diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2014-04-10 09:19:44 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2014-04-10 09:19:44 -0700 |
commit | e7990d45bb88c2f0565b5ee4c32eefe81653faff (patch) | |
tree | 852ab6988b7226083fefa0e0e851dbff0e7ec7f2 | |
parent | 190a3998be3ede25d6145e187d6d321f504d28fb (diff) | |
parent | a5065eb6da55b226661456e6a7435f605df98111 (diff) | |
download | linux-e7990d45bb88c2f0565b5ee4c32eefe81653faff.tar.bz2 |
Merge tag 'sound-fix-3.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here is a bunch of small fixes that have been collected since the
previous pull request. In addition to various misc fixes, the
following are included:
- HD-audio quirks for Dell, HP, Chromebook, and ALC28x codecs
- HD-audio AMD HDMI regression fix
- Continued PM support/fixes for ice1712 driver
- Multiplatform fixes for ASoC samsung drivers
- Addition of device id tables to a few ASoC drivers
- Bit clock polarity config and error flag fixes in ASoC fsl_sai"
* tag 'sound-fix-3.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (32 commits)
ALSA: usb-audio: Suppress repetitive debug messages from retire_playback_urb()
ALSA: hda - Make full_reset boolean
ALSA: hda - add headset mic detect quirk for a Dell laptop
sound: dmasound: use module_platform_driver_probe()
ALSA: au1x00: use module_platform_driver()
ALSA: hda - Use runtime helper to check active state.
ALSA: ice1712: Fix boundary checks in PCM pointer ops
ASoC: davinci-mcasp: Fix bit clock polarity settings
ASoC: samsung: Fix build on multiplatform
ASoC: fsl_sai: Fix Bit Clock Polarity configurations
ALSA: hda - Do not assign streams in reverse order
ALSA: hda/realtek - Add eapd shutup to ALC283
ALSA: hda/realtek - Change model name alias for ChromeOS
ASoC: da732x: Print correct major id
ALSA: hda/realtek - Improve HP depop when system change power state on Chromebook
ASoC: cs42l52: Fix mask for REVID
sound/oss: Remove uncompilable DBG macro use
ALSA: ice1712: Save/restore routing and rate registers
ALSA: ice1712: restore AK4xxx volumes on resume
ASoC: alc56(23|32): fix undefined return value of probing code
...
41 files changed, 323 insertions, 183 deletions
diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt index b93e9a91e30e..3aa4a8f528f4 100644 --- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt +++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt @@ -20,15 +20,6 @@ Required properties: have. - interrupt-parent: The phandle for the interrupt controller that services interrupts for this device. -- fsl,mode: The operating mode for the SSI interface. - "i2s-slave" - I2S mode, SSI is clock slave - "i2s-master" - I2S mode, SSI is clock master - "lj-slave" - left-justified mode, SSI is clock slave - "lj-master" - l.j. mode, SSI is clock master - "rj-slave" - right-justified mode, SSI is clock slave - "rj-master" - r.j., SSI is clock master - "ac97-slave" - AC97 mode, SSI is clock slave - "ac97-master" - AC97 mode, SSI is clock master - fsl,playback-dma: Phandle to a node for the DMA channel to use for playback of audio. This is typically dictated by SOC design. See the notes below. @@ -47,6 +38,9 @@ Required properties: be connected together, and SRFS and STFS be connected together. This would still allow different sample sizes, but not different sample rates. + - clocks: "ipg" - Required clock for the SSI unit + "baud" - Required clock for SSI master mode. Otherwise this + clock is not used Required are also ac97 link bindings if ac97 is used. See Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary @@ -64,6 +58,15 @@ Optional properties: Documentation/devicetree/bindings/dma/dma.txt. - dma-names: Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq is not defined. +- fsl,mode: The operating mode for the SSI interface. + "i2s-slave" - I2S mode, SSI is clock slave + "i2s-master" - I2S mode, SSI is clock master + "lj-slave" - left-justified mode, SSI is clock slave + "lj-master" - l.j. mode, SSI is clock master + "rj-slave" - right-justified mode, SSI is clock slave + "rj-master" - r.j., SSI is clock master + "ac97-slave" - AC97 mode, SSI is clock slave + "ac97-master" - AC97 mode, SSI is clock master Child 'codec' node required properties: - compatible: Compatible list, contains the name of the codec diff --git a/include/sound/cs8427.h b/include/sound/cs8427.h index f862cfff5f6a..0b6a1876639b 100644 --- a/include/sound/cs8427.h +++ b/include/sound/cs8427.h @@ -188,6 +188,7 @@ struct snd_pcm_substream; +int snd_cs8427_init(struct snd_i2c_bus *bus, struct snd_i2c_device *device); int snd_cs8427_create(struct snd_i2c_bus *bus, unsigned char addr, unsigned int reset_timeout, struct snd_i2c_device **r_cs8427); int snd_cs8427_reg_write(struct snd_i2c_device *device, unsigned char reg, diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 6c2dc3863ac0..7e21621e492a 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -150,10 +150,8 @@ static void snd_cs8427_free(struct snd_i2c_device *device) kfree(device->private_data); } -int snd_cs8427_create(struct snd_i2c_bus *bus, - unsigned char addr, - unsigned int reset_timeout, - struct snd_i2c_device **r_cs8427) +int snd_cs8427_init(struct snd_i2c_bus *bus, + struct snd_i2c_device *device) { static unsigned char initvals1[] = { CS8427_REG_CONTROL1 | CS8427_REG_AUTOINC, @@ -200,22 +198,10 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, Inhibit E->F transfers. */ CS8427_UD | CS8427_EFTUI | CS8427_DETUI, }; + struct cs8427 *chip = device->private_data; int err; - struct cs8427 *chip; - struct snd_i2c_device *device; unsigned char buf[24]; - if ((err = snd_i2c_device_create(bus, "CS8427", - CS8427_ADDR | (addr & 7), - &device)) < 0) - return err; - chip = device->private_data = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) { - snd_i2c_device_free(device); - return -ENOMEM; - } - device->private_free = snd_cs8427_free; - snd_i2c_lock(bus); err = snd_cs8427_reg_read(device, CS8427_REG_ID_AND_VER); if (err != CS8427_VER8427A) { @@ -264,10 +250,44 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, snd_i2c_unlock(bus); /* turn on run bit and rock'n'roll */ + snd_cs8427_reset(device); + + return 0; + +__fail: + snd_i2c_unlock(bus); + + return err; +} +EXPORT_SYMBOL(snd_cs8427_init); + +int snd_cs8427_create(struct snd_i2c_bus *bus, + unsigned char addr, + unsigned int reset_timeout, + struct snd_i2c_device **r_cs8427) +{ + int err; + struct cs8427 *chip; + struct snd_i2c_device *device; + + err = snd_i2c_device_create(bus, "CS8427", CS8427_ADDR | (addr & 7), + &device); + if (err < 0) + return err; + chip = device->private_data = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + snd_i2c_device_free(device); + return -ENOMEM; + } + device->private_free = snd_cs8427_free; + if (reset_timeout < 1) reset_timeout = 1; chip->reset_timeout = reset_timeout; - snd_cs8427_reset(device); + + err = snd_cs8427_init(bus, device); + if (err) + goto __fail; #if 0 // it's nice for read tests { @@ -286,7 +306,6 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, return 0; __fail: - snd_i2c_unlock(bus); snd_i2c_device_free(device); return err < 0 ? err : -EIO; } diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index a7cc49e96068..d10ef7675268 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -725,15 +725,4 @@ struct platform_driver au1000_ac97c_driver = { .remove = au1000_ac97_remove, }; -static int __init au1000_ac97_load(void) -{ - return platform_driver_register(&au1000_ac97c_driver); -} - -static void __exit au1000_ac97_unload(void) -{ - platform_driver_unregister(&au1000_ac97c_driver); -} - -module_init(au1000_ac97_load); -module_exit(au1000_ac97_unload); +module_platform_driver(au1000_ac97c_driver); diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 4918b7145b73..ec1ee07df59d 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -50,8 +50,6 @@ #include <linux/pnp.h> #include <linux/spinlock.h> -#define DEB(x) -#define DEB1(x) #include "sound_config.h" #include "ad1848.h" @@ -1016,8 +1014,6 @@ static void ad1848_close(int dev) ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; - DEB(printk("ad1848_close(void)\n")); - devc->intr_active = 0; ad1848_halt(dev); diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c index 87910e992133..c2d45a5848bc 100644 --- a/sound/oss/dmasound/dmasound_paula.c +++ b/sound/oss/dmasound/dmasound_paula.c @@ -733,19 +733,7 @@ static struct platform_driver amiga_audio_driver = { }, }; -static int __init amiga_audio_init(void) -{ - return platform_driver_probe(&amiga_audio_driver, amiga_audio_probe); -} - -module_init(amiga_audio_init); - -static void __exit amiga_audio_exit(void) -{ - platform_driver_unregister(&amiga_audio_driver); -} - -module_exit(amiga_audio_exit); +module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:amiga-audio"); diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index c5c24409ceb0..4709e592e2cc 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -275,7 +275,6 @@ static int opl3_kill_note (int devno, int voice, int note, int velocity) devc->v_alloc->map[voice] = 0; map = &pv_map[devc->lv_map[voice]]; - DEB(printk("Kill note %d\n", voice)); if (map->voice_mode == 0) return 0; @@ -873,8 +872,6 @@ static void opl3_aftertouch(int dev, int voice, int pressure) map = &pv_map[devc->lv_map[voice]]; - DEB(printk("Aftertouch %d\n", voice)); - if (map->voice_mode == 0) return; diff --git a/sound/oss/pas2_mixer.c b/sound/oss/pas2_mixer.c index a0bcb85c3904..50b5bd501247 100644 --- a/sound/oss/pas2_mixer.c +++ b/sound/oss/pas2_mixer.c @@ -21,10 +21,6 @@ #include "pas2.h" -#ifndef DEB -#define DEB(what) /* (what) */ -#endif - extern int pas_translate_code; extern char pas_model; extern int *pas_osp; @@ -120,8 +116,6 @@ pas_mixer_set(int whichDev, unsigned int level) { int left, right, devmask, changed, i, mixer = 0; - DEB(printk("static int pas_mixer_set(int whichDev = %d, unsigned int level = %X)\n", whichDev, level)); - left = level & 0x7f; right = (level & 0x7f00) >> 8; @@ -207,8 +201,6 @@ pas_mixer_reset(void) { int foo; - DEB(printk("pas2_mixer.c: void pas_mixer_reset(void)\n")); - for (foo = 0; foo < SOUND_MIXER_NRDEVICES; foo++) pas_mixer_set(foo, levels[foo]); @@ -220,7 +212,6 @@ static int pas_mixer_ioctl(int dev, unsigned int cmd, void __user *arg) int level,v ; int __user *p = (int __user *)arg; - DEB(printk("pas2_mixer.c: int pas_mixer_ioctl(unsigned int cmd = %X, unsigned int arg = %X)\n", cmd, arg)); if (cmd == SOUND_MIXER_PRIVATE1) { /* Set loudness bit */ if (get_user(level, p)) return -EFAULT; diff --git a/sound/oss/pas2_pcm.c b/sound/oss/pas2_pcm.c index 6f13ab4afc6b..474803b52f7d 100644 --- a/sound/oss/pas2_pcm.c +++ b/sound/oss/pas2_pcm.c @@ -22,10 +22,6 @@ #include "pas2.h" -#ifndef DEB -#define DEB(WHAT) -#endif - #define PAS_PCM_INTRBITS (0x08) /* * Sample buffer timer interrupt enable @@ -156,8 +152,6 @@ static int pas_audio_ioctl(int dev, unsigned int cmd, void __user *arg) int val, ret; int __user *p = arg; - DEB(printk("pas2_pcm.c: static int pas_audio_ioctl(unsigned int cmd = %X, unsigned int arg = %X)\n", cmd, arg)); - switch (cmd) { case SOUND_PCM_WRITE_RATE: @@ -204,8 +198,6 @@ static int pas_audio_ioctl(int dev, unsigned int cmd, void __user *arg) static void pas_audio_reset(int dev) { - DEB(printk("pas2_pcm.c: static void pas_audio_reset(void)\n")); - pas_write(pas_read(0xF8A) & ~0x40, 0xF8A); /* Disable PCM */ } @@ -214,8 +206,6 @@ static int pas_audio_open(int dev, int mode) int err; unsigned long flags; - DEB(printk("pas2_pcm.c: static int pas_audio_open(int mode = %X)\n", mode)); - spin_lock_irqsave(&pas_lock, flags); if (pcm_busy) { @@ -239,8 +229,6 @@ static void pas_audio_close(int dev) { unsigned long flags; - DEB(printk("pas2_pcm.c: static void pas_audio_close(void)\n")); - spin_lock_irqsave(&pas_lock, flags); pas_audio_reset(dev); @@ -256,8 +244,6 @@ static void pas_audio_output_block(int dev, unsigned long buf, int count, { unsigned long flags, cnt; - DEB(printk("pas2_pcm.c: static void pas_audio_output_block(char *buf = %P, int count = %X)\n", buf, count)); - cnt = count; if (audio_devs[dev]->dmap_out->dma > 3) cnt >>= 1; @@ -303,8 +289,6 @@ static void pas_audio_start_input(int dev, unsigned long buf, int count, unsigned long flags; int cnt; - DEB(printk("pas2_pcm.c: static void pas_audio_start_input(char *buf = %P, int count = %X)\n", buf, count)); - cnt = count; if (audio_devs[dev]->dmap_out->dma > 3) cnt >>= 1; @@ -388,8 +372,6 @@ static struct audio_driver pas_audio_driver = void __init pas_pcm_init(struct address_info *hw_config) { - DEB(printk("pas2_pcm.c: long pas_pcm_init()\n")); - pcm_bitsok = 8; if (pas_read(0xEF8B) & 0x08) pcm_bitsok |= 16; diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c index 851a1da46be1..3d50fb4236ed 100644 --- a/sound/oss/sb_common.c +++ b/sound/oss/sb_common.c @@ -226,8 +226,6 @@ int sb_dsp_reset(sb_devc * devc) { int loopc; - DEB(printk("Entered sb_dsp_reset()\n")); - if (devc->model == MDL_ESS) return ess_dsp_reset (devc); /* This is only for non-ESS chips */ @@ -246,8 +244,6 @@ int sb_dsp_reset(sb_devc * devc) return 0; /* Sorry */ } - DEB(printk("sb_dsp_reset() OK\n")); - return 1; } diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index 0e7254bde4c2..b47a69026f1b 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -865,8 +865,6 @@ printk(KERN_INFO "FKS: ess_dsp_reset 1\n"); ess_show_mixerregs (devc); #endif - DEB(printk("Entered ess_dsp_reset()\n")); - outb(3, DSP_RESET); /* Reset FIFO too */ udelay(10); @@ -881,8 +879,6 @@ ess_show_mixerregs (devc); } ess_extended (devc); - DEB(printk("sb_dsp_reset() OK\n")); - #ifdef FKS_LOGGING printk(KERN_INFO "FKS: dsp_reset 2\n"); ess_show_mixerregs (devc); diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index 9b9f7d385134..c0eea1dfe90f 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -216,8 +216,6 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun dev = dev >> 4; - DEB(printk("sequencer_write(dev=%d, count=%d)\n", dev, count)); - if (mode == OPEN_READ) return -EIO; @@ -959,8 +957,6 @@ int sequencer_open(int dev, struct file *file) dev = dev >> 4; mode = translate_mode(file); - DEB(printk("sequencer_open(dev=%d)\n", dev)); - if (!sequencer_ok) { /* printk("Sound card: sequencer not initialized\n");*/ @@ -1133,8 +1129,6 @@ void sequencer_release(int dev, struct file *file) dev = dev >> 4; - DEB(printk("sequencer_release(dev=%d)\n", dev)); - /* * Wait until the queue is empty (if we don't have nonblock) */ diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h index 9d35c4c65b9b..f2554ab78f5e 100644 --- a/sound/oss/sound_config.h +++ b/sound/oss/sound_config.h @@ -123,10 +123,6 @@ static inline int translate_mode(struct file *file) #include "sound_calls.h" #include "dev_table.h" -#ifndef DEB -#define DEB(x) -#endif - #ifndef DDB #define DDB(x) do {} while (0) #endif diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index e7780349cc55..b70c7c8f9c5d 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -154,7 +154,6 @@ static ssize_t sound_read(struct file *file, char __user *buf, size_t count, lof mutex_lock(&soundcard_mutex); - DEB(printk("sound_read(dev=%d, count=%d)\n", dev, count)); switch (dev & 0x0f) { case SND_DEV_DSP: case SND_DEV_DSP16: @@ -180,7 +179,6 @@ static ssize_t sound_write(struct file *file, const char __user *buf, size_t cou int ret = -EINVAL; mutex_lock(&soundcard_mutex); - DEB(printk("sound_write(dev=%d, count=%d)\n", dev, count)); switch (dev & 0x0f) { case SND_DEV_SEQ: case SND_DEV_SEQ2: @@ -206,7 +204,6 @@ static int sound_open(struct inode *inode, struct file *file) int dev = iminor(inode); int retval; - DEB(printk("sound_open(dev=%d)\n", dev)); if ((dev >= SND_NDEVS) || (dev < 0)) { printk(KERN_ERR "Invalid minor device %d\n", dev); return -ENXIO; @@ -257,7 +254,6 @@ static int sound_release(struct inode *inode, struct file *file) int dev = iminor(inode); mutex_lock(&soundcard_mutex); - DEB(printk("sound_release(dev=%d)\n", dev)); switch (dev & 0x0f) { case SND_DEV_CTL: module_put(mixer_devs[dev >> 4]->owner); @@ -351,7 +347,6 @@ static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) if (!access_ok(VERIFY_WRITE, p, len)) return -EFAULT; } - DEB(printk("sound_ioctl(dev=%d, cmd=0x%x, arg=0x%x)\n", dev, cmd, arg)); if (cmd == OSS_GETVERSION) return __put_user(SOUND_VERSION, (int __user *)p); @@ -409,7 +404,6 @@ static unsigned int sound_poll(struct file *file, poll_table * wait) struct inode *inode = file_inode(file); int dev = iminor(inode); - DEB(printk("sound_poll(dev=%d)\n", dev)); switch (dev & 0x0f) { case SND_DEV_SEQ: case SND_DEV_SEQ2: diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c index 5433c6f5eca2..62b8869f5a4c 100644 --- a/sound/oss/uart401.c +++ b/sound/oss/uart401.c @@ -274,19 +274,12 @@ static int reset_uart401(uart401_devc * devc) } } - + /* Flush input before enabling interrupts */ if (ok) - { - DEB(printk("Reset UART401 OK\n")); - } + uart401_input_loop(devc); else DDB(printk("Reset UART401 failed - No hardware detected.\n")); - if (ok) - uart401_input_loop(devc); /* - * Flush input before enabling interrupts - */ - return ok; } diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 97993e17f46a..248b90abb882 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -187,13 +187,14 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) struct azx_dev *azx_dev = &chip->azx_dev[dev]; dsp_lock(azx_dev); if (!azx_dev->opened && !dsp_is_locked(azx_dev)) { - res = azx_dev; - if (res->assigned_key == key) { - res->opened = 1; - res->assigned_key = key; + if (azx_dev->assigned_key == key) { + azx_dev->opened = 1; + azx_dev->assigned_key = key; dsp_unlock(azx_dev); return azx_dev; } + if (!res) + res = azx_dev; } dsp_unlock(azx_dev); } @@ -1604,7 +1605,7 @@ static void azx_exit_link_reset(struct azx *chip) } /* reset codec link */ -static int azx_reset(struct azx *chip, int full_reset) +static int azx_reset(struct azx *chip, bool full_reset) { if (!full_reset) goto __skip; @@ -1701,7 +1702,7 @@ static void azx_int_clear(struct azx *chip) /* * reset and start the controller registers */ -void azx_init_chip(struct azx *chip, int full_reset) +void azx_init_chip(struct azx *chip, bool full_reset) { if (chip->initialized) return; @@ -1758,7 +1759,7 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) #ifdef CONFIG_PM_RUNTIME if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) - if (chip->card->dev->power.runtime_status != RPM_ACTIVE) + if (!pm_runtime_active(chip->card->dev)) return IRQ_NONE; #endif @@ -1841,7 +1842,7 @@ static void azx_bus_reset(struct hda_bus *bus) bus->in_reset = 1; azx_stop_chip(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); #ifdef CONFIG_PM if (chip->initialized) { struct azx_pcm *p; @@ -1948,7 +1949,7 @@ int azx_codec_create(struct azx *chip, const char *model, * get back to the sanity state. */ azx_stop_chip(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); } } } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 1d2e3be2bae6..baf0e77330af 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -37,7 +37,7 @@ int azx_alloc_stream_pages(struct azx *chip); void azx_free_stream_pages(struct azx *chip); /* Low level azx interface */ -void azx_init_chip(struct azx *chip, int full_reset); +void azx_init_chip(struct azx *chip, bool full_reset); void azx_stop_chip(struct azx *chip); void azx_enter_link_reset(struct azx *chip); irqreturn_t azx_interrupt(int irq, void *dev_id); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 77ca894f8284..d6bca62ef387 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -636,7 +636,7 @@ static int azx_resume(struct device *dev) return -EIO; azx_init_pci(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); @@ -689,7 +689,7 @@ static int azx_runtime_resume(struct device *dev) status = azx_readw(chip, STATESTS); azx_init_pci(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); bus = chip->bus; if (status && bus) { diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ea2351d119f0..14ae979a92ea 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3026,6 +3026,11 @@ static void alc283_init(struct hda_codec *codec) bool hp_pin_sense; int val; + if (!spec->gen.autocfg.hp_outs) { + if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) + hp_pin = spec->gen.autocfg.line_out_pins[0]; + } + alc283_restore_default_value(codec); if (!hp_pin) @@ -3062,6 +3067,11 @@ static void alc283_shutup(struct hda_codec *codec) bool hp_pin_sense; int val; + if (!spec->gen.autocfg.hp_outs) { + if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) + hp_pin = spec->gen.autocfg.line_out_pins[0]; + } + if (!hp_pin) { alc269_shutup(codec); return; @@ -3085,6 +3095,7 @@ static void alc283_shutup(struct hda_codec *codec) if (hp_pin_sense) msleep(100); + alc_auto_setup_eapd(codec, false); snd_hda_shutup_pins(codec); alc_write_coef_idx(codec, 0x43, 0x9614); } @@ -3361,8 +3372,9 @@ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) if (spec->mute_led_polarity) enabled = !enabled; - pinval = AC_PINCTL_IN_EN | - (enabled ? AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80); + pinval = snd_hda_codec_get_pin_target(codec, spec->mute_led_nid); + pinval &= ~AC_PINCTL_VREFEN; + pinval |= enabled ? AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80; if (spec->mute_led_nid) snd_hda_set_pin_ctl_cache(codec, spec->mute_led_nid, pinval); } @@ -3994,6 +4006,10 @@ static void alc283_fixup_chromebook(struct hda_codec *codec, spec->gen.mixer_nid = 0; break; case HDA_FIXUP_ACT_INIT: + /* MIC2-VREF control */ + /* Set to manual mode */ + val = alc_read_coef_idx(codec, 0x06); + alc_write_coef_idx(codec, 0x06, val & ~0x000c); /* Enable Line1 input control by verb */ val = alc_read_coef_idx(codec, 0x1a); alc_write_coef_idx(codec, 0x1a, val | (1 << 4)); @@ -4602,6 +4618,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x065f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0667, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0668, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0669, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), @@ -4768,7 +4785,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, - {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-chrome"}, + {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {} }; diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index ed2144eee38a..496dbd0ad5db 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -579,12 +579,37 @@ static struct snd_ak4xxx_private akm_vx442_priv = { #ifdef CONFIG_PM_SLEEP static int snd_ice1712_delta_resume(struct snd_ice1712 *ice) { - unsigned char akm_backup[AK4XXX_IMAGE_SIZE]; + unsigned char akm_img_bak[AK4XXX_IMAGE_SIZE]; + unsigned char akm_vol_bak[AK4XXX_IMAGE_SIZE]; + + /* init spdif */ + switch (ice->eeprom.subvendor) { + case ICE1712_SUBDEVICE_AUDIOPHILE: + case ICE1712_SUBDEVICE_DELTA410: + case ICE1712_SUBDEVICE_DELTA1010E: + case ICE1712_SUBDEVICE_DELTA1010LT: + case ICE1712_SUBDEVICE_VX442: + case ICE1712_SUBDEVICE_DELTA66E: + snd_cs8427_init(ice->i2c, ice->cs8427); + break; + case ICE1712_SUBDEVICE_DELTA1010: + case ICE1712_SUBDEVICE_MEDIASTATION: + /* nothing */ + break; + case ICE1712_SUBDEVICE_DELTADIO2496: + case ICE1712_SUBDEVICE_DELTA66: + /* Set spdif defaults */ + snd_ice1712_delta_cs8403_spdif_write(ice, ice->spdif.cs8403_bits); + break; + } + /* init codec and restore registers */ if (ice->akm_codecs) { - memcpy(akm_backup, ice->akm->images, sizeof(akm_backup)); + memcpy(akm_img_bak, ice->akm->images, sizeof(akm_img_bak)); + memcpy(akm_vol_bak, ice->akm->volumes, sizeof(akm_vol_bak)); snd_akm4xxx_init(ice->akm); - memcpy(ice->akm->images, akm_backup, sizeof(akm_backup)); + memcpy(ice->akm->images, akm_img_bak, sizeof(akm_img_bak)); + memcpy(ice->akm->volumes, akm_vol_bak, sizeof(akm_vol_bak)); snd_akm4xxx_reset(ice->akm, 0); } diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 291672fc4a99..d9b9e4595f17 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -685,9 +685,10 @@ static snd_pcm_uframes_t snd_ice1712_playback_pointer(struct snd_pcm_substream * if (!(snd_ice1712_read(ice, ICE1712_IREG_PBK_CTRL) & 1)) return 0; ptr = runtime->buffer_size - inw(ice->ddma_port + 4); + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static snd_pcm_uframes_t snd_ice1712_playback_ds_pointer(struct snd_pcm_substream *substream) @@ -704,9 +705,10 @@ static snd_pcm_uframes_t snd_ice1712_playback_ds_pointer(struct snd_pcm_substrea addr = ICE1712_DSC_ADDR0; ptr = snd_ice1712_ds_read(ice, substream->number * 2, addr) - ice->playback_con_virt_addr[substream->number]; + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static snd_pcm_uframes_t snd_ice1712_capture_pointer(struct snd_pcm_substream *substream) @@ -717,9 +719,10 @@ static snd_pcm_uframes_t snd_ice1712_capture_pointer(struct snd_pcm_substream *s if (!(snd_ice1712_read(ice, ICE1712_IREG_CAP_CTRL) & 1)) return 0; ptr = inl(ICEREG(ice, CONCAP_ADDR)) - ice->capture_con_virt_addr; + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static const struct snd_pcm_hardware snd_ice1712_playback = { @@ -1048,6 +1051,8 @@ __out: old = inb(ICEMT(ice, RATE)); if (!force && old == val) goto __out; + + ice->cur_rate = rate; outb(val, ICEMT(ice, RATE)); spin_unlock_irqrestore(&ice->reg_lock, flags); @@ -1114,9 +1119,10 @@ static snd_pcm_uframes_t snd_ice1712_playback_pro_pointer(struct snd_pcm_substre if (!(inl(ICEMT(ice, PLAYBACK_CONTROL)) & ICE1712_PLAYBACK_START)) return 0; ptr = ice->playback_pro_size - (inw(ICEMT(ice, PLAYBACK_SIZE)) << 2); + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static snd_pcm_uframes_t snd_ice1712_capture_pro_pointer(struct snd_pcm_substream *substream) @@ -1127,9 +1133,10 @@ static snd_pcm_uframes_t snd_ice1712_capture_pro_pointer(struct snd_pcm_substrea if (!(inl(ICEMT(ice, PLAYBACK_CONTROL)) & ICE1712_CAPTURE_START_SHADOW)) return 0; ptr = ice->capture_pro_size - (inw(ICEMT(ice, CAPTURE_SIZE)) << 2); + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static const struct snd_pcm_hardware snd_ice1712_playback_pro = { @@ -2832,6 +2839,12 @@ static int snd_ice1712_suspend(struct device *dev) snd_pcm_suspend_all(ice->pcm_ds); snd_ac97_suspend(ice->ac97); + spin_lock_irq(&ice->reg_lock); + ice->pm_saved_is_spdif_master = is_spdif_master(ice); + ice->pm_saved_spdif_ctrl = inw(ICEMT(ice, ROUTE_SPDOUT)); + ice->pm_saved_route = inw(ICEMT(ice, ROUTE_PSDOUT03)); + spin_unlock_irq(&ice->reg_lock); + if (ice->pm_suspend) ice->pm_suspend(ice); @@ -2846,6 +2859,7 @@ static int snd_ice1712_resume(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct snd_card *card = dev_get_drvdata(dev); struct snd_ice1712 *ice = card->private_data; + int rate; if (!ice->pm_suspend_enabled) return 0; @@ -2860,14 +2874,37 @@ static int snd_ice1712_resume(struct device *dev) pci_set_master(pci); + if (ice->cur_rate) + rate = ice->cur_rate; + else + rate = PRO_RATE_DEFAULT; + if (snd_ice1712_chip_init(ice) < 0) { snd_card_disconnect(card); return -EIO; } + ice->cur_rate = rate; + if (ice->pm_resume) ice->pm_resume(ice); + if (ice->pm_saved_is_spdif_master) { + /* switching to external clock via SPDIF */ + spin_lock_irq(&ice->reg_lock); + outb(inb(ICEMT(ice, RATE)) | ICE1712_SPDIF_MASTER, + ICEMT(ice, RATE)); + spin_unlock_irq(&ice->reg_lock); + snd_ice1712_set_input_clock_source(ice, 1); + } else { + /* internal on-card clock */ + snd_ice1712_set_pro_rate(ice, rate, 1); + snd_ice1712_set_input_clock_source(ice, 0); + } + + outw(ice->pm_saved_spdif_ctrl, ICEMT(ice, ROUTE_SPDOUT)); + outw(ice->pm_saved_route, ICEMT(ice, ROUTE_PSDOUT03)); + if (ice->ac97) snd_ac97_resume(ice->ac97); diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 09f7e773bafb..f500905e9373 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -902,7 +902,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; alc5623_reset(codec); @@ -961,7 +960,7 @@ static int alc5623_probe(struct snd_soc_codec *codec) return -EINVAL; } - return ret; + return 0; } /* power down chip */ diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index ec071a6306ef..85942ca36cbf 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1061,7 +1061,6 @@ static int alc5632_resume(struct snd_soc_codec *codec) static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); - int ret; /* power on device */ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1075,7 +1074,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) return -EINVAL; } - return ret; + return 0; } /* power down chip */ @@ -1191,11 +1190,18 @@ static const struct i2c_device_id alc5632_i2c_table[] = { }; MODULE_DEVICE_TABLE(i2c, alc5632_i2c_table); +static const struct of_device_id alc5632_of_match[] = { + { .compatible = "realtek,alc5632", }, + { } +}; +MODULE_DEVICE_TABLE(of, alc5632_of_match); + /* i2c codec control layer */ static struct i2c_driver alc5632_i2c_driver = { .driver = { .name = "alc5632", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(alc5632_of_match), }, .probe = alc5632_i2c_probe, .remove = alc5632_i2c_remove, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index f0ca6bee6771..460d35547a68 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1259,7 +1259,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, } dev_info(&i2c_client->dev, "Cirrus Logic CS42L52, Revision: %02X\n", - reg & 0xFF); + reg & CS42L52_CHIP_REV_MASK); /* Set Platform Data */ if (cs42l52->pdata.mica_diff_cfg) diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 6fb8f00f4191..ac445993e6bf 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -37,7 +37,7 @@ #define CS42L52_CHIP_REV_A0 0x00 #define CS42L52_CHIP_REV_A1 0x01 #define CS42L52_CHIP_REV_B0 0x02 -#define CS42L52_CHIP_REV_MASK 0x03 +#define CS42L52_CHIP_REV_MASK 0x07 #define CS42L52_PWRCTL1 0x02 #define CS42L52_PWRCTL1_PDN_ALL 0x9F diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 082299a4e2fa..85020322eee7 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -495,17 +495,16 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) regcache_cache_bypass(cs42xx8->regmap, true); /* Validate the chip ID */ - regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); - if (val < 0) { - dev_err(dev, "failed to get device ID: %x", val); - ret = -EINVAL; + ret = regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); + if (ret < 0) { + dev_err(dev, "failed to get device ID, ret = %d", ret); goto err_enable; } /* The top four bits of the chip ID should be 0000 */ - if ((val & CS42XX8_CHIPID_CHIP_ID_MASK) != 0x00) { + if (((val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4) != 0x00) { dev_err(dev, "unmatched chip ID: %d\n", - val & CS42XX8_CHIPID_CHIP_ID_MASK); + (val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4); ret = -EINVAL; goto err_enable; } diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 7d168ec71cd7..48f3fef68484 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1571,7 +1571,8 @@ static int da732x_i2c_probe(struct i2c_client *i2c, } dev_info(&i2c->dev, "Revision: %d.%d\n", - (reg & DA732X_ID_MAJOR_MASK), (reg & DA732X_ID_MINOR_MASK)); + (reg & DA732X_ID_MAJOR_MASK) >> 4, + (reg & DA732X_ID_MINOR_MASK)); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da732x, da732x_dai, ARRAY_SIZE(da732x_dai)); diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 98c6e104357c..f7b0b37aa858 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2399,11 +2399,18 @@ static const struct i2c_device_id max98090_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); +static const struct of_device_id max98090_of_match[] = { + { .compatible = "maxim,max98090", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98090_of_match); + static struct i2c_driver max98090_i2c_driver = { .driver = { .name = "max98090", .owner = THIS_MODULE, .pm = &max98090_pm, + .of_match_table = of_match_ptr(max98090_of_match), }, .probe = max98090_i2c_probe, .remove = max98090_i2c_remove, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0061ae6b6716..68b4dd622b87 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2074,6 +2074,14 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +#if defined(CONFIG_OF) +static const struct of_device_id rt5640_of_match[] = { + { .compatible = "realtek,rt5640", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5640_of_match); +#endif + #ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, @@ -2203,6 +2211,7 @@ static struct i2c_driver rt5640_i2c_driver = { .name = "rt5640", .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt5640_acpi_match), + .of_match_table = of_match_ptr(rt5640_of_match), }, .probe = rt5640_i2c_probe, .remove = rt5640_i2c_remove, diff --git a/sound/soc/codecs/tlv320aic23-i2c.c b/sound/soc/codecs/tlv320aic23-i2c.c index 20fc46092c2c..b73c94ebcc2a 100644 --- a/sound/soc/codecs/tlv320aic23-i2c.c +++ b/sound/soc/codecs/tlv320aic23-i2c.c @@ -43,9 +43,16 @@ static const struct i2c_device_id tlv320aic23_id[] = { MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); +static const struct of_device_id tlv320aic23_of_match[] = { + { .compatible = "ti,tlv320aic23", }, + { } +}; +MODULE_DEVICE_TABLE(of, tlv320aic23_of_match); + static struct i2c_driver tlv320aic23_i2c_driver = { .driver = { .name = "tlv320aic23-codec", + .of_match_table = of_match_ptr(tlv320aic23_of_match), }, .probe = tlv320aic23_i2c_probe, .remove = __exit_p(tlv320aic23_i2c_remove), diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index a01ae97c90aa..4f75cac462d1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -336,7 +336,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; @@ -344,7 +344,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; @@ -352,7 +352,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index c4a423111673..56da8c8c5960 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -23,6 +23,71 @@ #include "fsl_sai.h" +#define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\ + FSL_SAI_CSR_FEIE) + +static irqreturn_t fsl_sai_isr(int irq, void *devid) +{ + struct fsl_sai *sai = (struct fsl_sai *)devid; + struct device *dev = &sai->pdev->dev; + u32 xcsr, mask; + + /* Only handle those what we enabled */ + mask = (FSL_SAI_FLAGS >> FSL_SAI_CSR_xIE_SHIFT) << FSL_SAI_CSR_xF_SHIFT; + + /* Tx IRQ */ + regmap_read(sai->regmap, FSL_SAI_TCSR, &xcsr); + xcsr &= mask; + + if (xcsr & FSL_SAI_CSR_WSF) + dev_dbg(dev, "isr: Start of Tx word detected\n"); + + if (xcsr & FSL_SAI_CSR_SEF) + dev_warn(dev, "isr: Tx Frame sync error detected\n"); + + if (xcsr & FSL_SAI_CSR_FEF) { + dev_warn(dev, "isr: Transmit underrun detected\n"); + /* FIFO reset for safety */ + xcsr |= FSL_SAI_CSR_FR; + } + + if (xcsr & FSL_SAI_CSR_FWF) + dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n"); + + if (xcsr & FSL_SAI_CSR_FRF) + dev_dbg(dev, "isr: Transmit FIFO watermark has been reached\n"); + + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_xF_W_MASK | FSL_SAI_CSR_FR, xcsr); + + /* Rx IRQ */ + regmap_read(sai->regmap, FSL_SAI_RCSR, &xcsr); + xcsr &= mask; + + if (xcsr & FSL_SAI_CSR_WSF) + dev_dbg(dev, "isr: Start of Rx word detected\n"); + + if (xcsr & FSL_SAI_CSR_SEF) + dev_warn(dev, "isr: Rx Frame sync error detected\n"); + + if (xcsr & FSL_SAI_CSR_FEF) { + dev_warn(dev, "isr: Receive overflow detected\n"); + /* FIFO reset for safety */ + xcsr |= FSL_SAI_CSR_FR; + } + + if (xcsr & FSL_SAI_CSR_FWF) + dev_dbg(dev, "isr: Enabled receive FIFO is full\n"); + + if (xcsr & FSL_SAI_CSR_FRF) + dev_dbg(dev, "isr: Receive FIFO watermark has been reached\n"); + + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_xF_W_MASK | FSL_SAI_CSR_FR, xcsr); + + return IRQ_HANDLED; +} + static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -114,7 +179,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * that is, together with the last bit of the previous * data word. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 |= FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_LEFT_J: @@ -122,7 +187,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * Frame high, one word length for frame sync, * frame sync asserts with the first bit of the frame. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); break; case SND_SOC_DAIFMT_DSP_A: @@ -132,7 +197,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * that is, together with the last bit of the previous * data word. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 &= ~FSL_SAI_CR4_FSP; val_cr4 |= FSL_SAI_CR4_FSE; sai->is_dsp_mode = true; @@ -142,7 +207,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * Frame high, one bit for frame sync, * frame sync asserts with the first bit of the frame. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); sai->is_dsp_mode = true; break; @@ -373,8 +438,8 @@ 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); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, FSL_SAI_FLAGS); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); 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, @@ -490,12 +555,14 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int ret; + int irq, ret; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) return -ENOMEM; + sai->pdev = pdev; + 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; @@ -514,6 +581,18 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(sai->regmap); } + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, fsl_sai_isr, 0, np->name, sai); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + return ret; + } + 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 e432260be598..a264185c7138 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -37,7 +37,21 @@ /* SAI Transmit/Recieve Control Register */ #define FSL_SAI_CSR_TERE BIT(31) +#define FSL_SAI_CSR_FR BIT(25) +#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) +#define FSL_SAI_CSR_xF_W_MASK (0x7 << FSL_SAI_CSR_xF_W_SHIFT) +#define FSL_SAI_CSR_WSF BIT(20) +#define FSL_SAI_CSR_SEF BIT(19) +#define FSL_SAI_CSR_FEF BIT(18) #define FSL_SAI_CSR_FWF BIT(17) +#define FSL_SAI_CSR_FRF BIT(16) +#define FSL_SAI_CSR_xIE_SHIFT 8 +#define FSL_SAI_CSR_WSIE BIT(12) +#define FSL_SAI_CSR_SEIE BIT(11) +#define FSL_SAI_CSR_FEIE BIT(10) +#define FSL_SAI_CSR_FWIE BIT(9) #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) @@ -99,6 +113,7 @@ #define FSL_SAI_MAXBURST_RX 6 struct fsl_sai { + struct platform_device *pdev; struct regmap *regmap; bool big_endian_regs; diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 4a88e36c82ec..76b072bd4ba2 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -39,15 +39,15 @@ struct s3c_ac97_info { }; static struct s3c_ac97_info s3c_ac97; -static struct s3c2410_dma_client s3c_dma_client_out = { +static struct s3c_dma_client s3c_dma_client_out = { .name = "AC97 PCMOut" }; -static struct s3c2410_dma_client s3c_dma_client_in = { +static struct s3c_dma_client s3c_dma_client_in = { .name = "AC97 PCMIn" }; -static struct s3c2410_dma_client s3c_dma_client_micin = { +static struct s3c_dma_client s3c_dma_client_micin = { .name = "AC97 MicIn" }; diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 225e5378014e..ad7c0f04f00d 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -14,8 +14,12 @@ #include <sound/dmaengine_pcm.h> +struct s3c_dma_client { + char *name; +}; + struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ + struct s3c_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; int dma_size; /* Size of the DMA transfer */ diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0a9b44c940ce..048ead967199 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1211,10 +1211,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_playback.dma_addr = regs_base + I2STXD; pri_dai->dma_capture.dma_addr = regs_base + I2SRXD; pri_dai->dma_playback.client = - (struct s3c2410_dma_client *)&pri_dai->dma_playback; + (struct s3c_dma_client *)&pri_dai->dma_playback; pri_dai->dma_playback.ch_name = "tx"; pri_dai->dma_capture.client = - (struct s3c2410_dma_client *)&pri_dai->dma_capture; + (struct s3c_dma_client *)&pri_dai->dma_capture; pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; @@ -1233,7 +1233,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.client = - (struct s3c2410_dma_client *)&sec_dai->dma_playback; + (struct s3c_dma_client *)&sec_dai->dma_playback; sec_dai->dma_playback.ch_name = "tx-sec"; if (!np) { diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 6a5e4bf6ac96..ab54e297957c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "pcm.h" @@ -132,11 +131,11 @@ struct s3c_pcm_info { struct s3c_dma_params *dma_capture; }; -static struct s3c2410_dma_client s3c_pcm_dma_client_out = { +static struct s3c_dma_client s3c_pcm_dma_client_out = { .name = "PCM Stereo out" }; -static struct s3c2410_dma_client s3c_pcm_dma_client_in = { +static struct s3c_dma_client s3c_pcm_dma_client_in = { .name = "PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d0794458963a..e9bb5d7a71ee 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,11 +33,11 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c2410_dma_client s3c2412_dma_client_out = { +static struct s3c_dma_client s3c2412_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c2412_dma_client_in = { +static struct s3c_dma_client s3c2412_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index f31e916dd8c4..d7b8457b5650 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,11 +31,11 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c2410_dma_client s3c24xx_dma_client_out = { +static struct s3c_dma_client s3c24xx_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c24xx_dma_client_in = { +static struct s3c_dma_client s3c24xx_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 28487dcc4538..cfe63b7bcc9f 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -18,7 +18,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "spdif.h" @@ -94,7 +93,7 @@ struct samsung_spdif_info { struct s3c_dma_params *dma_playback; }; -static struct s3c2410_dma_client spdif_dma_client_out = { +static struct s3c_dma_client spdif_dma_client_out = { .name = "S/PDIF Stereo out", }; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 49de5c1284f6..131336d40492 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1501,7 +1501,8 @@ static void retire_playback_urb(struct snd_usb_substream *subs, * The error should be lower than 2ms since the estimate relies * on two reads of a counter updated every ms. */ - if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) + if (printk_ratelimit() && + abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) dev_dbg(&subs->dev->dev, "delay: estimated %d, actual %d\n", est_delay, subs->last_delay); |