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authorLinus Torvalds <torvalds@linux-foundation.org>2014-04-10 09:19:44 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2014-04-10 09:19:44 -0700
commite7990d45bb88c2f0565b5ee4c32eefe81653faff (patch)
tree852ab6988b7226083fefa0e0e851dbff0e7ec7f2
parent190a3998be3ede25d6145e187d6d321f504d28fb (diff)
parenta5065eb6da55b226661456e6a7435f605df98111 (diff)
downloadlinux-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 ...
-rw-r--r--Documentation/devicetree/bindings/sound/fsl,ssi.txt21
-rw-r--r--include/sound/cs8427.h1
-rw-r--r--sound/i2c/cs8427.c57
-rw-r--r--sound/mips/au1x00.c13
-rw-r--r--sound/oss/ad1848.c4
-rw-r--r--sound/oss/dmasound/dmasound_paula.c14
-rw-r--r--sound/oss/opl3.c3
-rw-r--r--sound/oss/pas2_mixer.c9
-rw-r--r--sound/oss/pas2_pcm.c18
-rw-r--r--sound/oss/sb_common.c4
-rw-r--r--sound/oss/sb_ess.c4
-rw-r--r--sound/oss/sequencer.c6
-rw-r--r--sound/oss/sound_config.h4
-rw-r--r--sound/oss/soundcard.c6
-rw-r--r--sound/oss/uart401.c11
-rw-r--r--sound/pci/hda/hda_controller.c19
-rw-r--r--sound/pci/hda/hda_controller.h2
-rw-r--r--sound/pci/hda/hda_intel.c4
-rw-r--r--sound/pci/hda/patch_realtek.c23
-rw-r--r--sound/pci/ice1712/delta.c31
-rw-r--r--sound/pci/ice1712/ice1712.c47
-rw-r--r--sound/soc/codecs/alc5623.c3
-rw-r--r--sound/soc/codecs/alc5632.c10
-rw-r--r--sound/soc/codecs/cs42l52.c2
-rw-r--r--sound/soc/codecs/cs42l52.h2
-rw-r--r--sound/soc/codecs/cs42xx8.c11
-rw-r--r--sound/soc/codecs/da732x.c3
-rw-r--r--sound/soc/codecs/max98090.c7
-rw-r--r--sound/soc/codecs/rt5640.c9
-rw-r--r--sound/soc/codecs/tlv320aic23-i2c.c7
-rw-r--r--sound/soc/davinci/davinci-mcasp.c6
-rw-r--r--sound/soc/fsl/fsl_sai.c93
-rw-r--r--sound/soc/fsl/fsl_sai.h15
-rw-r--r--sound/soc/samsung/ac97.c6
-rw-r--r--sound/soc/samsung/dma.h6
-rw-r--r--sound/soc/samsung/i2s.c6
-rw-r--r--sound/soc/samsung/pcm.c5
-rw-r--r--sound/soc/samsung/s3c2412-i2s.c4
-rw-r--r--sound/soc/samsung/s3c24xx-i2s.c4
-rw-r--r--sound/soc/samsung/spdif.c3
-rw-r--r--sound/usb/pcm.c3
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);