* [GIT PULL] Sound fixes for 3.9-rc6
@ 2013-04-05 7:46 Takashi Iwai
2013-04-05 16:06 ` Linus Torvalds
0 siblings, 1 reply; 5+ messages in thread
From: Takashi Iwai @ 2013-04-05 7:46 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Mark Brown, Liam Girdwood, linux-kernel
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;
}
^ permalink raw reply related [flat|nested] 5+ messages in thread* Re: [GIT PULL] Sound fixes for 3.9-rc6
2013-04-05 7:46 [GIT PULL] Sound fixes for 3.9-rc6 Takashi Iwai
@ 2013-04-05 16:06 ` Linus Torvalds
2013-04-05 16:12 ` Takashi Iwai
2013-04-05 17:00 ` Mark Brown
0 siblings, 2 replies; 5+ messages in thread
From: Linus Torvalds @ 2013-04-05 16:06 UTC (permalink / raw)
To: Takashi Iwai; +Cc: Mark Brown, Liam Girdwood, Linux Kernel Mailing List
On Fri, Apr 5, 2013 at 12:46 AM, Takashi Iwai <tiwai@suse.de> wrote:
>
> please pull sound fixes for v3.9-rc6 from:
>
> git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus
Argh, Takashi, you're usually so reliable...
But you actually meant for me to pull the sound-3.9 tag, didn't you?
That "for-linus" branch isn't a signed tag..
Please double-check your scripts,
Linus
^ permalink raw reply [flat|nested] 5+ messages in thread* Re: [GIT PULL] Sound fixes for 3.9-rc6
2013-04-05 16:06 ` Linus Torvalds
@ 2013-04-05 16:12 ` Takashi Iwai
2013-04-05 17:00 ` Mark Brown
1 sibling, 0 replies; 5+ messages in thread
From: Takashi Iwai @ 2013-04-05 16:12 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Mark Brown, Liam Girdwood, Linux Kernel Mailing List
At Fri, 5 Apr 2013 09:06:43 -0700,
Linus Torvalds wrote:
>
> On Fri, Apr 5, 2013 at 12:46 AM, Takashi Iwai <tiwai@suse.de> wrote:
> >
> > please pull sound fixes for v3.9-rc6 from:
> >
> > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus
>
> Argh, Takashi, you're usually so reliable...
>
> But you actually meant for me to pull the sound-3.9 tag, didn't you?
> That "for-linus" branch isn't a signed tag..
Gah, sorry! Yes, please pull from tags/sound-3.9.
> Please double-check your scripts,
I fixed it once but I couldn't use it during migration to a new
workstation, which I ended up installing freshly.
Not the next time (tm)
Takashi
^ permalink raw reply [flat|nested] 5+ messages in thread
* Re: [GIT PULL] Sound fixes for 3.9-rc6
2013-04-05 16:06 ` Linus Torvalds
2013-04-05 16:12 ` Takashi Iwai
@ 2013-04-05 17:00 ` Mark Brown
1 sibling, 0 replies; 5+ messages in thread
From: Mark Brown @ 2013-04-05 17:00 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Takashi Iwai, Liam Girdwood, Linux Kernel Mailing List
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On Fri, Apr 05, 2013 at 09:06:43AM -0700, Linus Torvalds wrote:
> Argh, Takashi, you're usually so reliable...
> But you actually meant for me to pull the sound-3.9 tag, didn't you?
> That "for-linus" branch isn't a signed tag..
> Please double-check your scripts,
Probably needs a git upgrade - for a while git would silently substitute
a branch if it had the same head as a signed tag when generating pull
requests (which was very annoying if you'd had a public branch which you
were just getting round to signing). Current versions should get this
right and insist on the tag if that was what was supplied on the command
line.
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^ permalink raw reply [flat|nested] 5+ messages in thread
* [GIT PULL] Sound fixes for 3.9-rc6
@ 2013-04-12 14:34 Takashi Iwai
0 siblings, 0 replies; 5+ messages in thread
From: Takashi Iwai @ 2013-04-12 14:34 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Mark Brown, Liam Girdwood, linux-kernel
Linus,
please pull sound fixes for v3.9-rc7 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/sound-3.9
The topmost commit is 316d89e103c596a93c807fc84a35c08677730fb1
----------------------------------------------------------------
Sound fixes for 3.9-rc7
This contains a few small ASoC fixes (wm8903, wm5102, samsung-i2s,
tegra, and soc-compress) and an endian fix for NI USB-audio devices,
update for Mark's e-mail address.
No scary changes, AFAIS.
----------------------------------------------------------------
Alban Bedel (1):
ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is running
Charles Keepax (1):
ASoC: compress: Cancel delayed power down if needed
Eldad Zack (1):
ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*
Joonyoung Shim (1):
ASoC: core: Fix to check return value of snd_soc_update_bits_locked()
Lars-Peter Clausen (1):
ASoC: tegra: Don't claim to support PCM pause and resume
Mark Brown (2):
ASoC: wm5102: Correct lookup of arizona struct in SYSCLK event
MAINTAINERS: Update e-mail address
Prathyush K (2):
ASoC: Samsung: return error if drvdata is not set
ASoC: Samsung: set drvdata before adding secondary device
---
MAINTAINERS | 8 ++++----
sound/soc/codecs/wm5102.c | 2 +-
sound/soc/codecs/wm8903.c | 2 ++
sound/soc/samsung/i2s.c | 17 ++++++++++++-----
sound/soc/soc-compress.c | 14 +++++++++++---
sound/soc/soc-core.c | 2 +-
sound/soc/tegra/tegra_pcm.c | 24 +-----------------------
sound/usb/mixer_quirks.c | 4 ++--
sound/usb/quirks.c | 2 +-
9 files changed, 35 insertions(+), 40 deletions(-)
diff --git a/MAINTAINERS b/MAINTAINERS
index 61708c6..8bdd7a7 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -6631,7 +6631,7 @@ S: Supported
F: fs/reiserfs/
REGISTER MAP ABSTRACTION
-M: Mark Brown <broonie@opensource.wolfsonmicro.com>
+M: Mark Brown <broonie@kernel.org>
T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regmap.git
S: Supported
F: drivers/base/regmap/
@@ -7379,7 +7379,7 @@ F: sound/
SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)
M: Liam Girdwood <lgirdwood@gmail.com>
-M: Mark Brown <broonie@opensource.wolfsonmicro.com>
+M: Mark Brown <broonie@kernel.org>
T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
W: http://alsa-project.org/main/index.php/ASoC
@@ -7468,7 +7468,7 @@ F: drivers/clk/spear/
SPI SUBSYSTEM
M: Grant Likely <grant.likely@secretlab.ca>
-M: Mark Brown <broonie@opensource.wolfsonmicro.com>
+M: Mark Brown <broonie@kernel.org>
L: spi-devel-general@lists.sourceforge.net
Q: http://patchwork.kernel.org/project/spi-devel-general/list/
T: git git://git.secretlab.ca/git/linux-2.6.git
@@ -8713,7 +8713,7 @@ F: drivers/scsi/vmw_pvscsi.h
VOLTAGE AND CURRENT REGULATOR FRAMEWORK
M: Liam Girdwood <lrg@ti.com>
-M: Mark Brown <broonie@opensource.wolfsonmicro.com>
+M: Mark Brown <broonie@kernel.org>
W: http://opensource.wolfsonmicro.com/node/15
W: http://www.slimlogic.co.uk/?p=48
T: git git://git.kernel.org/pub/scm/linux/kernel/git/lrg/regulator.git
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index b82bbf5..34d0201 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -584,7 +584,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
- struct arizona *arizona = dev_get_drvdata(codec->dev);
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
struct regmap *regmap = codec->control_data;
const struct reg_default *patch = NULL;
int i, patch_size;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 134e41c..f8a31ad 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1083,6 +1083,8 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = {
{ "ROP", NULL, "Right Speaker PGA" },
{ "RON", NULL, "Right Speaker PGA" },
+ { "Charge Pump", NULL, "CLK_DSP" },
+
{ "Left Headphone Output PGA", NULL, "Charge Pump" },
{ "Right Headphone Output PGA", NULL, "Charge Pump" },
{ "Left Line Output PGA", NULL, "Charge Pump" },
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index d7231e3..6bbeb0b 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -972,6 +972,7 @@ static const struct snd_soc_dai_ops samsung_i2s_dai_ops = {
static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec)
{
struct i2s_dai *i2s;
+ int ret;
i2s = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dai), GFP_KERNEL);
if (i2s == NULL)
@@ -996,15 +997,17 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec)
i2s->i2s_dai_drv.capture.channels_max = 2;
i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES;
i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS;
+ dev_set_drvdata(&i2s->pdev->dev, i2s);
} else { /* Create a new platform_device for Secondary */
- i2s->pdev = platform_device_register_resndata(NULL,
- "samsung-i2s-sec", -1, NULL, 0, NULL, 0);
+ i2s->pdev = platform_device_alloc("samsung-i2s-sec", -1);
if (IS_ERR(i2s->pdev))
return NULL;
- }
- /* Pre-assign snd_soc_dai_set_drvdata */
- dev_set_drvdata(&i2s->pdev->dev, i2s);
+ platform_set_drvdata(i2s->pdev, i2s);
+ ret = platform_device_add(i2s->pdev);
+ if (ret < 0)
+ return NULL;
+ }
return i2s;
}
@@ -1107,6 +1110,10 @@ static int samsung_i2s_probe(struct platform_device *pdev)
if (samsung_dai_type == TYPE_SEC) {
sec_dai = dev_get_drvdata(&pdev->dev);
+ if (!sec_dai) {
+ dev_err(&pdev->dev, "Unable to get drvdata\n");
+ return -EFAULT;
+ }
snd_soc_register_dai(&sec_dai->pdev->dev,
&sec_dai->i2s_dai_drv);
asoc_dma_platform_register(&pdev->dev);
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index b5b3db7..ed0bfb0 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -211,19 +211,27 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) {
ret = platform->driver->compr_ops->set_params(cstream, params);
if (ret < 0)
- goto out;
+ goto err;
}
if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) {
ret = rtd->dai_link->compr_ops->set_params(cstream);
if (ret < 0)
- goto out;
+ goto err;
}
snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
SND_SOC_DAPM_STREAM_START);
-out:
+ /* cancel any delayed stream shutdown that is pending */
+ rtd->pop_wait = 0;
+ mutex_unlock(&rtd->pcm_mutex);
+
+ cancel_delayed_work_sync(&rtd->delayed_work);
+
+ return ret;
+
+err:
mutex_unlock(&rtd->pcm_mutex);
return ret;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 507d251..ff4b45a5 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2963,7 +2963,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
val = val << shift;
ret = snd_soc_update_bits_locked(codec, reg, val_mask, val);
- if (ret != 0)
+ if (ret < 0)
return ret;
if (snd_soc_volsw_is_stereo(mc)) {
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index c925ab0..5e2c55c 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -43,8 +43,6 @@
static const struct snd_pcm_hardware tegra_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_INTERLEAVED,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.channels_min = 2,
@@ -127,26 +125,6 @@ static int tegra_pcm_hw_free(struct snd_pcm_substream *substream)
return 0;
}
-static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- return snd_dmaengine_pcm_trigger(substream,
- SNDRV_PCM_TRIGGER_START);
-
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- return snd_dmaengine_pcm_trigger(substream,
- SNDRV_PCM_TRIGGER_STOP);
- default:
- return -EINVAL;
- }
- return 0;
-}
-
static int tegra_pcm_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
{
@@ -164,7 +142,7 @@ static struct snd_pcm_ops tegra_pcm_ops = {
.ioctl = snd_pcm_lib_ioctl,
.hw_params = tegra_pcm_hw_params,
.hw_free = tegra_pcm_hw_free,
- .trigger = tegra_pcm_trigger,
+ .trigger = snd_dmaengine_pcm_trigger,
.pointer = snd_dmaengine_pcm_pointer,
.mmap = tegra_pcm_mmap,
};
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 497d274..ebe9144 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -509,7 +509,7 @@ static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol,
else
ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest,
USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN,
- 0, cpu_to_le16(wIndex),
+ 0, wIndex,
&tmp, sizeof(tmp), 1000);
up_read(&mixer->chip->shutdown_rwsem);
@@ -540,7 +540,7 @@ static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol,
else
ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), bRequest,
USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT,
- cpu_to_le16(wValue), cpu_to_le16(wIndex),
+ wValue, wIndex,
NULL, 0, 1000);
up_read(&mixer->chip->shutdown_rwsem);
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 5325a38..9c5ab22 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -486,7 +486,7 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev)
{
int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0),
0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE,
- cpu_to_le16(1), 0, NULL, 0, 1000);
+ 1, 0, NULL, 0, 1000);
if (ret < 0)
return ret;
^ permalink raw reply related [flat|nested] 5+ messages in thread
end of thread, other threads:[~2013-04-12 14:34 UTC | newest]
Thread overview: 5+ messages (download: mbox.gz follow: Atom feed
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2013-04-05 7:46 [GIT PULL] Sound fixes for 3.9-rc6 Takashi Iwai
2013-04-05 16:06 ` Linus Torvalds
2013-04-05 16:12 ` Takashi Iwai
2013-04-05 17:00 ` Mark Brown
-- strict thread matches above, loose matches on Subject: below --
2013-04-12 14:34 Takashi Iwai
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