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Message-ID: <s5hzjxd2zec.wl%tiwai@suse.de>
Date: Fri, 05 Apr 2013 09:46:19 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Mark Brown <broonie@...nsource.wolfsonmicro.com>,
Liam Girdwood <lgirdwood@...il.com>,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] Sound fixes for 3.9-rc6
Linus,
please pull sound fixes for v3.9-rc6 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus
The topmost commit is fed61fb5275cf1fa4915d6415b4e376c87089d83
----------------------------------------------------------------
Sound fixes for 3.9-rc6
This contains slightly more volumes than usual at this stage, mostly
because of my vacation in the last week.
Nothing to scare, all small and/or trivial fixes:
- Fix loop path handling in ASoC DAPM
- Some memory handling fixes in ASoC core
- Fix spear_pcm to adapt to the updated API
- HD-audio HDMI ELD handling fixes
- Fix for CM6331 USB-audio SRC change bugs
- Revert power_save_controller option change due to user-space usage
- A few other small ASoC and HD-audio fixes
Thanks!
Takashi
----------------------------------------------------------------
Axel Lin (1):
ASoC: si476x: Add missing break for SNDRV_PCM_FORMAT_S8 switch case
David Henningsson (1):
ALSA: hda - fix typo in proc output
Jiri Slaby (1):
ALSA: hda/generic - fix uninitialized variable
Joe Perches (1):
ASoC:: max98090: Remove executable bit
Lars-Peter Clausen (2):
ASoC: spear_pcm: Update to new pcm_new() API
ASoC: dma-sh7760: Fix compile error
Mark Brown (1):
ASoC: dapm: Fix handling of loops
Markus Pargmann (1):
ASoC: pcm030 audio fabric: remove __init from probe
Mengdong Lin (2):
ALSA: hda - bug fix on return value when getting HDMI ELD info
ALSA: hda - bug fix on HDMI ELD debug message
Peter Ujfalusi (1):
ASoC: dapm: Fix pointer dereference in is_connected_output_ep()
Rainer Koenig (1):
ALSA: hda - Enabling Realtek ALC 671 codec
Sascha Hauer (1):
ASoC: imx-ssi: Fix occasional AC97 reset failure
Silviu-Mihai Popescu (1):
ASoC: core: fix invalid free of devm_ allocated data
Takashi Iwai (1):
Revert "ALSA: hda - Allow power_save_controller option override DCAPS"
Torstein Hegge (1):
ALSA: usb: Work around CM6631 sample rate change bug
Wei Yongjun (2):
ASoC: wm_adsp: fix possible memory leak in wm_adsp_load_coeff()
ASoC: core: fix possible memory leak in snd_soc_bytes_put()
---
Documentation/sound/alsa/ALSA-Configuration.txt | 5 ++-
include/sound/max98090.h | 0
include/sound/soc-dapm.h | 1 +
sound/pci/hda/hda_codec.c | 2 +-
sound/pci/hda/hda_eld.c | 2 +-
sound/pci/hda/hda_generic.c | 2 +-
sound/pci/hda/hda_intel.c | 6 ++--
sound/pci/hda/patch_hdmi.c | 2 +-
sound/pci/hda/patch_realtek.c | 4 ++-
sound/soc/codecs/max98090.c | 0
sound/soc/codecs/max98090.h | 0
sound/soc/codecs/si476x.c | 1 +
sound/soc/codecs/wm_adsp.c | 5 +--
sound/soc/fsl/imx-ssi.c | 5 +++
sound/soc/fsl/pcm030-audio-fabric.c | 2 +-
sound/soc/sh/dma-sh7760.c | 4 +--
sound/soc/soc-core.c | 8 ++---
sound/soc/soc-dapm.c | 14 ++++++++
sound/soc/spear/spear_pcm.c | 12 +++----
sound/usb/clock.c | 45 +++++++++++++++++++------
20 files changed, 83 insertions(+), 37 deletions(-)
mode change 100755 => 100644 include/sound/max98090.h
mode change 100755 => 100644 sound/soc/codecs/max98090.c
mode change 100755 => 100644 sound/soc/codecs/max98090.h
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 4499bd9..95731a0 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -890,9 +890,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
enable_msi - Enable Message Signaled Interrupt (MSI) (default = off)
power_save - Automatic power-saving timeout (in second, 0 =
disable)
- power_save_controller - Support runtime D3 of HD-audio controller
- (-1 = on for supported chip (default), false = off,
- true = force to on even for unsupported hardware)
+ power_save_controller - Reset HD-audio controller in power-saving mode
+ (default = on)
align_buffer_size - Force rounding of buffer/period sizes to multiples
of 128 bytes. This is more efficient in terms of memory
access but isn't required by the HDA spec and prevents
diff --git a/include/sound/max98090.h b/include/sound/max98090.h
old mode 100755
new mode 100644
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index e1ef63d..44a30b1 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -488,6 +488,7 @@ struct snd_soc_dapm_path {
/* status */
u32 connect:1; /* source and sink widgets are connected */
u32 walked:1; /* path has been walked */
+ u32 walking:1; /* path is in the process of being walked */
u32 weak:1; /* path ignored for power management */
int (*connected)(struct snd_soc_dapm_widget *source,
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index ecdf30e..4aba764 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -173,7 +173,7 @@ const char *snd_hda_get_jack_type(u32 cfg)
"Line Out", "Speaker", "HP Out", "CD",
"SPDIF Out", "Digital Out", "Modem Line", "Modem Hand",
"Line In", "Aux", "Mic", "Telephony",
- "SPDIF In", "Digitial In", "Reserved", "Other"
+ "SPDIF In", "Digital In", "Reserved", "Other"
};
return jack_types[(cfg & AC_DEFCFG_DEVICE)
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 7dd8463..d0d7ac1 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -320,7 +320,7 @@ int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid,
unsigned char *buf, int *eld_size)
{
int i;
- int ret;
+ int ret = 0;
int size;
/*
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 43c2ea5..2dbe767 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -740,7 +740,7 @@ EXPORT_SYMBOL_HDA(snd_hda_activate_path);
static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path)
{
struct hda_gen_spec *spec = codec->spec;
- bool changed;
+ bool changed = false;
int i;
if (!spec->power_down_unused || path->active)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 418bfc0..bcd40ee 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -134,8 +134,8 @@ MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
* this may give more power-saving, but will take longer time to
* wake up.
*/
-static int power_save_controller = -1;
-module_param(power_save_controller, bint, 0644);
+static bool power_save_controller = 1;
+module_param(power_save_controller, bool, 0644);
MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
#endif /* CONFIG_PM */
@@ -2931,8 +2931,6 @@ static int azx_runtime_idle(struct device *dev)
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip = card->private_data;
- if (power_save_controller > 0)
- return 0;
if (!power_save_controller ||
!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME))
return -EBUSY;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 78e1827..de8ac5c 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1196,7 +1196,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
_snd_printd(SND_PR_VERBOSE,
"HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
- codec->addr, pin_nid, eld->monitor_present, eld->eld_valid);
+ codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid);
if (eld->eld_valid) {
if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 563c24d..f15c36b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3440,7 +3440,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
const hda_nid_t *ssids;
if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 ||
- codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670)
+ codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670 ||
+ codec->vendor_id == 0x10ec0671)
ssids = alc663_ssids;
else
ssids = alc662_ssids;
@@ -3894,6 +3895,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 },
{ .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 },
{ .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 },
+ { .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 },
{ .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 },
{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
old mode 100755
new mode 100644
diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h
old mode 100755
new mode 100644
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index f2d61a1..566ea32 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -159,6 +159,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
width = SI476X_PCM_FORMAT_S8;
+ break;
case SNDRV_PCM_FORMAT_S16_LE:
width = SI476X_PCM_FORMAT_S16_LE;
break;
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f3f7e75..9af1bdd 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -828,7 +828,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
&buf_list);
if (!buf) {
adsp_err(dsp, "Out of memory\n");
- return -ENOMEM;
+ ret = -ENOMEM;
+ goto out_fw;
}
adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n",
@@ -865,7 +866,7 @@ out_fw:
wm_adsp_buf_free(&buf_list);
out:
kfree(file);
- return 0;
+ return ret;
}
int wm_adsp1_init(struct wm_adsp *adsp)
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 55464a5..810c7ee 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -496,6 +496,8 @@ static void imx_ssi_ac97_reset(struct snd_ac97 *ac97)
if (imx_ssi->ac97_reset)
imx_ssi->ac97_reset(ac97);
+ /* First read sometimes fails, do a dummy read */
+ imx_ssi_ac97_read(ac97, 0);
}
static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
@@ -504,6 +506,9 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
if (imx_ssi->ac97_warm_reset)
imx_ssi->ac97_warm_reset(ac97);
+
+ /* First read sometimes fails, do a dummy read */
+ imx_ssi_ac97_read(ac97, 0);
}
struct snd_ac97_bus_ops soc_ac97_ops = {
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index 8e52c14..eb43738 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -51,7 +51,7 @@ static struct snd_soc_card pcm030_card = {
.num_links = ARRAY_SIZE(pcm030_fabric_dai),
};
-static int __init pcm030_fabric_probe(struct platform_device *op)
+static int pcm030_fabric_probe(struct platform_device *op)
{
struct device_node *np = op->dev.of_node;
struct device_node *platform_np;
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 19eff8f..1a8b03e 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -342,8 +342,8 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd)
return 0;
}
-static struct snd_soc_platform sh7760_soc_platform = {
- .pcm_ops = &camelot_pcm_ops,
+static struct snd_soc_platform_driver sh7760_soc_platform = {
+ .ops = &camelot_pcm_ops,
.pcm_new = camelot_pcm_new,
.pcm_free = camelot_pcm_free,
};
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b7e84a7..507d251 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3140,7 +3140,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
if (params->mask) {
ret = regmap_read(codec->control_data, params->base, &val);
if (ret != 0)
- return ret;
+ goto out;
val &= params->mask;
@@ -3158,13 +3158,15 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
((u32 *)data)[0] |= cpu_to_be32(val);
break;
default:
- return -EINVAL;
+ ret = -EINVAL;
+ goto out;
}
}
ret = regmap_raw_write(codec->control_data, params->base,
data, len);
+out:
kfree(data);
return ret;
@@ -4197,7 +4199,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
dev_err(card->dev,
"ASoC: Property '%s' index %d could not be read: %d\n",
propname, 2 * i, ret);
- kfree(routes);
return -EINVAL;
}
ret = of_property_read_string_index(np, propname,
@@ -4206,7 +4207,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
dev_err(card->dev,
"ASoC: Property '%s' index %d could not be read: %d\n",
propname, (2 * i) + 1, ret);
- kfree(routes);
return -EINVAL;
}
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1d6a9b3..d6d9ba2 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -831,6 +831,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
if (path->weak)
continue;
+ if (path->walking)
+ return 1;
+
if (path->walked)
continue;
@@ -838,6 +841,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
if (path->sink && path->connect) {
path->walked = 1;
+ path->walking = 1;
/* do we need to add this widget to the list ? */
if (list) {
@@ -847,11 +851,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
dev_err(widget->dapm->dev,
"ASoC: could not add widget %s\n",
widget->name);
+ path->walking = 0;
return con;
}
}
con += is_connected_output_ep(path->sink, list);
+
+ path->walking = 0;
}
}
@@ -931,6 +938,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
if (path->weak)
continue;
+ if (path->walking)
+ return 1;
+
if (path->walked)
continue;
@@ -938,6 +948,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
if (path->source && path->connect) {
path->walked = 1;
+ path->walking = 1;
/* do we need to add this widget to the list ? */
if (list) {
@@ -947,11 +958,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
dev_err(widget->dapm->dev,
"ASoC: could not add widget %s\n",
widget->name);
+ path->walking = 0;
return con;
}
}
con += is_connected_input_ep(path->source, list);
+
+ path->walking = 0;
}
}
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index 9b76cc5..5e7aebe 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm)
static u64 spear_pcm_dmamask = DMA_BIT_MASK(32);
-static int spear_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
int ret;
if (!card->dev->dma_mask)
@@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card,
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
- if (dai->driver->playback.channels_min) {
- ret = spear_pcm_preallocate_dma_buffer(pcm,
+ if (rtd->cpu_dai->driver->playback.channels_min) {
+ ret = spear_pcm_preallocate_dma_buffer(rtd->pcm,
SNDRV_PCM_STREAM_PLAYBACK,
spear_pcm_hardware.buffer_bytes_max);
if (ret)
return ret;
}
- if (dai->driver->capture.channels_min) {
- ret = spear_pcm_preallocate_dma_buffer(pcm,
+ if (rtd->cpu_dai->driver->capture.channels_min) {
+ ret = spear_pcm_preallocate_dma_buffer(rtd->pcm,
SNDRV_PCM_STREAM_CAPTURE,
spear_pcm_hardware.buffer_bytes_max);
if (ret)
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 5e634a2..9e2703a 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -253,7 +253,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
{
struct usb_device *dev = chip->dev;
unsigned char data[4];
- int err, crate;
+ int err, cur_rate, prev_rate;
int clock = snd_usb_clock_find_source(chip, fmt->clock);
if (clock < 0)
@@ -266,6 +266,19 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
return -ENXIO;
}
+ err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+ USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+ UAC2_CS_CONTROL_SAM_FREQ << 8,
+ snd_usb_ctrl_intf(chip) | (clock << 8),
+ data, sizeof(data));
+ if (err < 0) {
+ snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
+ dev->devnum, iface, fmt->altsetting);
+ prev_rate = 0;
+ } else {
+ prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
+ }
+
data[0] = rate;
data[1] = rate >> 8;
data[2] = rate >> 16;
@@ -280,19 +293,31 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
return err;
}
- if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
- USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
- UAC2_CS_CONTROL_SAM_FREQ << 8,
- snd_usb_ctrl_intf(chip) | (clock << 8),
- data, sizeof(data))) < 0) {
+ err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+ USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+ UAC2_CS_CONTROL_SAM_FREQ << 8,
+ snd_usb_ctrl_intf(chip) | (clock << 8),
+ data, sizeof(data));
+ if (err < 0) {
snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
dev->devnum, iface, fmt->altsetting);
- return err;
+ cur_rate = 0;
+ } else {
+ cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
}
- crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
- if (crate != rate)
- snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
+ if (cur_rate != rate) {
+ snd_printd(KERN_WARNING
+ "current rate %d is different from the runtime rate %d\n",
+ cur_rate, rate);
+ }
+
+ /* Some devices doesn't respond to sample rate changes while the
+ * interface is active. */
+ if (rate != prev_rate) {
+ usb_set_interface(dev, iface, 0);
+ usb_set_interface(dev, iface, fmt->altsetting);
+ }
return 0;
}
--
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