65fc0987cfb..b7a7c805d63f 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -100,15 +100,19 @@ static int control_put(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *cdev = caiaqdev(chip->card); int pos = kcontrol->private_value; int v = ucontrol->value.integer.value[0]; - unsigned char cmd = EP1_CMD_WRITE_IO; + unsigned char cmd; - if (cdev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1)) - cmd = EP1_CMD_DIMM_LEDS; - - if (cdev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER)) + switch (cdev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): cmd = EP1_CMD_DIMM_LEDS; + break; + default: + cmd = EP1_CMD_WRITE_IO; + break; + } if (pos & CNT_INTVAL) { int i = pos & ~CNT_INTVAL; |