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Subject[GIT PULL] sound fixes for 2.6.31-rc4
Linus,

please pull ALSA fixes for v2.6.31-rc4 from:

git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus

containing the following small patches.


Thanks!

Takashi

===

Alexey Fisher (1):
ALSA: usb-audio - Volume control quirk for QuickCam E 3500

Chaithrika U S (1):
ASoC: tlv320aic3x: Enable PLL when not bypassed

Daniel Mack (1):
ALSA: snd_usb_caiaq: add support for Audio2DJ

Takashi Iwai (8):
ALSA: pcm - Fix regressions with VMware
ALSA: hda - Use snprintf() to be safer
ALSA: ctxfi - Fix uninitialized error checks
ALSA: hda - Restore GPIO1 properly at resume with AD1984A
ALSA: pcm - Add logging of hwptr updates and interrupt updates
ALSA: pcm - Fix warnings in debug loggings
ALSA: pcm - Fix hwptr buffer-size overlap bug
ALSA: hda - Fix mute control with some ALC262 models

---
Documentation/sound/alsa/Procfile.txt | 5 ++++
sound/core/pcm_lib.c | 36 ++++++++++++++++++++++++++++++++-
sound/pci/ctxfi/ctamixer.c | 14 +++++-------
sound/pci/ctxfi/ctsrc.c | 7 ++---
sound/pci/hda/patch_analog.c | 2 +-
sound/pci/hda/patch_realtek.c | 33 ++++++++++++++---------------
sound/pci/hda/patch_sigmatel.c | 2 +-
sound/soc/codecs/tlv320aic3x.c | 11 +++++++++-
sound/usb/Kconfig | 1 +
sound/usb/caiaq/audio.c | 1 +
sound/usb/caiaq/device.c | 8 ++++++-
sound/usb/caiaq/device.h | 1 +
sound/usb/usbmixer.c | 25 ++++++++++++++++++----
13 files changed, 107 insertions(+), 39 deletions(-)

diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt
index 381908d..719a819 100644
--- a/Documentation/sound/alsa/Procfile.txt
+++ b/Documentation/sound/alsa/Procfile.txt
@@ -101,6 +101,8 @@ card*/pcm*/xrun_debug
bit 0 = Enable XRUN/jiffies debug messages
bit 1 = Show stack trace at XRUN / jiffies check
bit 2 = Enable additional jiffies check
+ bit 3 = Log hwptr update at each period interrupt
+ bit 4 = Log hwptr update at each snd_pcm_update_hw_ptr()

When the bit 0 is set, the driver will show the messages to
kernel log when an xrun is detected. The debug message is
@@ -117,6 +119,9 @@ card*/pcm*/xrun_debug
buggy) hardware that doesn't give smooth pointer updates.
This feature is enabled via the bit 2.

+ Bits 3 and 4 are for logging the hwptr records. Note that
+ these will give flood of kernel messages.
+
card*/pcm*/sub*/info
The general information of this PCM sub-stream.

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 333e4dd..72cfd47 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -233,6 +233,18 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
xrun(substream);
return -EPIPE;
}
+ if (xrun_debug(substream, 8)) {
+ char name[16];
+ pcm_debug_name(substream, name, sizeof(name));
+ snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, "
+ "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n",
+ name, (unsigned int)pos,
+ (unsigned int)runtime->period_size,
+ (unsigned int)runtime->buffer_size,
+ (unsigned long)old_hw_ptr,
+ (unsigned long)runtime->hw_ptr_base,
+ (unsigned long)runtime->hw_ptr_interrupt);
+ }
hw_base = runtime->hw_ptr_base;
new_hw_ptr = hw_base + pos;
hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size;
@@ -244,18 +256,27 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
delta = new_hw_ptr - hw_ptr_interrupt;
}
if (delta < 0) {
- delta += runtime->buffer_size;
+ if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr)
+ delta += runtime->buffer_size;
if (delta < 0) {
hw_ptr_error(substream,
"Unexpected hw_pointer value "
"(stream=%i, pos=%ld, intr_ptr=%ld)\n",
substream->stream, (long)pos,
(long)hw_ptr_interrupt);
+#if 1
+ /* simply skipping the hwptr update seems more
+ * robust in some cases, e.g. on VMware with
+ * inaccurate timer source
+ */
+ return 0; /* skip this update */
+#else
/* rebase to interrupt position */
hw_base = new_hw_ptr = hw_ptr_interrupt;
/* align hw_base to buffer_size */
hw_base -= hw_base % runtime->buffer_size;
delta = 0;
+#endif
} else {
hw_base += runtime->buffer_size;
if (hw_base >= runtime->boundary)
@@ -344,6 +365,19 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
xrun(substream);
return -EPIPE;
}
+ if (xrun_debug(substream, 16)) {
+ char name[16];
+ pcm_debug_name(substream, name, sizeof(name));
+ snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, "
+ "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n",
+ name, (unsigned int)pos,
+ (unsigned int)runtime->period_size,
+ (unsigned int)runtime->buffer_size,
+ (unsigned long)old_hw_ptr,
+ (unsigned long)runtime->hw_ptr_base,
+ (unsigned long)runtime->hw_ptr_interrupt);
+ }
+
hw_base = runtime->hw_ptr_base;
new_hw_ptr = hw_base + pos;

diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c
index a1db51b..a7f4a67 100644
--- a/sound/pci/ctxfi/ctamixer.c
+++ b/sound/pci/ctxfi/ctamixer.c
@@ -242,13 +242,12 @@ static int get_amixer_rsc(struct amixer_mgr *mgr,

/* Allocate mem for amixer resource */
amixer = kzalloc(sizeof(*amixer), GFP_KERNEL);
- if (NULL == amixer) {
- err = -ENOMEM;
- return err;
- }
+ if (!amixer)
+ return -ENOMEM;

/* Check whether there are sufficient
* amixer resources to meet request. */
+ err = 0;
spin_lock_irqsave(&mgr->mgr_lock, flags);
for (i = 0; i < desc->msr; i++) {
err = mgr_get_resource(&mgr->mgr, 1, &idx);
@@ -397,12 +396,11 @@ static int get_sum_rsc(struct sum_mgr *mgr,

/* Allocate mem for sum resource */
sum = kzalloc(sizeof(*sum), GFP_KERNEL);
- if (NULL == sum) {
- err = -ENOMEM;
- return err;
- }
+ if (!sum)
+ return -ENOMEM;

/* Check whether there are sufficient sum resources to meet request. */
+ err = 0;
spin_lock_irqsave(&mgr->mgr_lock, flags);
for (i = 0; i < desc->msr; i++) {
err = mgr_get_resource(&mgr->mgr, 1, &idx);
diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c
index e1c145d..df43a5c 100644
--- a/sound/pci/ctxfi/ctsrc.c
+++ b/sound/pci/ctxfi/ctsrc.c
@@ -724,12 +724,11 @@ static int get_srcimp_rsc(struct srcimp_mgr *mgr,

/* Allocate mem for SRCIMP resource */
srcimp = kzalloc(sizeof(*srcimp), GFP_KERNEL);
- if (NULL == srcimp) {
- err = -ENOMEM;
- return err;
- }
+ if (!srcimp)
+ return -ENOMEM;

/* Check whether there are sufficient SRCIMP resources. */
+ err = 0;
spin_lock_irqsave(&mgr->mgr_lock, flags);
for (i = 0; i < desc->msr; i++) {
err = mgr_get_resource(&mgr->mgr, 1, &idx);
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index be7d25f..3da85ca 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3754,7 +3754,7 @@ static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
int mute = (!ucontrol->value.integer.value[0] &&
!ucontrol->value.integer.value[1]);
/* toggle GPIO1 according to the mute state */
- snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+ snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
mute ? 0x02 : 0x0);
return ret;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7e99763..8c8b273 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10631,6 +10631,18 @@ static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec,
alc262_lenovo_3000_automute(codec, 1);
}

+static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid,
+ int dir, int idx, long *valp)
+{
+ int i, change = 0;
+
+ for (i = 0; i < 2; i++, valp++)
+ change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx,
+ HDA_AMP_MUTE,
+ *valp ? 0 : HDA_AMP_MUTE);
+ return change;
+}
+
/* bind hp and internal speaker mute (with plug check) */
static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -10639,13 +10651,8 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;

- change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp ? 0 : HDA_AMP_MUTE);
- change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp ? 0 : HDA_AMP_MUTE);
-
+ change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
+ change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
if (change)
alc262_fujitsu_automute(codec, 0);
return change;
@@ -10680,10 +10687,7 @@ static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;

- change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp ? 0 : HDA_AMP_MUTE);
-
+ change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
if (change)
alc262_lenovo_3000_automute(codec, 0);
return change;
@@ -11854,12 +11858,7 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;

- change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[0] ? 0 : HDA_AMP_MUTE);
- change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[1] ? 0 : HDA_AMP_MUTE);
+ change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
if (change)
alc268_acer_automute(codec, 0);
return change;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index da7f9f6..512f3b9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4066,7 +4066,7 @@ static int stac92xx_add_jack(struct hda_codec *codec,
jack->nid = nid;
jack->type = type;

- sprintf(name, "%s at %s %s Jack",
+ snprintf(name, sizeof(name), "%s at %s %s Jack",
snd_hda_get_jack_type(def_conf),
snd_hda_get_jack_connectivity(def_conf),
snd_hda_get_jack_location(def_conf));
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index ab099f4..cb0d1bf 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -767,6 +767,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
u16 pll_d = 1;
+ u8 reg;

/* select data word length */
data =
@@ -801,8 +802,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
pll_q &= 0xf;
aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT);
aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV);
- } else
+ /* disable PLL if it is bypassed */
+ reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+ aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE);
+
+ } else {
aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV);
+ /* enable PLL when it is used */
+ reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+ aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE);
+ }

/* Route Left DAC to left channel input and
* right DAC to right channel input */
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index 523aec1..73525c0 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -48,6 +48,7 @@ config SND_USB_CAIAQ
* Native Instruments Kore Controller
* Native Instruments Kore Controller 2
* Native Instruments Audio Kontrol 1
+ * Native Instruments Audio 2 DJ
* Native Instruments Audio 4 DJ
* Native Instruments Audio 8 DJ
* Native Instruments Guitar Rig Session I/O
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 8f9b60c..121af06 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -646,6 +646,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE):
dev->samplerates |= SNDRV_PCM_RATE_192000;
/* fall thru */
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ):
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
dev->samplerates |= SNDRV_PCM_RATE_88200;
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index de38108..83e6c13 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,13 +35,14 @@
#include "input.h"

MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
"{Native Instruments, RigKontrol3},"
"{Native Instruments, Kore Controller},"
"{Native Instruments, Kore Controller 2},"
"{Native Instruments, Audio Kontrol 1},"
+ "{Native Instruments, Audio 2 DJ},"
"{Native Instruments, Audio 4 DJ},"
"{Native Instruments, Audio 8 DJ},"
"{Native Instruments, Session I/O},"
@@ -121,6 +122,11 @@ static struct usb_device_id snd_usb_id_table[] = {
.idVendor = USB_VID_NATIVEINSTRUMENTS,
.idProduct = USB_PID_AUDIO4DJ
},
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = USB_VID_NATIVEINSTRUMENTS,
+ .idProduct = USB_PID_AUDIO2DJ
+ },
{ /* terminator */ }
};

diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h
index ece7351..44e3edf 100644
--- a/sound/usb/caiaq/device.h
+++ b/sound/usb/caiaq/device.h
@@ -10,6 +10,7 @@
#define USB_PID_KORECONTROLLER 0x4711
#define USB_PID_KORECONTROLLER2 0x4712
#define USB_PID_AK1 0x0815
+#define USB_PID_AUDIO2DJ 0x041c
#define USB_PID_AUDIO4DJ 0x0839
#define USB_PID_AUDIO8DJ 0x1978
#define USB_PID_SESSIONIO 0x1915
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index 4bd3a7a..ec9cdf9 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -990,20 +990,35 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
break;
}

- /* quirk for UDA1321/N101 */
- /* note that detection between firmware 2.1.1.7 (N101) and later 2.1.1.21 */
- /* is not very clear from datasheets */
- /* I hope that the min value is -15360 for newer firmware --jk */
+ /* volume control quirks */
switch (state->chip->usb_id) {
case USB_ID(0x0471, 0x0101):
case USB_ID(0x0471, 0x0104):
case USB_ID(0x0471, 0x0105):
case USB_ID(0x0672, 0x1041):
+ /* quirk for UDA1321/N101.
+ * note that detection between firmware 2.1.1.7 (N101)
+ * and later 2.1.1.21 is not very clear from datasheets.
+ * I hope that the min value is -15360 for newer firmware --jk
+ */
if (!strcmp(kctl->id.name, "PCM Playback Volume") &&
cval->min == -15616) {
- snd_printk(KERN_INFO "using volume control quirk for the UDA1321/N101 chip\n");
+ snd_printk(KERN_INFO
+ "set volume quirk for UDA1321/N101 chip\n");
cval->max = -256;
}
+ break;
+
+ case USB_ID(0x046d, 0x09a4):
+ if (!strcmp(kctl->id.name, "Mic Capture Volume")) {
+ snd_printk(KERN_INFO
+ "set volume quirk for QuickCam E3500\n");
+ cval->min = 6080;
+ cval->max = 8768;
+ cval->res = 192;
+ }
+ break;
+
}

snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",

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