December 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: HDA: Fix auto-mute on Lenovo Edge 14
by David Henningsson 15 Dec '10

15 Dec '10

The SKU value of this machine dictates that auto-mute should be disabled. Since the SKU value is similar to the PCI SSID, the most likely conclusion is that the SKU value should be ignored. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[PATCH] ALSA: HDA: Fix auto-mute on Lenovo Edge 14
by David Henningsson 15 Dec '10

15 Dec '10

BugLink: http://launchpad.net/bugs/690530 The SKU value of this machine dictates that auto-mute should be disabled. Since the SKU value is similar to the PCI SSID, the most likely conclusion is that the SKU value should be ignored. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cd2d3a5..3aeda3a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14808,6 +14808,7 @@ static int alc269_resume(struct hda_codec *codec) enum { ALC269_FIXUP_SONY_VAIO, ALC269_FIXUP_DELL_M101Z, + ALC269_FIXUP_LENOVO_EDGE14, }; static const struct alc_fixup alc269_fixups[] = { @@ -14825,11 +14826,15 @@ static const struct alc_fixup alc269_fixups[] = { {} } }, + [ALC269_FIXUP_LENOVO_EDGE14] = { + .sku = ALC_FIXUP_SKU_IGNORE, + }, }; static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_LENOVO_EDGE14), {} }; -- 1.7.1 --------------070009030306060104090905--

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Re: [alsa-devel] [RFC/PATCH v6 03/12] media: Entities, pads and links
by Clemens Ladisch 15 Dec '10

15 Dec '10

Laurent Pinchart wrote: > On Monday 13 December 2010 17:10:51 Clemens Ladisch wrote: >> TYPE_EXT describes entities that represent some interface to the >> external world, TYPE_INT those that are internal to the entire device. >> (I'm not sure if that distinction is very useful, but TYPE_SUBDEV seems >> to be an even more meaningless name.) > > SUBDEV comes from the V4L2 world, and I agree that it might not be a very good > name. > > I'm not sure I would split entities in internal/external categories. I would > create a category for connectors though. I'm not disagreeing, but what is actually the distinction between types and subtypes? ;-) >> * Entity properties >> >> There needs to be a mechanism to associate meta-information (properties) >> with entities. This information should be optional and extensible, but, >> when being handled inside the kernel, doesn't need to be more than >> a read-only blob. I think that something like ALSA's TLV format (used >> for mixer controls) can be used here. (I'm not mentioning the X-word >> here, except to note that the "M" stands for "markup".) > > I've been thinking of adding a new ioctl for that. It's something we need to > draft. The UVC driver will need it, and I'm pretty sure other V4L2 drivers > would find it useful as well. I'm imagining a "read-the-properties" ioctl that just returns the entity's blob. >> EXT_SPEAKER also includes headphones; there might be made a case for >> having those as a separate subtype. > > Shouldn't headphones be represented by an EXT_JACK_ANALOG ? Headphone jacks are jacks; there are also USB headphones. >> EXT_BROADCAST represents devices like TV tuners, satellite receivers, >> cable tuners, or radios. > > There's clearly an overlap with V4L here. These come from the USB audio spec. Video devices are indeed likely to be more detailed than just a single audio source. :) >> INT_CONTROLS may have multiple independent controls (this is USB's >> Feature Unit); INT_EFFECT may have multiple controls that affect one >> single algorithm. > > I'd describe this as a feature unit/processing unit then. I was aiming for more descriptive names, but I agree that the original names might be more useful. > Should we have an AUDIO category ? Probably not, because there are combined audio/video jacks, any maybe other entities. >> * Entity specifications >> >> While TYPE_DEVICE entities can be identified by their device node, other >> entities typcially have just a numeric ID. > > In V4L2 sub-devices have (or rather will have once the media controller > patches will be integrated) device nodes as well, so exposing that information > is required. USB and HDA entities already have numeric IDs. >> For that, it would be useful to make do without separate identification and >> let the driver choose the entity ID. > > How would drivers do that ? What if you have two instances of the same chip > (a video sensor, audio mixer, ...) on the same board ? Then those would get different IDs; USB descriptors always describe the entire device. Regards, Clemens

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Re: [alsa-devel] [PATCH] pxa2xx-ac97: Use platform device IRQ resource.
by Eric Miao 15 Dec '10

15 Dec '10

On Wed, Dec 15, 2010 at 1:19 AM, Lennert Buytenhek <buytenh(a)wantstofly.org> wrote: > Instead of using the IRQ_AC97 define, which forces a dependency on > platform include files, take the IRQ number to use from the specified > platform device's first IRQ resource. > > Signed-off-by: Lennert Buytenhek <buytenh(a)secretlab.ca> Acked-by: Eric Miao <eric.y.miao(a)gmail.com> > > diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c > index 88eec38..3921f4a 100644 > --- a/sound/arm/pxa2xx-ac97-lib.c > +++ b/sound/arm/pxa2xx-ac97-lib.c > @@ -20,7 +20,6 @@ > #include <sound/ac97_codec.h> > #include <sound/pxa2xx-lib.h> > > -#include <asm/irq.h> > #include <mach/regs-ac97.h> > #include <mach/audio.h> > > @@ -314,9 +313,17 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); > > int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) > { > + int irq; > int ret; > pxa2xx_audio_ops_t *pdata = dev->dev.platform_data; > > + irq = platform_get_irq(dev, 0); > + if (irq < 0) { > + dev_err(&dev->dev, "no IRQ resource defined\n"); > + ret = -EINVAL; > + goto err_conf; > + } > + > if (pdata) { > switch (pdata->reset_gpio) { > case 95: > @@ -359,7 +366,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) > if (ret) > goto err_clk2; > > - ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL); > + ret = request_irq(irq, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL); > if (ret < 0) > goto err_irq; > > @@ -383,7 +390,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_probe); > void pxa2xx_ac97_hw_remove(struct platform_device *dev) > { > GCR |= GCR_ACLINK_OFF; > - free_irq(IRQ_AC97, NULL); > + free_irq(platform_get_irq(dev, 0), NULL); > if (ac97conf_clk) { > clk_put(ac97conf_clk); > ac97conf_clk = NULL; >

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[alsa-devel] [PATCH v2 00/16] twl6040 updates
by Olaya, Margarita 14 Dec '10

14 Dec '10

This patch series contains several fixes and adds new features for twl6040 (phoenix) codec. v2: Use dev_info to report detection events Use individual variables for headset and handsfree controls Split patch according to the feature/fix done Francois Mazard (1): ASoC: twl6040: Fix analog Mic L & R mux controls Jorge Eduardo Candelaria (4): ASoC: twl6040: Add headset and handset mux controls ASoC: twl6040: Set default gains to minimun value ASoC: twl6040: Clear interrupt status at boot time ASoC: twl6040: Increase timeout for power up Margarita Olaya Cabrera (9): ASoC: twl6040: Update twl IO macro ASoC: twl6040: Modify the IRQ handler ASoC: twl6040: Support all DAIs ASoC: twl6040: Restore bias level at resume ASoC: twl6040: Fix PCM error handling ops ASoC: twl6040: Support other sample rates. ASoC: twl6040: Enable automatic power for phoenix 1.1 ASoC: twl6040: Add ramp up/down volume for HS and HF ASoC: twl6040: Fix TLV dB step values for gains Misael Lopez Cruz (2): ASoC: twl6040: Enable plug detection interrupts ASoC: twl6040: Use correct offset for LineInAmp Right sound/soc/codecs/twl6040.c | 821 +++++++++++++++++++++++++++++++++++++------- sound/soc/codecs/twl6040.h | 1 + 2 files changed, 700 insertions(+), 122 deletions(-)

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[alsa-devel] [PATCH v4 0/3] Add Jack support for
by Olaya, Margarita 14 Dec '10

14 Dec '10

This patch series adds the support for headset jack detection v4: Remove unused structure Use scheduling work to debounce the jack detection Add verification for sdp Jorge Eduardo Candelaria (3): ASoC: sdp4430: Enable FM stereo pins ASoC: twl6040: Add jack support for headset and handset ASoC: sdp4430: Add Jack support sound/soc/codecs/twl6040.c | 67 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 7 ++++ sound/soc/omap/sdp4430.c | 43 +++++++++++++++++++++++++-- 3 files changed, 113 insertions(+), 4 deletions(-)

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[alsa-devel] [PATCH] ASoC: TWL4030: Fix 24bit support
by Peter Ujfalusi 14 Dec '10

14 Dec '10

twl4030 series of codecs supports S32_LE with msbits=24. Replace the S24_LE with S32_LE format, and add constraint for 24msbit in case of 32 S32_LE format. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c173cf0..e4d464b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1724,6 +1724,7 @@ static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); + snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); if (twl4030->master_substream) { twl4030->slave_substream = substream; /* The DAI has one configuration for playback and capture, so @@ -1848,7 +1849,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: format |= TWL4030_DATA_WIDTH_16S_16W; break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: format |= TWL4030_DATA_WIDTH_32S_24W; break; default: @@ -2181,7 +2182,7 @@ static int twl4030_voice_set_tristate(struct snd_soc_dai *dai, int tristate) } #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) -#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) +#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops twl4030_dai_hifi_ops = { .startup = twl4030_startup, -- 1.7.3.3

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[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 14 Dec '10

14 Dec '10

The following changes since commit fdea0571ddca8e3f22448f66d72a034575abea28: ASoC: Fix merge errors with flush_scheduled_work() removal (2010-12-13 12:58:59 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Dimitris Papastamos (1): ASoC: soc-cache: A few minor stylistic changes Jarkko Nikula (2): ASoC: Fix bias power down of non-DAPM codec ASoC: Fix build error caused by merging a fix for 2.6.37 into 2.6.38 Mario Becroft (1): ASoC: Optimise WM9081 FLL performance Mark Brown (11): ASoC: Implement WM8994/58 DAC and ADC oversampling control ASoC: Implement WM8903 high pass filter support ASoC: Implement WM8903 oversampling rate controls ASoC: Remove open coded symmetry implementation from WM8903 ASoC: Automatically manage WM8903 deemphasis rate ASoC: Fix widgets for WM8994/58 AIF2 source control Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.38 Merge branch 'for-2.6.37' into for-2.6.38 ASoC: Fix AC'97 registration unwind regulator: Update LDO2 for WM8958 ASoC: Explicitly clear WM8993 ramp controls on power down Olaya, Margarita (1): ASoC: dapm: Add output driver widget Vasily Khoruzhick (1): ASoC: Add HP iPAQ H1940 support drivers/regulator/wm8994-regulator.c | 23 +++- include/sound/soc-dapm.h | 10 ++ sound/soc/codecs/wm8903.c | 182 +++++++++++---------- sound/soc/codecs/wm8993.c | 6 + sound/soc/codecs/wm8994.c | 21 ++- sound/soc/codecs/wm9081.c | 4 + sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/h1940_uda1380.c | 296 ++++++++++++++++++++++++++++++++++ sound/soc/soc-cache.c | 14 +- sound/soc/soc-core.c | 2 +- sound/soc/soc-dapm.c | 9 + 12 files changed, 477 insertions(+), 100 deletions(-) create mode 100644 sound/soc/samsung/h1940_uda1380.c

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[alsa-devel] ASoC update for 2.6.37
by Mark Brown 14 Dec '10

14 Dec '10

The following changes since commit a0968628097380be52db8b4664da98fc425546a5: ASoC: WM8580: Fix R8 initial value (2010-12-09 10:55:56 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.37 One small fix - only affects a very small set of users which are clearly broken without the fix, at worst they'll continue to be broken in the same way if there's a problem. Jarkko Nikula (1): ASoC: Fix bias power down of non-DAPM codec sound/soc/soc-dapm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-)

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[alsa-devel] [PATCH] ASoC: soc-cache: A few minor stylistic changes
by Dimitris Papastamos 14 Dec '10

14 Dec '10

Remove redundant parentheses/spaces in the use of the sizeof operator. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 14 +++++++------- 1 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 678fd75..0e17b40 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -85,7 +85,7 @@ static int snd_soc_4_12_spi_write(void *control_data, const char *data, msg[1] = data[0]; spi_message_init(&m); - memset(&t, 0, (sizeof t)); + memset(&t, 0, sizeof t); t.tx_buf = &msg[0]; t.len = len; @@ -166,7 +166,7 @@ static int snd_soc_7_9_spi_write(void *control_data, const char *data, msg[1] = data[1]; spi_message_init(&m); - memset(&t, 0, (sizeof t)); + memset(&t, 0, sizeof t); t.tx_buf = &msg[0]; t.len = len; @@ -246,7 +246,7 @@ static int snd_soc_8_8_spi_write(void *control_data, const char *data, msg[1] = data[1]; spi_message_init(&m); - memset(&t, 0, (sizeof t)); + memset(&t, 0, sizeof t); t.tx_buf = &msg[0]; t.len = len; @@ -326,7 +326,7 @@ static int snd_soc_8_16_spi_write(void *control_data, const char *data, msg[2] = data[2]; spi_message_init(&m); - memset(&t, 0, (sizeof t)); + memset(&t, 0, sizeof t); t.tx_buf = &msg[0]; t.len = len; @@ -513,7 +513,7 @@ static int snd_soc_16_8_spi_write(void *control_data, const char *data, msg[2] = data[2]; spi_message_init(&m); - memset(&t, 0, (sizeof t)); + memset(&t, 0, sizeof t); t.tx_buf = &msg[0]; t.len = len; @@ -633,7 +633,7 @@ static int snd_soc_16_16_spi_write(void *control_data, const char *data, msg[3] = data[3]; spi_message_init(&m); - memset(&t, 0, (sizeof t)); + memset(&t, 0, sizeof t); t.tx_buf = &msg[0]; t.len = len; @@ -1342,7 +1342,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) * that register. */ bmp_size = codec_drv->reg_cache_size; - sync_bmp = kmalloc(BITS_TO_LONGS(bmp_size) * sizeof (long), + sync_bmp = kmalloc(BITS_TO_LONGS(bmp_size) * sizeof(long), GFP_KERNEL); if (!sync_bmp) { ret = -ENOMEM; -- 1.7.3.3

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