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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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[alsa-devel] XMOS Board Follow Up
by George Ioakimedes 14 Dec '10

14 Dec '10

I saw in the archives from around May some talk of getting this board to enumerate properly. I wonder if anyone can share their experience with this and if they were able to get it working. Thanks, George

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[alsa-devel] regarding setting of fragment size in usbaudio driver
by Amit Nagal 14 Dec '10

14 Dec '10

Hi , i am using a OSS based usbaudio application for usb audio streaming . my usb audio device is capture only device . i want to set the fragment size of usbaudio alsa driver to reduce latency problem coming in my application . currently even if my application reads 4k data from userspace , i observed driver will buffer for 100 ms data ( period size data )= 16 k at 44100 frquency and after driver reads 16k data , read call will return with 4k data . so read() call remains blocked for 100ms which is causing some latency problem . i used SNDCTL_DSP_SETFRAGMENT ioctl , but still driver behaviour to buffer 100ms data remained same . how driver's fragment size in driver can be set to some less value in OSS based application . Can some patch be implemented in kernel space to do that .? Thanx & Regards Amit Nagal

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[alsa-devel] Calculated average playback latency
by Jean-Yves Avenard 14 Dec '10

14 Dec '10

Hi there. I am involved in maintaining the mythtv audio code. Today one of our user complained that he couldn't get any audio out , even though everything was properly configured on his side. By default, Myth uses mmap access ; unless we found out that the audio device doesn't allow mmap (e.g. snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_MMAP_INTERLEAVED) errored) Problem, turned out his was using a USB audio device, and snd_pcm_hw_params_set_access didn't return any error. It's not the first time we're having such issues, so I decided to stop using mmap and instead use snd_pcm_writei. Now on my system, this has added quite a bit of latency and the A/V sync isn't perfect anymore... Is there an official way to calculate an average system latency (doesn't need to be perfectly accurate). I couldn't find obvious references on how to perform such task. I find some references in program like mplayer or jack using snd_pcm_status_get_trigger_tstamp Would anyone be kind enough to point me to some documentation or explain to me how I could calculate the average latency between the time you start playback and the time sounds actually comes out. Thank you in advance Jean-Yves

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[alsa-devel] [PATCH] ASoC: Explicitly clear WM8993 ramp controls on power down
by Mark Brown 14 Dec '10

14 Dec '10

This helps ensure that the ramp logic is reset when powering back up. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 77b1d44..15f34a2 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1030,6 +1030,12 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, WM8993_VMID_SEL_MASK | WM8993_BIAS_ENA, 0); + snd_soc_update_bits(codec, WM8993_ANTIPOP2, + WM8993_STARTUP_BIAS_ENA | + WM8993_VMID_BUF_ENA | + WM8993_VMID_RAMP_MASK | + WM8993_BIAS_SRC, 0); + #ifdef CONFIG_REGULATOR /* Post 2.6.34 we will be able to get a callback when * the regulators are disabled which we can use but -- 1.7.1

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Re: [alsa-devel] [RFC/PATCH v6 03/12] media: Entities, pads and links
by Laurent Pinchart 14 Dec '10

14 Dec '10

Hi Hans, On Tuesday 14 December 2010 15:51:08 Hans Verkuil wrote: > > 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? ;-) > > The type tells what the behavior is of an entity. E.g., type DEVNODE > represents device node(s) in userspace, V4L2_SUBDEV represents a v4l2 > sub-device, etc. The subtype tells whether a V4L2_SUBDEV is a sensor or a > receiver or whatever. Nice to know, but it doesn't change the way > sub-devices work. > > In the case of connectors you would create a CONNECTOR type and have a > bunch of subtypes for all the variations of connectors. > > That said, I'm not sure whether the distinction is useful for DEVNODEs. > You do need to know the subtype in order to interpret the union correctly. > > Laurent, does the MC code test against the DEVNODE type? I.e., does the MC > code ignore the subtype of a DEVNODE, or does it always use it? The MC code uses the DEVNODE type, ignoring the subtype, for power management. When a device node is opened all entities in the chain need to be powered up. -- Regards, Laurent Pinchart

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[alsa-devel] [PATCH v2] ASoC: dapm: Add output driver widget
by Olaya, Margarita 14 Dec '10

14 Dec '10

In some cases it was not possible to follow the appropiate power ON/OFF sequence like in cases where the PGA needs to be enabled before the driver and disabled before the PGA for pop reduction. Add a widget to support output driver (speaker, haptic, vibra, etc) drivers where power ON/OFF ordering is important. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- include/sound/soc-dapm.h | 10 ++++++++++ sound/soc/soc-dapm.c | 6 ++++++ 2 files changed, 16 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 041e98b..2f76a51 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -71,6 +71,10 @@ wcontrols, wncontrols) \ { .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols} +#define SND_SOC_DAPM_OUT_DRV(wname, wreg, wshift, winvert,\ + wcontrols, wncontrols) \ +{ .id = snd_soc_dapm_out_drv, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols} #define SND_SOC_DAPM_MIXER(wname, wreg, wshift, winvert, \ wcontrols, wncontrols)\ { .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ @@ -115,6 +119,11 @@ { .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ .event = wevent, .event_flags = wflags} +#define SND_SOC_DAPM_OUT_DRV_E(wname, wreg, wshift, winvert, wcontrols, \ + wncontrols, wevent, wflags) \ +{ .id = snd_soc_dapm_out_drv, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ + .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_MIXER_E(wname, wreg, wshift, winvert, wcontrols, \ wncontrols, wevent, wflags) \ { .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ @@ -368,6 +377,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_mixer, /* mixes several analog signals together */ snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */ snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */ + snd_soc_dapm_out_drv, /* output driver */ snd_soc_dapm_adc, /* analog to digital converter */ snd_soc_dapm_dac, /* digital to analog converter */ snd_soc_dapm_micbias, /* microphone bias (power) */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6a29d59..31e9681 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -62,6 +62,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_mixer_named_ctl] = 7, [snd_soc_dapm_pga] = 8, [snd_soc_dapm_adc] = 9, + [snd_soc_dapm_out_drv] = 10, [snd_soc_dapm_hp] = 10, [snd_soc_dapm_spk] = 10, [snd_soc_dapm_post] = 11, @@ -72,6 +73,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_adc] = 1, [snd_soc_dapm_hp] = 2, [snd_soc_dapm_spk] = 2, + [snd_soc_dapm_out_drv] = 2, [snd_soc_dapm_pga] = 4, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, @@ -231,6 +233,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, break; /* does not effect routing - always connected */ case snd_soc_dapm_pga: + case snd_soc_dapm_out_drv: case snd_soc_dapm_output: case snd_soc_dapm_adc: case snd_soc_dapm_input: @@ -1243,6 +1246,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_dac: case snd_soc_dapm_adc: case snd_soc_dapm_pga: + case snd_soc_dapm_out_drv: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: @@ -1421,6 +1425,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_adc: case snd_soc_dapm_dac: case snd_soc_dapm_pga: + case snd_soc_dapm_out_drv: case snd_soc_dapm_input: case snd_soc_dapm_output: case snd_soc_dapm_micbias: @@ -1538,6 +1543,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) w->power_check = dapm_dac_check_power; break; case snd_soc_dapm_pga: + case snd_soc_dapm_out_drv: w->power_check = dapm_generic_check_power; dapm_new_pga(dapm, w); break; -- 1.7.0.4

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Re: [alsa-devel] [PATCH 5/5] sound: ml403-ac97cr: Use vsprintf extension %pR for struct resource
by Takashi Iwai 14 Dec '10

14 Dec '10

At Mon, 13 Dec 2010 13:42:22 -0800, Joe Perches wrote: > > Signed-off-by: Joe Perches <joe(a)perches.com> Applied now. Thanks. Takashi > --- > sound/drivers/ml403-ac97cr.c | 4 ++-- > 1 files changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c > index a1282c1..5cfcb90 100644 > --- a/sound/drivers/ml403-ac97cr.c > +++ b/sound/drivers/ml403-ac97cr.c > @@ -1143,8 +1143,8 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, > (resource->start) + 1); > if (ml403_ac97cr->port == NULL) { > snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " > - "unable to remap memory region (%x to %x)\n", > - resource->start, resource->end); > + "unable to remap memory region (%pR)\n", > + resource); > snd_ml403_ac97cr_free(ml403_ac97cr); > return -EBUSY; > } > -- > 1.7.3.3.398.g0b0cd.dirty >

1 0

[alsa-devel] [PATCH] ASoC: Fix AC'97 registration unwind
by Mark Brown 13 Dec '10

13 Dec '10

soc_unregister_ac97_dai_link() takes a CODEC as an argument, not a rtd like the registration function, so give it what it's looking for. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 68edc69..1dc4b11 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1830,7 +1830,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) { printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); while (--i >= 0) - soc_unregister_ac97_dai_link(&card->rtd[i]); + soc_unregister_ac97_dai_link(card->rtd[i].codec); goto probe_aux_dev_err; } } -- 1.7.1

2 1

[alsa-devel] [PATCH] regulator: Update LDO2 for WM8958
by Mark Brown 13 Dec '10

13 Dec '10

LDO2 has a slightly different range of supported voltages on WM8958 so update the selector<->voltage mappings to match. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- As the WM8958 support is currently sitting in the ASoC tree for the next merge window this should get merged that way. I've done some test merges here and git seems to be able to figure out the overlaps with the code in your tree so are you OK with applying to ASoC? drivers/regulator/wm8994-regulator.c | 23 +++++++++++++++++++++-- 1 files changed, 21 insertions(+), 2 deletions(-) diff --git a/drivers/regulator/wm8994-regulator.c b/drivers/regulator/wm8994-regulator.c index 03713bc..21fefef 100644 --- a/drivers/regulator/wm8994-regulator.c +++ b/drivers/regulator/wm8994-regulator.c @@ -131,10 +131,19 @@ static struct regulator_ops wm8994_ldo1_ops = { static int wm8994_ldo2_list_voltage(struct regulator_dev *rdev, unsigned int selector) { + struct wm8994_ldo *ldo = rdev_get_drvdata(rdev); + if (selector > WM8994_LDO2_MAX_SELECTOR) return -EINVAL; - return (selector * 100000) + 900000; + switch (ldo->wm8994->type) { + case WM8994: + return (selector * 100000) + 900000; + case WM8958: + return (selector * 100000) + 1000000; + default: + return -EINVAL; + } } static int wm8994_ldo2_get_voltage(struct regulator_dev *rdev) @@ -157,7 +166,17 @@ static int wm8994_ldo2_set_voltage(struct regulator_dev *rdev, struct wm8994_ldo *ldo = rdev_get_drvdata(rdev); int selector, v; - selector = (min_uV - 900000) / 100000; + switch (ldo->wm8994->type) { + case WM8994: + selector = (min_uV - 900000) / 100000; + break; + case WM8958: + selector = (min_uV - 1000000) / 100000; + break; + default: + return -EINVAL; + } + v = wm8994_ldo2_list_voltage(rdev, selector); if (v < 0 || v > max_uV) return -EINVAL; -- 1.7.1

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[alsa-devel] [RFC 2.6.38] ASoC: Fix bias power down of non-DAPM codec
by Jarkko Nikula 13 Dec '10

13 Dec '10

Currently bias of non-DAPM codec will be powered down (standby/off) whenever there is a stream stop. This is wrong in simultaneous playback/capture since the bias is put down immediately after stopping the first stream. Fix this by using the codec->active count when figuring out the needed bias level after stream stop. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- RFC since does this look a valid fix and how to deal with 2.6.37? I think patching the .37 first will result in bisect build errors in mainline when later merging upcoming .38 changes. Would it be better to patch .37 after it's released? --- sound/soc/soc-dapm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9af2d8a..b521a13 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -981,6 +981,9 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) case SND_SOC_DAPM_STREAM_RESUME: sys_power = 1; break; + case SND_SOC_DAPM_STREAM_STOP: + sys_power = !!dapm->codec->active; + break; case SND_SOC_DAPM_STREAM_SUSPEND: sys_power = 0; break; -- 1.7.2.3

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Re: [alsa-devel] [RFC/PATCH v6 03/12] media: Entities, pads and links
by Clemens Ladisch 13 Dec '10

13 Dec '10

Laurent Pinchart wrote: > A link is a point-to-point oriented connection between two pads, either > on the same entity or on different entities. Data flows from a source > pad to a sink pad. > > Links are stored in the source entity. In the descriptors of USB Audio and HDAudio devices, the links are stored only in the sink. But AFAICS this doesn't matter in the media driver API. > +Links have flags that describe the link capabilities and state. > + > + MEDIA_LINK_FLAG_ACTIVE indicates that the link is active and can be > + used to transfer media data. When two or more links target a sink pad, > + only one of them can be active at a time. > + MEDIA_LINK_FLAG_IMMUTABLE indicates that the link active state can't > + be modified at runtime. If MEDIA_LINK_FLAG_IMMUTABLE is set, then > + MEDIA_LINK_FLAG_ACTIVE must also be set since an immutable link is > + always active. So the intended userspace API for controlling routing is to change the ACTIVE flag of links? In USB and HD audio devices, all links are immutable, and the routing is controlled by 'selector' entities that activate exactly one of their input pads. In userspace, this entity shows up as a mixer control. I guess it would be possible to map the ACTIVE flag onto these controls. Alternatively, entities can have 'mute' mixer controls associated with their pads. In this case, multiple unmuted inputs would be mixed together. > +#define MEDIA_ENTITY_TYPE_MASK 0x00ff0000 > +#define MEDIA_ENTITY_SUBTYPE_MASK 0x0000ffff > + > +#define MEDIA_ENTITY_TYPE_NODE (1 << MEDIA_ENTITY_TYPE_SHIFT) > ... > +#define MEDIA_ENTITY_TYPE_NODE_ALSA (MEDIA_ENTITY_TYPE_NODE + 3) ALSA has PCM and MIDI devices, and several types of mixer controls. (It also has hardware dependent and timer devices, but I don't think these would need topology information.) So we need at least these: MEDIA_ENTITY_TYPE_NODE_ALSA_PCM MEDIA_ENTITY_TYPE_NODE_ALSA_MIDI MEDIA_ENTITY_TYPE_SUBDEV_ALSA_CONTROL Furthermore, topology information is also needed for entities not associated with a mixer control, such as microphones, speakers, jacks/ connectors, and effect units. These entities are defined in the USB and HD audio specifications, but are not yet handled by ALSA. > +struct media_entity { > ... > + union { > + /* Node specifications */ > + struct { > + u32 major; > + u32 minor; > + } v4l; > + struct { > + u32 major; > + u32 minor; > + } fb; > + int alsa; > + int dvb; > + > + /* Sub-device specifications */ > + /* Nothing needed yet */ > + }; ALSA devices are not addressed by their device node but with card/device/ subdevice numbers; mixer controls have numeric IDs, unique per card: struct { int card; int device; int subdevice; } alsa_device; struct { int card; int numid; } alsa_control; Regards, Clemens

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[alsa-devel] Joining the development
by Ramon Fried 13 Dec '10

13 Dec '10

Hello, First, I'm sorry if I posted it in the wrong place. I would like to participate in the development of ALSA drivers. If there is any project that need assitance I would be happy to contribute. Please notify me. Warm regards, Ramon Fried.

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[alsa-devel] low volume with hda base card (see attached alsa-info output
by Halim Sahin 13 Dec '10

13 Dec '10

Hi, I need to set all mixer stuff to 100% and the output isn't ok. Volume is too low. When I switch to another input on my amp. it's about 500% louder. Please have a look to attached ala-info output. I tested with opensuse 11.2 and also with a newer kernel for the distro 2.6.34. Can you please help? Thx. Halim -- Halim Sahin E-Mail: halim.sahin (at) t-online.de

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[alsa-devel] [PATCH] ASoC: Fix widgets for WM8994/58 AIF2 source control
by Mark Brown 13 Dec '10

13 Dec '10

The compiler really ought to have been warning about unreferenced variables... Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index da48802..b599616 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -780,13 +780,13 @@ SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME, SOC_ENUM("AIF1ADCL Source", aif1adcl_src), SOC_ENUM("AIF1ADCR Source", aif1adcr_src), -SOC_ENUM("AIF2ADCL Source", aif1adcl_src), -SOC_ENUM("AIF2ADCR Source", aif1adcr_src), +SOC_ENUM("AIF2ADCL Source", aif2adcl_src), +SOC_ENUM("AIF2ADCR Source", aif2adcr_src), SOC_ENUM("AIF1DACL Source", aif1dacl_src), SOC_ENUM("AIF1DACR Source", aif1dacr_src), -SOC_ENUM("AIF2DACL Source", aif1dacl_src), -SOC_ENUM("AIF2DACR Source", aif1dacr_src), +SOC_ENUM("AIF2DACL Source", aif2dacl_src), +SOC_ENUM("AIF2DACR Source", aif2dacr_src), SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), -- 1.7.1

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Re: [alsa-devel] [Alsa-user] regarding setting of fragment size in usbaudio driver
by Bill Unruh 13 Dec '10

13 Dec '10

On Mon, 13 Dec 2010, Amit Nagal wrote: > Hi , > > Thanx for the reply . > > Actually on my embedded target , i am not using arecord , but reading > from (/dev/dsp0) capture data . He asked you if arecord works, not whether you are using it. Why are you using the OSS emulation in alsa, rather than using the native alsa? That adds an extra layer and more code. > > i have 2 question : > > i used SNDCTL_DSP_GETBLKSIZE ioctl to find the fragment size of driver > . i found it to be 16k . > > 1) now if the application reads 4k data from userspace , read call > will remain blocked for audio driver's fragment size usb data read ( > 16k) > or app_request_size(4k) ? So read in all 16K into your buffer. But you do not say how your application reads that 4k in? Why do you not publish the fragment of you code that you use so people can see what y ou do. As I understand OSS, the driver reads the usb and puts the output into a buffer. YOur call to the oss system, then reads from that buffer and delivers it to you. That should not block, unless there is nothing in the buffer (ie you forgot to tell the driver to start reading the sound card). > > 2)time taken for read(4k) = time_taken(fragment size read ) or > time_taken(app_request_size=4k) ? > > thanx & regards > amit nagal > > > On Mon, Dec 13, 2010 at 6:04 PM, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> Amit Nagal wrote: >>> i am using a OSS based usbaudio application for usb audio streaming . >>> >>> my usb audio device is capture only device . >>> >>> i want to set the fragment size of usbaudio alsa driver to reduce >>> latency problem coming in my application . >>> >>> currently even if my application reads 4k data from userspace , i >>> observed driver will buffer for 100 ms data ( period size data )= 16 k >>> at 44100 frquency >>> >>> and after driver reads 16k data , read call will return with 4k data >>> . so read() call remains blocked for 100ms which is causing some >>> latency problem . >> >> There is no extra buffer in the driver. >> There seems to be something wrong with your algorithm. >> >> Does recording with the arecord tool work? >> >> >> Regards, >> Clemens >> > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > Alsa-user mailing list > Alsa-user(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-user > -- William G. Unruh | Canadian Institute for| Tel: +1(604)822-3273 Physics&Astronomy | Advanced Research | Fax: +1(604)822-5324 UBC, Vancouver,BC | Program in Cosmology | unruh(a)physics.ubc.ca Canada V6T 1Z1 | and Gravity | www.theory.physics.ubc.ca/

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Re: [alsa-devel] 10 Sec sleep in wait_for_avail_min function
by Clemens Ladisch 13 Dec '10

13 Dec '10

Nitin Arora wrote: > I need to stream audio from my audio device to the host through USB. > While audio is being streamed and songs are being played on the host > and during that I plug out the audio device, Disconnect event takes > too long to get reported and displayed on the screen (on GUI). > > I dug this problem a little deeper and found that when my application is > continuous invoking read on the audio device to capture and audio data > it follows this path: > > .... .. snd_pcm_lib_read -> snd_pcm_lib_read1 -> snd_pcm_capture_avail > > and in the case when there is no data left (MY DISCONNECT CASE) > it invokes as follows: > > .... .. snd_pcm_lib_read -> snd_pcm_lib_read1 -> wait_for_avail_min > > and in wait_for_avail_min it sleeps for 10 sec. This ten-second timeout is intended for devices where the interrupt does not work for some reason. In the case of USB disconnection, the driver should stop the stream immediately. Does this happen if you try to record with the arecord tool? Regards, Clemens -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] Custom IOCTL call
by Radivoje Jovanovic 13 Dec '10

13 Dec '10

I am writing custom IOCTL calls for my card. I have made hwdep interface and set ioctl ops call for it. But, now the question is which device I need to open in my user space app to talk to correct ioctl? I have tried /dev/snd/controlC0 and /dev/snd/pcmC0D0p Both of theses returned invalid IOCTL. Cheers Ogi

3 2

[alsa-devel] Unable to set ALSA buffer size
by Irfan Shaikh 13 Dec '10

13 Dec '10

Hello All, I am trying to set my alsa buffer size as 1024*8. But my playback device is not able to set to requested ALSA buffer size. snd_pcm_hw_params_set_periods(handle, params, 8, 0); snd_pcm_hw_params_set_period_size(handle, params, 1024, 0); snd_pcm_uframes_t frames1; frames1=1024*8; retVal=snd_pcm_hw_params_set_buffer_size_near(handle, params, &frames1); snd_pcm_hw_params_get_buffer_size(params, (snd_pcm_uframes_t *) &val); /*Expected Output*/ ==============>>>>>>> buffer size = 8192 frames /*Output*/ ==============>>>>>>> buffer size = 1325 frames Return value for above APIs is coming as zero. but still i am unable to get expected result. Is there any other forcible way to do the above. Regards, SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

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[alsa-devel] Gigabyte 880GM-USB3 MIDI and Dolby 5.1/7.1 issues
by Jonathan Gardner 13 Dec '10

13 Dec '10

I'm a developer who spends most of his time in Python writing web apps but I know my way around C. I've been using Linux for over 10 years now and I thought it's time to get serious about making Linux work for the hardware I use. I'm self-employed now, so I don't have to deal with any legal departments and can help out to my heart's desire. Disclaimer: I intend to learn how to help debug these issues and hack on some fairly low-level pieces of software and get a clear sense of how hardware really works. Mostly this is a "scratch-my-own-itch" thing, but it's always bothered me that I've never truly understood what happens below the KDE interface. I recently bought a Gigabyte 880GM-USB3 motherboard. http://www.gigabyte.us/products/product-page.aspx?pid=3412#sp I think "Realtek ALC892" describes the hardware for audio. I'm running Fedora 13 with KDE which seems to have alsa 1.0.23 running. I don't know much about hardware or audio, but I do know that it's not working 100% for me with this motherboard. There are two issues that stand out. First, with my older motherboard, I could control the various output and input levels of the channels. With this motherboard, I only have one channel "Internal Audio Analog Stereo". This board is supposed to support up to Dolby 7.1 surround sound. I want to make this computer my hub for all my devices. Obviously this isn't possible. Second, I never really got MIDI working, at least not that I recall in my 10 years of Linux. I know MIDI is possible, I just don't know what it takes. I've been playing with musescore which apparently plays the notes over MIDI so you can hear what you wrote, which would be nice. RTFM is certainly acceptable advice, if you can show me which FM is preferred. I'll also write some documentation if it helps. -- Jonathan Gardner jgardner(a)jonathangardner.net

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[alsa-devel] "Unknown symbol" with alsa snapshot (tiwai)
by Simos Xenitellis 12 Dec '10

12 Dec '10

Hi, I am trying out the latest snapshot of Alsa with Ubuntu 10.10. It compiles without errors. However, when the modules are to be loaded, there are Unknown symbol errors: [ 332.243113] snd: Unknown symbol unregister_sound_special (err 0) [ 332.243379] snd: Unknown symbol register_sound_special_device (err 0) [ 332.245467] snd_timer: Unknown symbol snd_info_register (err 0) [ 332.245631] snd_timer: Unknown symbol snd_info_create_module_entry (err 0) [ 332.245839] snd_timer: Unknown symbol snd_info_free_entry (err 0) [ 332.246210] snd_timer: Unknown symbol __snd_printk (err 0) [ 332.246373] snd_timer: Unknown symbol snd_iprintf (err 0) [ 332.246607] snd_timer: Unknown symbol snd_ecards_limit (err 0) [ 332.246834] snd_timer: Unknown symbol snd_oss_info_register (err 0) [ 332.246997] snd_timer: Unknown symbol snd_unregister_device (err 0) [ 332.247224] snd_timer: Unknown symbol snd_device_new (err 0) [ 332.247625] snd_timer: Unknown symbol snd_register_device_for_dev (err 0) Is this a module loading order issue? Has there been a change in the latest kernel that causes the problem? Simos

2 1

[alsa-devel] [PATCH v2 04/16] ASoC: twl6040: Set default gains to minimun value
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Updated default values to improve power consumption. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b575fd3..27d6d53 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -86,19 +86,19 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x1B, /* TWL6040_LINEGAIN 0x0F */ 0x00, /* TWL6040_HSLCTL 0x10 */ 0x00, /* TWL6040_HSRCTL 0x11 */ - 0x00, /* TWL6040_HSGAIN 0x12 */ - 0x00, /* TWL6040_EARCTL 0x13 */ + 0xFF, /* TWL6040_HSGAIN 0x12 */ + 0x1E, /* TWL6040_EARCTL 0x13 */ 0x00, /* TWL6040_HFLCTL 0x14 */ - 0x00, /* TWL6040_HFLGAIN 0x15 */ + 0x1D, /* TWL6040_HFLGAIN 0x15 */ 0x00, /* TWL6040_HFRCTL 0x16 */ - 0x00, /* TWL6040_HFRGAIN 0x17 */ + 0x1D, /* TWL6040_HFRGAIN 0x17 */ 0x00, /* TWL6040_VIBCTLL 0x18 */ 0x00, /* TWL6040_VIBDATL 0x19 */ 0x00, /* TWL6040_VIBCTLR 0x1A */ 0x00, /* TWL6040_VIBDATR 0x1B */ 0x00, /* TWL6040_HKCTL1 0x1C */ 0x00, /* TWL6040_HKCTL2 0x1D */ - 0x00, /* TWL6040_GPOCTL 0x1E */ + 0x02, /* TWL6040_GPOCTL 0x1E */ 0x00, /* TWL6040_ALB 0x1F */ 0x00, /* TWL6040_DLB 0x20 */ 0x00, /* not used 0x21 */ -- 1.7.0.4

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[alsa-devel] [PATCH v2 03/16] ASoC: twl6040: Add headset and handset mux controls
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> This patch adds support for the twl6040 headset and handset MUX controls. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 100 ++++++++++++++++++++++++++++++++------------ 1 files changed, 73 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 8a6c623..b575fd3 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -476,6 +476,12 @@ static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); /* + * AFMGAIN volume control: + * from 18 to 24 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(afm_amp_tlv, 1800, 600, 0); + +/* * HSGAIN volume control: * from -30 to 0 dB in 2 dB steps */ @@ -506,6 +512,28 @@ static const struct soc_enum twl6040_enum[] = { SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts), }; +static const char *twl6040_hs_texts[] = { + "Off", "HS DAC", "Line-In amp" +}; + +static const struct soc_enum twl6040_hs_enum[] = { + SOC_ENUM_SINGLE(TWL6040_REG_HSLCTL, 5, ARRAY_SIZE(twl6040_hs_texts), + twl6040_hs_texts), + SOC_ENUM_SINGLE(TWL6040_REG_HSRCTL, 5, ARRAY_SIZE(twl6040_hs_texts), + twl6040_hs_texts), +}; + +static const char *twl6040_hf_texts[] = { + "Off", "HF DAC", "Line-In amp" +}; + +static const struct soc_enum twl6040_hf_enum[] = { + SOC_ENUM_SINGLE(TWL6040_REG_HFLCTL, 2, ARRAY_SIZE(twl6040_hf_texts), + twl6040_hf_texts), + SOC_ENUM_SINGLE(TWL6040_REG_HFRCTL, 2, ARRAY_SIZE(twl6040_hf_texts), + twl6040_hf_texts), +}; + static const struct snd_kcontrol_new amicl_control = SOC_DAPM_ENUM("Route", twl6040_enum[0]); @@ -513,18 +541,18 @@ static const struct snd_kcontrol_new amicr_control = SOC_DAPM_ENUM("Route", twl6040_enum[1]); /* Headset DAC playback switches */ -static const struct snd_kcontrol_new hsdacl_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 5, 1, 0); +static const struct snd_kcontrol_new hsl_mux_controls = + SOC_DAPM_ENUM("Route", twl6040_hs_enum[0]); -static const struct snd_kcontrol_new hsdacr_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 5, 1, 0); +static const struct snd_kcontrol_new hsr_mux_controls = + SOC_DAPM_ENUM("Route", twl6040_hs_enum[1]); /* Handsfree DAC playback switches */ -static const struct snd_kcontrol_new hfdacl_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 2, 1, 0); +static const struct snd_kcontrol_new hfl_mux_controls = + SOC_DAPM_ENUM("Route", twl6040_hf_enum[0]); -static const struct snd_kcontrol_new hfdacr_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); +static const struct snd_kcontrol_new hfr_mux_controls = + SOC_DAPM_ENUM("Route", twl6040_hf_enum[1]); static const struct snd_kcontrol_new ep_driver_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0); @@ -536,6 +564,10 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { SOC_DOUBLE_TLV("Capture Volume", TWL6040_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + /* AFM gains */ + SOC_DOUBLE_TLV("Aux FM Volume", + TWL6040_REG_LINEGAIN, 0, 5, 0xF, 0, afm_amp_tlv), + /* Playback gains */ SOC_DOUBLE_TLV("Headset Playback Volume", TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), @@ -572,6 +604,12 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_PGA("MicAmpR", TWL6040_REG_MICRCTL, 0, 0, NULL, 0), + /* Auxiliary FM PGAs */ + SND_SOC_DAPM_PGA("AFMAmpL", + TWL6040_REG_MICLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("AFMAmpR", + TWL6040_REG_MICRCTL, 1, 0, NULL, 0), + /* ADCs */ SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", TWL6040_REG_MICLCTL, 2, 0), @@ -606,15 +644,15 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - /* Analog playback switches */ - SND_SOC_DAPM_SWITCH("HSDAC Left Playback", - SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), - SND_SOC_DAPM_SWITCH("HSDAC Right Playback", - SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), - SND_SOC_DAPM_SWITCH("HFDAC Left Playback", - SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), - SND_SOC_DAPM_SWITCH("HFDAC Right Playback", - SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + SND_SOC_DAPM_MUX("HF Left Playback", + SND_SOC_NOPM, 0, 0, &hfl_mux_controls), + SND_SOC_DAPM_MUX("HF Right Playback", + SND_SOC_NOPM, 0, 0, &hfr_mux_controls), + /* Analog playback Muxes */ + SND_SOC_DAPM_MUX("HS Left Playback", + SND_SOC_NOPM, 0, 0, &hsl_mux_controls), + SND_SOC_DAPM_MUX("HS Right Playback", + SND_SOC_NOPM, 0, 0, &hsr_mux_controls), /* Analog playback drivers */ SND_SOC_DAPM_PGA_E("Handsfree Left Driver", @@ -658,12 +696,18 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Left", NULL, "MicAmpL"}, {"ADC Right", NULL, "MicAmpR"}, - /* Headset playback path */ - {"HSDAC Left Playback", "Switch", "HSDAC Left"}, - {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + /* AFM path */ + {"AFMAmpL", "NULL", "AFML"}, + {"AFMAmpR", "NULL", "AFMR"}, + + {"HS Left Playback", "HS DAC", "HSDAC Left"}, + {"HS Left Playback", "Line-In amp", "AFMAmpL"}, - {"Headset Left Driver", NULL, "HSDAC Left Playback"}, - {"Headset Right Driver", NULL, "HSDAC Right Playback"}, + {"HS Right Playback", "HS DAC", "HSDAC Right"}, + {"HS Right Playback", "Line-In amp", "AFMAmpR"}, + + {"Headset Left Driver", "NULL", "HS Left Playback"}, + {"Headset Right Driver", "NULL", "HS Right Playback"}, {"HSOL", NULL, "Headset Left Driver"}, {"HSOR", NULL, "Headset Right Driver"}, @@ -672,12 +716,14 @@ static const struct snd_soc_dapm_route intercon[] = { {"Earphone Driver", "Switch", "HSDAC Left"}, {"EP", NULL, "Earphone Driver"}, - /* Handsfree playback path */ - {"HFDAC Left Playback", "Switch", "HFDAC Left"}, - {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + {"HF Left Playback", "HF DAC", "HFDAC Left"}, + {"HF Left Playback", "Line-In amp", "AFMAmpL"}, + + {"HF Right Playback", "HF DAC", "HFDAC Right"}, + {"HF Right Playback", "Line-In amp", "AFMAmpR"}, - {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, - {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + {"HFDAC Left PGA", NULL, "HF Left Playback"}, + {"HFDAC Right PGA", NULL, "HF Right Playback"}, {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, -- 1.7.0.4

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[alsa-devel] [PATCH v4 2/3] ASoC: twl6040: Add jack support for headset and handset
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> This patch adds support for reporting twl6040 headset and handset jack events. The machine driver retrieves and report the status through twl6040_hs_jack_detect. A workq is used to debounce of the irq. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 67 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 7 ++++ 2 files changed, 74 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b92f2b7..5d7e2f7 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -42,6 +42,11 @@ #define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) +struct twl6040_jack_data { + struct snd_soc_jack *jack; + int report; +}; + /* codec private data */ struct twl6040_data { int audpwron; @@ -52,6 +57,11 @@ struct twl6040_data { unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; struct completion ready; + struct twl6040_jack_data hs_jack; + struct snd_soc_codec *codec; + struct workqueue_struct *workqueue; + struct delayed_work delayed_work; + struct mutex mutex; }; /* @@ -381,6 +391,47 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, return 0; } +void twl6040_hs_jack_report(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int report) +{ + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + int status; + + mutex_lock(&priv->mutex); + + /* Sync status */ + status = twl6040_read_reg_volatile(codec, TWL6040_REG_STATUS); + if (status & TWL6040_PLUGCOMP) + snd_soc_jack_report(jack, report, report); + else + snd_soc_jack_report(jack, 0, report); + + mutex_unlock(&priv->mutex); +} + +void twl6040_hs_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int report) +{ + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_jack_data *hs_jack = &priv->hs_jack; + + hs_jack->jack = jack; + hs_jack->report = report; + + twl6040_hs_jack_report(codec, hs_jack->jack, hs_jack->report); +} +EXPORT_SYMBOL_GPL(twl6040_hs_jack_detect); + +static void twl6040_accessory_work(struct work_struct *work) +{ + struct twl6040_data *priv = container_of(work, + struct twl6040_data, delayed_work.work); + struct snd_soc_codec *codec = priv->codec; + struct twl6040_jack_data *hs_jack = &priv->hs_jack; + + twl6040_hs_jack_report(codec, hs_jack->jack, hs_jack->report); +} + /* audio interrupt handler */ static irqreturn_t twl6040_naudint_handler(int irq, void *data) { @@ -396,6 +447,9 @@ static irqreturn_t twl6040_naudint_handler(int irq, void *data) break; case TWL6040_PLUGINT: case TWL6040_UNPLUGINT: + queue_delayed_work(priv->workqueue, &priv->delayed_work, + msecs_to_jiffies(200)); + break; case TWL6040_HOOKINT: break; case TWL6040_HFINT: @@ -1023,6 +1077,8 @@ static int twl6040_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, priv); + priv->codec = codec; + if (twl_codec) { audpwron = twl_codec->audpwron_gpio; naudint = twl_codec->naudint_irq; @@ -1033,6 +1089,14 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->audpwron = audpwron; priv->naudint = naudint; + priv->workqueue = create_singlethread_workqueue("twl6040-codec"); + + if (!priv->workqueue) + goto work_err; + + INIT_DELAYED_WORK(&priv->delayed_work, twl6040_accessory_work); + + mutex_init(&priv->mutex); init_completion(&priv->ready); @@ -1089,6 +1153,8 @@ gpio2_err: if (gpio_is_valid(audpwron)) gpio_free(audpwron); gpio1_err: + destroy_workqueue(priv->workqueue); +work_err: kfree(priv); return ret; } @@ -1107,6 +1173,7 @@ static int twl6040_remove(struct snd_soc_codec *codec) if (naudint) free_irq(naudint, codec); + destroy_workqueue(priv->workqueue); kfree(priv); return 0; diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h index f7c77fa..67396f6 100644 --- a/sound/soc/codecs/twl6040.h +++ b/sound/soc/codecs/twl6040.h @@ -135,4 +135,11 @@ #define TWL6040_HPPLL_ID 1 #define TWL6040_LPPLL_ID 2 +/* STATUS (0x2E) fields */ + +#define TWL6040_PLUGCOMP 0x02 + +void twl6040_hs_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int report); + #endif /* End of __TWL6040_H__ */ -- 1.7.0.4

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[alsa-devel] [PATCH v2 02/16] ASoC: twl6040: Modify the IRQ handler
by Olaya, Margarita 11 Dec '10

11 Dec '10

Multiples interrupts can be received. The irq handler is modified to attend all of them. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 30 ++++++++++++------------------ 1 files changed, 12 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index d33d2b4..8a6c623 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -441,30 +441,24 @@ static irqreturn_t twl6040_naudint_handler(int irq, void *data) twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); - switch (intid) { - case TWL6040_THINT: + if (intid & TWL6040_THINT) dev_alert(codec->dev, "die temp over-limit detection\n"); - break; - case TWL6040_PLUGINT: - case TWL6040_UNPLUGINT: + + if ((intid & TWL6040_PLUGINT) || (intid & TWL6040_UNPLUGINT)) queue_delayed_work(priv->workqueue, &priv->delayed_work, msecs_to_jiffies(200)); - break; - case TWL6040_HOOKINT: - break; - case TWL6040_HFINT: + + if (intid & TWL6040_HOOKINT) + dev_info(codec->dev, "hook detection\n"); + + if (intid & TWL6040_HFINT) dev_alert(codec->dev, "hf drivers over current detection\n"); - break; - case TWL6040_VIBINT: + + if (intid & TWL6040_VIBINT) dev_alert(codec->dev, "vib drivers over current detection\n"); - break; - case TWL6040_READYINT: + + if (intid & TWL6040_READYINT) complete(&priv->ready); - break; - default: - dev_err(codec->dev, "unknown audio interrupt %d\n", intid); - break; - } return IRQ_HANDLED; } -- 1.7.0.4

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[alsa-devel] [PATCH v2 16/16] ASoC: twl6040: Fix TLV dB step values for gains
by Olaya, Margarita 11 Dec '10

11 Dec '10

Some gains were incorrectly configured for dB values. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 6e0839a..315ce19 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -954,9 +954,9 @@ static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); /* * MICGAIN volume control: - * from 6 to 30 dB in 6 dB steps + * from -6 to 30 dB in 6 dB steps */ -static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -600, 600, 0); /* * AFMGAIN volume control: -- 1.7.0.4

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[alsa-devel] [PATCH v2 15/16] ASoC: twl6040: Use correct offset for LineInAmp Right
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Misael Lopez Cruz <misael.lopez(a)ti.com> Gain for LineInAmp Right uses LINEGAIN[5:3], which means that offset for right channel should be 4. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index d5af1d6..6e0839a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1049,7 +1049,7 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* AFM gains */ SOC_DOUBLE_TLV("Aux FM Volume", - TWL6040_REG_LINEGAIN, 0, 5, 0xF, 0, afm_amp_tlv), + TWL6040_REG_LINEGAIN, 0, 4, 0xF, 0, afm_amp_tlv), /* Playback gains */ SOC_TWL6040_DOUBLE_TLV("Headset Playback Volume", -- 1.7.0.4

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[alsa-devel] [PATCH v2 14/16] ASoC: twl6040: Increase timeout for power up
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> After coming back from suspend, the timeout waiting for Phoenix chip to complete its power up sequence is not enough, which leaves the codec cache value for some registers in an outdated state. Increase the timeout value to wait for the power up sequence to correclty complete. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 71d345c..d5af1d6 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1239,7 +1239,7 @@ static int twl6040_power_up_completion(struct snd_soc_codec *codec, u8 intid; time_left = wait_for_completion_timeout(&priv->ready, - msecs_to_jiffies(48)); + msecs_to_jiffies(144)); if (!time_left) { twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &intid, -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 13/16] ASoC: twl6040: Enable plug detection interrupts
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Misael Lopez Cruz <misael.lopez(a)ti.com> Enable plug detection interrupt mask in order to get headset PLUGINT/UNPLUGINT interrupts. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 10 +++------- sound/soc/codecs/twl6040.h | 1 + 2 files changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index cd11a3d..71d345c 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1623,7 +1623,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) struct twl6040_data *priv; int audpwron, naudint; int ret = 0; - u8 icrev; + u8 icrev, intmr = TWL6040_ALLINT_MSK; priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); if (priv == NULL) @@ -1671,16 +1671,12 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->codec_powered = 0; /* enable only codec ready interrupt */ - twl6040_write(codec, TWL6040_REG_INTMR, - ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); + intmr &= ~(TWL6040_READYMSK | TWL6040_PLUGMSK); /* reset interrupt status to allow correct power up sequence */ twl6040_read_reg_volatile(codec, TWL6040_REG_INTID); - } else { - /* no interrupts at all */ - twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, - TWL6040_ALLINT_MSK); } + twl6040_write(codec, TWL6040_REG_INTMR, intmr); if (naudint) { /* audio interrupt */ diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h index 67396f6..23aeed0 100644 --- a/sound/soc/codecs/twl6040.h +++ b/sound/soc/codecs/twl6040.h @@ -79,6 +79,7 @@ /* INTMR (0x04) fields */ +#define TWL6040_PLUGMSK 0x02 #define TWL6040_READYMSK 0x40 #define TWL6040_ALLINT_MSK 0x7B -- 1.7.0.4

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[alsa-devel] [PATCH v2 11/16] ASoC: twl6040: Add ramp up/down volume for HS and HF
by Olaya, Margarita 11 Dec '10

11 Dec '10

Add ramp functions for the headset and handsfree outputs in order to reduce the pops during power on/off sequences. In order to give more control to volume ramp, step size and delay between steps can be specified. The patches are based on wm8350 implementation from Liam Girdwood. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 533 +++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 521 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 46e8e93..eaab21d 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -40,7 +40,35 @@ #include "twl6040.h" #define TWL6040_RATES SNDRV_PCM_RATE_8000_96000 -#define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) +#define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +#define TWL6040_OUTHS_0dB 0x00 +#define TWL6040_OUTHS_M30dB 0x0F +#define TWL6040_OUTHF_0dB 0x03 +#define TWL6040_OUTHF_M52dB 0x1D + +#define TWL6040_RAMP_NONE 0 +#define TWL6040_RAMP_UP 1 +#define TWL6040_RAMP_DOWN 2 + +#define TWL6040_HSL_VOL_MASK 0x0F +#define TWL6040_HSL_VOL_SHIFT 0 +#define TWL6040_HSR_VOL_MASK 0xF0 +#define TWL6040_HSR_VOL_SHIFT 4 +#define TWL6040_HF_VOL_MASK 0x1F +#define TWL6040_HF_VOL_SHIFT 0 + +struct twl6040_output { + u16 active; + u16 left_vol; + u16 right_vol; + u16 left_step; + u16 right_step; + unsigned int step_delay; + u16 ramp; + u16 mute; + struct completion ramp_done; +}; struct twl6040_jack_data { struct snd_soc_jack *jack; @@ -62,6 +90,12 @@ struct twl6040_data { struct workqueue_struct *workqueue; struct delayed_work delayed_work; struct mutex mutex; + struct twl6040_output headset; + struct twl6040_output handsfree; + struct workqueue_struct *hf_workqueue; + struct workqueue_struct *hs_workqueue; + struct delayed_work hs_delayed_work; + struct delayed_work hf_delayed_work; }; /* @@ -263,6 +297,305 @@ static void twl6040_init_vdd_regs(struct snd_soc_codec *codec) } } +/* + * Ramp HS PGA volume to minimise pops at stream startup and shutdown. + */ +static inline int twl6040_hs_ramp_step(struct snd_soc_codec *codec, + unsigned int left_step, unsigned int right_step) +{ + + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *headset = &priv->headset; + int left_complete = 0, right_complete = 0; + u8 reg, val; + + /* left channel */ + left_step = (left_step > 0xF) ? 0xF : left_step; + reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN); + val = (~reg & TWL6040_HSL_VOL_MASK); + + if (headset->ramp == TWL6040_RAMP_UP) { + /* ramp step up */ + if (val < headset->left_vol) { + val += left_step; + reg &= ~TWL6040_HSL_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HSGAIN, + (reg | (~val & TWL6040_HSL_VOL_MASK))); + } else { + left_complete = 1; + } + } else if (headset->ramp == TWL6040_RAMP_DOWN) { + /* ramp step down */ + if (val > 0x0) { + val -= left_step; + reg &= ~TWL6040_HSL_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HSGAIN, reg | + (~val & TWL6040_HSL_VOL_MASK)); + } else { + left_complete = 1; + } + } + + /* right channel */ + right_step = (right_step > 0xF) ? 0xF : right_step; + reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN); + val = (~reg & TWL6040_HSR_VOL_MASK) >> TWL6040_HSR_VOL_SHIFT; + + if (headset->ramp == TWL6040_RAMP_UP) { + /* ramp step up */ + if (val < headset->right_vol) { + val += right_step; + reg &= ~TWL6040_HSR_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HSGAIN, + (reg | (~val << TWL6040_HSR_VOL_SHIFT))); + } else { + right_complete = 1; + } + } else if (headset->ramp == TWL6040_RAMP_DOWN) { + /* ramp step down */ + if (val > 0x0) { + val -= right_step; + reg &= ~TWL6040_HSR_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HSGAIN, + reg | (~val << TWL6040_HSR_VOL_SHIFT)); + } else { + right_complete = 1; + } + } + + return left_complete & right_complete; +} + +/* + * Ramp HF PGA volume to minimise pops at stream startup and shutdown. + */ +static inline int twl6040_hf_ramp_step(struct snd_soc_codec *codec, + unsigned int left_step, unsigned int right_step) +{ + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *handsfree = &priv->handsfree; + int left_complete = 0, right_complete = 0; + u16 reg, val; + + /* left channel */ + left_step = (left_step > 0x1D) ? 0x1D : left_step; + reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFLGAIN); + reg = 0x1D - reg; + val = (reg & TWL6040_HF_VOL_MASK); + if (handsfree->ramp == TWL6040_RAMP_UP) { + /* ramp step up */ + if (val < handsfree->left_vol) { + val += left_step; + reg &= ~TWL6040_HF_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HFLGAIN, + reg | (0x1D - val)); + } else { + left_complete = 1; + } + } else if (handsfree->ramp == TWL6040_RAMP_DOWN) { + /* ramp step down */ + if (val > 0) { + val -= left_step; + reg &= ~TWL6040_HF_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HFLGAIN, + reg | (0x1D - val)); + } else { + left_complete = 1; + } + } + + /* right channel */ + right_step = (right_step > 0x1D) ? 0x1D : right_step; + reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFRGAIN); + reg = 0x1D - reg; + val = (reg & TWL6040_HF_VOL_MASK); + if (handsfree->ramp == TWL6040_RAMP_UP) { + /* ramp step up */ + if (val < handsfree->right_vol) { + val += right_step; + reg &= ~TWL6040_HF_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HFRGAIN, + reg | (0x1D - val)); + } else { + right_complete = 1; + } + } else if (handsfree->ramp == TWL6040_RAMP_DOWN) { + /* ramp step down */ + if (val > 0) { + val -= right_step; + reg &= ~TWL6040_HF_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HFRGAIN, + reg | (0x1D - val)); + } + } + + return left_complete & right_complete; +} + +/* + * This work ramps both output PGAs at stream start/stop time to + * minimise pop associated with DAPM power switching. + */ +static void twl6040_pga_hs_work(struct work_struct *work) +{ + struct twl6040_data *priv = + container_of(work, struct twl6040_data, hs_delayed_work.work); + struct snd_soc_codec *codec = priv->codec; + struct twl6040_output *headset = &priv->headset; + unsigned int delay = headset->step_delay; + int i, headset_complete; + + /* do we need to ramp at all ? */ + if (headset->ramp == TWL6040_RAMP_NONE) + return; + + /* HS PGA volumes have 4 bits of resolution to ramp */ + for (i = 0; i <= 16; i++) { + headset_complete = 1; + if (headset->ramp != TWL6040_RAMP_NONE) + headset_complete = twl6040_hs_ramp_step(codec, + headset->left_step, + headset->right_step); + + /* ramp finished ? */ + if (headset_complete) + break; + + /* + * TODO: tune: delay is longer over 0dB + * as increases are larger. + */ + if (i >= 8) + schedule_timeout_interruptible(msecs_to_jiffies(delay + + (delay >> 1))); + else + schedule_timeout_interruptible(msecs_to_jiffies(delay)); + } + + if (headset->ramp == TWL6040_RAMP_DOWN) { + headset->active = 0; + complete(&headset->ramp_done); + } else { + headset->active = 1; + } + headset->ramp = TWL6040_RAMP_NONE; +} + +static void twl6040_pga_hf_work(struct work_struct *work) +{ + struct twl6040_data *priv = + container_of(work, struct twl6040_data, hf_delayed_work.work); + struct snd_soc_codec *codec = priv->codec; + struct twl6040_output *handsfree = &priv->handsfree; + unsigned int delay = handsfree->step_delay; + int i, handsfree_complete; + + /* do we need to ramp at all ? */ + if (handsfree->ramp == TWL6040_RAMP_NONE) + return; + + /* HF PGA volumes have 5 bits of resolution to ramp */ + for (i = 0; i <= 32; i++) { + handsfree_complete = 1; + if (handsfree->ramp != TWL6040_RAMP_NONE) + handsfree_complete = twl6040_hf_ramp_step(codec, + handsfree->left_step, + handsfree->right_step); + + /* ramp finished ? */ + if (handsfree_complete) + break; + + /* + * TODO: tune: delay is longer over 0dB + * as increases are larger. + */ + if (i >= 16) + schedule_timeout_interruptible(msecs_to_jiffies(delay + + (delay >> 1))); + else + schedule_timeout_interruptible(msecs_to_jiffies(delay)); + } + + + if (handsfree->ramp == TWL6040_RAMP_DOWN) { + handsfree->active = 0; + complete(&handsfree->ramp_done); + } else + handsfree->active = 1; + handsfree->ramp = TWL6040_RAMP_NONE; +} + +static int pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *out; + struct delayed_work *work; + struct workqueue_struct *queue; + + switch (w->shift) { + case 2: + case 3: + out = &priv->headset; + work = &priv->hs_delayed_work; + queue = priv->hs_workqueue; + out->step_delay = 5; /* 5 ms between volume ramp steps */ + break; + case 4: + out = &priv->handsfree; + work = &priv->hf_delayed_work; + queue = priv->hf_workqueue; + out->step_delay = 5; /* 5 ms between volume ramp steps */ + if (SND_SOC_DAPM_EVENT_ON(event)) + priv->non_lp++; + else + priv->non_lp--; + break; + default: + return -1; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (out->active) + break; + + /* don't use volume ramp for power-up */ + out->left_step = out->left_vol; + out->right_step = out->right_vol; + + if (!delayed_work_pending(work)) { + out->ramp = TWL6040_RAMP_UP; + queue_delayed_work(queue, work, + msecs_to_jiffies(1)); + } + break; + + case SND_SOC_DAPM_PRE_PMD: + if (!out->active) + break; + + if (!delayed_work_pending(work)) { + /* use volume ramp for power-down */ + out->left_step = 1; + out->right_step = 1; + out->ramp = TWL6040_RAMP_DOWN; + INIT_COMPLETION(out->ramp_done); + + queue_delayed_work(queue, work, + msecs_to_jiffies(1)); + + wait_for_completion_timeout(&out->ramp_done, + msecs_to_jiffies(2000)); + } + break; + } + + return 0; +} + /* twl6040 codec manual power-up sequence */ static void twl6040_power_up(struct snd_soc_codec *codec) { @@ -463,6 +796,156 @@ static irqreturn_t twl6040_naudint_handler(int irq, void *data) return IRQ_HANDLED; } +static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *out = NULL; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int ret; + unsigned int reg = mc->reg; + + /* For HS and HF we shadow the values and only actually write + * them out when active in order to ensure the amplifier comes on + * as quietly as possible. */ + switch (reg) { + case TWL6040_REG_HSGAIN: + out = &twl6040_priv->headset; + break; + default: + break; + } + + if (out) { + out->left_vol = ucontrol->value.integer.value[0]; + out->right_vol = ucontrol->value.integer.value[1]; + if (!out->active) + return 1; + } + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + return 1; +} + +static int twl6040_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *out = &twl6040_priv->headset; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + + switch (reg) { + case TWL6040_REG_HSGAIN: + out = &twl6040_priv->headset; + ucontrol->value.integer.value[0] = out->left_vol; + ucontrol->value.integer.value[1] = out->right_vol; + return 0; + + default: + break; + } + + return snd_soc_get_volsw(kcontrol, ucontrol); +} + +static int twl6040_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *out = NULL; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int ret; + unsigned int reg = mc->reg; + + /* For HS and HF we shadow the values and only actually write + * them out when active in order to ensure the amplifier comes on + * as quietly as possible. */ + switch (reg) { + case TWL6040_REG_HFLGAIN: + case TWL6040_REG_HFRGAIN: + out = &twl6040_priv->handsfree; + break; + default: + break; + } + + if (out) { + out->left_vol = ucontrol->value.integer.value[0]; + out->right_vol = ucontrol->value.integer.value[1]; + if (!out->active) + return 1; + } + + ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + if (ret < 0) + return ret; + + return 1; +} + +static int twl6040_get_volsw_2r(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *out = &twl6040_priv->handsfree; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + + /* If these are cached registers use the cache */ + switch (reg) { + case TWL6040_REG_HFLGAIN: + case TWL6040_REG_HFRGAIN: + out = &twl6040_priv->handsfree; + ucontrol->value.integer.value[0] = out->left_vol; + ucontrol->value.integer.value[1] = out->right_vol; + return 0; + + default: + break; + } + + return snd_soc_get_volsw_2r(kcontrol, ucontrol); +} + +/* double control with volume update */ +#define SOC_TWL6040_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax,\ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, .get = twl6040_get_volsw, \ + .put = twl6040_put_volsw, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right,\ + .max = xmax, .platform_max = xmax, .invert = xinvert} } + +/* double control with volume update */ +#define SOC_TWL6040_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = twl6040_get_volsw_2r, .put = twl6040_put_volsw_2r_vu, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .rshift = xshift, .max = xmax, .invert = xinvert}, } + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -569,9 +1052,9 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { TWL6040_REG_LINEGAIN, 0, 5, 0xF, 0, afm_amp_tlv), /* Playback gains */ - SOC_DOUBLE_TLV("Headset Playback Volume", + SOC_TWL6040_DOUBLE_TLV("Headset Playback Volume", TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), - SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + SOC_TWL6040_DOUBLE_R_TLV("Handsfree Playback Volume", TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), SOC_SINGLE_TLV("Earphone Playback Volume", TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv), @@ -657,16 +1140,20 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { /* Analog playback drivers */ SND_SOC_DAPM_PGA_E("Handsfree Left Driver", TWL6040_REG_HFLCTL, 4, 0, NULL, 0, - twl6040_power_mode_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Handsfree Right Driver", TWL6040_REG_HFRCTL, 4, 0, NULL, 0, - twl6040_power_mode_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_PGA("Headset Left Driver", - TWL6040_REG_HSLCTL, 2, 0, NULL, 0), - SND_SOC_DAPM_PGA("Headset Right Driver", - TWL6040_REG_HSRCTL, 2, 0, NULL, 0), + pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Headset Left Driver", + TWL6040_REG_HSLCTL, 2, 0, NULL, 0, + pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Headset Right Driver", + TWL6040_REG_HSRCTL, 2, 0, NULL, 0, + pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SWITCH_E("Earphone Driver", SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls, twl6040_power_mode_event, @@ -1169,6 +1656,8 @@ static int twl6040_probe(struct snd_soc_codec *codec) mutex_init(&priv->mutex); init_completion(&priv->ready); + init_completion(&priv->headset.ramp_done); + init_completion(&priv->handsfree.ramp_done); if (gpio_is_valid(audpwron)) { ret = gpio_request(audpwron, "audpwron"); @@ -1205,10 +1694,24 @@ static int twl6040_probe(struct snd_soc_codec *codec) /* init vio registers */ twl6040_init_vio_regs(codec); + priv->hf_workqueue = create_singlethread_workqueue("twl6040-hf"); + if (priv->hf_workqueue == NULL) { + ret = -ENOMEM; + goto irq_err; + } + priv->hs_workqueue = create_singlethread_workqueue("twl6040-hs"); + if (priv->hs_workqueue == NULL) { + ret = -ENOMEM; + goto wq_err; + } + + INIT_DELAYED_WORK(&priv->hs_delayed_work, twl6040_pga_hs_work); + INIT_DELAYED_WORK(&priv->hf_delayed_work, twl6040_pga_hf_work); + /* power on device */ ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret) - goto irq_err; + goto bias_err; snd_soc_add_controls(codec, twl6040_snd_controls, ARRAY_SIZE(twl6040_snd_controls)); @@ -1216,6 +1719,10 @@ static int twl6040_probe(struct snd_soc_codec *codec) return 0; +bias_err: + destroy_workqueue(priv->hs_workqueue); +wq_err: + destroy_workqueue(priv->hf_workqueue); irq_err: if (naudint) free_irq(naudint, codec); @@ -1244,6 +1751,8 @@ static int twl6040_remove(struct snd_soc_codec *codec) free_irq(naudint, codec); destroy_workqueue(priv->workqueue); + destroy_workqueue(priv->hf_workqueue); + destroy_workqueue(priv->hs_workqueue); kfree(priv); return 0; -- 1.7.0.4

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[alsa-devel] [PATCH v2 12/16] ASoC: twl6040: Clear interrupt status at boot time
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> On Phoenix 1.1, the INTID register default value is an invalid one, causing the interrupt handler to think the phoenix power on sequence is ready before it actually finishes. This causes some i2c errors when trying to configure twl. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 21 +++++++++++---------- 1 files changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index eaab21d..cd11a3d 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1669,6 +1669,17 @@ static int twl6040_probe(struct snd_soc_codec *codec) goto gpio2_err; priv->codec_powered = 0; + + /* enable only codec ready interrupt */ + twl6040_write(codec, TWL6040_REG_INTMR, + ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); + + /* reset interrupt status to allow correct power up sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_INTID); + } else { + /* no interrupts at all */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + TWL6040_ALLINT_MSK); } if (naudint) { @@ -1679,16 +1690,6 @@ static int twl6040_probe(struct snd_soc_codec *codec) "twl6040_codec", codec); if (ret) goto gpio2_err; - } else { - if (gpio_is_valid(audpwron)) { - /* enable only codec ready interrupt */ - twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, - ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); - } else { - /* no interrupts at all */ - twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, - TWL6040_ALLINT_MSK); - } } /* init vio registers */ -- 1.7.0.4

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[alsa-devel] [PATCH v2 10/16] ASoC: twl6040: Enable automatic power for phoenix 1.1
by Olaya, Margarita 11 Dec '10

11 Dec '10

Phoenix 1.1 supports automatic power on sequence, a verification is added to use it with new revision of the chip. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 13 +++++++++---- 1 files changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 27fc1b9..46e8e93 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1136,6 +1136,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) struct twl6040_data *priv; int audpwron, naudint; int ret = 0; + u8 icrev; priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); if (priv == NULL) @@ -1144,13 +1145,17 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->codec = codec; - if (twl_codec) { + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &icrev, TWL6040_REG_ASICREV); + + if (twl_codec && (icrev > 0)) audpwron = twl_codec->audpwron_gpio; - naudint = twl_codec->naudint_irq; - } else { + else audpwron = -EINVAL; + + if (twl_codec) + naudint = twl_codec->naudint_irq; + else naudint = 0; - } priv->audpwron = audpwron; priv->naudint = naudint; -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 09/16] ASoC: twl6040: Fix analog Mic L & R mux controls
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Francois Mazard <f-mazard(a)ti.com> The mux control has 4 elements not 3 Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 9776715..27fc1b9 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -508,8 +508,8 @@ static const char *twl6040_amicr_texts[] = {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; static const struct soc_enum twl6040_enum[] = { - SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 3, twl6040_amicl_texts), - SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 4, twl6040_amicl_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 4, twl6040_amicr_texts), }; static const char *twl6040_hs_texts[] = { -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 08/16] ASoC: twl6040: Support other sample rates.
by Olaya, Margarita 11 Dec '10

11 Dec '10

The twl6040 can support more sample rates other than 88.2 and 96k. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index a89e7b8..9776715 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -39,7 +39,7 @@ #include "twl6040.h" -#define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define TWL6040_RATES SNDRV_PCM_RATE_8000_96000 #define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) struct twl6040_jack_data { @@ -890,10 +890,17 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); switch (rate) { + case 11250: + case 22500: + case 44100: case 88200: lppllctl |= TWL6040_LPLLFIN; priv->sysclk = 17640000; break; + case 8000: + case 16000: + case 32000: + case 48000: case 96000: lppllctl &= ~TWL6040_LPLLFIN; priv->sysclk = 19200000; -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 07/16] ASoC: twl6040: Fix PCM error handling ops
by Olaya, Margarita 11 Dec '10

11 Dec '10

This patch moves all the PCM error handling for clock config out of trigger() and startup() and into prepare(). Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 55 +++++++++++++++++-------------------------- 1 files changed, 22 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index d15ea44..a89e7b8 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -865,23 +865,6 @@ static int twl6040_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - if (!priv->sysclk) { - dev_err(codec->dev, - "no mclk configured, call set_sysclk() on init\n"); - return -EINVAL; - } - - /* - * capture is not supported at 17.64 MHz, - * it's reserved for headset low-power playback scenario - */ - if ((priv->sysclk == 17640000) && substream->stream) { - dev_err(codec->dev, - "capture mode is not supported at %dHz\n", - priv->sysclk); - return -EINVAL; - } - snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, priv->sysclk_constraints); @@ -925,31 +908,37 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, return 0; } -static int twl6040_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) +static int twl6040_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - /* - * low-power playback mode is restricted - * for headset path only - */ - if ((priv->sysclk == 17640000) && priv->non_lp) { + if (!priv->sysclk) { + dev_err(codec->dev, + "no mclk configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + /* + * capture is not supported at 17.64 MHz, + * it's reserved for headset low-power playback scenario + */ + if ((priv->sysclk == 17640000) && + substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + dev_err(codec->dev, + "capture mode is not supported at %dHz\n", + priv->sysclk); + return -EINVAL; + } + + if ((priv->sysclk == 17640000) && priv->non_lp) { dev_err(codec->dev, "some enabled paths aren't supported at %dHz\n", priv->sysclk); return -EPERM; - } - break; - default: - break; } - return 0; } @@ -1063,7 +1052,7 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai, static struct snd_soc_dai_ops twl6040_dai_ops = { .startup = twl6040_startup, .hw_params = twl6040_hw_params, - .trigger = twl6040_trigger, + .prepare = twl6040_prepare, .set_sysclk = twl6040_set_dai_sysclk, }; -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 06/16] ASoC: twl6040: Restore bias level at resume
by Olaya, Margarita 11 Dec '10

11 Dec '10

This patch restores the CODEC bias level at resume(). Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b713d13..d15ea44 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1125,6 +1125,7 @@ static int twl6040_suspend(struct snd_soc_codec *codec, pm_message_t state) static int twl6040_resume(struct snd_soc_codec *codec) { twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6040_set_bias_level(codec, codec->dapm.suspend_bias_level); return 0; } -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 05/16] ASoC: twl6040: Support all DAIs
by Olaya, Margarita 11 Dec '10

11 Dec '10

This patch adds support for all the TWL6040 DAIs. - twl6040-ul - twl6040-dl1 - twl6040-dl2 - twl6040-vib Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 43 ++++++++++++++++++++++++++++++++++++------- 1 files changed, 36 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 27d6d53..b713d13 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1067,23 +1067,51 @@ static struct snd_soc_dai_ops twl6040_dai_ops = { .set_sysclk = twl6040_set_dai_sysclk, }; -static struct snd_soc_dai_driver twl6040_dai = { - .name = "twl6040-hifi", +static struct snd_soc_dai_driver twl6040_dai[] = { +{ + .name = "twl6040-ul", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}, +{ + .name = "twl6040-dl1", .playback = { - .stream_name = "Playback", + .stream_name = "Headset Playback", .channels_min = 1, - .channels_max = 4, + .channels_max = 2, .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, - .capture = { - .stream_name = "Capture", + .ops = &twl6040_dai_ops, +}, +{ + .name = "twl6040-dl2", + .playback = { + .stream_name = "Handsfree Playback", .channels_min = 1, .channels_max = 2, .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, .ops = &twl6040_dai_ops, +}, +{ + .name = "twl6040-vib", + .playback = { + .stream_name = "Vibra Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}, }; #ifdef CONFIG_PM @@ -1235,7 +1263,8 @@ static struct snd_soc_codec_driver soc_codec_dev_twl6040 = { static int __devinit twl6040_codec_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, - &soc_codec_dev_twl6040, &twl6040_dai, 1); + &soc_codec_dev_twl6040, twl6040_dai, + ARRAY_SIZE(twl6040_dai)); } static int __devexit twl6040_codec_remove(struct platform_device *pdev) -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 01/16] ASoC: twl6040: Update twl IO macro
by Olaya, Margarita 11 Dec '10

11 Dec '10

Update the codec to use the new twl core register macros Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 5d7e2f7..d33d2b4 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -210,7 +210,7 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, if (reg >= TWL6040_CACHEREGNUM) return -EIO; - twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg); + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &value, reg); twl6040_write_reg_cache(codec, reg, value); return value; @@ -226,7 +226,7 @@ static int twl6040_write(struct snd_soc_codec *codec, return -EIO; twl6040_write_reg_cache(codec, reg, value); - return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); + return twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, value, reg); } static void twl6040_init_vio_regs(struct snd_soc_codec *codec) @@ -439,7 +439,7 @@ static irqreturn_t twl6040_naudint_handler(int irq, void *data) struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); u8 intid; - twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); switch (intid) { case TWL6040_THINT: @@ -715,7 +715,7 @@ static int twl6040_power_up_completion(struct snd_soc_codec *codec, msecs_to_jiffies(48)); if (!time_left) { - twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); if (!(intid & TWL6040_READYINT)) { dev_err(codec->dev, "timeout waiting for READYINT\n"); -- 1.7.0.4

1 0

[alsa-devel] [PATCH v4 3/3] ASoC: sdp4430: Add Jack support
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Use jack framework to enable detection for the headset microphone and stereo output in the sdp4430. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: David Anders <x0132446(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/omap/sdp4430.c | 32 ++++++++++++++++++++++++++++++++ 1 files changed, 32 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 0c37c51..189e039 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -24,6 +24,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <plat/hardware.h> @@ -65,6 +66,21 @@ static struct snd_soc_ops sdp4430_ops = { .hw_params = sdp4430_hw_params, }; +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -160,6 +176,22 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); ret = snd_soc_dapm_sync(dapm); + if (ret) + return ret; + + /* Headset jack detection */ + ret = snd_soc_jack_new(codec, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + + if (machine_is_omap_4430sdp()) + twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); + else + snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET); return ret; } -- 1.7.0.4

1 0

[alsa-devel] [PATCHv4 1/3] ASoC: sdp4430: Enable FM stereo pins
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Add FM stereo pins to the machine driver and add them as a dapm widget. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/omap/sdp4430.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index ebbd62f..0c37c51 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -101,6 +101,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), + SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -123,6 +124,10 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Earphone speaker */ {"Earphone Spk", NULL, "EP"}, + + /* Aux/FM Stereo In: AFML, AFMR */ + {"AFML", NULL, "Aux/FM Stereo In"}, + {"AFMR", NULL, "Aux/FM Stereo In"}, }; static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) @@ -149,13 +154,11 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) /* SDP4430 connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Mic"); snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "AFML"); + snd_soc_dapm_enable_pin(dapm, "AFMR"); snd_soc_dapm_enable_pin(dapm, "Headset Mic"); snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - /* TWL6040 not connected pins */ - snd_soc_dapm_nc_pin(dapm, "AFML"); - snd_soc_dapm_nc_pin(dapm, "AFMR"); - ret = snd_soc_dapm_sync(dapm); return ret; -- 1.7.0.4

1 0

[alsa-devel] [PATCH 1/3] ASoC: Implement WM8903 oversampling rate controls
by Mark Brown 11 Dec '10

11 Dec '10

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 797992c..542c7c4 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -481,6 +481,16 @@ static const char *hpf_mode_text[] = { static const struct soc_enum hpf_mode = SOC_ENUM_SINGLE(WM8903_ADC_DIGITAL_0, 5, 4, hpf_mode_text); +static const char *osr_text[] = { + "Low power", "High performance" +}; + +static const struct soc_enum adc_osr = + SOC_ENUM_SINGLE(WM8903_ANALOGUE_ADC_0, 0, 2, osr_text); + +static const struct soc_enum dac_osr = + SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 0, 2, osr_text); + static const char *drc_slope_text[] = { "1", "1/2", "1/4", "1/8", "1/16", "0" }; @@ -619,6 +629,7 @@ SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, 6, 1, 0), /* ADCs */ +SOC_ENUM("ADC OSR", adc_osr), SOC_SINGLE("HPF Switch", WM8903_ADC_DIGITAL_0, 4, 1, 0), SOC_ENUM("HPF Mode", hpf_mode), SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0), @@ -650,6 +661,7 @@ SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8903_DAC_DIGITAL_0, 4, 8, 12, 0, digital_sidetone_tlv), /* DAC */ +SOC_ENUM("DAC OSR", dac_osr), SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT, WM8903_DAC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv), SOC_ENUM("DAC Soft Mute Rate", soft_mute), -- 1.7.1

2 5

[alsa-devel] [PATCH] ASoC: Implement WM8903 high pass filter support
by Mark Brown 11 Dec '10

11 Dec '10

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 620793e..797992c 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -474,6 +474,13 @@ static const DECLARE_TLV_DB_SCALE(drc_tlv_min, 0, 600, 0); static const DECLARE_TLV_DB_SCALE(drc_tlv_max, 1200, 600, 0); static const DECLARE_TLV_DB_SCALE(drc_tlv_startup, -300, 50, 0); +static const char *hpf_mode_text[] = { + "Hi-fi", "Voice 1", "Voice 2", "Voice 3" +}; + +static const struct soc_enum hpf_mode = + SOC_ENUM_SINGLE(WM8903_ADC_DIGITAL_0, 5, 4, hpf_mode_text); + static const char *drc_slope_text[] = { "1", "1/2", "1/4", "1/8", "1/16", "0" }; @@ -612,6 +619,8 @@ SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, 6, 1, 0), /* ADCs */ +SOC_SINGLE("HPF Switch", WM8903_ADC_DIGITAL_0, 4, 1, 0), +SOC_ENUM("HPF Mode", hpf_mode), SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0), SOC_ENUM("DRC Compressor Slope R0", drc_slope_r0), SOC_ENUM("DRC Compressor Slope R1", drc_slope_r1), -- 1.7.1

2 1

[alsa-devel] [PATCH v2] ASoC: tlv320dac33: Power down digital parts, when not needed
by Peter Ujfalusi 11 Dec '10

11 Dec '10

If the following scenario has been followed: 1. Enable analog bypass amixer sset 'Analog Left Bypass' on amixer sset 'Analog Right Bypass' on 2. Start playback aplay -fdat -d3 /dev/zero After the playback stopped (3 sec), and the soc timeout (5 sec), the digital parts of the codec will remain powered up. This means that the DAI clocks are continue to run, the oscillator remain operational, etc. Use the SND_SOC_DAPM_POST_PMD widget to get notification about the stopped stream, and power down the digital part of the codec. If the analog bypass is enabled, than the codec will remain in BIAS_ON level, and things will work correctly. In case, if the bypass is disabled, than the codec will fall to BIAS_STANDBY than to BIAS_OFF level, as it used to. The digital part of DAC33 is initialized at every stream start (DAPM_PRE:PRE_PMU event), so subsequent streams (within 5 sec) will have working DAI. When the codec is coming out from BIAS_OFF, the full power-up sequence followed by the same DAPM_PRE widget event will power up the digital part. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320dac33.c | 23 +++++++++++++++++++++-- 1 files changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index b3445b3..776ac80 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -354,6 +354,21 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) dac33_write(codec, DAC33_PWR_CTRL, reg); } +static inline void dac33_disable_digital(struct snd_soc_codec *codec) +{ + u8 reg; + + /* Stop the DAI clock */ + reg = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + reg &= ~DAC33_BCLKON; + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg); + + /* Power down the Oscillator, and DACs */ + reg = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + reg &= ~(DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB); + dac33_write(codec, DAC33_PWR_CTRL, reg); +} + static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); @@ -402,7 +417,7 @@ exit: return ret; } -static int playback_event(struct snd_soc_dapm_widget *w, +static int dac33_playback_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(w->codec); @@ -414,6 +429,9 @@ static int playback_event(struct snd_soc_dapm_widget *w, dac33_prepare_chip(dac33->substream); } break; + case SND_SOC_DAPM_POST_PMD: + dac33_disable_digital(w->codec); + break; } return 0; } @@ -609,7 +627,8 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Right DAC Power", DAC33_RDAC_PWR_CTRL, 2, 0, NULL, 0), - SND_SOC_DAPM_PRE("Prepare Playback", playback_event), + SND_SOC_DAPM_PRE("Pre Playback", dac33_playback_event), + SND_SOC_DAPM_POST("Post Playback", dac33_playback_event), }; static const struct snd_soc_dapm_route audio_map[] = { -- 1.7.3.3

2 1

[alsa-devel] [PATCH v4] ASoC: Add HP iPAQ H1940 support
by Vasily Khoruzhick 10 Dec '10

10 Dec '10

Add glue driver to make s3c24xx-i2s and uda1380 produce some sound on H1940. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- v4: Adapt v3 version to latest -next tree v3-v1: History lost :( sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/h1940_uda1380.c | 296 +++++++++++++++++++++++++++++++++++++ 3 files changed, 306 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/h1940_uda1380.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index eb45cf9..67cdad4 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -116,6 +116,14 @@ config ASOC_SAMSUNG_SIMTEC_HERMES select SND_SOC_TLV320AIC3X select ASOC_SAMSUNG_SIMTEC +config ASOC_SAMSUNG_H1940_UDA1380 + tristate "Audio support for the HP iPAQ H1940" + depends on ASOC_SAMSUNG && ARCH_H1940 + select SND_S3C24XX_I2S + select SND_SOC_UDA1380 + help + This driver provides audio support for HP iPAQ h1940 PDA. + config ASOC_SAMSUNG_RX1950_UDA1380 tristate "Audio support for the HP iPAQ RX1950" depends on ASOC_SAMSUNG && MACH_RX1950 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 0d24f95..622e76e 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -27,6 +27,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-h1940-uda1380-objs := h1940_uda1380.o snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o snd-soc-smdk-wm8580-objs := smdk_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o @@ -43,6 +44,7 @@ obj-$(CONFIG_ASOC_SAMSUNG_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_ASOC_SAMSUNG_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_ASOC_SAMSUNG_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_ASOC_SAMSUNG_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_ASOC_SAMSUNG_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(CONFIG_ASOC_SAMSUNG_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o obj-$(CONFIG_ASOC_SAMSUNG_SMDK_WM8580) += snd-soc-smdk-wm8580.o obj-$(CONFIG_ASOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c new file mode 100644 index 0000000..c45f7ce --- /dev/null +++ b/sound/soc/samsung/h1940_uda1380.c @@ -0,0 +1,296 @@ +/* + * h1940-uda1380.c -- ALSA Soc Audio Layer + * + * Copyright (c) 2010 Arnaud Patard <arnaud.patard(a)rtp-net.org> + * Copyright (c) 2010 Vasily Khoruzhick <anarsoul(a)gmail.com> + * + * Based on version from Arnaud Patard <arnaud.patard(a)rtp-net.org> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> + +#include <sound/soc.h> +#include <sound/uda1380.h> +#include <sound/jack.h> + +#include <plat/regs-iis.h> + +#include <mach/h1940-latch.h> + +#include <asm/mach-types.h> + +#include "dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda1380.h" + +static unsigned int rates[] = { + 11025, + 22050, + 44100, +}; + +static struct snd_pcm_hw_constraint_list hw_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static struct snd_soc_jack hp_jack; + +static struct snd_soc_jack_pin hp_jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, +}; + +static struct snd_soc_jack_gpio hp_jack_gpios[] = { + { + .gpio = S3C2410_GPG(4), + .name = "hp-gpio", + .report = SND_JACK_HEADPHONE, + .invert = 1, + .debounce_time = 200, + }, +}; + +static int h1940_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.rate_min = hw_rates.list[0]; + runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1]; + runtime->hw.rates = SNDRV_PCM_RATE_KNOT; + + return snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_rates); +} + +static int h1940_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int div; + int ret; + unsigned int rate = params_rate(params); + + switch (rate) { + case 11025: + case 22050: + case 44100: + div = s3c24xx_i2s_get_clockrate() / (384 * rate); + if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate)) + div++; + break; + default: + dev_err(&rtd->dev, "%s: rate %d is not supported\n", + __func__, rate); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* select clock source */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_384FS); + if (ret < 0) + return ret; + + /* set BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops h1940_ops = { + .startup = h1940_startup, + .hw_params = h1940_hw_params, +}; + +static int h1940_spk_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + gpio_set_value(H1940_LATCH_AUDIO_POWER, 1); + else + gpio_set_value(H1940_LATCH_AUDIO_POWER, 0); + + return 0; +} + +/* h1940 machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", h1940_spk_power), +}; + +/* h1940 machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = { + /* headphone connected to VOUTLHP, VOUTRHP */ + {"Headphone Jack", NULL, "VOUTLHP"}, + {"Headphone Jack", NULL, "VOUTRHP"}, + + /* ext speaker connected to VOUTL, VOUTR */ + {"Speaker", NULL, "VOUTL"}, + {"Speaker", NULL, "VOUTR"}, + + /* mic is connected to VINM */ + {"VINM", NULL, "Mic Jack"}, +}; + +static struct platform_device *s3c24xx_snd_device; + +static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int err; + + /* Add h1940 specific widgets */ + err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + if (err) + return err; + + /* Set up h1940 specific audio path audio_mapnects */ + err = snd_soc_dapm_add_routes(dapm, audio_map, + ARRAY_SIZE(audio_map)); + if (err) + return err; + + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + + snd_soc_dapm_sync(dapm); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack); + + snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), + hp_jack_pins); + + snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + +/* s3c24xx digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link h1940_uda1380_dai[] = { + { + .name = "uda1380", + .stream_name = "UDA1380 Duplex", + .cpu_dai_name = "s3c24xx-iis", + .codec_dai_name = "uda1380-hifi", + .init = h1940_uda1380_init, + .platform_name = "samsung-audio", + .codec_name = "uda1380-codec.0-001a", + .ops = &h1940_ops, + }, +}; + +static struct snd_soc_card h1940_asoc = { + .name = "h1940", + .dai_link = h1940_uda1380_dai, + .num_links = ARRAY_SIZE(h1940_uda1380_dai), +}; + +static int __init h1940_init(void) +{ + int ret; + + if (!machine_is_h1940()) + return -ENODEV; + + /* configure some gpios */ + ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power"); + if (ret) + goto err_out; + + ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0); + if (ret) + goto err_gpio; + + s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_snd_device) { + ret = -ENOMEM; + goto err_gpio; + } + + platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc); + ret = platform_device_add(s3c24xx_snd_device); + + if (ret) + goto err_plat; + + return 0; + +err_plat: + platform_device_put(s3c24xx_snd_device); +err_gpio: + gpio_free(H1940_LATCH_AUDIO_POWER); + +err_out: + return ret; +} + +static void __exit h1940_exit(void) +{ + platform_device_unregister(s3c24xx_snd_device); + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + gpio_free(H1940_LATCH_AUDIO_POWER); +} + +module_init(h1940_init); +module_exit(h1940_exit); + +/* Module information */ +MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); +MODULE_DESCRIPTION("ALSA SoC H1940"); +MODULE_LICENSE("GPL"); -- 1.7.3.2

3 2

[alsa-devel] [PATCH] ASoC: Implement WM8994/58 DAC and ADC oversampling control
by Mark Brown 10 Dec '10

10 Dec '10

The oversampling rate of the DAC and ADC can be controlled to optimise for either low power consumption or maximum performance. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b599616..af104ac 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -558,6 +558,16 @@ static const struct soc_enum aif2dacl_src = static const struct soc_enum aif2dacr_src = SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 14, 2, aif_chan_src_text); +static const char *osr_text[] = { + "Low Power", "High Performance", +}; + +static const struct soc_enum dac_osr = + SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 0, 2, osr_text); + +static const struct soc_enum adc_osr = + SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 1, 2, osr_text); + static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -834,6 +844,9 @@ SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0), SOC_ENUM("AIF2ADC HPF Mode", aif2adc_hpf), SOC_DOUBLE("AIF2ADC HPF Switch", WM8994_AIF2_ADC_FILTERS, 12, 11, 1, 0), +SOC_ENUM("ADC OSR", adc_osr), +SOC_ENUM("DAC OSR", dac_osr), + SOC_DOUBLE_R_TLV("DAC1 Volume", WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), SOC_DOUBLE_R("DAC1 Switch", WM8994_DAC1_LEFT_VOLUME, -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: Optimise WM9081 FLL performance
by Mark Brown 10 Dec '10

10 Dec '10

From: Mario Becroft <mb(a)gem.win.co.nz> Tune the FLL gain for optimal performance according to evaluation results. Signed-off-by: Mario Becroft <mb(a)gem.win.co.nz> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index c706077..e5055b2 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -590,6 +590,10 @@ static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id, reg5 |= fll_div.fll_clk_ref_div << WM9081_FLL_CLK_REF_DIV_SHIFT; snd_soc_write(codec, WM9081_FLL_CONTROL_5, reg5); + /* Set gain to the recommended value */ + snd_soc_update_bits(codec, WM9081_FLL_CONTROL_4, + WM9081_FLL_GAIN_MASK, 0); + /* Enable the FLL */ snd_soc_write(codec, WM9081_FLL_CONTROL_1, reg1 | WM9081_FLL_ENA); -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: tlv320dac33: Power down digital parts, when not needed
by Peter Ujfalusi 10 Dec '10

10 Dec '10

If the following scenarion has been followed: 1. Enable analog bypass amixer sset 'Analog Left Bypass' on amixer sset 'Analog Right Bypass' on 2. Start playback aplay -fdat -d3 /dev/zero After the playback stopped (3 sec), and the soc timeout (5 sec), the digital parts of the codec will remain powered up. This means that the DAI clocks are continue to run, the oscillator remain operational, etc. Use the SND_SOC_DAPM_POST_PMD widget to get notification about the stopped stream, and power down the digital part of the codec. If the analog bypass is enabled, than the codec will remain in BIAS_ON level, and things will work correctly. In case, if the bypass is disabled, than the codec will fall to BIAS_STANDBY than to BIAS_OFF level, as it used to. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 23 +++++++++++++++++++++-- 1 files changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index b3445b3..776ac80 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -354,6 +354,21 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) dac33_write(codec, DAC33_PWR_CTRL, reg); } +static inline void dac33_disable_digital(struct snd_soc_codec *codec) +{ + u8 reg; + + /* Stop the DAI clock */ + reg = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + reg &= ~DAC33_BCLKON; + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg); + + /* Power down the Oscillator, and DACs */ + reg = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + reg &= ~(DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB); + dac33_write(codec, DAC33_PWR_CTRL, reg); +} + static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); @@ -402,7 +417,7 @@ exit: return ret; } -static int playback_event(struct snd_soc_dapm_widget *w, +static int dac33_playback_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(w->codec); @@ -414,6 +429,9 @@ static int playback_event(struct snd_soc_dapm_widget *w, dac33_prepare_chip(dac33->substream); } break; + case SND_SOC_DAPM_POST_PMD: + dac33_disable_digital(w->codec); + break; } return 0; } @@ -609,7 +627,8 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Right DAC Power", DAC33_RDAC_PWR_CTRL, 2, 0, NULL, 0), - SND_SOC_DAPM_PRE("Prepare Playback", playback_event), + SND_SOC_DAPM_PRE("Pre Playback", dac33_playback_event), + SND_SOC_DAPM_POST("Post Playback", dac33_playback_event), }; static const struct snd_soc_dapm_route audio_map[] = { -- 1.7.3.3

2 3

[alsa-devel] [PATCH] ALSA: aoa: Remove wrong i2c_set_clientdata in onyx_i2c_remove()
by Axel Lin 10 Dec '10

10 Dec '10

It does not make sense to set clientdata to onyx in onyx_i2c_remove() as we are going to kfree onyx. What we really want here is i2c_set_clientdata(client, NULL); Since the i2c core will take care of it now, so this patch just removes it. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/aoa/codecs/onyx.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 91852e4..3687a6c 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1114,7 +1114,6 @@ static int onyx_i2c_remove(struct i2c_client *client) of_node_put(onyx->codec.node); if (onyx->codec_info) kfree(onyx->codec_info); - i2c_set_clientdata(client, onyx); kfree(onyx); return 0; } -- 1.7.2

2 1

Re: [alsa-devel] Help wanted: emachines em350 internal mic
by Raymond Yau 09 Dec '10

09 Dec '10

>> Greetings, >> There is only a single mic on this netbook. However, the alsa device >> shows up as stereo, and the right channel carries an inverted copy of >> the left channel. http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=59c774ed5ee00e9623a2… >> So when the two channels are summed to make a mono (e.g. for >> telephony, or playback on mono internal speaker) it appears that there >> is no signal. >> Is there a way to "cure" this problem in the driver? I can think of >> * Force internal mic capture to be mono (left only) >> * Copy left to right capture >> * Invert right capture >> but don't know how easy any are to implement.

4 6

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 09 Dec '10

09 Dec '10

The following changes since commit 146fd574ec06b1c593805738b1c2c8c5a4128681: ASoC: Add ADC high pass filter support to WM8994 (2010-12-08 15:46:49 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Alexander Sverdlin (1): ASoC: EP93xx: sampling rate range extended Dmitry Artamonow (1): ASoC: fix deemphasis control in wm8904/55/60 codecs Mark Brown (1): Merge branch 'for-2.6.37' into for-2.6.38 Seungwhan Youn (2): ASoC: WM8580: Fix R8 initial value ASoC: Remove unnecessary structure definitions include/sound/soc.h | 3 --- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8904.c | 3 ++- sound/soc/codecs/wm8955.c | 3 ++- sound/soc/codecs/wm8960.c | 3 ++- sound/soc/ep93xx/ep93xx-i2s.c | 4 ++-- sound/soc/ep93xx/ep93xx-pcm.c | 4 ++-- 7 files changed, 11 insertions(+), 11 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.37
by Mark Brown 09 Dec '10

09 Dec '10

The following changes since commit 2a7b1a00206895cfa444fd83477dca67a88a9d25: ASoC: Correct WM8962 interrupt mask register read (2010-12-07 15:49:42 +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 The usual small, focused bug fixes. Dmitry Artamonow (1): ASoC: fix deemphasis control in wm8904/55/60 codecs Seungwhan Youn (1): ASoC: WM8580: Fix R8 initial value sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8904.c | 3 ++- sound/soc/codecs/wm8955.c | 3 ++- sound/soc/codecs/wm8960.c | 3 ++- 4 files changed, 7 insertions(+), 4 deletions(-)

2 1

[alsa-devel] [PATCH] asoc, codec: Add SGTL5000 codec support to asoc
by zhaoming.zeng＠freescale.com 09 Dec '10

09 Dec '10

From: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> Add Freescale SGTL5000 codec support. Signed-off-by: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sgtl5000.c | 1052 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sgtl5000.h | 403 +++++++++++++++++ 4 files changed, 1460 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/sgtl5000.c create mode 100644 sound/soc/codecs/sgtl5000.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3b5690d..f12ef39 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -318,3 +318,6 @@ config SND_SOC_WM2000 config SND_SOC_WM9090 tristate + +config SND_SOC_SGTL5000 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f67a2d6..116ec3d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -65,6 +65,7 @@ snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-jz4740-codec-objs := jz4740.o +snd-soc-sgtl5000-objs := sgtl5000.o # Amp snd-soc-max9877-objs := max9877.o @@ -139,6 +140,7 @@ obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o +obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c new file mode 100644 index 0000000..a053f19 --- /dev/null +++ b/sound/soc/codecs/sgtl5000.c @@ -0,0 +1,1052 @@ +/* + * sgtl5000.c -- SGTL5000 ALSA SoC Audio driver + * + * Copyright 2008-2010 Freescale Semiconductor, Inc. All Rights Reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <mach/hardware.h> + +#include "sgtl5000.h" + +static const u16 sgtl5000_regs[] = { + 0xa011, 0x0000, 0x0000, 0x0000, 0x0008, 0x0000, 0x0010, 0x0000, + 0x0010, 0x0000, 0x0010, 0x0000, 0x0000, 0x0000, 0x323c, 0x0000, + 0x3c3c, 0x0000, 0x3c3c, 0x0000, 0x555f, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x408c, 0x0000, 0x0008, 0x0000, + 0x0000, 0x0000, 0x1818, 0x0000, 0x0111, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0404, 0x0000, + 0x7060, 0x0000, 0x5000, 0x0000, 0x0000, 0x0000, 0x0017, 0x0000, + 0x01c0, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, +}; + +enum sgtl5000_regulator_supplies { + VDDA, + VDDIO, + VDDD, + SGTL5000_SUPPLY_NUM +}; + +static const char* supply_names[SGTL5000_SUPPLY_NUM] = { + "VDDA", + "VDDIO", + "VDDD" +}; + +struct sgtl5000_priv { + int sysclk; + int master; + int fmt; + int rev; + int lrclk; + int capture_channels; + int playback_active; + int capture_active; + struct regulator *supplies[SGTL5000_SUPPLY_NUM]; + int regulator_volt[SGTL5000_SUPPLY_NUM]; + struct regulator *reg_vddio; + struct regulator *reg_vdda; + struct regulator *reg_vddd; + int vddio; /* voltage of VDDIO (mv) */ + int vdda; /* voltage of vdda (mv) */ + int vddd; /* voltage of vddd (mv), 0 if not connected */ + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; +}; + +static const char *adc_mux_text[] = { + "MIC_IN", "LINE_IN" +}; + +static const char *dac_mux_text[] = { + "DAC", "LINE_IN" +}; + +static const struct soc_enum adc_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text); + +static const struct soc_enum dac_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text); + +static const struct snd_kcontrol_new adc_mux = +SOC_DAPM_ENUM("Capture Mux", adc_enum); + +static const struct snd_kcontrol_new dac_mux = +SOC_DAPM_ENUM("Headphone Mux", dac_enum); + +static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MIC_IN"), + + SND_SOC_DAPM_OUTPUT("HP_OUT"), + SND_SOC_DAPM_OUTPUT("LINE_OUT"), + + SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_CTRL, 4, 1, NULL, 0), + SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_CTRL, 8, 1, NULL, 0), + + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), + SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + + SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_DIG_POWER, 6, 0), + SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Capture Mux", "LINE_IN", "LINE_IN"}, + {"Capture Mux", "MIC_IN", "MIC_IN"}, + {"ADC", NULL, "Capture Mux"}, + {"Headphone Mux", "DAC", "DAC"}, + {"Headphone Mux", "LINE_IN", "LINE_IN"}, + {"LO", NULL, "DAC"}, + {"HP", NULL, "Headphone Mux"}, + {"LINE_OUT", NULL, "LO"}, + {"HP_OUT", NULL, "HP"}, +}; + +static int dac_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xfc - 0x3c; + return 0; +} + +static int dac_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg, l, r; + + reg = snd_soc_read(codec, SGTL5000_CHIP_DAC_VOL); + l = (reg & SGTL5000_DAC_VOL_LEFT_MASK) >> SGTL5000_DAC_VOL_LEFT_SHIFT; + r = (reg & SGTL5000_DAC_VOL_RIGHT_MASK) >> SGTL5000_DAC_VOL_RIGHT_SHIFT; + l = l < 0x3c ? 0x3c : l; + l = l > 0xfc ? 0xfc : l; + r = r < 0x3c ? 0x3c : r; + r = r > 0xfc ? 0xfc : r; + l = 0xfc - l; + r = 0xfc - r; + + ucontrol->value.integer.value[0] = l; + ucontrol->value.integer.value[1] = r; + + return 0; +} + +static int dac_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg, l, r; + + l = ucontrol->value.integer.value[0]; + r = ucontrol->value.integer.value[1]; + + l = l < 0 ? 0 : l; + l = l > 0xfc - 0x3c ? 0xfc - 0x3c : l; + r = r < 0 ? 0 : r; + r = r > 0xfc - 0x3c ? 0xfc - 0x3c : r; + l = 0xfc - l; + r = 0xfc - r; + + reg = l << SGTL5000_DAC_VOL_LEFT_SHIFT | + r << SGTL5000_DAC_VOL_RIGHT_SHIFT; + + snd_soc_write(codec, SGTL5000_CHIP_DAC_VOL, reg); + + return 0; +} + +static const char *mic_gain_text[] = { + "0dB", "20dB", "30dB", "40dB" +}; + +static const char *adc_m6db_text[] = { + "No Change", "Reduced by 6dB" +}; + +static const struct soc_enum mic_gain = +SOC_ENUM_SINGLE(SGTL5000_CHIP_MIC_CTRL, 0, 4, mic_gain_text); + +static const struct soc_enum adc_m6db = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_ADC_CTRL, 8, 2, adc_m6db_text); + +static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { + SOC_ENUM("MIC GAIN", mic_gain), + SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0), + SOC_ENUM("Capture Vol Reduction", adc_m6db), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = dac_info_volsw, + .get = dac_get_volsw, + .put = dac_put_volsw, + }, + SOC_DOUBLE("Headphone Playback Volume", SGTL5000_CHIP_ANA_HP_CTRL, 0, 8, 0x7f, 1), +}; + +static int __sgtl5000_digital_mute(struct snd_soc_codec *codec, int mute) +{ + u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT; + + if (mute) + snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, + adcdac_ctrl, adcdac_ctrl); + else + snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, + adcdac_ctrl, 0); + + return 0; +} + +static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + return __sgtl5000_digital_mute(codec, mute); +} + +static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + u16 i2sctl = 0; + + sgtl5000->master = 0; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2sctl |= SGTL5000_I2S_MASTER; + sgtl5000->master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + return -EINVAL; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + i2sctl |= SGTL5000_I2S_MODE_PCM; + break; + case SND_SOC_DAIFMT_DSP_B: + i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + case SND_SOC_DAIFMT_I2S: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + break; + case SND_SOC_DAIFMT_RIGHT_J: + i2sctl |= SGTL5000_I2S_MODE_RJ; + i2sctl |= SGTL5000_I2S_LRPOL; + break; + case SND_SOC_DAIFMT_LEFT_J: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + default: + return -EINVAL; + } + sgtl5000->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_NB_IF: + break; + case SND_SOC_DAIFMT_IB_IF: + case SND_SOC_DAIFMT_IB_NF: + i2sctl |= SGTL5000_I2S_SCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, i2sctl); + + return 0; +} + +static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case SGTL5000_SYSCLK: + sgtl5000->sysclk = freq; + break; + case SGTL5000_LRCLK: + sgtl5000->lrclk = freq; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int sgtl5000_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int reg; + + reg = snd_soc_read(codec, SGTL5000_CHIP_DIG_POWER); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg |= SGTL5000_I2S_IN_POWERUP; + else + reg |= SGTL5000_I2S_OUT_POWERUP; + snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER, reg); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + reg = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + reg |= SGTL5000_ADC_POWERUP; + if (sgtl5000->capture_channels == 1) + reg &= ~SGTL5000_ADC_STEREO; + else + reg |= SGTL5000_ADC_STEREO; + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, reg); + } + + return 0; +} + +static int sgtl5000_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + struct snd_pcm_runtime *master_runtime; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sgtl5000->playback_active++; + else + sgtl5000->capture_active++; + + /* The DAI has shared clocks so if we already have a playback or + * capture going then constrain this substream to match it. + */ + if (sgtl5000->master_substream) { + master_runtime = sgtl5000->master_substream->runtime; + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + sgtl5000->slave_substream = substream; + } else + sgtl5000->master_substream = substream; + + return 0; +} + +static void sgtl5000_pcm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int reg, dig_pwr, ana_pwr; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sgtl5000->playback_active--; + else + sgtl5000->capture_active--; + + if (sgtl5000->master_substream == substream) + sgtl5000->master_substream = sgtl5000->slave_substream; + + sgtl5000->slave_substream = NULL; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + ana_pwr = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + ana_pwr &= ~(SGTL5000_ADC_POWERUP | SGTL5000_ADC_STEREO); + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); + } + + dig_pwr = snd_soc_read(codec, SGTL5000_CHIP_DIG_POWER); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dig_pwr &= ~SGTL5000_I2S_IN_POWERUP; + else + dig_pwr &= ~SGTL5000_I2S_OUT_POWERUP; + snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER, dig_pwr); + + if (!sgtl5000->playback_active && !sgtl5000->capture_active) { + reg = snd_soc_read(codec, SGTL5000_CHIP_I2S_CTRL); + reg &= ~SGTL5000_I2S_MASTER; + snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, reg); + } +} + +/* + * Set PCM DAI bit size and sample rate. + * input: params_rate, params_fmt + */ +static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int channels = params_channels(params); + int clk_ctl = 0; + int pll_ctl = 0; + int i2s_ctl; + int div2 = 0; + int reg; + int sys_fs; + + if (!sgtl5000->sysclk) { + dev_err(codec->dev, "%s: set sysclk first!\n", __func__); + return -EFAULT; + } + + if (substream == sgtl5000->slave_substream) + return 0; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + sgtl5000->capture_channels = channels; + + switch (sgtl5000->lrclk) { + case 8000: + case 16000: + sys_fs = 32000; + break; + case 11025: + case 22050: + sys_fs = 44100; + break; + default: + sys_fs = sgtl5000->lrclk; + break; + } + + switch (sys_fs / sgtl5000->lrclk) { + case 4: + clk_ctl |= SGTL5000_RATE_MODE_DIV_4 << SGTL5000_RATE_MODE_SHIFT; + break; + case 2: + clk_ctl |= SGTL5000_RATE_MODE_DIV_2 << SGTL5000_RATE_MODE_SHIFT; + break; + default: + break; + } + + switch (sys_fs) { + case 32000: + clk_ctl |= SGTL5000_SYS_FS_32k << SGTL5000_SYS_FS_SHIFT; + break; + case 44100: + clk_ctl |= SGTL5000_SYS_FS_44_1k << SGTL5000_SYS_FS_SHIFT; + break; + case 48000: + clk_ctl |= SGTL5000_SYS_FS_48k << SGTL5000_SYS_FS_SHIFT; + break; + case 96000: + clk_ctl |= SGTL5000_SYS_FS_96k << SGTL5000_SYS_FS_SHIFT; + break; + default: + dev_err(codec->dev, "sample rate %d not supported\n", sgtl5000->lrclk); + return -EFAULT; + } + + /* SGTL5000 rev1 has a IC bug to prevent switching to MCLK from PLL. */ + if (!sgtl5000->master) { + sys_fs = sgtl5000->lrclk; + clk_ctl = SGTL5000_RATE_MODE_DIV_1 << SGTL5000_RATE_MODE_SHIFT; + if (sys_fs * 256 == sgtl5000->sysclk) + clk_ctl |= SGTL5000_MCLK_FREQ_256FS << \ + SGTL5000_MCLK_FREQ_SHIFT; + else if (sys_fs * 384 == sgtl5000->sysclk && sys_fs != 96000) + clk_ctl |= SGTL5000_MCLK_FREQ_384FS << \ + SGTL5000_MCLK_FREQ_SHIFT; + else if (sys_fs * 512 == sgtl5000->sysclk && sys_fs != 96000) + clk_ctl |= SGTL5000_MCLK_FREQ_512FS << \ + SGTL5000_MCLK_FREQ_SHIFT; + else { + pr_err("%s: PLL not supported in slave mode\n", + __func__); + return -EINVAL; + } + } else + clk_ctl |= SGTL5000_MCLK_FREQ_PLL << SGTL5000_MCLK_FREQ_SHIFT; + + if ((clk_ctl & SGTL5000_MCLK_FREQ_MASK) == SGTL5000_MCLK_FREQ_PLL) { + u64 out, t; + unsigned int in, int_div, frac_div; + if (sgtl5000->sysclk > 17000000) { + div2 = 1; + in = sgtl5000->sysclk / 2; + } else { + div2 = 0; + in = sgtl5000->sysclk; + } + if (sys_fs == 44100) + out = 180633600; + else + out = 196608000; + t = do_div(out, in); + int_div = out; + t *= 2048; + do_div(t, in); + frac_div = t; + pll_ctl = int_div << SGTL5000_PLL_INT_DIV_SHIFT | + frac_div << SGTL5000_PLL_FRAC_DIV_SHIFT; + } + + i2s_ctl = snd_soc_read(codec, SGTL5000_CHIP_I2S_CTRL); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "fs=%d,clk_ctl=%d,pll_ctl=%d,i2s_ctl=%d,div2=%d\n", + sgtl5000->lrclk, clk_ctl, pll_ctl, i2s_ctl, div2); + + if ((clk_ctl & SGTL5000_MCLK_FREQ_MASK) == SGTL5000_MCLK_FREQ_PLL) { + snd_soc_write(codec, SGTL5000_CHIP_PLL_CTRL, pll_ctl); + reg = snd_soc_read(codec, SGTL5000_CHIP_CLK_TOP_CTRL); + if (div2) + reg |= SGTL5000_INPUT_FREQ_DIV2; + else + reg &= ~SGTL5000_INPUT_FREQ_DIV2; + snd_soc_write(codec, SGTL5000_CHIP_CLK_TOP_CTRL, reg); + reg = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + reg |= SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP; + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, reg); + } + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl); + + return 0; +} + +static void sgtl5000_mic_bias(struct snd_soc_codec *codec, int enable) +{ + int reg, bias_r = 0; + if (enable) + bias_r = SGTL5000_BIAS_R_4k << SGTL5000_BIAS_R_SHIFT; + reg = snd_soc_read(codec, SGTL5000_CHIP_MIC_CTRL); + if ((reg & SGTL5000_BIAS_R_MASK) != bias_r) { + reg &= ~SGTL5000_BIAS_R_MASK; + reg |= bias_r; + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, reg); + } +} + +static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg, ana_pwr; + u16 *cache = codec->reg_cache; + + if (codec->bias_level == level) + return 0; + + switch (level) { + case SND_SOC_BIAS_ON: + sgtl5000_mic_bias(codec, 1); + + snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_MASK, SGTL5000_BIAS_R_MASK); + + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); + + __sgtl5000_digital_mute(codec, 0); + break; + + case SND_SOC_BIAS_PREPARE: /* partial On */ + snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_MASK, 0); + + /* must power up hp/line out before vag & dac to + avoid pops. */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + + reg &= ~SGTL5000_VAG_POWERUP; + reg |= SGTL5000_DAC_POWERUP; + reg |= SGTL5000_HP_POWERUP; + reg |= SGTL5000_LINE_OUT_POWERUP; + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, reg); + + snd_soc_update_bits(codec, SGTL5000_CHIP_DIG_POWER, + SGTL5000_DAC_EN, SGTL5000_DAC_EN); + + break; + + case SND_SOC_BIAS_STANDBY: + /* soc calls digital_mute to unmute before record but doesn't + call digital_mute to mute after record. */ + __sgtl5000_digital_mute(codec, 1); + + snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_MASK, 0); + + reg = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + if (reg & SGTL5000_VAG_POWERUP) { + reg &= ~SGTL5000_VAG_POWERUP; + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, reg); + } + reg &= ~SGTL5000_DAC_POWERUP; + reg &= ~SGTL5000_HP_POWERUP; + reg &= ~SGTL5000_LINE_OUT_POWERUP; + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, reg); + + reg = snd_soc_read(codec, SGTL5000_CHIP_DIG_POWER); + reg &= ~SGTL5000_DAC_EN; + snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER, reg); + + break; + + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* mute hp/Line_out first to avoid pops. */ + __sgtl5000_digital_mute(codec, 1); + sgtl5000_mic_bias(codec, 0); + + reg = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + ana_pwr = reg; + reg &= ~SGTL5000_VAG_POWERUP; + + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, reg); + + reg &= ~SGTL5000_HP_POWERUP; + reg &= ~SGTL5000_LINE_OUT_POWERUP; + reg &= ~SGTL5000_DAC_POWERUP; + reg &= ~SGTL5000_ADC_POWERUP; + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, reg); + + /* save ANA POWER register value for resume */ + cache[SGTL5000_CHIP_ANA_POWER >> 1] = ana_pwr; + break; + } + codec->bias_level = level; + return 0; +} + +#define SGTL5000_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000) + +#define SGTL5000_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +struct snd_soc_dai_ops sgtl5000_ops = { + .prepare = sgtl5000_pcm_prepare, + .startup = sgtl5000_pcm_startup, + .shutdown = sgtl5000_pcm_shutdown, + .hw_params = sgtl5000_pcm_hw_params, + .digital_mute = sgtl5000_digital_mute, + .set_fmt = sgtl5000_set_dai_fmt, + .set_sysclk = sgtl5000_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver sgtl5000_dai = { + .name = "sgtl5000", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SGTL5000_RATES, + .formats = SGTL5000_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SGTL5000_RATES, + .formats = SGTL5000_FORMATS, + }, + .ops = &sgtl5000_ops, + .symmetric_rates = 1, +}; + +static int sgtl5000_volatile_register(unsigned int reg) +{ + if (reg == SGTL5000_CHIP_ID || + reg == SGTL5000_CHIP_ADCDAC_CTRL || + reg == SGTL5000_CHIP_ANA_STATUS) + return 1; + return 0; +} + +static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int sgtl5000_restore_reg(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + return snd_soc_write(codec, reg, cache[reg >> 1]); +} + +static int sgtl5000_resume(struct snd_soc_codec *codec) +{ + int i; + + /* Restore refs first in same order as in sgtl5000_probe */ + sgtl5000_restore_reg(codec, SGTL5000_CHIP_LINREG_CTRL); + sgtl5000_restore_reg(codec, SGTL5000_CHIP_ANA_POWER); + msleep(10); + sgtl5000_restore_reg(codec, SGTL5000_CHIP_REF_CTRL); + sgtl5000_restore_reg(codec, SGTL5000_CHIP_LINE_OUT_CTRL); + + /* Restore everythine else */ + for (i = 0; i < ARRAY_SIZE(sgtl5000_regs); i++) + sgtl5000_restore_reg(codec, i); + + snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); + + /* Bring the codec back up to standby first to minimise pop/clicks */ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + sgtl5000_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static int sgtl5000_probe(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + u16 reg, ana_pwr, lreg_ctrl; + int vag; + int ret; + int vddd, vdda, vddio; + + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + vddd = sgtl5000->regulator_volt[VDDD]; + vdda = sgtl5000->regulator_volt[VDDA]; + vddio = sgtl5000->regulator_volt[VDDIO]; + + reg = snd_soc_read(codec, SGTL5000_CHIP_ID); + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(codec->dev, "Device with ID register %x is not a sgtl5000\n", reg); + return -ENODEV; + } + + sgtl5000->rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(codec->dev, "sgtl5000 revision %d\n", sgtl5000->rev); + + /* reset value */ + ana_pwr = SGTL5000_DAC_STEREO | + SGTL5000_LINREG_SIMPLE_POWERUP | + SGTL5000_STARTUP_POWERUP | + SGTL5000_ADC_STEREO | SGTL5000_REFTOP_POWERUP; + lreg_ctrl = 0; + + /* workaround for rev 0x11: use vddd linear regulator */ + if (!vddd || (sgtl5000->rev >= 0x11)) { + /* set VDDD to 1.2v */ + lreg_ctrl |= 0x8 << SGTL5000_LINREG_VDDD_SHIFT; + /* power internal linear regulator */ + ana_pwr |= SGTL5000_LINEREG_D_POWERUP; + } else { + /* turn of startup power */ + ana_pwr &= ~SGTL5000_STARTUP_POWERUP; + ana_pwr &= ~SGTL5000_LINREG_SIMPLE_POWERUP; + } + + if (vddio < 3100 && vdda < 3100) { + /* Enable VDDC charge pump */ + ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP; + } + + if (vddio >= 3100 && vdda >= 3100) { + /* VDDC use VDDIO rail */ + lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; + lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << + SGTL5000_VDDC_MAN_ASSN_SHIFT; + } + + /* If PLL is powered up (such as on power cycle) leave it on. */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + ana_pwr |= reg & (SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + + /* set ADC/DAC ref voltage to vdda / 2 */ + vag = vdda / 2; + if (vag <= SGTL5000_ANA_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_ANA_GND_BASE + SGTL5000_ANA_GND_STP * + (SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT)) + vag = SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT; + else + vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; + + /* set line out ref voltage to vddio / 2 */ + vag = vddio / 2; + if (vag <= SGTL5000_LINE_OUT_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_LINE_OUT_GND_BASE + SGTL5000_LINE_OUT_GND_STP * + SGTL5000_LINE_OUT_GND_MAX) + vag = SGTL5000_LINE_OUT_GND_MAX; + else + vag = (vag - SGTL5000_LINE_OUT_GND_BASE) / + SGTL5000_LINE_OUT_GND_STP; + + snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl); + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); + msleep(10); + + /* For rev 0x11, if vddd linear reg has been enabled, we have + to disable simple reg to get proper VDDD voltage. */ + if ((ana_pwr & SGTL5000_LINEREG_D_POWERUP) && (sgtl5000->rev >= 0x11)) { + ana_pwr &= ~SGTL5000_LINREG_SIMPLE_POWERUP; + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); + msleep(10); + } + + snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, + vag << SGTL5000_ANA_GND_SHIFT); + + snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, + vag << SGTL5000_LINE_OUT_GND_SHIFT | + SGTL5000_LINE_OUT_CURRENT_360u << + SGTL5000_LINE_OUT_CURRENT_SHIFT); + + snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_SSS_CTRL, + SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT); + snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER, 0); + + snd_soc_write(codec, SGTL5000_CHIP_ADCDAC_CTRL, + SGTL5000_DAC_VOL_RAMP_EN | + SGTL5000_DAC_MUTE_RIGHT | + SGTL5000_DAC_MUTE_LEFT); + + snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, 0x015f); + + reg = snd_soc_read(codec, SGTL5000_CHIP_ANA_ADC_CTRL); + reg &= ~SGTL5000_ADC_VOL_M6DB; + reg &= ~(SGTL5000_ADC_VOL_LEFT_MASK | SGTL5000_ADC_VOL_RIGHT_MASK); + reg |= (0xf << SGTL5000_ADC_VOL_LEFT_SHIFT) + | (0xf << SGTL5000_ADC_VOL_RIGHT_SHIFT); + snd_soc_write(codec, SGTL5000_CHIP_ANA_ADC_CTRL, reg); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_LINE_OUT_MUTE | + SGTL5000_HP_MUTE | + SGTL5000_HP_ZCD_EN | + SGTL5000_ADC_ZCD_EN); + + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_CLK_TOP_CTRL, 0); + + /* disable DAP */ + snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); + + snd_soc_add_controls(codec, sgtl5000_snd_controls, + ARRAY_SIZE(sgtl5000_snd_controls)); + + snd_soc_dapm_new_controls(codec, sgtl5000_dapm_widgets, + ARRAY_SIZE(sgtl5000_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int sgtl5000_remove(struct snd_soc_codec *codec) +{ + if (codec->control_data) + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_dapm_free(codec); + + return 0; +} + +struct snd_soc_codec_driver sgtl5000_driver = { + .probe = sgtl5000_probe, + .remove = sgtl5000_remove, + .suspend = sgtl5000_suspend, + .resume = sgtl5000_resume, + .set_bias_level = sgtl5000_set_bias_level, + .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), + .reg_word_size = sizeof(u16), + .reg_cache_default = sgtl5000_regs, + .volatile_register = sgtl5000_volatile_register, +}; + +static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct sgtl5000_priv *sgtl5000; + int ret; + int i; + + sgtl5000 = kzalloc(sizeof(struct sgtl5000_priv), GFP_KERNEL); + if (!sgtl5000) + return -ENOMEM; + + i2c_set_clientdata(client, sgtl5000); + + for (i = 0; i < SGTL5000_SUPPLY_NUM; i++) { + struct regulator *reg; + reg = regulator_get(&client->dev, supply_names[i]); + if (IS_ERR(reg)) + continue; + + sgtl5000->regulator_volt[i] = regulator_get_voltage(reg) / 1000; + regulator_enable(reg); + + sgtl5000->supplies[i] = reg; + } + + sgtl5000->regulator_volt[VDDA] = 3300; + sgtl5000->regulator_volt[VDDIO] = 3300; + + msleep(1); + + ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) { + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + goto err_out; + } + + return 0; + +err_out: + for (i=0; i < SGTL5000_SUPPLY_NUM; i++) { + if( !sgtl5000->supplies[i] ) + continue; + + regulator_disable(sgtl5000->supplies[i]); + regulator_put(sgtl5000->supplies[i]); + } + + kfree(sgtl5000); + return ret; +} + +static __devexit int sgtl5000_i2c_remove(struct i2c_client *client) +{ + struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + int i; + + if (client->dev.platform_data) + clk_disable((struct clk *)client->dev.platform_data); + + snd_soc_unregister_codec(&client->dev); + + for (i = 0; i < SGTL5000_SUPPLY_NUM; i++){ + if( !sgtl5000->supplies[i] ) + continue; + + regulator_disable(sgtl5000->supplies[i]); + regulator_put(sgtl5000->supplies[i]); + } + + kfree(sgtl5000); + return 0; +} + +static const struct i2c_device_id sgtl5000_id[] = { + {"sgtl5000-i2c", 0}, + {}, +}; + +MODULE_DEVICE_TABLE(i2c, sgtl5000_id); + +static struct i2c_driver sgtl5000_i2c_driver = { + .driver = { + .name = "sgtl5000-i2c", + .owner = THIS_MODULE, + }, + .probe = sgtl5000_i2c_probe, + .remove = __devexit_p(sgtl5000_i2c_remove), + .id_table = sgtl5000_id, +}; + +static int __init sgtl5000_modinit(void) +{ + return i2c_add_driver(&sgtl5000_i2c_driver); +} +module_init(sgtl5000_modinit); + +static void __exit sgtl5000_exit(void) +{ + i2c_del_driver(&sgtl5000_i2c_driver); +} +module_exit(sgtl5000_exit); + +MODULE_DESCRIPTION("ASoC sgtl5000 driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h new file mode 100644 index 0000000..7f24655 --- /dev/null +++ b/sound/soc/codecs/sgtl5000.h @@ -0,0 +1,403 @@ +/* + * sgtl5000.h - SGTL5000 audio codec interface + * + * Copyright 2008-2009 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _SGTL5000_H +#define _SGTL5000_H + +#include <linux/i2c.h> + +/* + * Register values. + */ +#define SGTL5000_CHIP_ID 0x0000 +#define SGTL5000_CHIP_DIG_POWER 0x0002 +#define SGTL5000_CHIP_CLK_CTRL 0x0004 +#define SGTL5000_CHIP_I2S_CTRL 0x0006 +#define SGTL5000_CHIP_SSS_CTRL 0x000a +#define SGTL5000_CHIP_ADCDAC_CTRL 0x000e +#define SGTL5000_CHIP_DAC_VOL 0x0010 +#define SGTL5000_CHIP_PAD_STRENGTH 0x0014 +#define SGTL5000_CHIP_ANA_ADC_CTRL 0x0020 +#define SGTL5000_CHIP_ANA_HP_CTRL 0x0022 +#define SGTL5000_CHIP_ANA_CTRL 0x0024 +#define SGTL5000_CHIP_LINREG_CTRL 0x0026 +#define SGTL5000_CHIP_REF_CTRL 0x0028 +#define SGTL5000_CHIP_MIC_CTRL 0x002a +#define SGTL5000_CHIP_LINE_OUT_CTRL 0x002c +#define SGTL5000_CHIP_LINE_OUT_VOL 0x002e +#define SGTL5000_CHIP_ANA_POWER 0x0030 +#define SGTL5000_CHIP_PLL_CTRL 0x0032 +#define SGTL5000_CHIP_CLK_TOP_CTRL 0x0034 +#define SGTL5000_CHIP_ANA_STATUS 0x0036 +#define SGTL5000_CHIP_SHORT_CTRL 0x003c +#define SGTL5000_CHIP_ANA_TEST2 0x003a +#define SGTL5000_DAP_CTRL 0x0100 +#define SGTL5000_DAP_PEQ 0x0102 +#define SGTL5000_DAP_BASS_ENHANCE 0x0104 +#define SGTL5000_DAP_BASS_ENHANCE_CTRL 0x0106 +#define SGTL5000_DAP_AUDIO_EQ 0x0108 +#define SGTL5000_DAP_SURROUND 0x010a +#define SGTL5000_DAP_FLT_COEF_ACCESS 0x010c +#define SGTL5000_DAP_COEF_WR_B0_MSB 0x010e +#define SGTL5000_DAP_COEF_WR_B0_LSB 0x0110 +#define SGTL5000_DAP_EQ_BASS_BAND0 0x0116 +#define SGTL5000_DAP_EQ_BASS_BAND1 0x0118 +#define SGTL5000_DAP_EQ_BASS_BAND2 0x011a +#define SGTL5000_DAP_EQ_BASS_BAND3 0x011c +#define SGTL5000_DAP_EQ_BASS_BAND4 0x011e +#define SGTL5000_DAP_MAIN_CHAN 0x0120 +#define SGTL5000_DAP_MIX_CHAN 0x0122 +#define SGTL5000_DAP_AVC_CTRL 0x0124 +#define SGTL5000_DAP_AVC_THRESHOLD 0x0126 +#define SGTL5000_DAP_AVC_ATTACK 0x0128 +#define SGTL5000_DAP_AVC_DECAY 0x012a +#define SGTL5000_DAP_COEF_WR_B1_MSB 0x012c +#define SGTL5000_DAP_COEF_WR_B1_LSB 0x012e +#define SGTL5000_DAP_COEF_WR_B2_MSB 0x0130 +#define SGTL5000_DAP_COEF_WR_B2_LSB 0x0132 +#define SGTL5000_DAP_COEF_WR_A1_MSB 0x0134 +#define SGTL5000_DAP_COEF_WR_A1_LSB 0x0136 +#define SGTL5000_DAP_COEF_WR_A2_MSB 0x0138 +#define SGTL5000_DAP_COEF_WR_A2_LSB 0x013a + +/* + * Field Definitions. + */ + +/* + * SGTL5000_CHIP_ID + */ +#define SGTL5000_PARTID_MASK 0xff00 +#define SGTL5000_PARTID_SHIFT 8 +#define SGTL5000_PARTID_WIDTH 8 +#define SGTL5000_PARTID_PART_ID 0xa0 +#define SGTL5000_REVID_MASK 0x00ff +#define SGTL5000_REVID_SHIFT 0 +#define SGTL5000_REVID_WIDTH 8 + +/* + * SGTL5000_CHIP_DIG_POWER + */ +#define SGTL5000_ADC_EN 0x0040 +#define SGTL5000_DAC_EN 0x0020 +#define SGTL5000_DAP_POWERUP 0x0010 +#define SGTL5000_I2S_OUT_POWERUP 0x0002 +#define SGTL5000_I2S_IN_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_CLK_CTRL + */ +#define SGTL5000_RATE_MODE_MASK 0x0030 +#define SGTL5000_RATE_MODE_SHIFT 4 +#define SGTL5000_RATE_MODE_WIDTH 2 +#define SGTL5000_RATE_MODE_DIV_1 0 +#define SGTL5000_RATE_MODE_DIV_2 1 +#define SGTL5000_RATE_MODE_DIV_4 2 +#define SGTL5000_RATE_MODE_DIV_6 3 +#define SGTL5000_SYS_FS_MASK 0x000c +#define SGTL5000_SYS_FS_SHIFT 2 +#define SGTL5000_SYS_FS_WIDTH 2 +#define SGTL5000_SYS_FS_32k 0x0 +#define SGTL5000_SYS_FS_44_1k 0x1 +#define SGTL5000_SYS_FS_48k 0x2 +#define SGTL5000_SYS_FS_96k 0x3 +#define SGTL5000_MCLK_FREQ_MASK 0x0003 +#define SGTL5000_MCLK_FREQ_SHIFT 0 +#define SGTL5000_MCLK_FREQ_WIDTH 2 +#define SGTL5000_MCLK_FREQ_256FS 0x0 +#define SGTL5000_MCLK_FREQ_384FS 0x1 +#define SGTL5000_MCLK_FREQ_512FS 0x2 +#define SGTL5000_MCLK_FREQ_PLL 0x3 + +/* + * SGTL5000_CHIP_I2S_CTRL + */ +#define SGTL5000_I2S_SCLKFREQ_MASK 0x0100 +#define SGTL5000_I2S_SCLKFREQ_SHIFT 8 +#define SGTL5000_I2S_SCLKFREQ_WIDTH 1 +#define SGTL5000_I2S_SCLKFREQ_64FS 0x0 +#define SGTL5000_I2S_SCLKFREQ_32FS 0x1 /* Not for RJ mode */ +#define SGTL5000_I2S_MASTER 0x0080 +#define SGTL5000_I2S_SCLK_INV 0x0040 +#define SGTL5000_I2S_DLEN_MASK 0x0030 +#define SGTL5000_I2S_DLEN_SHIFT 4 +#define SGTL5000_I2S_DLEN_WIDTH 2 +#define SGTL5000_I2S_DLEN_32 0x0 +#define SGTL5000_I2S_DLEN_24 0x1 +#define SGTL5000_I2S_DLEN_20 0x2 +#define SGTL5000_I2S_DLEN_16 0x3 +#define SGTL5000_I2S_MODE_MASK 0x000c +#define SGTL5000_I2S_MODE_SHIFT 2 +#define SGTL5000_I2S_MODE_WIDTH 2 +#define SGTL5000_I2S_MODE_I2S_LJ 0x0 +#define SGTL5000_I2S_MODE_RJ 0x1 +#define SGTL5000_I2S_MODE_PCM 0x2 +#define SGTL5000_I2S_LRALIGN 0x0002 +#define SGTL5000_I2S_LRPOL 0x0001 /* set for which mode */ + +/* + * SGTL5000_CHIP_SSS_CTRL + */ +#define SGTL5000_DAP_MIX_LRSWAP 0x4000 +#define SGTL5000_DAP_LRSWAP 0x2000 +#define SGTL5000_DAC_LRSWAP 0x1000 +#define SGTL5000_I2S_OUT_LRSWAP 0x0400 +#define SGTL5000_DAP_MIX_SEL_MASK 0x0300 +#define SGTL5000_DAP_MIX_SEL_SHIFT 8 +#define SGTL5000_DAP_MIX_SEL_WIDTH 2 +#define SGTL5000_DAP_MIX_SEL_ADC 0x0 +#define SGTL5000_DAP_MIX_SEL_I2S_IN 0x1 +#define SGTL5000_DAP_SEL_MASK 0x00c0 +#define SGTL5000_DAP_SEL_SHIFT 6 +#define SGTL5000_DAP_SEL_WIDTH 2 +#define SGTL5000_DAP_SEL_ADC 0x0 +#define SGTL5000_DAP_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_MASK 0x0030 +#define SGTL5000_DAC_SEL_SHIFT 4 +#define SGTL5000_DAC_SEL_WIDTH 2 +#define SGTL5000_DAC_SEL_ADC 0x0 +#define SGTL5000_DAC_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_DAP 0x3 +#define SGTL5000_I2S_OUT_SEL_MASK 0x0003 +#define SGTL5000_I2S_OUT_SEL_SHIFT 0 +#define SGTL5000_I2S_OUT_SEL_WIDTH 2 +#define SGTL5000_I2S_OUT_SEL_ADC 0x0 +#define SGTL5000_I2S_OUT_SEL_I2S_IN 0x1 +#define SGTL5000_I2S_OUT_SEL_DAP 0x3 + +/* + * SGTL5000_CHIP_ADCDAC_CTRL + */ +#define SGTL5000_VOL_BUSY_DAC_RIGHT 0x2000 +#define SGTL5000_VOL_BUSY_DAC_LEFT 0x1000 +#define SGTL5000_DAC_VOL_RAMP_EN 0x0200 +#define SGTL5000_DAC_VOL_RAMP_EXPO 0x0100 +#define SGTL5000_DAC_MUTE_RIGHT 0x0008 +#define SGTL5000_DAC_MUTE_LEFT 0x0004 +#define SGTL5000_ADC_HPF_FREEZE 0x0002 +#define SGTL5000_ADC_HPF_BYPASS 0x0001 + +/* + * SGTL5000_CHIP_DAC_VOL + */ +#define SGTL5000_DAC_VOL_RIGHT_MASK 0xff00 +#define SGTL5000_DAC_VOL_RIGHT_SHIFT 8 +#define SGTL5000_DAC_VOL_RIGHT_WIDTH 8 +#define SGTL5000_DAC_VOL_LEFT_MASK 0x00ff +#define SGTL5000_DAC_VOL_LEFT_SHIFT 0 +#define SGTL5000_DAC_VOL_LEFT_WIDTH 8 + +/* + * SGTL5000_CHIP_PAD_STRENGTH + */ +#define SGTL5000_PAD_I2S_LRCLK_MASK 0x0300 +#define SGTL5000_PAD_I2S_LRCLK_SHIFT 8 +#define SGTL5000_PAD_I2S_LRCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_SCLK_MASK 0x00c0 +#define SGTL5000_PAD_I2S_SCLK_SHIFT 6 +#define SGTL5000_PAD_I2S_SCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_DOUT_MASK 0x0030 +#define SGTL5000_PAD_I2S_DOUT_SHIFT 4 +#define SGTL5000_PAD_I2S_DOUT_WIDTH 2 +#define SGTL5000_PAD_I2C_SDA_MASK 0x000c +#define SGTL5000_PAD_I2C_SDA_SHIFT 2 +#define SGTL5000_PAD_I2C_SDA_WIDTH 2 +#define SGTL5000_PAD_I2C_SCL_MASK 0x0003 +#define SGTL5000_PAD_I2C_SCL_SHIFT 0 +#define SGTL5000_PAD_I2C_SCL_WIDTH 2 + +/* + * SGTL5000_CHIP_ANA_ADC_CTRL + */ +#define SGTL5000_ADC_VOL_M6DB 0x0100 +#define SGTL5000_ADC_VOL_RIGHT_MASK 0x00f0 +#define SGTL5000_ADC_VOL_RIGHT_SHIFT 4 +#define SGTL5000_ADC_VOL_RIGHT_WIDTH 4 +#define SGTL5000_ADC_VOL_LEFT_MASK 0x000f +#define SGTL5000_ADC_VOL_LEFT_SHIFT 0 +#define SGTL5000_ADC_VOL_LEFT_WIDTH 4 + +/* + * SGTL5000_CHIP_ANA_HP_CTRL + */ +#define SGTL5000_HP_VOL_RIGHT_MASK 0x7f00 +#define SGTL5000_HP_VOL_RIGHT_SHIFT 8 +#define SGTL5000_HP_VOL_RIGHT_WIDTH 7 +#define SGTL5000_HP_VOL_LEFT_MASK 0x007f +#define SGTL5000_HP_VOL_LEFT_SHIFT 0 +#define SGTL5000_HP_VOL_LEFT_WIDTH 7 + +/* + * SGTL5000_CHIP_ANA_CTRL + */ +#define SGTL5000_LINE_OUT_MUTE 0x0100 +#define SGTL5000_HP_SEL_MASK 0x0040 +#define SGTL5000_HP_SEL_SHIFT 6 +#define SGTL5000_HP_SEL_WIDTH 1 +#define SGTL5000_HP_SEL_DAC 0x0 +#define SGTL5000_HP_SEL_LINE_IN 0x1 +#define SGTL5000_HP_ZCD_EN 0x0020 +#define SGTL5000_HP_MUTE 0x0010 +#define SGTL5000_ADC_SEL_MASK 0x0004 +#define SGTL5000_ADC_SEL_SHIFT 2 +#define SGTL5000_ADC_SEL_WIDTH 1 +#define SGTL5000_ADC_SEL_MIC 0x0 +#define SGTL5000_ADC_SEL_LINE_IN 0x1 +#define SGTL5000_ADC_ZCD_EN 0x0002 +#define SGTL5000_ADC_MUTE 0x0001 + +/* + * SGTL5000_CHIP_LINREG_CTRL + */ +#define SGTL5000_VDDC_MAN_ASSN_MASK 0x0040 +#define SGTL5000_VDDC_MAN_ASSN_SHIFT 6 +#define SGTL5000_VDDC_MAN_ASSN_WIDTH 1 +#define SGTL5000_VDDC_MAN_ASSN_VDDA 0x0 +#define SGTL5000_VDDC_MAN_ASSN_VDDIO 0x1 +#define SGTL5000_VDDC_ASSN_OVRD 0x0020 +#define SGTL5000_LINREG_VDDD_MASK 0x000f +#define SGTL5000_LINREG_VDDD_SHIFT 0 +#define SGTL5000_LINREG_VDDD_WIDTH 4 + +/* + * SGTL5000_CHIP_REF_CTRL + */ +#define SGTL5000_ANA_GND_MASK 0x01f0 +#define SGTL5000_ANA_GND_SHIFT 4 +#define SGTL5000_ANA_GND_WIDTH 5 +#define SGTL5000_ANA_GND_BASE 800 /* mv */ +#define SGTL5000_ANA_GND_STP 25 /*mv */ +#define SGTL5000_BIAS_CTRL_MASK 0x000e +#define SGTL5000_BIAS_CTRL_SHIFT 1 +#define SGTL5000_BIAS_CTRL_WIDTH 3 +#define SGTL5000_SMALL_POP 0x0001 + +/* + * SGTL5000_CHIP_MIC_CTRL + */ +#define SGTL5000_BIAS_R_MASK 0x0200 +#define SGTL5000_BIAS_R_SHIFT 8 +#define SGTL5000_BIAS_R_WIDTH 2 +#define SGTL5000_BIAS_R_off 0x0 +#define SGTL5000_BIAS_R_2K 0x1 +#define SGTL5000_BIAS_R_4k 0x2 +#define SGTL5000_BIAS_R_8k 0x3 +#define SGTL5000_BIAS_VOLT_MASK 0x0070 +#define SGTL5000_BIAS_VOLT_SHIFT 4 +#define SGTL5000_BIAS_VOLT_WIDTH 3 +#define SGTL5000_MIC_GAIN_MASK 0x0003 +#define SGTL5000_MIC_GAIN_SHIFT 0 +#define SGTL5000_MIC_GAIN_WIDTH 2 + +/* + * SGTL5000_CHIP_LINE_OUT_CTRL + */ +#define SGTL5000_LINE_OUT_CURRENT_MASK 0x0f00 +#define SGTL5000_LINE_OUT_CURRENT_SHIFT 8 +#define SGTL5000_LINE_OUT_CURRENT_WIDTH 4 +#define SGTL5000_LINE_OUT_CURRENT_180u 0x0 +#define SGTL5000_LINE_OUT_CURRENT_270u 0x1 +#define SGTL5000_LINE_OUT_CURRENT_360u 0x3 +#define SGTL5000_LINE_OUT_CURRENT_450u 0x7 +#define SGTL5000_LINE_OUT_CURRENT_540u 0xf +#define SGTL5000_LINE_OUT_GND_MASK 0x003f +#define SGTL5000_LINE_OUT_GND_SHIFT 0 +#define SGTL5000_LINE_OUT_GND_WIDTH 6 +#define SGTL5000_LINE_OUT_GND_BASE 800 /* mv */ +#define SGTL5000_LINE_OUT_GND_STP 25 +#define SGTL5000_LINE_OUT_GND_MAX 0x23 + +/* + * SGTL5000_CHIP_LINE_OUT_VOL + */ +#define SGTL5000_LINE_OUT_VOL_RIGHT_MASK 0x1f00 +#define SGTL5000_LINE_OUT_VOL_RIGHT_SHIFT 8 +#define SGTL5000_LINE_OUT_VOL_RIGHT_WIDTH 5 +#define SGTL5000_LINE_OUT_VOL_LEFT_MASK 0x001f +#define SGTL5000_LINE_OUT_VOL_LEFT_SHIFT 0 +#define SGTL5000_LINE_OUT_VOL_LEFT_WIDTH 5 + +/* + * SGTL5000_CHIP_ANA_POWER + */ +#define SGTL5000_DAC_STEREO 0x4000 +#define SGTL5000_LINREG_SIMPLE_POWERUP 0x2000 +#define SGTL5000_STARTUP_POWERUP 0x1000 +#define SGTL5000_VDDC_CHRGPMP_POWERUP 0x0800 +#define SGTL5000_PLL_POWERUP 0x0400 +#define SGTL5000_LINEREG_D_POWERUP 0x0200 +#define SGTL5000_VCOAMP_POWERUP 0x0100 +#define SGTL5000_VAG_POWERUP 0x0080 +#define SGTL5000_ADC_STEREO 0x0040 +#define SGTL5000_REFTOP_POWERUP 0x0020 +#define SGTL5000_HP_POWERUP 0x0010 +#define SGTL5000_DAC_POWERUP 0x0008 +#define SGTL5000_CAPLESS_HP_POWERUP 0x0004 +#define SGTL5000_ADC_POWERUP 0x0002 +#define SGTL5000_LINE_OUT_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_PLL_CTRL + */ +#define SGTL5000_PLL_INT_DIV_MASK 0xf800 +#define SGTL5000_PLL_INT_DIV_SHIFT 11 +#define SGTL5000_PLL_INT_DIV_WIDTH 5 +#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 +#define SGTL5000_PLL_FRAC_DIV_SHIFT 0 +#define SGTL5000_PLL_FRAC_DIV_WIDTH 11 + +/* + * SGTL5000_CHIP_CLK_TOP_CTRL + */ +#define SGTL5000_INT_OSC_EN 0x0800 +#define SGTL5000_INPUT_FREQ_DIV2 0x0008 + +/* + * SGTL5000_CHIP_ANA_STATUS + */ +#define SGTL5000_HP_LRSHORT 0x0200 +#define SGTL5000_CAPLESS_SHORT 0x0100 +#define SGTL5000_PLL_LOCKED 0x0010 + +/* + * SGTL5000_CHIP_SHORT_CTRL + */ +#define SGTL5000_LVLADJR_MASK 0x7000 +#define SGTL5000_LVLADJR_SHIFT 12 +#define SGTL5000_LVLADJR_WIDTH 3 +#define SGTL5000_LVLADJL_MASK 0x0700 +#define SGTL5000_LVLADJL_SHIFT 8 +#define SGTL5000_LVLADJL_WIDTH 3 +#define SGTL5000_LVLADJC_MASK 0x0070 +#define SGTL5000_LVLADJC_SHIFT 4 +#define SGTL5000_LVLADJC_WIDTH 3 +#define SGTL5000_LR_SHORT_MOD_MASK 0x000c +#define SGTL5000_LR_SHORT_MOD_SHIFT 2 +#define SGTL5000_LR_SHORT_MOD_WIDTH 2 +#define SGTL5000_CM_SHORT_MOD_MASK 0x0003 +#define SGTL5000_CM_SHORT_MOD_SHIFT 0 +#define SGTL5000_CM_SHORT_MOD_WIDTH 2 + +/* + *SGTL5000_CHIP_ANA_TEST2 + */ +#define SGTL5000_MONO_DAC 0x1000 + +/* + * SGTL5000_DAP_CTRL + */ +#define SGTL5000_DAP_MIX_EN 0x0010 +#define SGTL5000_DAP_EN 0x0001 + +#define SGTL5000_SYSCLK 0x00 +#define SGTL5000_LRCLK 0x01 + +#endif -- 1.7.0.4

4 4

[alsa-devel] [PATCH] ASoC: WM8580: Fix R8 initial value
by Jassi Brar 09 Dec '10

09 Dec '10

From: Seungwhan Youn <sw.youn(a)samsung.com> Acc to WM8580 manual, the default value for R8 is 0x10, not 0x1c. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/codecs/wm8580.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6e95332..6be373d 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -179,7 +179,7 @@ static const u16 wm8580_reg[] = { 0x0121, 0x017e, 0x007d, 0x0014, /*R3*/ 0x0121, 0x017e, 0x007d, 0x0194, /*R7*/ - 0x001c, 0x0002, 0x0002, 0x00c2, /*R11*/ + 0x0010, 0x0002, 0x0002, 0x00c2, /*R11*/ 0x0182, 0x0082, 0x000a, 0x0024, /*R15*/ 0x0009, 0x0000, 0x00ff, 0x0000, /*R19*/ 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R23*/ -- 1.6.2.5

4 3

[alsa-devel] [PATCH] ASoC: Remove unnecessary structure definitions
by Seungwhan Youn 09 Dec '10

09 Dec '10

This patch removes some legacy structure definitions which are not using in current ASoC drivers. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- include/sound/soc.h | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5c3bce8..6458c71 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -222,10 +222,8 @@ enum snd_soc_bias_level { struct snd_jack; struct snd_soc_card; -struct snd_soc_device; struct snd_soc_pcm_stream; struct snd_soc_ops; -struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; struct snd_soc_dai; struct snd_soc_dai_driver; @@ -235,7 +233,6 @@ struct snd_soc_platform_driver; struct snd_soc_codec; struct snd_soc_codec_driver; struct soc_enum; -struct snd_soc_ac97_ops; struct snd_soc_jack; struct snd_soc_jack_pin; -- 1.6.2.5

3 2

[alsa-devel] [PATCH] ASoC: fix deemphasis control in wm8904/55/60 codecs
by Dmitry Artamonow 09 Dec '10

09 Dec '10

Deemphasis control's .get callback should update control's value instead of returning it - return value of callback function is used for indicating error or success of operation. Signed-off-by: Dmitry Artamonow <mad_soft(a)inbox.ru> --- I discovered this while trying to adapt this deemphasis code to ak4641 codec. sound/soc/codecs/wm8904.c | 3 ++- sound/soc/codecs/wm8955.c | 3 ++- sound/soc/codecs/wm8960.c | 3 ++- 3 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index fca60a0..9001cc4 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -818,7 +818,8 @@ static int wm8904_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - return wm8904->deemph; + ucontrol->value.enumerated.item[0] = wm8904->deemph; + return 0; } static int wm8904_put_deemph(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index f89ad6c..9cbab8e 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -380,7 +380,8 @@ static int wm8955_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); - return wm8955->deemph; + ucontrol->value.enumerated.item[0] = wm8955->deemph; + return 0; } static int wm8955_put_deemph(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 8d5efb3..21986c4 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -138,7 +138,8 @@ static int wm8960_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - return wm8960->deemph; + ucontrol->value.enumerated.item[0] = wm8960->deemph; + return 0; } static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, -- 1.7.0.4

3 2

[alsa-devel] [PATCH v3 0/3] Add Jack support for
by Olaya, Margarita 09 Dec '10

09 Dec '10

This patch series adds the support for headset jack detection v3: Reorder patches to avoid building issues. Really add the jack detection in the codec 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 | 42 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 12 ++++++++++++ sound/soc/omap/sdp4430.c | 39 +++++++++++++++++++++++++++++++++++---- 3 files changed, 89 insertions(+), 4 deletions(-)

2 1

[alsa-devel] [PATCH 1/2] ASoC: tlv320dac33: Fix compillation error
by Peter Ujfalusi 09 Dec '10

09 Dec '10

Fix the compilation error introduced by patch: ASoC: tlv320dac33: Avoid multiple soft power up Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index a0ba5d1..d13fda0 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -638,6 +638,7 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { -- 1.7.3.2

3 5

[alsa-devel] [PATCH v2 1/2] ASoC: tlv320dac33: Rename outpup amplifier widget
by Peter Ujfalusi 09 Dec '10

09 Dec '10

Use better name for the widget, and remove the 'Power' from it's name. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index d13fda0..8e340a5 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -599,9 +599,9 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Analog Right Bypass", SND_SOC_NOPM, 0, 0, &dac33_dapm_abypassr_control), - SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amp Power", + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amplifier", DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amp Power", + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amplifier", DAC33_OUT_AMP_PWR_CTRL, 4, 3, 3, 0), SND_SOC_DAPM_PRE("Prepare Playback", playback_event), @@ -612,15 +612,15 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Analog Left Bypass", "Switch", "LINEL"}, {"Analog Right Bypass", "Switch", "LINER"}, - {"Output Left Amp Power", NULL, "DACL"}, - {"Output Right Amp Power", NULL, "DACR"}, + {"Output Left Amplifier", NULL, "DACL"}, + {"Output Right Amplifier", NULL, "DACR"}, - {"Output Left Amp Power", NULL, "Analog Left Bypass"}, - {"Output Right Amp Power", NULL, "Analog Right Bypass"}, + {"Output Left Amplifier", NULL, "Analog Left Bypass"}, + {"Output Right Amplifier", NULL, "Analog Right Bypass"}, /* output */ - {"LEFT_LO", NULL, "Output Left Amp Power"}, - {"RIGHT_LO", NULL, "Output Right Amp Power"}, + {"LEFT_LO", NULL, "Output Left Amplifier"}, + {"RIGHT_LO", NULL, "Output Right Amplifier"}, }; static int dac33_add_widgets(struct snd_soc_codec *codec) -- 1.7.3.2

3 3

[alsa-devel] [PATCH] [ALSA] ice1712 - working M-Audio Delta 66E support
by Brian Bloniarz 09 Dec '10

09 Dec '10

Rev. E of the M-Audio Delta 66 is partially supported (commit ef2cd2ccad66b4aba518eca7514eface267ee0f3), but the layout of the GPIO pins was still unclear. This patch adds the GPIO definitions so that communication to the CS8247 & 2x AK4524 works correctly. ALSA bug#3327 has more details; users cap & jhunt report there that the GPIO wiring is similar to the Digigram VX442 (chip select: pin 4 = CS8427, pin 5 = AK4524 #0, pin 6 = AK4524 #1). There has been a lot of conflicting information in the bug, but given these definitions, my Delta 66E works; I tested analog in&out at 44.1kHz & 96kHz, analog gain settings, S/PDIF clock sync, and S/PDIF in&out at 44.1kHz. Signed-off-by: Brian Bloniarz <brian.bloniarz(a)gmail.com> --- sound/pci/ice1712/delta.c | 49 ++++++++++++++++++++++++++++++++++++++++++++- sound/pci/ice1712/delta.h | 11 ++++++++++ 2 files changed, 59 insertions(+), 1 deletions(-) diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 712c171..7b62de0 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -96,6 +96,11 @@ static unsigned char ap_cs8427_codec_select(struct snd_ice1712 *ice) tmp |= ICE1712_DELTA_AP_CCLK | ICE1712_DELTA_AP_CS_CODEC; tmp &= ~ICE1712_DELTA_AP_CS_DIGITAL; break; + case ICE1712_SUBDEVICE_DELTA66E: + tmp |= ICE1712_DELTA_66E_CCLK | ICE1712_DELTA_66E_CS_CHIP_A | + ICE1712_DELTA_66E_CS_CHIP_B; + tmp &= ~ICE1712_DELTA_66E_CS_CS8427; + break; case ICE1712_SUBDEVICE_VX442: tmp |= ICE1712_VX442_CCLK | ICE1712_VX442_CODEC_CHIP_A | ICE1712_VX442_CODEC_CHIP_B; tmp &= ~ICE1712_VX442_CS_DIGITAL; @@ -119,6 +124,9 @@ static void ap_cs8427_codec_deassert(struct snd_ice1712 *ice, unsigned char tmp) case ICE1712_SUBDEVICE_DELTA410: tmp |= ICE1712_DELTA_AP_CS_DIGITAL; break; + case ICE1712_SUBDEVICE_DELTA66E: + tmp |= ICE1712_DELTA_66E_CS_CS8427; + break; case ICE1712_SUBDEVICE_VX442: tmp |= ICE1712_VX442_CS_DIGITAL; break; @@ -276,6 +284,20 @@ static void delta1010lt_ak4524_lock(struct snd_akm4xxx *ak, int chip) } /* + * AK4524 on Delta66 rev E to choose the chip address + */ +static void delta66e_ak4524_lock(struct snd_akm4xxx *ak, int chip) +{ + struct snd_ak4xxx_private *priv = (void *)ak->private_value[0]; + struct snd_ice1712 *ice = ak->private_data[0]; + + snd_ice1712_save_gpio_status(ice); + priv->cs_mask = + priv->cs_addr = chip == 0 ? ICE1712_DELTA_66E_CS_CHIP_A : + ICE1712_DELTA_66E_CS_CHIP_B; +} + +/* * AK4528 on VX442 to choose the chip mask */ static void vx442_ak4524_lock(struct snd_akm4xxx *ak, int chip) @@ -487,6 +509,29 @@ static struct snd_ak4xxx_private akm_delta1010lt_priv __devinitdata = { .mask_flags = 0, }; +static struct snd_akm4xxx akm_delta66e __devinitdata = { + .type = SND_AK4524, + .num_adcs = 4, + .num_dacs = 4, + .ops = { + .lock = delta66e_ak4524_lock, + .set_rate_val = delta_ak4524_set_rate_val + } +}; + +static struct snd_ak4xxx_private akm_delta66e_priv __devinitdata = { + .caddr = 2, + .cif = 0, /* the default level of the CIF pin from AK4524 */ + .data_mask = ICE1712_DELTA_66E_DOUT, + .clk_mask = ICE1712_DELTA_66E_CCLK, + .cs_mask = 0, + .cs_addr = 0, /* set later */ + .cs_none = 0, + .add_flags = 0, + .mask_flags = 0, +}; + + static struct snd_akm4xxx akm_delta44 __devinitdata = { .type = SND_AK4524, .num_adcs = 4, @@ -644,9 +689,11 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_init(ak, &akm_delta44, &akm_delta44_priv, ice); break; case ICE1712_SUBDEVICE_VX442: - case ICE1712_SUBDEVICE_DELTA66E: err = snd_ice1712_akm4xxx_init(ak, &akm_vx442, &akm_vx442_priv, ice); break; + case ICE1712_SUBDEVICE_DELTA66E: + err = snd_ice1712_akm4xxx_init(ak, &akm_delta66e, &akm_delta66e_priv, ice); + break; default: snd_BUG(); return -EINVAL; diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h index 1a0ac6c..11a9c3a 100644 --- a/sound/pci/ice1712/delta.h +++ b/sound/pci/ice1712/delta.h @@ -144,6 +144,17 @@ extern struct snd_ice1712_card_info snd_ice1712_delta_cards[]; #define ICE1712_DELTA_1010LT_CS_NONE 0x50 /* nothing */ #define ICE1712_DELTA_1010LT_WORDCLOCK 0x80 /* sample clock source: 0 = Word Clock Input, 1 = S/PDIF Input ??? */ +/* M-Audio Delta 66 rev. E definitions. + * Newer revisions of Delta 66 have CS8427 over SPI for + * S/PDIF transceiver instead of CS8404/CS8414. */ +/* 0x01 = DFS */ +#define ICE1712_DELTA_66E_CCLK 0x02 /* SPI clock */ +#define ICE1712_DELTA_66E_DIN 0x04 /* data input */ +#define ICE1712_DELTA_66E_DOUT 0x08 /* data output */ +#define ICE1712_DELTA_66E_CS_CS8427 0x10 /* chip select, low = CS8427 */ +#define ICE1712_DELTA_66E_CS_CHIP_A 0x20 /* AK4524 #0 */ +#define ICE1712_DELTA_66E_CS_CHIP_B 0x40 /* AK4524 #1 */ + /* Digigram VX442 definitions */ #define ICE1712_VX442_CCLK 0x02 /* SPI clock */ #define ICE1712_VX442_DIN 0x04 /* data input */ -- 1.7.1

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[alsa-devel] [PATCH] ALSA: HDA: Quirk for Dell Vostro 320 to make microphone work
by David Henningsson 09 Dec '10

09 Dec '10

BugLink: http://launchpad.net/497546 Confirmed that the ideapad model works better than the current quirk for Dell Vostro 320. With this patch, there is noone left using the DELL_LAPTOP model, should we remove it completely? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

1 0

[PATCH] ALSA: HDA: Quirk for Dell Vostro 320 to make microphone work
by David Henningsson 09 Dec '10

09 Dec '10

BugLink: http://launchpad.net/497546 Confirmed that the ideapad model works better than the current quirk for Dell Vostro 320. Cc: stable(a)kernel.org (2.6.35+) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 46bbf14..ee9c3ed 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3097,8 +3097,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", - CXT5066_DELL_LAPTOP), + SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), -- 1.7.1 --------------070307010403030000030409--

1 0

[alsa-devel] [PATCH v3 3/3] ASoC: sdp4430: Add Jack support
by Olaya, Margarita 08 Dec '10

08 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Use jack framework to enable detection for the headset microphone and stereo output. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/omap/sdp4430.c | 28 ++++++++++++++++++++++++++++ 1 files changed, 28 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 0c37c51..ca7fb90 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -24,6 +24,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <plat/hardware.h> @@ -65,6 +66,21 @@ static struct snd_soc_ops sdp4430_ops = { .hw_params = sdp4430_hw_params, }; +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -160,7 +176,19 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); ret = snd_soc_dapm_sync(dapm); + if (ret) + return ret; + + /* Headset jack detection */ + ret = snd_soc_jack_new(codec, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); return ret; } -- 1.7.0.4

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[alsa-devel] [PATCH v3 2/3] ASoC: twl6040: Add jack support for headset and handset
by Olaya, Margarita 08 Dec '10

08 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> This patch adds support for reporting twl6040 headset and handset jack events. The machine driver retrieves and report the status through twl6040_hs_jack_detect in case of the early interrupt. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 42 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 12 ++++++++++++ 2 files changed, 54 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b92f2b7..c3799d7 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -42,10 +42,16 @@ #define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) +struct twl6040_jack_data { + struct snd_soc_jack *jack; + int report; +}; + /* codec private data */ struct twl6040_data { int audpwron; int naudint; + struct twl6040_jack_data hs_jack; int codec_powered; int pll; int non_lp; @@ -386,6 +392,8 @@ static irqreturn_t twl6040_naudint_handler(int irq, void *data) { struct snd_soc_codec *codec = data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_jack_data *jack = &priv->hs_jack; + int report = 0; u8 intid; twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); @@ -395,7 +403,23 @@ static irqreturn_t twl6040_naudint_handler(int irq, void *data) dev_alert(codec->dev, "die temp over-limit detection\n"); break; case TWL6040_PLUGINT: + /* Debounce */ + msleep(200); + report = jack->report; + /* + * Early interrupt, CODEC driver cannot report jack status + * since jack is not registered yet. MACHINE driver will + * register jack and report status thru twl6040_hs_jack_detect + */ + if (jack->jack) + snd_soc_jack_report(jack->jack, report, jack->report); + break; case TWL6040_UNPLUGINT: + /* Debounce */ + msleep(200); + if (jack->jack) + snd_soc_jack_report(jack->jack, report, jack->report); + break; case TWL6040_HOOKINT: break; case TWL6040_HFINT: @@ -1011,6 +1035,24 @@ static int twl6040_resume(struct snd_soc_codec *codec) #define twl6040_resume NULL #endif +void twl6040_hs_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int report) +{ + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + int status; + + priv->hs_jack.jack = jack; + priv->hs_jack.report = report; + + /* Sync status */ + status = twl6040_read_reg_volatile(codec, TWL6040_REG_STATUS); + if (status & TWL6040_PLUGCOMP) + snd_soc_jack_report(jack, report, report); + else + snd_soc_jack_report(jack, 0, report); +} +EXPORT_SYMBOL_GPL(twl6040_hs_jack_detect); + static int twl6040_probe(struct snd_soc_codec *codec) { struct twl4030_codec_data *twl_codec = codec->dev->platform_data; diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h index f7c77fa..5456c18 100644 --- a/sound/soc/codecs/twl6040.h +++ b/sound/soc/codecs/twl6040.h @@ -135,4 +135,16 @@ #define TWL6040_HPPLL_ID 1 #define TWL6040_LPPLL_ID 2 +/* STATUS (0x2E) fields */ + +#define TWL6040_PLUGCOMP 0x02 + +struct twl6040_setup_data { + void (*codec_enable)(int enable); + void *jack; +}; + +void twl6040_hs_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int report); + #endif /* End of __TWL6040_H__ */ -- 1.7.0.4

2 1

[alsa-devel] [PATCH v3 1/3] ASoC: sdp4430: Enable FM stereo pins
by Olaya, Margarita 08 Dec '10

08 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Add FM stereo pins to the machine driver and add them as a dapm widget. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/omap/sdp4430.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index ebbd62f..0c37c51 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -101,6 +101,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), + SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -123,6 +124,10 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Earphone speaker */ {"Earphone Spk", NULL, "EP"}, + + /* Aux/FM Stereo In: AFML, AFMR */ + {"AFML", NULL, "Aux/FM Stereo In"}, + {"AFMR", NULL, "Aux/FM Stereo In"}, }; static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) @@ -149,13 +154,11 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) /* SDP4430 connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Mic"); snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "AFML"); + snd_soc_dapm_enable_pin(dapm, "AFMR"); snd_soc_dapm_enable_pin(dapm, "Headset Mic"); snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - /* TWL6040 not connected pins */ - snd_soc_dapm_nc_pin(dapm, "AFML"); - snd_soc_dapm_nc_pin(dapm, "AFMR"); - ret = snd_soc_dapm_sync(dapm); return ret; -- 1.7.0.4

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[alsa-devel] ASoC updaets for 2.6.38
by Mark Brown 08 Dec '10

08 Dec '10

The following changes since commit 0afc8c733e95642ee9200966081da82564af8f8f: Merge branch 'for-2.6.37' into for-2.6.38 (2010-12-06 14:14:47 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Axel Lin (2): ASoC: Fix resource leak if soc_register_ac97_dai_link failed ASoC: Fix snd_soc_instantiate_card error path Dimitris Papastamos (3): ASoC: soc-core: Replace use of strncpy() with strlcpy() ASoC: soc-core: Remove useless inline function construct ASoC: soc-core: Fix null pointer dereference Jarkko Nikula (1): ASoC: Merge common code in DAI link and auxiliary codec probing/removal Jassi Brar (1): ASoC: WM8580: Debug BCLK and sample size Mark Brown (3): ASoC: Correct WM8962 interrupt mask register read Merge branch 'for-2.6.37' into for-2.6.38 ASoC: Support WM8994 mono AIF configurations Uk Kim (1): ASoC: Add ADC high pass filter support to WM8994 sound/soc/codecs/wm8580.c | 8 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8994.c | 43 ++++++- sound/soc/soc-core.c | 308 +++++++++++++++++++++------------------------ 4 files changed, 187 insertions(+), 174 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.37
by Mark Brown 08 Dec '10

08 Dec '10

The following changes since commit 1dcb4f38e5bc28dfce0f8c7eef184a090b03bfc7: ASoC: Hold client_mutex while calling snd_soc_instantiate_cards() (2010-12-06 12:53:43 +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 Axel Lin (1): ASoC: Fix resource leak if soc_register_ac97_dai_link failed Jassi Brar (1): ASoC: WM8580: Debug BCLK and sample size Mark Brown (1): ASoC: Correct WM8962 interrupt mask register read More of the same small bug fixes. There's a conflict with 2.6.38 which I've already resolved - I'll send a pull request for that later. sound/soc/codecs/wm8580.c | 8 ++++---- sound/soc/codecs/wm8962.c | 2 +- sound/soc/soc-core.c | 12 +++++++----- 3 files changed, 12 insertions(+), 10 deletions(-)

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[alsa-devel] [PATCH 2/3] ASoC: CS4271 codec support
by Alexander 08 Dec '10

08 Dec '10

From: Alexander Sverdlin <subaparts(a)yandex.ru> Added support for CS4271 codec to ASoC. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- include/sound/cs4271.h | 25 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4271.c | 607 +++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 638 insertions(+), 0 deletions(-) diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h new file mode 100644 index 0000000..16f8d32 --- /dev/null +++ b/include/sound/cs4271.h @@ -0,0 +1,25 @@ +/* + * Definitions for CS4271 ASoC codec driver + * + * Copyright (c) 2010 Alexander Sverdlin <subaparts(a)yandex.ru> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __CS4271_H +#define __CS4271_H + +struct cs4271_platform_data { + int gpio_nreset; /* GPIO driving Reset pin, if any */ + int gpio_disable; /* GPIO that disable serial bus, if any */ +}; + +#endif /* __CS4271_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0f33db2..b7d7d3c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 @@ -154,6 +155,9 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_CS4271 + tristate + config SND_SOC_CX20442 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 10e5e09..ec8d0ff 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -12,6 +12,7 @@ snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs4270-objs := cs4270.o +snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-l3-objs := l3.o @@ -89,6 +90,7 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c new file mode 100644 index 0000000..c0d74da --- /dev/null +++ b/sound/soc/codecs/cs4271.c @@ -0,0 +1,607 @@ +/* + * CS4271 ASoC codec driver + * + * Copyright (c) 2010 Alexander Sverdlin <subaparts(a)yandex.ru> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <sound/cs4271.h> + +#define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* + * CS4271 registers + * High byte represents SPI chip address (0x10) + write command (0) + * Low byte - codec register address + */ +#define CS4271_MODE1 0x2001 /* Mode Control 1 */ +#define CS4271_DACCTL 0x2002 /* DAC Control */ +#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x2006 /* ADC Control */ +#define CS4271_MODE2 0x2007 /* Mode Control 2 */ +#define CS4271_CHIPID 0x2008 /* Chip ID */ + +#define CS4271_FIRSTREG CS4271_MODE1 +#define CS4271_LASTREG CS4271_MODE2 +#define CS4271_NR_REGS ((CS4271_LASTREG & 0xFF) + 1) + +/* Bit masks for the CS4271 registers */ +#define CS4271_MODE1_MODE_MASK 0xC0 +#define CS4271_MODE1_MODE_1X 0x00 +#define CS4271_MODE1_MODE_2X 0x80 +#define CS4271_MODE1_MODE_4X 0xC0 + +#define CS4271_MODE1_DIV_MASK 0x30 +#define CS4271_MODE1_DIV_1 0x00 +#define CS4271_MODE1_DIV_15 0x10 +#define CS4271_MODE1_DIV_2 0x20 +#define CS4271_MODE1_DIV_3 0x30 + +#define CS4271_MODE1_MASTER 0x08 + +#define CS4271_MODE1_DAC_DIF_MASK 0x07 +#define CS4271_MODE1_DAC_DIF_LJ 0x00 +#define CS4271_MODE1_DAC_DIF_I2S 0x01 +#define CS4271_MODE1_DAC_DIF_RJ16 0x02 +#define CS4271_MODE1_DAC_DIF_RJ24 0x03 +#define CS4271_MODE1_DAC_DIF_RJ20 0x04 +#define CS4271_MODE1_DAC_DIF_RJ18 0x05 + +#define CS4271_DACCTL_AMUTE 0x80 +#define CS4271_DACCTL_IF_SLOW 0x40 + +#define CS4271_DACCTL_DEM_MASK 0x30 +#define CS4271_DACCTL_DEM_DIS 0x00 +#define CS4271_DACCTL_DEM_441 0x10 +#define CS4271_DACCTL_DEM_48 0x20 +#define CS4271_DACCTL_DEM_32 0x30 + +#define CS4271_DACCTL_SVRU 0x08 +#define CS4271_DACCTL_SRD 0x04 +#define CS4271_DACCTL_INVA 0x02 +#define CS4271_DACCTL_INVB 0x01 + +#define CS4271_DACVOL_BEQUA 0x40 +#define CS4271_DACVOL_SOFT 0x20 +#define CS4271_DACVOL_ZEROC 0x10 + +#define CS4271_DACVOL_ATAPI_MASK 0x0F +#define CS4271_DACVOL_ATAPI_M_M 0x00 +#define CS4271_DACVOL_ATAPI_M_BR 0x01 +#define CS4271_DACVOL_ATAPI_M_BL 0x02 +#define CS4271_DACVOL_ATAPI_M_BLR2 0x03 +#define CS4271_DACVOL_ATAPI_AR_M 0x04 +#define CS4271_DACVOL_ATAPI_AR_BR 0x05 +#define CS4271_DACVOL_ATAPI_AR_BL 0x06 +#define CS4271_DACVOL_ATAPI_AR_BLR2 0x07 +#define CS4271_DACVOL_ATAPI_AL_M 0x08 +#define CS4271_DACVOL_ATAPI_AL_BR 0x09 +#define CS4271_DACVOL_ATAPI_AL_BL 0x0A +#define CS4271_DACVOL_ATAPI_AL_BLR2 0x0B +#define CS4271_DACVOL_ATAPI_ALR2_M 0x0C +#define CS4271_DACVOL_ATAPI_ALR2_BR 0x0D +#define CS4271_DACVOL_ATAPI_ALR2_BL 0x0E +#define CS4271_DACVOL_ATAPI_ALR2_BLR2 0x0F + +#define CS4271_VOLA_MUTE 0x80 +#define CS4271_VOLA_VOL_MASK 0x7F +#define CS4271_VOLB_MUTE 0x80 +#define CS4271_VOLB_VOL_MASK 0x7F + +#define CS4271_ADCCTL_DITHER16 0x20 + +#define CS4271_ADCCTL_ADC_DIF_MASK 0x10 +#define CS4271_ADCCTL_ADC_DIF_LJ 0x00 +#define CS4271_ADCCTL_ADC_DIF_I2S 0x10 + +#define CS4271_ADCCTL_MUTEA 0x08 +#define CS4271_ADCCTL_MUTEB 0x04 +#define CS4271_ADCCTL_HPFDA 0x02 +#define CS4271_ADCCTL_HPFDB 0x01 + +#define CS4271_MODE2_LOOP 0x10 +#define CS4271_MODE2_MUTECAEQUB 0x08 +#define CS4271_MODE2_FREEZE 0x04 +#define CS4271_MODE2_CPEN 0x02 +#define CS4271_MODE2_PDN 0x01 + +#define CS4271_CHIPID_PART_MASK 0xF0 +#define CS4271_CHIPID_REV_MASK 0x0F + +/* CS4271 default register settings, except auto-mute is off */ +static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { + 0, 0, 0, CS4271_DACVOL_ATAPI_AL_BR, 0, 0, 0, + CS4271_MODE2_CPEN | CS4271_MODE2_PDN, +}; + +struct cs4271_private { + /* SND_SOC_I2C or SND_SOC_SPI */ + enum snd_soc_control_type bus_type; + void *control_data; + unsigned int mclk; + /* SND_SOC_DAIFMT_I2S or SND_SOC_DAIFMT_LEFT_J */ + unsigned int mode; + unsigned int master; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; + /* GPIO that disable serial bus, if any */ + int gpio_disable; +}; + +struct cs4271_clk_cfg { + unsigned int ratio; /* MCLK / sample rate */ + u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ + u8 mclk_master; /* ratio bit mask for Master mode */ + u8 mclk_slave; /* ratio bit mask for Slave mode */ +}; + +static struct cs4271_clk_cfg cs4271_clk_tab[] = { + {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1}, + {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}, + {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3} +}; + +#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) + +/* + * @freq is the desired MCLK rate + * MCLK rate should (c) be the sample rate, multiplied by one of the + * ratios listed in cs4271_mclk_fs_ratios table + */ +static int cs4271_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->mclk = freq; + return 0; +} + +static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + case SND_SOC_DAIFMT_CBS_CFS: + cs4271->master = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs4271->master = 1; + } + + return 0; +} + +static int cs4271_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + unsigned int i, ratio, val; + + ratio = cs4271->mclk / params_rate(params); + for (i = 0; i < CS4171_NR_RATIOS; i++) + if (cs4271_clk_tab[i].ratio == ratio) + break; + + if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + /* Configure DAC */ + val = cs4271_clk_tab[i].speed_mode; + + if (cs4271->master) + val |= cs4271_clk_tab[i].mclk_master | CS4271_MODE1_MASTER; + else + val |= cs4271_clk_tab[i].mclk_slave; + + switch (cs4271->mode) { + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + val |= CS4271_MODE1_DAC_DIF_LJ; + break; + case SND_SOC_DAIFMT_I2S: + val |= CS4271_MODE1_DAC_DIF_I2S; + } + + snd_soc_update_bits(codec, CS4271_MODE1, + CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK | + CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); + + /* Configure ADC */ + snd_soc_update_bits(codec, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, + (cs4271->mode == SND_SOC_DAIFMT_I2S) ? + CS4271_ADCCTL_ADC_DIF_I2S : CS4271_ADCCTL_ADC_DIF_LJ); + + return 0; +} + +static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int val_a = 0; + int val_b = 0; + + if (mute) { + val_a = CS4271_VOLA_MUTE; + val_b = CS4271_VOLB_MUTE; + } + + snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + return 0; +} + +/* CS4271 controls */ +static DECLARE_TLV_DB_SCALE(cs4271_dac_tlv, -12700, 100, 0); + +static const char *cs4271_de_texts[] = {"None", "44.1KHz", "48KHz", "32KHz"}; +static const struct soc_enum cs4271_de_enum = + SOC_ENUM_SINGLE(CS4271_DACCTL, 4, 4, cs4271_de_texts); + +static const struct snd_kcontrol_new cs4271_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", CS4271_VOLA, CS4271_VOLB, + 0, 0x7F, 1, cs4271_dac_tlv), + SOC_SINGLE("Digital Loopback Switch", CS4271_MODE2, 4, 1, 0), + SOC_SINGLE("Soft Ramp Switch", CS4271_DACVOL, 5, 1, 0), + SOC_SINGLE("Zero Cross Switch", CS4271_DACVOL, 4, 1, 0), + SOC_ENUM("De-emphasis filter", cs4271_de_enum), + SOC_SINGLE("Auto-Mute Switch", CS4271_DACCTL, 7, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4271_DACCTL, 6, 1, 0), + SOC_SINGLE("Soft Volume Ramp-Up Switch", CS4271_DACCTL, 3, 1, 0), + SOC_SINGLE("Soft Ramp-Down Switch", CS4271_DACCTL, 2, 1, 0), + SOC_SINGLE("Left Channel Inversion Switch", CS4271_DACCTL, 1, 1, 0), + SOC_SINGLE("Right Channel Inversion Switch", CS4271_DACCTL, 0, 1, 0), + SOC_DOUBLE("Master Capture Switch", CS4271_ADCCTL, 3, 2, 1, 1), + SOC_SINGLE("Dither 16-Bit Data Switch", CS4271_ADCCTL, 5, 1, 0), + SOC_DOUBLE("High Pass Filter Switch", CS4271_ADCCTL, 1, 0, 1, 1), + SOC_DOUBLE_R("Master Playback Switch", CS4271_VOLA, CS4271_VOLB, + 7, 1, 1), +}; + +static struct snd_soc_dai_ops cs4271_dai_ops = { + .hw_params = cs4271_hw_params, + .set_sysclk = cs4271_set_dai_sysclk, + .set_fmt = cs4271_set_dai_fmt, + .digital_mute = cs4271_digital_mute, +}; + +struct snd_soc_dai_driver cs4271_dai = { + .name = "cs4271-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .rate_min = 8000, + .rate_max = 96000, + .formats = CS4271_PCM_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .rate_min = 8000, + .rate_max = 96000, + .formats = CS4271_PCM_FORMATS, + }, + .ops = &cs4271_dai_ops, +}; + +/* + * This function writes all writeble registers from cache to codec. + * It's used to setup initial config and restore after suspend. + */ +static int cs4271_write_cache(struct snd_soc_codec *codec) +{ + int i, ret; + + for (i = CS4271_LASTREG; i >= CS4271_FIRSTREG; i--) { + ret = snd_soc_write(codec, i, snd_soc_read(codec, i)); + if (ret) { + dev_err(codec->dev, "Cache write failed\n"); + return ret; + } + } + + return 0; +} + +#ifdef CONFIG_PM +static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + /* Set power-down bit */ + snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, + CS4271_MODE2_PDN); + + return 0; +} +#else +#define cs4271_soc_suspend NULL +#endif /* CONFIG_PM */ + +static int cs4271_soc_resume(struct snd_soc_codec *codec) +{ + int ret; + + /* Restore codec state */ + ret = cs4271_write_cache(codec); + if (ret < 0) + return ret; + + /* then disable the power-down bit */ + snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + + return 0; +} + +static int cs4271_probe(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; + int ret; + int gpio_nreset = -EINVAL; + int gpio_disable = -EINVAL; + + codec->control_data = cs4271->control_data; + + if (cs4271plat) { + if (gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; + if (gpio_is_valid(cs4271plat->gpio_disable)) + gpio_disable = cs4271plat->gpio_disable; + } + + if ((gpio_disable >= 0) && + gpio_request(gpio_disable, "CS4271 Disable")) + gpio_disable = -EINVAL; + if (gpio_disable >= 0) + gpio_direction_output(gpio_disable, 0); + + if ((gpio_nreset >= 0) && + gpio_request(gpio_nreset, "CS4271 Reset")) + gpio_nreset = -EINVAL; + if (gpio_nreset >= 0) { + gpio_direction_output(gpio_nreset, 1); + /* Give the codec time to wake up */ + udelay(1); + } + + cs4271->gpio_nreset = gpio_nreset; + cs4271->gpio_disable = gpio_disable; + + /* + * In case of I2C, chip address specified in board data. + * So cache IO operations use 8 bit codec register address. + * In case of SPI, chip address and register address + * passed together as 16 bit value. + * Anyway, register address is masked with 0xFF inside + * soc-cache code. + */ + ret = snd_soc_codec_set_cache_io(codec, + (cs4271->bus_type == SND_SOC_SPI) ? 16 : 8, 8, + cs4271->bus_type); + if (ret) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + ret = cs4271_soc_resume(codec); + if (ret < 0) + return ret; + + return snd_soc_add_controls(codec, cs4271_snd_controls, + ARRAY_SIZE(cs4271_snd_controls)); +} + +static int cs4271_remove(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + int gpio_nreset, gpio_disable; + + gpio_nreset = cs4271->gpio_nreset; + gpio_disable = cs4271->gpio_disable; + + if (gpio_is_valid(gpio_nreset)) { + /* Set codec to the reset state */ + gpio_set_value(gpio_nreset, 0); + gpio_free(gpio_nreset); + } + + if (gpio_is_valid(gpio_disable)) + gpio_free(gpio_disable); + + return 0; +}; + +struct snd_soc_codec_driver soc_codec_dev_cs4271 = { + .probe = cs4271_probe, + .remove = cs4271_remove, + .suspend = cs4271_soc_suspend, + .resume = cs4271_soc_resume, + .reg_cache_default = &cs4271_dflt_reg, + .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), + .reg_word_size = sizeof(cs4271_dflt_reg[0]), + .compress_type = SND_SOC_FLAT_COMPRESSION, + .reg_cache_step = 1, +}; + +#if defined(CONFIG_SPI_MASTER) +static int __devinit cs4271_spi_probe(struct spi_device *spi) +{ + struct cs4271_private *cs4271; + int ret; + + cs4271 = kzalloc(sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + spi_set_drvdata(spi, cs4271); + cs4271->control_data = spi; + cs4271->bus_type = SND_SOC_SPI; + + ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, + &cs4271_dai, 1); + if (ret < 0) + kfree(cs4271); + return ret; +} + +static int __devexit cs4271_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + kfree(spi_get_drvdata(spi)); + return 0; +} + +static struct spi_driver cs4271_spi_driver = { + .driver = { + .name = "cs4271-codec", + .owner = THIS_MODULE, + }, + .probe = cs4271_spi_probe, + .remove = __devexit_p(cs4271_spi_remove), +}; +#endif /* defined(CONFIG_SPI_MASTER) */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static struct i2c_device_id cs4271_i2c_id[] = { + {"cs4271", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); + +static int __devinit cs4271_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4271_private *cs4271; + int ret; + + cs4271 = kzalloc(sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + i2c_set_clientdata(client, cs4271); + cs4271->control_data = client; + cs4271->bus_type = SND_SOC_I2C; + + ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, + &cs4271_dai, 1); + if (ret < 0) + kfree(cs4271); + return ret; +} + +static int __devexit cs4271_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static struct i2c_driver cs4271_i2c_driver = { + .driver = { + .name = "cs4271-codec", + .owner = THIS_MODULE, + }, + .id_table = cs4271_i2c_id, + .probe = cs4271_i2c_probe, + .remove = __devexit_p(cs4271_i2c_remove), +}; +#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */ + +/* + * We only register our serial bus driver here without + * assignment to particular chip. So if any of the below + * fails, there is some problem with I2C or SPI subsystem. + * In most cases this module will be compiled with support + * of only one serial bus. + */ +static int __init cs4271_modinit(void) +{ + int ret; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&cs4271_i2c_driver); + if (ret) { + pr_err("Failed to register CS4271 I2C driver: %d\n", ret); + return ret; + } +#endif + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&cs4271_spi_driver); + if (ret) { + pr_err("Failed to register CS4271 SPI driver: %d\n", ret); + return ret; + } +#endif + + return 0; +} +module_init(cs4271_modinit); + +static void __exit cs4271_modexit(void) +{ +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&cs4271_spi_driver); +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&cs4271_i2c_driver); +#endif +} +module_exit(cs4271_modexit); + +MODULE_AUTHOR("Alexander Sverdlin <subaparts(a)yandex.ru>"); +MODULE_DESCRIPTION("Cirrus Logic CS4271 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL");

3 7

[alsa-devel] [PATCH] ASoC: soc-core: Fix null pointer dereference
by Dimitris Papastamos 08 Dec '10

08 Dec '10

In case the codec driver did not provide a read/write function, codec->driver->read|write will be NULL. Ensure that we use the one specified in codec->read|write to avoid oopsing when we access the debugfs entries. This is achieved by using snd_soc_read() and snd_soc_write(). Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 392d336..ecd9146 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -117,7 +117,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) * the register being volatile and the device being * powered off. */ - ret = codec->driver->read(codec, i); + ret = snd_soc_read(codec, i); if (ret >= 0) count += snprintf(buf + count, PAGE_SIZE - count, @@ -228,7 +228,7 @@ static ssize_t codec_reg_write_file(struct file *file, start++; if (strict_strtoul(start, 16, &value)) return -EINVAL; - codec->driver->write(codec, reg, value); + snd_soc_write(codec, reg, value); return buf_size; } -- 1.7.3.3

3 2

[alsa-devel] [PATCH 3/3] ASoC: EDB93xx machine sound driver
by Alexander 08 Dec '10

08 Dec '10

From: Alexander Sverdlin <subaparts(a)yandex.ru> Added support for EDB93xx sound with CS4271 codec. Features: - Playback, Capture - Sample rates from 8kHz to 96kHz tested Limitations: - Depends on ALSA auto sample format conversion - No DAPM support Depends on: CS4271 SoC codec support patch, EP93xx IS2 and PCM fixes patch Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- arch/arm/mach-ep93xx/edb93xx.c | 54 +++++++++++++++ sound/soc/ep93xx/Kconfig | 9 +++ sound/soc/ep93xx/Makefile | 2 + sound/soc/ep93xx/edb93xx.c | 142 ++++++++++++++++++++++++++++++++++++++++ 4 files changed, 207 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ep93xx/edb93xx.c b/arch/arm/mach-ep93xx/edb93xx.c index 4b04316..f453e5a 100644 --- a/arch/arm/mach-ep93xx/edb93xx.c +++ b/arch/arm/mach-ep93xx/edb93xx.c @@ -30,12 +30,21 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/i2c-gpio.h> +#include <linux/spi/spi.h> +#include <mach/ep93xx_spi.h> #include <mach/hardware.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> +#include <sound/cs4271.h> + +#define edb93xx_has_audio() (machine_is_edb9301() || \ + machine_is_edb9302() || \ + machine_is_edb9302a() || \ + machine_is_edb9307a() || \ + machine_is_edb9315a()) static void __init edb93xx_register_flash(void) { @@ -93,6 +102,49 @@ static void __init edb93xx_register_i2c(void) /************************************************************************* + * EDB93xx SPI peripheral handling + *************************************************************************/ +static struct cs4271_platform_data cs4271_edb93xx_gpio = { + .gpio_nreset = EP93XX_GPIO_LINE_EGPIO1, + .gpio_disable = EP93XX_GPIO_LINE_EGPIO6, +}; + +static struct spi_board_info edb93xx_spi_board_info[] = { + { + .modalias = "cs4271-codec", + .max_speed_hz = 6000000, + .bus_num = 0, + .chip_select = 0, + .mode = SPI_MODE_3, + .platform_data = &cs4271_edb93xx_gpio, + }, +}; + +static struct ep93xx_spi_info edb93xx_spi_info = { + .num_chipselect = ARRAY_SIZE(edb93xx_spi_board_info), +}; + +static void __init edb93xx_register_spi(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_spi(&edb93xx_spi_info, + edb93xx_spi_board_info, + ARRAY_SIZE(edb93xx_spi_board_info)); + } +} + +/************************************************************************* + * EDB93xx I2S + *************************************************************************/ +static void __init edb93xx_register_i2s(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_i2s(); + } +} + + +/************************************************************************* * EDB93xx pwm *************************************************************************/ static void __init edb93xx_register_pwm(void) @@ -117,6 +169,8 @@ static void __init edb93xx_init_machine(void) edb93xx_register_flash(); ep93xx_register_eth(&edb93xx_eth_data, 1); edb93xx_register_i2c(); + edb93xx_register_spi(); + edb93xx_register_i2s(); edb93xx_register_pwm(); } diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig index 5742904..91a28de 100644 --- a/sound/soc/ep93xx/Kconfig +++ b/sound/soc/ep93xx/Kconfig @@ -30,3 +30,12 @@ config SND_EP93XX_SOC_SIMONE help Say Y or M here if you want to add support for AC97 audio on the Simplemachines Sim.One board. + +config SND_EP93XX_SOC_EDB93XX + tristate "SoC Audio support for Cirrus Logic EDB93xx boards" + depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A) + select SND_EP93XX_SOC_I2S + select SND_SOC_CS4271 + help + Say Y or M here if you want to add support for I2S audio on the + Cirrus Logic EDB93xx boards. diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile index 8e7977f..5514146 100644 --- a/sound/soc/ep93xx/Makefile +++ b/sound/soc/ep93xx/Makefile @@ -10,6 +10,8 @@ obj-$(CONFIG_SND_EP93XX_SOC_AC97) += snd-soc-ep93xx-ac97.o # EP93XX Machine Support snd-soc-snappercl15-objs := snappercl15.o snd-soc-simone-objs := simone.o +snd-soc-edb93xx-objs := edb93xx.o obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15) += snd-soc-snappercl15.o obj-$(CONFIG_SND_EP93XX_SOC_SIMONE) += snd-soc-simone.o +obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX) += snd-soc-edb93xx.o diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c new file mode 100644 index 0000000..b270085 --- /dev/null +++ b/sound/soc/ep93xx/edb93xx.c @@ -0,0 +1,142 @@ +/* + * SoC audio for EDB93xx + * + * Copyright (c) 2010 Alexander Sverdlin <subaparts(a)yandex.ru> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include "ep93xx-pcm.h" + +#define edb93xx_has_audio() (machine_is_edb9301() || \ + machine_is_edb9302() || \ + machine_is_edb9302a() || \ + machine_is_edb9307a() || \ + machine_is_edb9315a()) + +static int edb93xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int err; + unsigned int rate = params_rate(params); + /* + * We set LRCLK equal to `rate' and SCLK = LRCLK * 64, + * because our sample size is 32 bit * 2 channels. + * I2S standard permits us to transmit more bits than + * the codec uses. + * MCLK = SCLK * 4 is the best recommended value, + * but we have to fall back to ratio 2 for higher + * sample rates. + */ + unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2); + + err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, + SND_SOC_CLOCK_IN); + if (err) + return err; + + return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate, + SND_SOC_CLOCK_OUT); +} + +static struct snd_soc_ops edb93xx_ops = { + .hw_params = edb93xx_hw_params, +}; + +static struct snd_soc_dai_link edb93xx_dai = { + .name = "CS4271", + .stream_name = "CS4271 HiFi", + .platform_name = "ep93xx-pcm-audio", + .cpu_dai_name = "ep93xx-i2s", + .codec_name = "spi0.0", + .codec_dai_name = "cs4271-hifi", + .ops = &edb93xx_ops, +}; + +static struct snd_soc_card snd_soc_edb93xx = { + .name = "EDB93XX", + .dai_link = &edb93xx_dai, + .num_links = 1, +}; + +static struct platform_device *edb93xx_snd_device; + +static int __init edb93xx_init(void) +{ + int ret; + + if (!edb93xx_has_audio()) + return -ENODEV; + + ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, + EP93XX_SYSCON_I2SCLKDIV_ORIDE | + EP93XX_SYSCON_I2SCLKDIV_SPOL); + if (ret) + return ret; + + edb93xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!edb93xx_snd_device) { + ret = -ENOMEM; + goto free_i2s; + } + + platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx); + ret = platform_device_add(edb93xx_snd_device); + if (ret) + goto device_put; + + return 0; + +device_put: + platform_device_put(edb93xx_snd_device); +free_i2s: + ep93xx_i2s_release(); + return ret; +} +module_init(edb93xx_init); + +static void __exit edb93xx_exit(void) +{ + platform_device_unregister(edb93xx_snd_device); + ep93xx_i2s_release(); +} +module_exit(edb93xx_exit); + +MODULE_AUTHOR("Alexander Sverdlin <subaparts(a)yandex.ru>"); +MODULE_DESCRIPTION("ALSA SoC EDB93xx"); +MODULE_LICENSE("GPL");

2 1

[alsa-devel] [PATCH] Fix possible sprintf overrun in snd_pcm_hw_open
by David Henningsson 08 Dec '10

08 Dec '10

I spotted this while reading code a few weeks ago, and I ran it through the Ubuntu security team just to be sure. They decided it was not needing any security embargo or similar, so here comes the patch. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 3

[alsa-devel] [PATCH 1/2] ASoC: omap: Add hardware constraint for HDMI and ABE
by Olaya, Margarita 08 Dec '10

08 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> The Audio Backend needs a step size of 24 * 4, while HDMI requires a step size of 32 * 4 bits to align the audio buffer. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/omap/omap-pcm.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 8caeb8d..6a21447 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -280,6 +280,15 @@ static int omap_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) goto out; + /* + * ABE needs a step of 24 * 4 data bits, and HDMI 32 * 4 + * Ensure buffer size satisfies both constraints. + */ + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 384); + if (ret < 0) + goto out; + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); if (prtd == NULL) { ret = -ENOMEM; -- 1.7.0.4

4 5

[alsa-devel] [PATCH] ASoC: Support WM8994 mono AIF configurations
by Mark Brown 08 Dec '10

08 Dec '10

The WM8994 supports mono signals - enable this in the driver. With DSP mode an automatic data channel selector is available, activate this when in mono mode. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 21 +++++++++++++++------ 1 files changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 78b96f6..da48802 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2121,10 +2121,12 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, struct wm8994 *control = codec->control_data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int aif1_reg; + int aif2_reg; int bclk_reg; int lrclk_reg; int rate_reg; int aif1 = 0; + int aif2 = 0; int bclk = 0; int lrclk = 0; int rate_val = 0; @@ -2135,6 +2137,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, switch (dai->id) { case 1: aif1_reg = WM8994_AIF1_CONTROL_1; + aif2_reg = WM8994_AIF1_CONTROL_2; bclk_reg = WM8994_AIF1_BCLK; rate_reg = WM8994_AIF1_RATE; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || @@ -2147,6 +2150,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, break; case 2: aif1_reg = WM8994_AIF2_CONTROL_1; + aif2_reg = WM8994_AIF2_CONTROL_2; bclk_reg = WM8994_AIF2_BCLK; rate_reg = WM8994_AIF2_RATE; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || @@ -2202,6 +2206,10 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "AIF%dCLK is %dHz, target BCLK %dHz\n", dai->id, wm8994->aifclk[id], bclk_rate); + if (params_channels(params) == 1 && + (snd_soc_read(codec, aif1_reg) & 0x18) == 0x18) + aif2 |= WM8994_AIF1_MONO; + if (wm8994->aifclk[id] == 0) { dev_err(dai->dev, "AIF%dCLK not configured\n", dai->id); return -EINVAL; @@ -2245,6 +2253,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, lrclk, bclk_rate / lrclk); snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); + snd_soc_update_bits(codec, aif2_reg, WM8994_AIF1_MONO, aif2); snd_soc_update_bits(codec, bclk_reg, WM8994_AIF1_BCLK_DIV_MASK, bclk); snd_soc_update_bits(codec, lrclk_reg, WM8994_AIF1DAC_RATE_MASK, lrclk); @@ -2400,14 +2409,14 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .id = 1, .playback = { .stream_name = "AIF1 Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, .capture = { .stream_name = "AIF1 Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, @@ -2419,14 +2428,14 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .id = 2, .playback = { .stream_name = "AIF2 Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, .capture = { .stream_name = "AIF2 Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, @@ -2438,14 +2447,14 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .id = 3, .playback = { .stream_name = "AIF3 Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, .capture = { .stream_name = "AIF3 Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: Correct WM8962 interrupt mask register read
by Mark Brown 08 Dec '10

08 Dec '10

Fix mismerge from the out of tree BSP where this support was developed. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e809274..1304ca9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3339,7 +3339,7 @@ static irqreturn_t wm8962_irq(int irq, void *data) int mask; int active; - mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); + mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK); active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; -- 1.7.1

2 1

[PATCH] Fix possible sprintf overrun in snd_pcm_hw_open
by David Henningsson 08 Dec '10

08 Dec '10

BugLink: http://launchpad.net/bugs/668487 Possible buffer overrun if the number of "card" and "device" are absurdly high, especially on 64-bit platforms. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm_hw.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 9d243d5..ce74ad4 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1270,7 +1270,7 @@ int snd_pcm_hw_open(snd_pcm_t **pcmp, const char *name, SNDERR("invalid stream %d", stream); return -EINVAL; } - sprintf(filename, filefmt, card, device); + snprintf(filename, sizeof(filename), filefmt, card, device); __again: if (attempt++ > 3) { -- 1.7.1 --------------080408060601020802060106--

1 0

[alsa-devel] [PATCH 1/3] ASoC: SAMSUNG: Fix initial return value
by Seungwhan Youn 08 Dec '10

08 Dec '10

This patch fixed intial return value to be a '0' as asuccess on set_audio_clock_heirachy(). This avoids unintended error on initialize. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/s3c24xx/smdk_spdif.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/smdk_spdif.c b/sound/soc/s3c24xx/smdk_spdif.c index c8bd904..cb658d1 100644 --- a/sound/soc/s3c24xx/smdk_spdif.c +++ b/sound/soc/s3c24xx/smdk_spdif.c @@ -28,7 +28,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev) { struct clk *fout_epll, *mout_epll, *sclk_audio0, *sclk_spdif; - int ret; + int ret = 0; fout_epll = clk_get(NULL, "fout_epll"); if (IS_ERR(fout_epll)) { -- 1.6.2.5

4 19

[alsa-devel] [PATCH] ASoC: WM8580: Debug BCLK and sample size
by Jassi Brar 08 Dec '10

08 Dec '10

In case of SNDRV_PCM_FORMAT_S32_LE, we need to set WM8580_AIF_LENGTH_32, rather than WM8580_AIF_LENGTH_24. Also, the BCLK has to be 64fs, for sample size of 20, 24 and 32 bits. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/codecs/wm8580.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index cb7765f..45efc94 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -491,16 +491,16 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, paifa |= 0x8; break; case SNDRV_PCM_FORMAT_S20_3LE: - paifa |= 0x10; + paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_20; break; case SNDRV_PCM_FORMAT_S24_LE: - paifa |= 0x10; + paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_24; break; case SNDRV_PCM_FORMAT_S32_LE: - paifa |= 0x10; - paifb |= WM8580_AIF_LENGTH_24; + paifa |= 0x0; + paifb |= WM8580_AIF_LENGTH_32; break; default: return -EINVAL; -- 1.6.2.5

4 4

[alsa-devel] [PATCH] ALSA: hda - Reset sample sizes and max bitrates when reading ELD
by Anssi Hannula 08 Dec '10

08 Dec '10

When a new HDMI/DP device is plugged in, hdmi_update_short_audio_desc() is called for every SAD (Short Audio Descriptor) in the ELD data. For LPCM coding type SAD defines the supported sample sizes. For several other coding types (such as AC-3), a maximum bitrate is defined. The maximum bitrate and sample size fields are not always cleared. Therefore, if a device is unplugged and a different one is plugged in, and the coding types of some SAD positions differ between the devices, the old max_bitrate or sample_bits values will persist if the new SADs do not define those values. The leftover max_bitrate and sample_bits do not cause any issues other than wrongly showing up in eld#X.Y procfs file and kernel log. Fix that by always clearing sample_bits and max_bitrate when reading SADs. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- sound/pci/hda/hda_eld.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 009031f..4a66347 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -189,6 +189,9 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, a->channels = GRAB_BITS(buf, 0, 0, 3); a->channels++; + a->sample_bits = 0; + a->max_bitrate = 0; + a->format = GRAB_BITS(buf, 0, 3, 4); switch (a->format) { case AUDIO_CODING_TYPE_REF_STREAM_HEADER: @@ -198,7 +201,6 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, case AUDIO_CODING_TYPE_LPCM: val = GRAB_BITS(buf, 2, 0, 3); - a->sample_bits = 0; for (i = 0; i < 3; i++) if (val & (1 << i)) a->sample_bits |= cea_sample_sizes[i + 1]; -- 1.7.3

2 2

Re: [alsa-devel] [PATCH] ALSA: hda - Do not wrongly restrict min_channels based on ELD
by Stephen Warren 08 Dec '10

08 Dec '10

Anssi Hannula wrote: > Commit bbbe33900d1f3c added functionality to restrict PCM parameters > based on ELD info (derived from EDID data) of the audio sink. > > However, it wrongly assumes that the bits 0-2 of the first byte of > CEA Short Audio Descriptors mean a supported number of channels. In > reality, they mean the maximum number of channels (as per CEA-861-D > 7.5.2). This means that the channel count can only be used to restrict > max_channels, not min_channels. > > Restricting min_channels causes us to deny opening the device in stereo > mode if the sink only has SADs that declare larger numbers of channels > (like Primare SP32 AV Processor does). > > Fix that by not restricting min_channels based on ELD information. Yes, re-reading the CEA-861-D spec, that makes sense. Comments inline though. BTW, I noticed that no SADs are required if only basic audio is supported. Is that case handled by the ELD parser? I.e. instead of: pcm->channels_max = 0; Should it be: pcm->channels_max = 0; if supports basic audio: pcm->channels_max = 2; or something like that? > Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> > Reported-by: Jean-Yves Avenard <jyavenard(a)gmail.com> > Tested-by: Jean-Yves Avenard <jyavenard(a)gmail.com> > Cc: stable(a)kernel.org > --- > > I think this should go to 2.6.36 stable tree as well. It does not affect > earlier stable releases. > > sound/pci/hda/hda_eld.c | 4 ---- > sound/pci/hda/patch_hdmi.c | 1 - > 2 files changed, 0 insertions(+), 5 deletions(-) > > diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c > index cb0c23a..47ef8aa 100644 > --- a/sound/pci/hda/hda_eld.c > +++ b/sound/pci/hda/hda_eld.c > @@ -601,13 +601,10 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct > hda_pcm_stream *pcm, > pcm->rates = 0; > pcm->formats = 0; > pcm->maxbps = 0; > - pcm->channels_min = -1; > pcm->channels_max = 0; > for (i = 0; i < eld->sad_count; i++) { > struct cea_sad *a = &eld->sad[i]; > pcm->rates |= a->rates; > - if (a->channels < pcm->channels_min) > - pcm->channels_min = a->channels; > if (a->channels > pcm->channels_max) > pcm->channels_max = a->channels; > if (a->format == AUDIO_CODING_TYPE_LPCM) { That all looks OK. > @@ -635,7 +632,6 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct > hda_pcm_stream *pcm, > /* restrict the parameters by the values the codec provides */ > pcm->rates &= codec_pars->rates; > pcm->formats &= codec_pars->formats; > - pcm->channels_min = max(pcm->channels_min, codec_pars->channels_min); Shouldn't that be: pcm->channels_min = pcm->channels_min; Or, is that assignment already made earlier as a default? > pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max); > pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps); > } > diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > index d3e49aa..31df774 100644 > --- a/sound/pci/hda/patch_hdmi.c > +++ b/sound/pci/hda/patch_hdmi.c > @@ -834,7 +834,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, > return -ENODEV; > } else { > /* fallback to the codec default */ > - hinfo->channels_min = codec_pars->channels_min; Isn't this still required to default the value? > hinfo->channels_max = codec_pars->channels_max; > hinfo->rates = codec_pars->rates; > hinfo->formats = codec_pars->formats; > -- > 1.7.3 -- nvpublic

2 5

[alsa-devel] [PATCH] ALSA: hda - Always allow basic audio irrespective of ELD info
by Anssi Hannula 07 Dec '10

07 Dec '10

Commit bbbe33900d1f3c added functionality to restrict PCM parameters based on ELD info (derived from EDID data) of the audio sink. However, according to CEA-861-D no SAD is needed for basic audio (32/44.1/48kHz stereo 16-bit audio), which is instead indicated with a basic audio flag in the CEA EDID Extension. The flag is not present in ELD. However, as all audio capable sinks are required to support basic audio, we can assume it to be always available. Fix allowed audio formats with sinks that have SADs (Short Audio Descriptors) which do not completely overlap with the basic audio formats (there are no reports of affected devices so far) by always assuming that basic audio is supported. Reported-by: Stephen Warren <swarren(a)nvidia.com> Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- Tested on 2.6.36 and I saw no obvious regressions. The issue affects stable 2.6.36 as well, but as there are no reports of affected devices, I'll leave the decision on whether to send this to stable@ for the ALSA subsystem maintainers. sound/pci/hda/hda_eld.c | 16 +++++++--------- 1 files changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 47ef8aa..009031f 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -598,21 +598,19 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, { int i; - pcm->rates = 0; - pcm->formats = 0; - pcm->maxbps = 0; - pcm->channels_max = 0; + /* assume basic audio support (the basic audio flag is not in ELD; + * however, all audio capable sinks are required to support basic + * audio) */ + pcm->rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; + pcm->formats = SNDRV_PCM_FMTBIT_S16_LE; + pcm->maxbps = 16; + pcm->channels_max = 2; for (i = 0; i < eld->sad_count; i++) { struct cea_sad *a = &eld->sad[i]; pcm->rates |= a->rates; if (a->channels > pcm->channels_max) pcm->channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) { - if (a->sample_bits & AC_SUPPCM_BITS_16) { - pcm->formats |= SNDRV_PCM_FMTBIT_S16_LE; - if (pcm->maxbps < 16) - pcm->maxbps = 16; - } if (a->sample_bits & AC_SUPPCM_BITS_20) { pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE; if (pcm->maxbps < 20) -- 1.7.3

2 3

[alsa-devel] [PATCH] ALSA: hda - Do not wrongly restrict min_channels based on ELD
by Anssi Hannula 07 Dec '10

07 Dec '10

Commit bbbe33900d1f3c added functionality to restrict PCM parameters based on ELD info (derived from EDID data) of the audio sink. However, it wrongly assumes that the bits 0-2 of the first byte of CEA Short Audio Descriptors mean a supported number of channels. In reality, they mean the maximum number of channels (as per CEA-861-D 7.5.2). This means that the channel count can only be used to restrict max_channels, not min_channels. Restricting min_channels causes us to deny opening the device in stereo mode if the sink only has SADs that declare larger numbers of channels (like Primare SP32 AV Processor does). Fix that by not restricting min_channels based on ELD information. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Reported-by: Jean-Yves Avenard <jyavenard(a)gmail.com> Tested-by: Jean-Yves Avenard <jyavenard(a)gmail.com> Cc: stable(a)kernel.org --- I think this should go to 2.6.36 stable tree as well. It does not affect earlier stable releases. sound/pci/hda/hda_eld.c | 4 ---- sound/pci/hda/patch_hdmi.c | 1 - 2 files changed, 0 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index cb0c23a..47ef8aa 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -601,13 +601,10 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, pcm->rates = 0; pcm->formats = 0; pcm->maxbps = 0; - pcm->channels_min = -1; pcm->channels_max = 0; for (i = 0; i < eld->sad_count; i++) { struct cea_sad *a = &eld->sad[i]; pcm->rates |= a->rates; - if (a->channels < pcm->channels_min) - pcm->channels_min = a->channels; if (a->channels > pcm->channels_max) pcm->channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) { @@ -635,7 +632,6 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, /* restrict the parameters by the values the codec provides */ pcm->rates &= codec_pars->rates; pcm->formats &= codec_pars->formats; - pcm->channels_min = max(pcm->channels_min, codec_pars->channels_min); pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max); pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps); } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d3e49aa..31df774 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -834,7 +834,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, return -ENODEV; } else { /* fallback to the codec default */ - hinfo->channels_min = codec_pars->channels_min; hinfo->channels_max = codec_pars->channels_max; hinfo->rates = codec_pars->rates; hinfo->formats = codec_pars->formats; -- 1.7.3

1 0

[alsa-devel] [PATCH] ASoC: soc-core: Fix null pointer dereference
by Dimitris Papastamos 07 Dec '10

07 Dec '10

In case the codec driver did not provide a read/write function, codec->driver->read|write will be NULL. Ensure that we use the one specified in codec->read|write to avoid oopsing when we access the debugfs entries. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 392d336..078f9df 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -117,7 +117,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) * the register being volatile and the device being * powered off. */ - ret = codec->driver->read(codec, i); + ret = codec->read(codec, i); if (ret >= 0) count += snprintf(buf + count, PAGE_SIZE - count, @@ -228,7 +228,7 @@ static ssize_t codec_reg_write_file(struct file *file, start++; if (strict_strtoul(start, 16, &value)) return -EINVAL; - codec->driver->write(codec, reg, value); + codec->write(codec, reg, value); return buf_size; } -- 1.7.3.3

2 1

[alsa-devel] [PATCH 2/2] ASoC: omap: Stop DMA re enabling in self linkage mode
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Hari Nagalla <hnagalla(a)ti.com> While using self linking, there is a chance that the DMA has re-enabled the channel just after disabling it. This patch stops the OMAP4 DMA re-enabling after stoping the DMA channel. Signed-off-by: Hari Nagalla <hnagalla(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/omap/omap-pcm.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 6a21447..afe91ad 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -234,6 +234,13 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_PUSH: prtd->period_index = -1; omap_stop_dma(prtd->dma_ch); + /* + * Since we are using self linking, there is a + * chance that the DMA as re-enabled the channel + * just after disabling it. + */ + while (omap_get_dma_active_status(prtd->dma_ch)) + omap_stop_dma(prtd->dma_ch); break; default: ret = -EINVAL; -- 1.7.0.4

5 7

[alsa-devel] [PATCH] ASoC: dapm: Add speaker driver widget.
by Olaya, Margarita 07 Dec '10

07 Dec '10

In some cases it was not possible to follow the appropiate power ON/OFF sequence. Add a widget to support speaker drivers where power ON/OFF ordering is important. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- include/sound/soc-dapm.h | 10 ++++++++++ sound/soc/soc-dapm.c | 6 ++++++ 2 files changed, 16 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 041e98b..3b438e5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -71,6 +71,10 @@ wcontrols, wncontrols) \ { .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols} +#define SND_SOC_DAPM_DRV(wname, wreg, wshift, winvert,\ + wcontrols, wncontrols) \ +{ .id = snd_soc_dapm_drv, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols} #define SND_SOC_DAPM_MIXER(wname, wreg, wshift, winvert, \ wcontrols, wncontrols)\ { .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ @@ -115,6 +119,11 @@ { .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ .event = wevent, .event_flags = wflags} +#define SND_SOC_DAPM_DRV_E(wname, wreg, wshift, winvert, wcontrols, \ + wncontrols, wevent, wflags) \ +{ .id = snd_soc_dapm_drv, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ + .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_MIXER_E(wname, wreg, wshift, winvert, wcontrols, \ wncontrols, wevent, wflags) \ { .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ @@ -368,6 +377,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_mixer, /* mixes several analog signals together */ snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */ snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */ + snd_soc_dapm_drv, /* speaker driver */ snd_soc_dapm_adc, /* analog to digital converter */ snd_soc_dapm_dac, /* digital to analog converter */ snd_soc_dapm_micbias, /* microphone bias (power) */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6a29d59..4726b65 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -61,6 +61,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_mixer] = 7, [snd_soc_dapm_mixer_named_ctl] = 7, [snd_soc_dapm_pga] = 8, + [snd_soc_dapm_drv] = 9, [snd_soc_dapm_adc] = 9, [snd_soc_dapm_hp] = 10, [snd_soc_dapm_spk] = 10, @@ -72,6 +73,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_adc] = 1, [snd_soc_dapm_hp] = 2, [snd_soc_dapm_spk] = 2, + [snd_soc_dapm_drv] = 3, [snd_soc_dapm_pga] = 4, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, @@ -231,6 +233,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, break; /* does not effect routing - always connected */ case snd_soc_dapm_pga: + case snd_soc_dapm_drv: case snd_soc_dapm_output: case snd_soc_dapm_adc: case snd_soc_dapm_input: @@ -1243,6 +1246,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_dac: case snd_soc_dapm_adc: case snd_soc_dapm_pga: + case snd_soc_dapm_drv: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: @@ -1421,6 +1425,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_adc: case snd_soc_dapm_dac: case snd_soc_dapm_pga: + case snd_soc_dapm_drv: case snd_soc_dapm_input: case snd_soc_dapm_output: case snd_soc_dapm_micbias: @@ -1538,6 +1543,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) w->power_check = dapm_dac_check_power; break; case snd_soc_dapm_pga: + case snd_soc_dapm_drv: w->power_check = dapm_generic_check_power; dapm_new_pga(dapm, w); break; -- 1.7.0.4

3 11

[alsa-devel] [PATCH] ASoC: Add ADC high pass filter support to WM8994
by Mark Brown 07 Dec '10

07 Dec '10

Signed-off-by: Uk Kim <w0806.kim(a)samsung.com> --- sound/soc/codecs/wm8994.c | 22 ++++++++++++++++++++++ 1 files changed, 22 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b37e95c..78b96f6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -310,6 +310,19 @@ static const char *sidetone_hpf_text[] = { static const struct soc_enum sidetone_hpf = SOC_ENUM_SINGLE(WM8994_SIDETONE, 7, 7, sidetone_hpf_text); +static const char *adc_hpf_text[] = { + "HiFi", "Voice 1", "Voice 2", "Voice 3" +}; + +static const struct soc_enum aif1adc1_hpf = + SOC_ENUM_SINGLE(WM8994_AIF1_ADC1_FILTERS, 13, 4, adc_hpf_text); + +static const struct soc_enum aif1adc2_hpf = + SOC_ENUM_SINGLE(WM8994_AIF1_ADC2_FILTERS, 13, 4, adc_hpf_text); + +static const struct soc_enum aif2adc_hpf = + SOC_ENUM_SINGLE(WM8994_AIF2_ADC_FILTERS, 13, 4, adc_hpf_text); + static const DECLARE_TLV_DB_SCALE(aif_tlv, 0, 600, 0); static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0); @@ -812,6 +825,15 @@ SOC_SINGLE_TLV("DAC2 Left Sidetone Volume", WM8994_DAC2_MIXER_VOLUMES, SOC_ENUM("Sidetone HPF Mux", sidetone_hpf), SOC_SINGLE("Sidetone HPF Switch", WM8994_SIDETONE, 6, 1, 0), +SOC_ENUM("AIF1ADC1 HPF Mode", aif1adc1_hpf), +SOC_DOUBLE("AIF1ADC1 HPF Switch", WM8994_AIF1_ADC1_FILTERS, 12, 11, 1, 0), + +SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf), +SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0), + +SOC_ENUM("AIF2ADC HPF Mode", aif2adc_hpf), +SOC_DOUBLE("AIF2ADC HPF Switch", WM8994_AIF2_ADC_FILTERS, 12, 11, 1, 0), + SOC_DOUBLE_R_TLV("DAC1 Volume", WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), SOC_DOUBLE_R("DAC1 Switch", WM8994_DAC1_LEFT_VOLUME, -- 1.7.1

1 0

[alsa-devel] [PATCH 0/2] Fixed issues/defects reported by Coverity tool.
by sudarshan.bisht＠nokia.com 07 Dec '10

07 Dec '10

From: Sudarshan <sudarshan.bisht(a)nokia.com> Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. * First patch here is just a cosmetic one as it removes tabs and trailing whitespaces from the code. * Second patch has got fixes for some minor issues/defects reported by Coverity tool and these fixes can prevent memory leaks and crashes in some cases. It also has some whitespaces and tabs removal differences, please ignore them. Sudarshan (1): alsa-lib: Removed trailing whitespaces as per pulseaudio coding guidelines. bisht (1): alsa-lib: Coverity tool has picked some minor issues in alsa-lib source code, fixed them. aserver/aserver.c | 1813 +++++++------- modules/mixer/simple/sbase.c | 837 ++++---- modules/mixer/simple/sbasedl.c | 126 +- src/alisp/alisp.c | 4 +- src/alisp/alisp_snd.c | 1493 ++++++------ src/conf.c | 5117 ++++++++++++++++++++-------------------- src/control/control_hw.c | 597 +++--- src/control/control_shm.c | 1019 ++++---- src/control/setup.c | 3 +- src/hwdep/hwdep.c | 496 ++-- src/mixer/mixer.c | 962 ++++---- src/mixer/simple_abst.c | 1 + src/mixer/simple_none.c | 15 +- src/pcm/pcm.c | 4812 +++++++++++++++++++------------------- src/pcm/pcm_file.c | 1452 ++++++------ src/pcm/pcm_hooks.c | 937 ++++---- src/pcm/pcm_ladspa.c | 2284 +++++++++--------- src/pcm/pcm_local.h | 3 +- src/pcm/pcm_meter.c | 1634 +++++++------- src/pcm/pcm_rate.c | 2166 +++++++++--------- src/pcm/pcm_share.c | 10 +- src/pcm/pcm_shm.c | 1329 ++++++------ src/pcm/pcm_simple.c | 404 ++-- src/rawmidi/rawmidi.c | 2 +- src/rawmidi/rawmidi_virt.c | 8 +- 25 files changed, 13782 insertions(+), 13742 deletions(-)

3 4

[alsa-devel] [PATCH 15/15] ASoC: twl6040: Fix TLV dB step values for gains
by Olaya, Margarita 07 Dec '10

07 Dec '10

Some gains were incorrectly configured for dB values. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 9b12219..feba59d 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -911,15 +911,15 @@ static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); /* * MICGAIN volume control: - * from 6 to 30 dB in 6 dB steps + * from -6 to 30 dB in 6 dB steps */ -static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -600, 600, 0); /* * AFMGAIN volume control: * from 18 to 24 dB in 6 dB steps */ -static DECLARE_TLV_DB_SCALE(afm_amp_tlv, 600, 600, 0); +static DECLARE_TLV_DB_SCALE(afm_amp_tlv, 1800, 600, 0); /* -- 1.7.0.4

2 1

[alsa-devel] [PATCH 12/15] ASoC: twl6040: Enable plug detection interrupts
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Misael Lopez Cruz <misael.lopez(a)ti.com> Enable plug detection interrupt mask in order to get headset PLUGINT/UNPLUGINT interrupts. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 10 +++------- sound/soc/codecs/twl6040.h | 1 + 2 files changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 17a8715..bfb8dec 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1589,7 +1589,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) struct twl6040_data *priv; int audpwron, naudint; int ret = 0; - u8 icrev; + u8 icrev, intmr = TWL6040_ALLINT_MSK; priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); if (priv == NULL) @@ -1629,16 +1629,12 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->codec_powered = 0; /* enable only codec ready interrupt */ - twl6040_write(codec, TWL6040_REG_INTMR, - ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); + intmr &= ~(TWL6040_READYMSK | TWL6040_PLUGMSK); /* reset interrupt status to allow correct power up sequence */ twl6040_read_reg_volatile(codec, TWL6040_REG_INTID); - } else { - /* no interrupts at all */ - twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, - TWL6040_ALLINT_MSK); } + twl6040_write(codec, TWL6040_REG_INTMR, intmr); if (naudint) { /* audio interrupt */ diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h index 5456c18..1b48857 100644 --- a/sound/soc/codecs/twl6040.h +++ b/sound/soc/codecs/twl6040.h @@ -79,6 +79,7 @@ /* INTMR (0x04) fields */ +#define TWL6040_PLUGMSK 0x02 #define TWL6040_READYMSK 0x40 #define TWL6040_ALLINT_MSK 0x7B -- 1.7.0.4

2 2

[alsa-devel] [PATCH 14/15] ASoC: twl6040: Use correct offset for LineInAmp Right
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Misael Lopez Cruz <misael.lopez(a)ti.com> Gain for LineInAmp Right uses LINEGAIN[5:3], which means that offset for right channel should be 4. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index a8bad49..9b12219 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -997,7 +997,7 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* AFM gains */ SOC_DOUBLE_TLV("Aux FM Volume", - TWL6040_REG_LINEGAIN, 0, 5, 0xF, 0, afm_amp_tlv), + TWL6040_REG_LINEGAIN, 0, 4, 0xF, 0, afm_amp_tlv), /* Playback gains */ SOC_TWL6040_DOUBLE_TLV("Headset Playback Volume", -- 1.7.0.4

2 1

[alsa-devel] [PATCH 13/15] ASoC: twl6040: Increase timeout for power up
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> After coming back from suspend, the timeout waiting for Phoenix chip to complete its power up sequence is not enough, which leaves the codec cache value for some registers in an outdated state. Increase the timeout value to wait for the power up sequence to correclty complete. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index bfb8dec..a8bad49 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1187,7 +1187,7 @@ static int twl6040_power_up_completion(struct snd_soc_codec *codec, u8 intid; time_left = wait_for_completion_timeout(&priv->ready, - msecs_to_jiffies(48)); + msecs_to_jiffies(144)); if (!time_left) { twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &intid, -- 1.7.0.4

2 1

[alsa-devel] [PATCH 11/15] ASoC: twl6040: Clear interrupt status at boot time
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> On Phoenix 1.1, the INTID register default value is an invalid one, causing the interrupt handler to think the phoenix power on sequence is ready before it actually finishes. This causes some i2c errors when trying to configure twl. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 21 +++++++++++---------- 1 files changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b2402ae..17a8715 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1627,6 +1627,17 @@ static int twl6040_probe(struct snd_soc_codec *codec) goto gpio2_err; priv->codec_powered = 0; + + /* enable only codec ready interrupt */ + twl6040_write(codec, TWL6040_REG_INTMR, + ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); + + /* reset interrupt status to allow correct power up sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_INTID); + } else { + /* no interrupts at all */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + TWL6040_ALLINT_MSK); } if (naudint) { @@ -1637,16 +1648,6 @@ static int twl6040_probe(struct snd_soc_codec *codec) "twl6040_codec", codec); if (ret) goto gpio2_err; - } else { - if (gpio_is_valid(audpwron)) { - /* enable only codec ready interrupt */ - twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, - ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); - } else { - /* no interrupts at all */ - twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, - TWL6040_ALLINT_MSK); - } } /* init vio registers */ -- 1.7.0.4

2 1

[alsa-devel] [PATCH 10/15] ASoC: twl6040: Add ramp up/down volume for HS and HF
by Olaya, Margarita 07 Dec '10

07 Dec '10

Add ramp functions for the headset and handsfree outputs in order to reduce the pops during power on/off sequences. In order to give more control to volume ramp, step size and delay between steps can be specified. The patches are based on wm8350 implementation from Liam Girdwood. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> --- sound/soc/codecs/twl6040.c | 536 +++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 524 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 0079e12..b2402ae 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -40,7 +40,35 @@ #include "twl6040.h" #define TWL6040_RATES SNDRV_PCM_RATE_8000_96000 -#define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) +#define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +#define TWL6040_OUTHS_0dB 0x00 +#define TWL6040_OUTHS_M30dB 0x0F +#define TWL6040_OUTHF_0dB 0x03 +#define TWL6040_OUTHF_M52dB 0x1D + +#define TWL6040_RAMP_NONE 0 +#define TWL6040_RAMP_UP 1 +#define TWL6040_RAMP_DOWN 2 + +#define TWL6040_HSL_VOL_MASK 0x0F +#define TWL6040_HSL_VOL_SHIFT 0 +#define TWL6040_HSR_VOL_MASK 0xF0 +#define TWL6040_HSR_VOL_SHIFT 4 +#define TWL6040_HF_VOL_MASK 0x1F +#define TWL6040_HF_VOL_SHIFT 0 + +struct twl6040_output { + u16 active; + u16 left_vol; + u16 right_vol; + u16 left_step; + u16 right_step; + unsigned int step_delay; + u16 ramp; + u16 mute; + struct completion ramp_done; +}; struct twl6040_jack_data { struct snd_soc_jack *jack; @@ -58,6 +86,13 @@ struct twl6040_data { unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; struct completion ready; + struct twl6040_output headset; + struct twl6040_output handsfree; + struct workqueue_struct *hf_workqueue; + struct workqueue_struct *hs_workqueue; + struct delayed_work hs_delayed_work; + struct delayed_work hf_delayed_work; + struct snd_soc_codec *codec; }; /* @@ -259,6 +294,305 @@ static void twl6040_init_vdd_regs(struct snd_soc_codec *codec) } } +/* + * Ramp HS PGA volume to minimise pops at stream startup and shutdown. + */ +static inline int twl6040_hs_ramp_step(struct snd_soc_codec *codec, + unsigned int left_step, unsigned int right_step) +{ + + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *headset = &priv->headset; + int left_complete = 0, right_complete = 0; + u8 reg, val; + + /* left channel */ + left_step = (left_step > 0xF) ? 0xF : left_step; + reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN); + val = (~reg & TWL6040_HSL_VOL_MASK); + + if (headset->ramp == TWL6040_RAMP_UP) { + /* ramp step up */ + if (val < headset->left_vol) { + val += left_step; + reg &= ~TWL6040_HSL_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HSGAIN, + (reg | (~val & TWL6040_HSL_VOL_MASK))); + } else { + left_complete = 1; + } + } else if (headset->ramp == TWL6040_RAMP_DOWN) { + /* ramp step down */ + if (val > 0x0) { + val -= left_step; + reg &= ~TWL6040_HSL_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HSGAIN, reg | + (~val & TWL6040_HSL_VOL_MASK)); + } else { + left_complete = 1; + } + } + + /* right channel */ + right_step = (right_step > 0xF) ? 0xF : right_step; + reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN); + val = (~reg & TWL6040_HSR_VOL_MASK) >> TWL6040_HSR_VOL_SHIFT; + + if (headset->ramp == TWL6040_RAMP_UP) { + /* ramp step up */ + if (val < headset->right_vol) { + val += right_step; + reg &= ~TWL6040_HSR_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HSGAIN, + (reg | (~val << TWL6040_HSR_VOL_SHIFT))); + } else { + right_complete = 1; + } + } else if (headset->ramp == TWL6040_RAMP_DOWN) { + /* ramp step down */ + if (val > 0x0) { + val -= right_step; + reg &= ~TWL6040_HSR_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HSGAIN, + reg | (~val << TWL6040_HSR_VOL_SHIFT)); + } else { + right_complete = 1; + } + } + + return left_complete & right_complete; +} + +/* + * Ramp HF PGA volume to minimise pops at stream startup and shutdown. + */ +static inline int twl6040_hf_ramp_step(struct snd_soc_codec *codec, + unsigned int left_step, unsigned int right_step) +{ + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *handsfree = &priv->handsfree; + int left_complete = 0, right_complete = 0; + u16 reg, val; + + /* left channel */ + left_step = (left_step > 0x1D) ? 0x1D : left_step; + reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFLGAIN); + reg = 0x1D - reg; + val = (reg & TWL6040_HF_VOL_MASK); + if (handsfree->ramp == TWL6040_RAMP_UP) { + /* ramp step up */ + if (val < handsfree->left_vol) { + val += left_step; + reg &= ~TWL6040_HF_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HFLGAIN, + reg | (0x1D - val)); + } else { + left_complete = 1; + } + } else if (handsfree->ramp == TWL6040_RAMP_DOWN) { + /* ramp step down */ + if (val > 0) { + val -= left_step; + reg &= ~TWL6040_HF_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HFLGAIN, + reg | (0x1D - val)); + } else { + left_complete = 1; + } + } + + /* right channel */ + right_step = (right_step > 0x1D) ? 0x1D : right_step; + reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFRGAIN); + reg = 0x1D - reg; + val = (reg & TWL6040_HF_VOL_MASK); + if (handsfree->ramp == TWL6040_RAMP_UP) { + /* ramp step up */ + if (val < handsfree->right_vol) { + val += right_step; + reg &= ~TWL6040_HF_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HFRGAIN, + reg | (0x1D - val)); + } else { + right_complete = 1; + } + } else if (handsfree->ramp == TWL6040_RAMP_DOWN) { + /* ramp step down */ + if (val > 0) { + val -= right_step; + reg &= ~TWL6040_HF_VOL_MASK; + twl6040_write(codec, TWL6040_REG_HFRGAIN, + reg | (0x1D - val)); + } + } + + return left_complete & right_complete; +} + +/* + * This work ramps both output PGAs at stream start/stop time to + * minimise pop associated with DAPM power switching. + */ +static void twl6040_pga_hs_work(struct work_struct *work) +{ + struct twl6040_data *priv = + container_of(work, struct twl6040_data, hs_delayed_work.work); + struct snd_soc_codec *codec = priv->codec; + struct twl6040_output *headset = &priv->headset; + unsigned int delay = headset->step_delay; + int i, headset_complete; + + /* do we need to ramp at all ? */ + if (headset->ramp == TWL6040_RAMP_NONE) + return; + + /* HS PGA volumes have 4 bits of resolution to ramp */ + for (i = 0; i <= 16; i++) { + headset_complete = 1; + if (headset->ramp != TWL6040_RAMP_NONE) + headset_complete = twl6040_hs_ramp_step(codec, + headset->left_step, + headset->right_step); + + /* ramp finished ? */ + if (headset_complete) + break; + + /* + * TODO: tune: delay is longer over 0dB + * as increases are larger. + */ + if (i >= 8) + schedule_timeout_interruptible(msecs_to_jiffies(delay + + (delay >> 1))); + else + schedule_timeout_interruptible(msecs_to_jiffies(delay)); + } + + if (headset->ramp == TWL6040_RAMP_DOWN) { + headset->active = 0; + complete(&headset->ramp_done); + } else { + headset->active = 1; + } + headset->ramp = TWL6040_RAMP_NONE; +} + +static void twl6040_pga_hf_work(struct work_struct *work) +{ + struct twl6040_data *priv = + container_of(work, struct twl6040_data, hf_delayed_work.work); + struct snd_soc_codec *codec = priv->codec; + struct twl6040_output *handsfree = &priv->handsfree; + unsigned int delay = handsfree->step_delay; + int i, handsfree_complete; + + /* do we need to ramp at all ? */ + if (handsfree->ramp == TWL6040_RAMP_NONE) + return; + + /* HF PGA volumes have 5 bits of resolution to ramp */ + for (i = 0; i <= 32; i++) { + handsfree_complete = 1; + if (handsfree->ramp != TWL6040_RAMP_NONE) + handsfree_complete = twl6040_hf_ramp_step(codec, + handsfree->left_step, + handsfree->right_step); + + /* ramp finished ? */ + if (handsfree_complete) + break; + + /* + * TODO: tune: delay is longer over 0dB + * as increases are larger. + */ + if (i >= 16) + schedule_timeout_interruptible(msecs_to_jiffies(delay + + (delay >> 1))); + else + schedule_timeout_interruptible(msecs_to_jiffies(delay)); + } + + + if (handsfree->ramp == TWL6040_RAMP_DOWN) { + handsfree->active = 0; + complete(&handsfree->ramp_done); + } else + handsfree->active = 1; + handsfree->ramp = TWL6040_RAMP_NONE; +} + +static int pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *out; + struct delayed_work *work; + struct workqueue_struct *queue; + + switch (w->shift) { + case 2: + case 3: + out = &priv->headset; + work = &priv->hs_delayed_work; + queue = priv->hs_workqueue; + out->step_delay = 5; /* 5 ms between volume ramp steps */ + break; + case 4: + out = &priv->handsfree; + work = &priv->hf_delayed_work; + queue = priv->hf_workqueue; + out->step_delay = 5; /* 5 ms between volume ramp steps */ + if (SND_SOC_DAPM_EVENT_ON(event)) + priv->non_lp++; + else + priv->non_lp--; + break; + default: + return -1; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (out->active) + break; + + /* don't use volume ramp for power-up */ + out->left_step = out->left_vol; + out->right_step = out->right_vol; + + if (!delayed_work_pending(work)) { + out->ramp = TWL6040_RAMP_UP; + queue_delayed_work(queue, work, + msecs_to_jiffies(1)); + } + break; + + case SND_SOC_DAPM_PRE_PMD: + if (!out->active) + break; + + if (!delayed_work_pending(work)) { + /* use volume ramp for power-down */ + out->left_step = 1; + out->right_step = 1; + out->ramp = TWL6040_RAMP_DOWN; + INIT_COMPLETION(out->ramp_done); + + queue_delayed_work(queue, work, + msecs_to_jiffies(1)); + + wait_for_completion_timeout(&out->ramp_done, + msecs_to_jiffies(2000)); + } + break; + } + + return 0; +} + /* twl6040 codec manual power-up sequence */ static void twl6040_power_up(struct snd_soc_codec *codec) { @@ -419,6 +753,156 @@ static irqreturn_t twl6040_naudint_handler(int irq, void *data) return IRQ_HANDLED; } +static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *out = NULL; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int ret; + unsigned int reg = mc->reg; + + /* For HS and HF we shadow the values and only actually write + * them out when active in order to ensure the amplifier comes on + * as quietly as possible. */ + switch (reg) { + case TWL6040_REG_HSGAIN: + out = &twl6040_priv->headset; + break; + default: + break; + } + + if (out) { + out->left_vol = ucontrol->value.integer.value[0]; + out->right_vol = ucontrol->value.integer.value[1]; + if (!out->active) + return 1; + } + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + return 1; +} + +static int twl6040_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *out = &twl6040_priv->headset; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + + switch (reg) { + case TWL6040_REG_HSGAIN: + out = &twl6040_priv->headset; + ucontrol->value.integer.value[0] = out->left_vol; + ucontrol->value.integer.value[1] = out->right_vol; + return 0; + + default: + break; + } + + return snd_soc_get_volsw(kcontrol, ucontrol); +} + +static int twl6040_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *out = NULL; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int ret; + unsigned int reg = mc->reg; + + /* For HS and HF we shadow the values and only actually write + * them out when active in order to ensure the amplifier comes on + * as quietly as possible. */ + switch (reg) { + case TWL6040_REG_HFLGAIN: + case TWL6040_REG_HFRGAIN: + out = &twl6040_priv->handsfree; + break; + default: + break; + } + + if (out) { + out->left_vol = ucontrol->value.integer.value[0]; + out->right_vol = ucontrol->value.integer.value[1]; + if (!out->active) + return 1; + } + + ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + if (ret < 0) + return ret; + + return 1; +} + +static int twl6040_get_volsw_2r(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_output *out = &twl6040_priv->handsfree; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + + /* If these are cached registers use the cache */ + switch (reg) { + case TWL6040_REG_HFLGAIN: + case TWL6040_REG_HFRGAIN: + out = &twl6040_priv->handsfree; + ucontrol->value.integer.value[0] = out->left_vol; + ucontrol->value.integer.value[1] = out->right_vol; + return 0; + + default: + break; + } + + return snd_soc_get_volsw_2r(kcontrol, ucontrol); +} + +/* double control with volume update */ +#define SOC_TWL6040_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax,\ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, .get = twl6040_get_volsw, \ + .put = twl6040_put_volsw, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right,\ + .max = xmax, .platform_max = xmax, .invert = xinvert} } + +/* double control with volume update */ +#define SOC_TWL6040_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = twl6040_get_volsw_2r, .put = twl6040_put_volsw_2r_vu, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .rshift = xshift, .max = xmax, .invert = xinvert}, } + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -516,9 +1000,9 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { TWL6040_REG_LINEGAIN, 0, 5, 0xF, 0, afm_amp_tlv), /* Playback gains */ - SOC_DOUBLE_TLV("Headset Playback Volume", + SOC_TWL6040_DOUBLE_TLV("Headset Playback Volume", TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), - SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + SOC_TWL6040_DOUBLE_R_TLV("Handsfree Playback Volume", TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), SOC_SINGLE_TLV("Earphone Playback Volume", TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv), @@ -604,16 +1088,20 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { /* Analog playback drivers */ SND_SOC_DAPM_PGA_E("Handsfree Left Driver", TWL6040_REG_HFLCTL, 4, 0, NULL, 0, - twl6040_power_mode_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Handsfree Right Driver", TWL6040_REG_HFRCTL, 4, 0, NULL, 0, - twl6040_power_mode_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_PGA("Headset Left Driver", - TWL6040_REG_HSLCTL, 2, 0, NULL, 0), - SND_SOC_DAPM_PGA("Headset Right Driver", - TWL6040_REG_HSRCTL, 2, 0, NULL, 0), + pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Headset Left Driver", + TWL6040_REG_HSLCTL, 2, 0, NULL, 0, + pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Headset Right Driver", + TWL6040_REG_HSRCTL, 2, 0, NULL, 0, + pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SWITCH_E("Earphone Driver", SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls, twl6040_power_mode_event, @@ -1108,6 +1596,8 @@ static int twl6040_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, priv); + priv->codec = codec; + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &icrev, TWL6040_REG_ASICREV); if (twl_codec && (icrev > 0)) @@ -1124,6 +1614,8 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->naudint = naudint; init_completion(&priv->ready); + init_completion(&priv->headset.ramp_done); + init_completion(&priv->handsfree.ramp_done); if (gpio_is_valid(audpwron)) { ret = gpio_request(audpwron, "audpwron"); @@ -1160,10 +1652,24 @@ static int twl6040_probe(struct snd_soc_codec *codec) /* init vio registers */ twl6040_init_vio_regs(codec); + priv->hf_workqueue = create_singlethread_workqueue("twl6040-hf"); + if (priv->hf_workqueue == NULL) { + ret = -ENOMEM; + goto irq_err; + } + priv->hs_workqueue = create_singlethread_workqueue("twl6040-hs"); + if (priv->hs_workqueue == NULL) { + ret = -ENOMEM; + goto wq_err; + } + + INIT_DELAYED_WORK(&priv->hs_delayed_work, twl6040_pga_hs_work); + INIT_DELAYED_WORK(&priv->hf_delayed_work, twl6040_pga_hf_work); + /* power on device */ ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret) - goto irq_err; + goto bias_err; snd_soc_add_controls(codec, twl6040_snd_controls, ARRAY_SIZE(twl6040_snd_controls)); @@ -1171,6 +1677,10 @@ static int twl6040_probe(struct snd_soc_codec *codec) return 0; +bias_err: + destroy_workqueue(priv->hs_workqueue); +wq_err: + destroy_workqueue(priv->hf_workqueue); irq_err: if (naudint) free_irq(naudint, codec); @@ -1196,6 +1706,8 @@ static int twl6040_remove(struct snd_soc_codec *codec) if (naudint) free_irq(naudint, codec); + destroy_workqueue(priv->hf_workqueue); + destroy_workqueue(priv->hs_workqueue); kfree(priv); return 0; -- 1.7.0.4

2 1

[alsa-devel] [PATCH 09/15] ASoC: twl6040: Enable automatic power for phoenix 1.1
by Olaya, Margarita 07 Dec '10

07 Dec '10

Phoenix 1.1 supports automatic power on sequence, a verification is added to use it with new revision of the chip. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 13 +++++++++---- 1 files changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index d92fb6d..0079e12 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1101,19 +1101,24 @@ static int twl6040_probe(struct snd_soc_codec *codec) struct twl6040_data *priv; int audpwron, naudint; int ret = 0; + u8 icrev; priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); if (priv == NULL) return -ENOMEM; snd_soc_codec_set_drvdata(codec, priv); - if (twl_codec) { + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &icrev, TWL6040_REG_ASICREV); + + if (twl_codec && (icrev > 0)) audpwron = twl_codec->audpwron_gpio; - naudint = twl_codec->naudint_irq; - } else { + else audpwron = -EINVAL; + + if (twl_codec) + naudint = twl_codec->naudint_irq; + else naudint = 0; - } priv->audpwron = audpwron; priv->naudint = naudint; -- 1.7.0.4

2 1

[alsa-devel] [PATCH 08/15] ASoC: twl6040: Fix analog Mic L & R mux controls
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Francois Mazard <f-mazard(a)ti.com> The mux control has 4 elements not 3. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index e6a7bf9..d92fb6d 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -473,8 +473,8 @@ static const char *twl6040_hf_texts[] = { }; static const struct soc_enum twl6040_enum[] = { - SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 3, twl6040_amicl_texts), - SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 4, twl6040_amicl_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 4, twl6040_amicr_texts), SOC_ENUM_SINGLE(TWL6040_REG_HSLCTL, 5, 3, twl6040_hs_texts), SOC_ENUM_SINGLE(TWL6040_REG_HSRCTL, 5, 3, twl6040_hs_texts), SOC_ENUM_SINGLE(TWL6040_REG_HFLCTL, 2, 3, twl6040_hf_texts), -- 1.7.0.4

2 1

[alsa-devel] [PATCH 07/15] ASoC: twl6040: Support other sample rates
by Olaya, Margarita 07 Dec '10

07 Dec '10

The twl6040 can support more sample rates other than 88.2 and 96k. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 45d6f2d..e6a7bf9 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -39,7 +39,7 @@ #include "twl6040.h" -#define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define TWL6040_RATES SNDRV_PCM_RATE_8000_96000 #define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) struct twl6040_jack_data { @@ -837,10 +837,17 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); switch (rate) { + case 11250: + case 22500: + case 44100: case 88200: lppllctl |= TWL6040_LPLLFIN; priv->sysclk = 17640000; break; + case 8000: + case 16000: + case 32000: + case 48000: case 96000: lppllctl &= ~TWL6040_LPLLFIN; priv->sysclk = 19200000; -- 1.7.0.4

2 1

[alsa-devel] [PATCH 06/15] ASoC: twl6040: Fix PCM error handling ops
by Olaya, Margarita 07 Dec '10

07 Dec '10

This patch moves all the PCM error handling for clock config out of trigger() and startup() and into prepare(). Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 55 +++++++++++++++++-------------------------- 1 files changed, 22 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 81bf781..45d6f2d 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -812,23 +812,6 @@ static int twl6040_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - if (!priv->sysclk) { - dev_err(codec->dev, - "no mclk configured, call set_sysclk() on init\n"); - return -EINVAL; - } - - /* - * capture is not supported at 17.64 MHz, - * it's reserved for headset low-power playback scenario - */ - if ((priv->sysclk == 17640000) && substream->stream) { - dev_err(codec->dev, - "capture mode is not supported at %dHz\n", - priv->sysclk); - return -EINVAL; - } - snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, priv->sysclk_constraints); @@ -872,31 +855,37 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, return 0; } -static int twl6040_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) +static int twl6040_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - /* - * low-power playback mode is restricted - * for headset path only - */ - if ((priv->sysclk == 17640000) && priv->non_lp) { + if (!priv->sysclk) { + dev_err(codec->dev, + "no mclk configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + /* + * capture is not supported at 17.64 MHz, + * it's reserved for headset low-power playback scenario + */ + if ((priv->sysclk == 17640000) && + substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + dev_err(codec->dev, + "capture mode is not supported at %dHz\n", + priv->sysclk); + return -EINVAL; + } + + if ((priv->sysclk == 17640000) && priv->non_lp) { dev_err(codec->dev, "some enabled paths aren't supported at %dHz\n", priv->sysclk); return -EPERM; - } - break; - default: - break; } - return 0; } @@ -1010,7 +999,7 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai, static struct snd_soc_dai_ops twl6040_dai_ops = { .startup = twl6040_startup, .hw_params = twl6040_hw_params, - .trigger = twl6040_trigger, + .prepare = twl6040_prepare, .set_sysclk = twl6040_set_dai_sysclk, }; -- 1.7.0.4

2 1

[alsa-devel] [PATCH 05/15] ASoC: twl6040: Restore bias level at resume
by Olaya, Margarita 07 Dec '10

07 Dec '10

This patch restores the CODEC bias level at resume(). Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 7417b50..81bf781 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1072,6 +1072,7 @@ static int twl6040_suspend(struct snd_soc_codec *codec, pm_message_t state) static int twl6040_resume(struct snd_soc_codec *codec) { twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6040_set_bias_level(codec, codec->dapm.suspend_bias_level); return 0; } -- 1.7.0.4

2 1

[alsa-devel] [PATCH 04/15] ASoC: twl6040: Support all DAIs
by Olaya, Margarita 07 Dec '10

07 Dec '10

This patch adds support for all the TWL6040 DAIs. - twl6040-ul - twl6040-dl1 - twl6040-dl2 - twl6040-vib Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 43 ++++++++++++++++++++++++++++++++++++------- 1 files changed, 36 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 8464bcc..7417b50 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1014,23 +1014,51 @@ static struct snd_soc_dai_ops twl6040_dai_ops = { .set_sysclk = twl6040_set_dai_sysclk, }; -static struct snd_soc_dai_driver twl6040_dai = { - .name = "twl6040-hifi", +static struct snd_soc_dai_driver twl6040_dai[] = { +{ + .name = "twl6040-ul", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}, +{ + .name = "twl6040-dl1", .playback = { - .stream_name = "Playback", + .stream_name = "Headset Playback", .channels_min = 1, - .channels_max = 4, + .channels_max = 2, .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, - .capture = { - .stream_name = "Capture", + .ops = &twl6040_dai_ops, +}, +{ + .name = "twl6040-dl2", + .playback = { + .stream_name = "Handsfree Playback", .channels_min = 1, .channels_max = 2, .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, .ops = &twl6040_dai_ops, +}, +{ + .name = "twl6040-vib", + .playback = { + .stream_name = "Vibra Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}, }; #ifdef CONFIG_PM @@ -1187,7 +1215,8 @@ static struct snd_soc_codec_driver soc_codec_dev_twl6040 = { static int __devinit twl6040_codec_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, - &soc_codec_dev_twl6040, &twl6040_dai, 1); + &soc_codec_dev_twl6040, twl6040_dai, + ARRAY_SIZE(twl6040_dai)); } static int __devexit twl6040_codec_remove(struct platform_device *pdev) -- 1.7.0.4

2 1

[alsa-devel] [PATCH 03/15] ASoC: twl6040: Add headset and handset mux controls
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> This patch adds support for the twl6040 headset and handset MUX controls. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 101 ++++++++++++++++++++++++++++++-------------- 1 files changed, 69 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index e908146..8464bcc 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -82,19 +82,19 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x1B, /* TWL6040_LINEGAIN 0x0F */ 0x00, /* TWL6040_HSLCTL 0x10 */ 0x00, /* TWL6040_HSRCTL 0x11 */ - 0x00, /* TWL6040_HSGAIN 0x12 */ - 0x00, /* TWL6040_EARCTL 0x13 */ + 0xFF, /* TWL6040_HSGAIN 0x12 */ + 0x1E, /* TWL6040_EARCTL 0x13 */ 0x00, /* TWL6040_HFLCTL 0x14 */ - 0x00, /* TWL6040_HFLGAIN 0x15 */ + 0x1D, /* TWL6040_HFLGAIN 0x15 */ 0x00, /* TWL6040_HFRCTL 0x16 */ - 0x00, /* TWL6040_HFRGAIN 0x17 */ + 0x1D, /* TWL6040_HFRGAIN 0x17 */ 0x00, /* TWL6040_VIBCTLL 0x18 */ 0x00, /* TWL6040_VIBDATL 0x19 */ 0x00, /* TWL6040_VIBCTLR 0x1A */ 0x00, /* TWL6040_VIBDATR 0x1B */ 0x00, /* TWL6040_HKCTL1 0x1C */ 0x00, /* TWL6040_HKCTL2 0x1D */ - 0x00, /* TWL6040_GPOCTL 0x1E */ + 0x02, /* TWL6040_GPOCTL 0x1E */ 0x00, /* TWL6040_ALB 0x1F */ 0x00, /* TWL6040_DLB 0x20 */ 0x00, /* not used 0x21 */ @@ -432,6 +432,13 @@ static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); /* + * AFMGAIN volume control: + * from 18 to 24 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(afm_amp_tlv, 600, 600, 0); + + +/* * HSGAIN volume control: * from -30 to 0 dB in 2 dB steps */ @@ -457,9 +464,21 @@ static const char *twl6040_amicl_texts[] = static const char *twl6040_amicr_texts[] = {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; +static const char *twl6040_hs_texts[] = { + "Off", "HS DAC", "Line-In amp" +}; + +static const char *twl6040_hf_texts[] = { + "Off", "HF DAC", "Line-In amp" +}; + static const struct soc_enum twl6040_enum[] = { SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 3, twl6040_amicl_texts), SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts), + SOC_ENUM_SINGLE(TWL6040_REG_HSLCTL, 5, 3, twl6040_hs_texts), + SOC_ENUM_SINGLE(TWL6040_REG_HSRCTL, 5, 3, twl6040_hs_texts), + SOC_ENUM_SINGLE(TWL6040_REG_HFLCTL, 2, 3, twl6040_hf_texts), + SOC_ENUM_SINGLE(TWL6040_REG_HFRCTL, 2, 3, twl6040_hf_texts), }; static const struct snd_kcontrol_new amicl_control = @@ -469,18 +488,18 @@ static const struct snd_kcontrol_new amicr_control = SOC_DAPM_ENUM("Route", twl6040_enum[1]); /* Headset DAC playback switches */ -static const struct snd_kcontrol_new hsdacl_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 5, 1, 0); +static const struct snd_kcontrol_new hsl_mux_controls = + SOC_DAPM_ENUM("Route", twl6040_enum[2]); -static const struct snd_kcontrol_new hsdacr_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 5, 1, 0); +static const struct snd_kcontrol_new hsr_mux_controls = + SOC_DAPM_ENUM("Route", twl6040_enum[3]); /* Handsfree DAC playback switches */ -static const struct snd_kcontrol_new hfdacl_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 2, 1, 0); +static const struct snd_kcontrol_new hfl_mux_controls = + SOC_DAPM_ENUM("Route", twl6040_enum[4]); -static const struct snd_kcontrol_new hfdacr_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); +static const struct snd_kcontrol_new hfr_mux_controls = + SOC_DAPM_ENUM("Route", twl6040_enum[5]); static const struct snd_kcontrol_new ep_driver_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0); @@ -492,6 +511,10 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { SOC_DOUBLE_TLV("Capture Volume", TWL6040_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + /* AFM gains */ + SOC_DOUBLE_TLV("Aux FM Volume", + TWL6040_REG_LINEGAIN, 0, 5, 0xF, 0, afm_amp_tlv), + /* Playback gains */ SOC_DOUBLE_TLV("Headset Playback Volume", TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), @@ -528,6 +551,12 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_PGA("MicAmpR", TWL6040_REG_MICRCTL, 0, 0, NULL, 0), + /* Auxiliary FM PGAs */ + SND_SOC_DAPM_PGA("AFMAmpL", + TWL6040_REG_MICLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("AFMAmpR", + TWL6040_REG_MICRCTL, 1, 0, NULL, 0), + /* ADCs */ SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", TWL6040_REG_MICLCTL, 2, 0), @@ -562,15 +591,15 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - /* Analog playback switches */ - SND_SOC_DAPM_SWITCH("HSDAC Left Playback", - SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), - SND_SOC_DAPM_SWITCH("HSDAC Right Playback", - SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), - SND_SOC_DAPM_SWITCH("HFDAC Left Playback", - SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), - SND_SOC_DAPM_SWITCH("HFDAC Right Playback", - SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + SND_SOC_DAPM_MUX("HF Left Playback", + SND_SOC_NOPM, 0, 0, &hfl_mux_controls), + SND_SOC_DAPM_MUX("HF Right Playback", + SND_SOC_NOPM, 0, 0, &hfr_mux_controls), + /* Analog playback Muxes */ + SND_SOC_DAPM_MUX("HS Left Playback", + SND_SOC_NOPM, 0, 0, &hsl_mux_controls), + SND_SOC_DAPM_MUX("HS Right Playback", + SND_SOC_NOPM, 0, 0, &hsr_mux_controls), /* Analog playback drivers */ SND_SOC_DAPM_PGA_E("Handsfree Left Driver", @@ -614,12 +643,18 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Left", NULL, "MicAmpL"}, {"ADC Right", NULL, "MicAmpR"}, - /* Headset playback path */ - {"HSDAC Left Playback", "Switch", "HSDAC Left"}, - {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + /* AFM path */ + {"AFMAmpL", "NULL", "AFML"}, + {"AFMAmpR", "NULL", "AFMR"}, - {"Headset Left Driver", NULL, "HSDAC Left Playback"}, - {"Headset Right Driver", NULL, "HSDAC Right Playback"}, + {"HS Left Playback", "HS DAC", "HSDAC Left"}, + {"HS Left Playback", "Line-In amp", "AFMAmpL"}, + + {"HS Right Playback", "HS DAC", "HSDAC Right"}, + {"HS Right Playback", "Line-In amp", "AFMAmpR"}, + + {"Headset Left Driver", "NULL", "HS Left Playback"}, + {"Headset Right Driver", "NULL", "HS Right Playback"}, {"HSOL", NULL, "Headset Left Driver"}, {"HSOR", NULL, "Headset Right Driver"}, @@ -628,12 +663,14 @@ static const struct snd_soc_dapm_route intercon[] = { {"Earphone Driver", "Switch", "HSDAC Left"}, {"EP", NULL, "Earphone Driver"}, - /* Handsfree playback path */ - {"HFDAC Left Playback", "Switch", "HFDAC Left"}, - {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + {"HF Left Playback", "HF DAC", "HFDAC Left"}, + {"HF Left Playback", "Line-In amp", "AFMAmpL"}, + + {"HF Right Playback", "HF DAC", "HFDAC Right"}, + {"HF Right Playback", "Line-In amp", "AFMAmpR"}, - {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, - {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + {"HFDAC Left PGA", NULL, "HF Left Playback"}, + {"HFDAC Right PGA", NULL, "HF Right Playback"}, {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, -- 1.7.0.4

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[alsa-devel] [PATCH 02/15] ASoC: twl6040: Modify the IRQ handler
by Olaya, Margarita 07 Dec '10

07 Dec '10

Multiples interrupts can be received. The irq handler is modified to attend all of them. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 30 +++++++++++++----------------- 1 files changed, 13 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 59018b3..e908146 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -398,27 +398,23 @@ static irqreturn_t twl6040_naudint_handler(int irq, void *data) twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); - switch (intid) { - case TWL6040_THINT: + if (intid & TWL6040_THINT) dev_alert(codec->dev, "die temp over-limit detection\n"); - break; - case TWL6040_PLUGINT: - case TWL6040_UNPLUGINT: - case TWL6040_HOOKINT: - break; - case TWL6040_HFINT: + + if (intid & TWL6040_PLUGINT) + dev_alert(codec->dev, "headset plug detection\n"); + + if (intid & TWL6040_UNPLUGINT) + dev_alert(codec->dev, "headset unplug detection\n"); + + if (intid & TWL6040_HFINT) dev_alert(codec->dev, "hf drivers over current detection\n"); - break; - case TWL6040_VIBINT: + + if (intid & TWL6040_VIBINT) dev_alert(codec->dev, "vib drivers over current detection\n"); - break; - case TWL6040_READYINT: + + if (intid & TWL6040_READYINT) complete(&priv->ready); - break; - default: - dev_err(codec->dev, "unknown audio interrupt %d\n", intid); - break; - } return IRQ_HANDLED; } -- 1.7.0.4

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[alsa-devel] [PATCH 01/15] ASoC: twl6040: Update twl IO macro
by Olaya, Margarita 07 Dec '10

07 Dec '10

Update the codec to use the new twl core register macros Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 5dd5336..59018b3 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -206,7 +206,7 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, if (reg >= TWL6040_CACHEREGNUM) return -EIO; - twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg); + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &value, reg); twl6040_write_reg_cache(codec, reg, value); return value; @@ -222,7 +222,7 @@ static int twl6040_write(struct snd_soc_codec *codec, return -EIO; twl6040_write_reg_cache(codec, reg, value); - return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); + return twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, value, reg); } static void twl6040_init_vio_regs(struct snd_soc_codec *codec) @@ -396,7 +396,7 @@ static irqreturn_t twl6040_naudint_handler(int irq, void *data) int report = 0; u8 intid; - twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); switch (intid) { case TWL6040_THINT: @@ -669,7 +669,7 @@ static int twl6040_power_up_completion(struct snd_soc_codec *codec, msecs_to_jiffies(48)); if (!time_left) { - twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); if (!(intid & TWL6040_READYINT)) { dev_err(codec->dev, "timeout waiting for READYINT\n"); -- 1.7.0.4

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[alsa-devel] [PATCH v2 3/3] ASoC: twl6040: Add jack support for headset and handset
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> This patch adds support for reporting twl6040 headset and handset jack events. The machine driver retrieves and report the status through twl6040_hs_jack_detect in case of the early interrupt. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 26 ++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 12 ++++++++++++ 2 files changed, 38 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b92f2b7..5dd5336 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -42,10 +42,16 @@ #define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) +struct twl6040_jack_data { + struct snd_soc_jack *jack; + int report; +}; + /* codec private data */ struct twl6040_data { int audpwron; int naudint; + struct twl6040_jack_data hs_jack; int codec_powered; int pll; int non_lp; @@ -386,6 +392,8 @@ static irqreturn_t twl6040_naudint_handler(int irq, void *data) { struct snd_soc_codec *codec = data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + struct twl6040_jack_data *jack = &priv->hs_jack; + int report = 0; u8 intid; twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); @@ -1011,6 +1019,24 @@ static int twl6040_resume(struct snd_soc_codec *codec) #define twl6040_resume NULL #endif +void twl6040_hs_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int report) +{ + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + int status; + + priv->hs_jack.jack = jack; + priv->hs_jack.report = report; + + /* Sync status */ + status = twl6040_read_reg_volatile(codec, TWL6040_REG_STATUS); + if (status & TWL6040_PLUGCOMP) + snd_soc_jack_report(jack, report, report); + else + snd_soc_jack_report(jack, 0, report); +} +EXPORT_SYMBOL_GPL(twl6040_hs_jack_detect); + static int twl6040_probe(struct snd_soc_codec *codec) { struct twl4030_codec_data *twl_codec = codec->dev->platform_data; diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h index f7c77fa..5456c18 100644 --- a/sound/soc/codecs/twl6040.h +++ b/sound/soc/codecs/twl6040.h @@ -135,4 +135,16 @@ #define TWL6040_HPPLL_ID 1 #define TWL6040_LPPLL_ID 2 +/* STATUS (0x2E) fields */ + +#define TWL6040_PLUGCOMP 0x02 + +struct twl6040_setup_data { + void (*codec_enable)(int enable); + void *jack; +}; + +void twl6040_hs_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int report); + #endif /* End of __TWL6040_H__ */ -- 1.7.0.4

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[alsa-devel] [PATCH v2 2/3] ASoC: sdp4430: Add Jack support
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Use jack framework to enable detection for the headset microphone and stereo output. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/omap/sdp4430.c | 28 ++++++++++++++++++++++++++++ 1 files changed, 28 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 0c37c51..ca7fb90 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -24,6 +24,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <plat/hardware.h> @@ -65,6 +66,21 @@ static struct snd_soc_ops sdp4430_ops = { .hw_params = sdp4430_hw_params, }; +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -160,7 +176,19 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); ret = snd_soc_dapm_sync(dapm); + if (ret) + return ret; + + /* Headset jack detection */ + ret = snd_soc_jack_new(codec, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); return ret; } -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: Fix snd_soc_instantiate_card error path
by Axel Lin 07 Dec '10

07 Dec '10

>From 0443551476ee6fabe45a0fbc7460dca8e0f6a965 Mon Sep 17 00:00:00 2001 From: Axel Lin <axel.lin(a)gmail.com> Date: Tue, 7 Dec 2010 16:12:29 +0800 Subject: [PATCH] ASoC: Fix snd_soc_instantiate_card error path Properly free the resources in the case of snd_card_register failure and soc_register_ac97_dai_link failure. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This patch is against linux-next tree. In the case of soc_register_ac97_dai_link failure, I think we need below fix for Linus's tree: Add while (--i >= 0) soc_unregister_ac97_dai_link(&card->rtd[i]); just before goto probe_dai_err; Should I also send a patch against Linus's tree? Regards, Axel sound/soc/soc-core.c | 14 ++++++++------ 1 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 392d336..b4c8c38 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1839,18 +1839,20 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) ret = snd_card_register(card->snd_card); if (ret < 0) { printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); - goto probe_dai_err; + goto probe_aux_dev_err; } #ifdef CONFIG_SND_SOC_AC97_BUS /* register any AC97 codecs */ for (i = 0; i < card->num_rtd; i++) { - ret = soc_register_ac97_dai_link(&card->rtd[i]); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); - goto probe_dai_err; - } + ret = soc_register_ac97_dai_link(&card->rtd[i]); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); + while (--i >= 0) + soc_unregister_ac97_dai_link(&card->rtd[i]); + goto probe_aux_dev_err; } + } #endif card->instantiated = 1; -- 1.7.2

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[alsa-devel] [RFC v2 0/3] ASoC: Option to reorder widget power sequence
by Peter Ujfalusi 07 Dec '10

07 Dec '10

Hello, Chasnges sicne v1: - Flag moved to per widget - Flag name changed to: route_change_before_powerup - Comments from Mark has been addressed in the soc-dapm.c file - New helper function to set the reorder power up sequence. - tlv320dac33 to be the first codec to use the reversed power up order Note: power down sequence remains the same for the reordered widgets as well. Intro from the v1: Quite some time ago I have faced similar issue with the sequence of the DAPM register writes [1] For that issue, I used a workaround within the twl4030 codec driver. Now I have again issue with the register write sequence. The dac33 codec emmits a pop noise, when user enable the analog bypass path. The route cause in this codec are the DAPM update power and register write sequence: 1. User enable the bypass 2. DAPM power up sequence takes place 2.1 codec DAPM widgets got powered on, codec is enabled 2.2 External speaker enabled 3. The bit enabling the bypass got changed. Step 3 cause a pop, which propagate to the speaker, since it is already enabled. The first patch implements a new flag for the core to reorder the sequence to: 1. User enable the bypass 2. The bit enabling the bypass got changed. 3. DAPM power up sequence takes place 3.1 codec DAPM widgets got powered on, codec is enabled 3.2 External speaker enabled The default sequence is kept, codec driver must ask to have the reordered sequence to avoid any side effect on other codecs. [1]: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-August/030024.html Peter --- Peter Ujfalusi (3): ASoC: core: DAPM widget flag for reversed power up sequence ASoC: core: Helper function to request reversed widget power up ASoC: tlv320dac33: Reverse the power up sequence for bypass switches include/sound/soc-dapm.h | 3 ++ sound/soc/codecs/tlv320dac33.c | 4 +++ sound/soc/soc-dapm.c | 54 ++++++++++++++++++++++++++++++++++++++- 3 files changed, 59 insertions(+), 2 deletions(-) -- 1.7.3.2

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[alsa-devel] snd_hda_intel error
by Jack Byer 07 Dec '10

07 Dec '10

I found this message in my system logs: Dec 04 07:25:52 [pulseaudio] alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 8. Dec 04 07:25:52 [pulseaudio] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Dec 04 07:25:52 [pulseaudio] alsa-util.c: snd_pcm_dump(): Dec 04 07:25:52 [pulseaudio] alsa-util.c: Soft volume PCM Dec 04 07:25:52 [pulseaudio] alsa-util.c: Control: PCM Playback Volume Dec 04 07:25:52 [pulseaudio] alsa-util.c: min_dB: -51 Dec 04 07:25:52 [pulseaudio] alsa-util.c: max_dB: 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: resolution: 256 Dec 04 07:25:52 [pulseaudio] alsa-util.c: Its setup is: Dec 04 07:25:52 [pulseaudio] alsa-util.c: stream : CAPTURE Dec 04 07:25:52 [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED Dec 04 07:25:52 [pulseaudio] alsa-util.c: format : S16_LE Dec 04 07:25:52 [pulseaudio] alsa-util.c: subformat : STD Dec 04 07:25:52 [pulseaudio] alsa-util.c: channels : 2 Dec 04 07:25:52 [pulseaudio] alsa-util.c: rate : 44100 Dec 04 07:25:52 [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) Dec 04 07:25:52 [pulseaudio] alsa-util.c: msbits : 16 Dec 04 07:25:52 [pulseaudio] alsa-util.c: buffer_size : 16384 Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_size : 8192 Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_time : 185759 Dec 04 07:25:52 [pulseaudio] alsa-util.c: tstamp_mode : ENABLE Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_step : 1 Dec 04 07:25:52 [pulseaudio] alsa-util.c: avail_min : 15502 Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_event : 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: start_threshold : -1 Dec 04 07:25:52 [pulseaudio] alsa-util.c: stop_threshold : 4611686018427387904 Dec 04 07:25:52 [pulseaudio] alsa-util.c: silence_threshold: 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: silence_size : 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: boundary : 4611686018427387904 Dec 04 07:25:52 [pulseaudio] alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: Its setup is: Dec 04 07:25:52 [pulseaudio] alsa-util.c: stream : CAPTURE Dec 04 07:25:52 [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED Dec 04 07:25:52 [pulseaudio] alsa-util.c: format : S16_LE Dec 04 07:25:52 [pulseaudio] alsa-util.c: subformat : STD Dec 04 07:25:52 [pulseaudio] alsa-util.c: channels : 2 Dec 04 07:25:52 [pulseaudio] alsa-util.c: rate : 44100 Dec 04 07:25:52 [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) Dec 04 07:25:52 [pulseaudio] alsa-util.c: msbits : 16 Dec 04 07:25:52 [pulseaudio] alsa-util.c: buffer_size : 16384 Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_size : 8192 Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_time : 185759 Dec 04 07:25:52 [pulseaudio] alsa-util.c: tstamp_mode : ENABLE Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_step : 1 Dec 04 07:25:52 [pulseaudio] alsa-util.c: avail_min : 15502 Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_event : 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: start_threshold : -1 Dec 04 07:25:52 [pulseaudio] alsa-util.c: stop_threshold : 4611686018427387904 Dec 04 07:25:52 [pulseaudio] alsa-util.c: silence_threshold: 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: silence_size : 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: boundary : 4611686018427387904 Dec 04 07:25:52 [pulseaudio] alsa-util.c: appl_ptr : 135040 Dec 04 07:25:52 [pulseaudio] alsa-util.c: hw_ptr : 135040 I am using the driver from the 2.6.31.1 kernel

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[alsa-devel] [PATCH] ASoC: mcbsp: Add McBSP support for OMAP4
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> This patch adds McBSP support for the OMAP4 CPU Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 38 +++++++++++++++++++++++++++++++++++--- sound/soc/omap/omap-mcbsp.h | 4 ++++ 2 files changed, 39 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7e84f24..132062c 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -102,6 +102,17 @@ static const int omap24xx_dma_reqs[][2] = { static const int omap24xx_dma_reqs[][2] = {}; #endif +#if defined(CONFIG_ARCH_OMAP4) +static const int omap44xx_dma_reqs[][2] = { + { OMAP44XX_DMA_MCBSP1_TX, OMAP44XX_DMA_MCBSP1_RX }, + { OMAP44XX_DMA_MCBSP2_TX, OMAP44XX_DMA_MCBSP2_RX }, + { OMAP44XX_DMA_MCBSP3_TX, OMAP44XX_DMA_MCBSP3_RX }, + { OMAP44XX_DMA_MCBSP4_TX, OMAP44XX_DMA_MCBSP4_RX }, +}; +#else +static const int omap44xx_dma_reqs[][2] = {}; +#endif + #if defined(CONFIG_ARCH_OMAP2420) static const unsigned long omap2420_mcbsp_port[][2] = { { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, @@ -147,6 +158,21 @@ static const unsigned long omap34xx_mcbsp_port[][2] = { static const unsigned long omap34xx_mcbsp_port[][2] = {}; #endif +#if defined(CONFIG_ARCH_OMAP4) +static const unsigned long omap44xx_mcbsp_port[][2] = { + { OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, + OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, + OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, + OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, + OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, +}; +#else +static const unsigned long omap44xx_mcbsp_port[][2] = {}; +#endif + static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -224,7 +250,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) */ - if (cpu_is_omap343x()) { + if (cpu_is_omap343x() || cpu_is_omap44xx()) { /* * Rule for the buffer size. We should not allow * smaller buffer than the FIFO size to avoid underruns @@ -332,6 +358,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } else if (cpu_is_omap343x()) { dma = omap24xx_dma_reqs[bus_id][substream->stream]; port = omap34xx_mcbsp_port[bus_id][substream->stream]; + } else if (cpu_is_omap44xx()) { + dma = omap44xx_dma_reqs[bus_id][substream->stream]; + port = omap44xx_mcbsp_port[bus_id][substream->stream]; } else { return -ENODEV; } @@ -498,11 +527,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->spcr2 |= XINTM(3) | FREE; regs->spcr1 |= RINTM(3); /* RFIG and XFIG are not defined in 34xx */ - if (!cpu_is_omap34xx()) { + if (!cpu_is_omap34xx() && !cpu_is_omap44xx()) { regs->rcr2 |= RFIG; regs->xcr2 |= XFIG; } - if (cpu_is_omap2430() || cpu_is_omap34xx()) { + if (cpu_is_omap2430() || cpu_is_omap34xx() || cpu_is_omap44xx()) { regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE; regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE; } @@ -623,6 +652,9 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, if (cpu_class_is_omap1()) { err = -EINVAL; break; + } else if (cpu_is_omap44xx()) { + regs->srgr2 |= CLKSM; + break; } err = omap2_mcbsp_set_clks_src(mcbsp_data->bus_id, MCBSP_CLKS_PRCM_SRC); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index ffdcc5a..110c106 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -50,6 +50,10 @@ enum omap_mcbsp_div { #undef NUM_LINKS #define NUM_LINKS 3 #endif +#if defined(CONFIG_ARCH_OMAP4) +#undef NUM_LINKS +#define NUM_LINKS 4 +#endif #if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3) #undef NUM_LINKS #define NUM_LINKS 5 -- 1.7.0.4

3 2

[alsa-devel] Bug in alsa drivers reported when launching pulseaudio
by kieran magee 07 Dec '10

07 Dec '10

Some strange anomalies were occuring in my audio systems, namely, volume fluctuations during playback. Attempted to access sound in preferences, had disabled pulseaudio a while back with no ill effects, (weeks back), I couldn't access the menu Item. I was instructed that pulseaudio not running may be the issue, so attempted to launch that and came upon this worrying error message, E: alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 16. E: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. E: alsa-util.c: snd_pcm_dump(): E: alsa-util.c: Soft volume PCM E: alsa-util.c: Control: PCM Playback Volume E: alsa-util.c: min_dB: -51 E: alsa-util.c: max_dB: 0 E: alsa-util.c: resolution: 256 E: alsa-util.c: Its setup is: E: alsa-util.c: stream : CAPTURE E: alsa-util.c: access : MMAP_INTERLEAVED E: alsa-util.c: format : S16_LE E: alsa-util.c: subformat : STD E: alsa-util.c: channels : 2 E: alsa-util.c: rate : 44100 E: alsa-util.c: exact rate : 44100 (44100/1) E: alsa-util.c: msbits : 16 E: alsa-util.c: buffer_size : 88192 E: alsa-util.c: period_size : 44096 E: alsa-util.c: period_time : 999909 E: alsa-util.c: tstamp_mode : ENABLE E: alsa-util.c: period_step : 1 E: alsa-util.c: avail_min : 87310 E: alsa-util.c: period_event : 0 E: alsa-util.c: start_threshold : -1 E: alsa-util.c: stop_threshold : 6205960286516543488 E: alsa-util.c: silence_threshold: 0 E: alsa-util.c: silence_size : 0 E: alsa-util.c: boundary : 6205960286516543488 E: alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 E: alsa-util.c: Its setup is: E: alsa-util.c: stream : CAPTURE E: alsa-util.c: access : MMAP_INTERLEAVED E: alsa-util.c: format : S16_LE E: alsa-util.c: subformat : STD E: alsa-util.c: channels : 2 E: alsa-util.c: rate : 44100 E: alsa-util.c: exact rate : 44100 (44100/1) E: alsa-util.c: msbits : 16 E: alsa-util.c: buffer_size : 88192 E: alsa-util.c: period_size : 44096 E: alsa-util.c: period_time : 999909 E: alsa-util.c: tstamp_mode : ENABLE E: alsa-util.c: period_step : 1 E: alsa-util.c: avail_min : 87310 E: alsa-util.c: period_event : 0 E: alsa-util.c: start_threshold : -1 E: alsa-util.c: stop_threshold : 6205960286516543488 E: alsa-util.c: silence_threshold: 0 E: alsa-util.c: silence_size : 0 E: alsa-util.c: boundary : 6205960286516543488 E: alsa-util.c: appl_ptr : 87328 E: alsa-util.c: hw_ptr : 87328 If you need any system information, please feel free to ask. I'm not subscribed to the mailing list, as a note, so responses need to be cc'd directly to me if I am to get them. Thanks for any help you can give me, Best, Kieran Brantner-Magee (the system is ubuntu 10.10 kernel 2.6.35-23 )

3 4

[alsa-devel] [PATCH 00/15] twl6040 updates
by Olaya, Margarita 07 Dec '10

07 Dec '10

This patch series contains several fixes and adds new features for twl6040 (phoenix) codec. Francois Mazard (1): ASoC: twl6040: Fix analog Mic L & R mux controls Jorge Eduardo Candelaria (3): ASoC: twl6040: Add headset and handset mux controls 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 | 815 +++++++++++++++++++++++++++++++++++++------- sound/soc/codecs/twl6040.h | 1 + 2 files changed, 695 insertions(+), 121 deletions(-) Regards, Margarita

1 0

[alsa-devel] [PATCH v2 1/3] ASoC: sdp4430: Enable FM stereo pins
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Add FM stereo pins to the machine driver and add them as a dapm widget. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/omap/sdp4430.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index ebbd62f..0c37c51 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -101,6 +101,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), + SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -123,6 +124,10 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Earphone speaker */ {"Earphone Spk", NULL, "EP"}, + + /* Aux/FM Stereo In: AFML, AFMR */ + {"AFML", NULL, "Aux/FM Stereo In"}, + {"AFMR", NULL, "Aux/FM Stereo In"}, }; static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) @@ -149,13 +154,11 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) /* SDP4430 connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Mic"); snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "AFML"); + snd_soc_dapm_enable_pin(dapm, "AFMR"); snd_soc_dapm_enable_pin(dapm, "Headset Mic"); snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - /* TWL6040 not connected pins */ - snd_soc_dapm_nc_pin(dapm, "AFML"); - snd_soc_dapm_nc_pin(dapm, "AFMR"); - ret = snd_soc_dapm_sync(dapm); return ret; -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 0/3] Add Jack support for
by Olaya, Margarita 07 Dec '10

07 Dec '10

This patch series adds the support for headset jack detection v2: Include codec patch in the series Jorge Eduardo Candelaria (3): ASoC: sdp4430: Enable FM stereo pins ASoC: sdp4430: Add Jack support ASoC: twl6040: Add jack support for headset and handset sound/soc/codecs/twl6040.c | 26 ++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 12 ++++++++++++ sound/soc/omap/sdp4430.c | 39 +++++++++++++++++++++++++++++++++++---- 3 files changed, 73 insertions(+), 4 deletions(-) Regards, Margarita

1 0

[alsa-devel] [PATCH 2/2] ASoC: sdp4430: Add Jack support
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Use jack framework to enable detection for the headset microphone and stereo output. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/omap/sdp4430.c | 28 ++++++++++++++++++++++++++++ 1 files changed, 28 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 0c37c51..ca7fb90 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -24,6 +24,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <plat/hardware.h> @@ -65,6 +66,21 @@ static struct snd_soc_ops sdp4430_ops = { .hw_params = sdp4430_hw_params, }; +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -160,7 +176,19 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); ret = snd_soc_dapm_sync(dapm); + if (ret) + return ret; + + /* Headset jack detection */ + ret = snd_soc_jack_new(codec, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); return ret; } -- 1.7.0.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: sdp4430: Enable FM stereo pins
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Add FM stereo pins to the machine driver and add them as a dapm widget. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/omap/sdp4430.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index ebbd62f..0c37c51 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -101,6 +101,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), + SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -123,6 +124,10 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Earphone speaker */ {"Earphone Spk", NULL, "EP"}, + + /* Aux/FM Stereo In: AFML, AFMR */ + {"AFML", NULL, "Aux/FM Stereo In"}, + {"AFMR", NULL, "Aux/FM Stereo In"}, }; static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) @@ -149,13 +154,11 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) /* SDP4430 connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Mic"); snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "AFML"); + snd_soc_dapm_enable_pin(dapm, "AFMR"); snd_soc_dapm_enable_pin(dapm, "Headset Mic"); snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - /* TWL6040 not connected pins */ - snd_soc_dapm_nc_pin(dapm, "AFML"); - snd_soc_dapm_nc_pin(dapm, "AFMR"); - ret = snd_soc_dapm_sync(dapm); return ret; -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: soc-core: Remove useless inline function construct
by Dimitris Papastamos 06 Dec '10

06 Dec '10

There is no need to mark this function as inline. Inline functions usually are small and concise functions that benefit from not needing to set up a stack frame and undergo a call/ret sequence upon each invocation. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e586616..1b4cf65 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3194,7 +3194,7 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) * Simplify DAI link configuration by removing ".-1" from device names * and sanitizing names. */ -static inline char *fmt_single_name(struct device *dev, int *id) +static char *fmt_single_name(struct device *dev, int *id) { char *found, name[NAME_SIZE]; int id1, id2; -- 1.7.3.3

3 2

[alsa-devel] [PATCH] ASoC: soc-core: Replace use of strncpy() with strlcpy()
by Dimitris Papastamos 06 Dec '10

06 Dec '10

By using strncpy() if the source string does not have a null byte in the first n bytes, then the destination string is not null-terminated. This can be fixed in a two-step process by manually null-terminating the array after the use of strncpy() or by using strlcpy(). Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fd284a7..e586616 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3202,7 +3202,7 @@ static inline char *fmt_single_name(struct device *dev, int *id) if (dev_name(dev) == NULL) return NULL; - strncpy(name, dev_name(dev), NAME_SIZE); + strlcpy(name, dev_name(dev), NAME_SIZE); /* are we a "%s.%d" name (platform and SPI components) */ found = strstr(name, dev->driver->name); @@ -3225,7 +3225,7 @@ static inline char *fmt_single_name(struct device *dev, int *id) /* sanitize component name for DAI link creation */ snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, name); - strncpy(name, tmp, NAME_SIZE); + strlcpy(name, tmp, NAME_SIZE); } else *id = 0; } -- 1.7.3.3

2 1

[alsa-devel] [PATCH] ASoC: Merge common code in DAI link and auxiliary codec probing/removal
by Jarkko Nikula 06 Dec '10

06 Dec '10

Commit 2eea392 "ASoC: Add support for optional auxiliary dailess codecs" added much of code that can be shared with DAI link codec probing/removal. Merge now this common code into new soc_probe_codec, soc_remove_codec and soc_post_component_init functions. Error prints in these functions are converted to use dev_err and to print the error code. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/soc-core.c | 284 +++++++++++++++++++++++--------------------------- 1 files changed, 132 insertions(+), 152 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fd284a7..c0cd9b8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1327,6 +1327,27 @@ out: return 1; } +static void soc_remove_codec(struct snd_soc_codec *codec) +{ + int err; + + if (codec->driver->remove) { + err = codec->driver->remove(codec); + if (err < 0) + dev_err(codec->dev, + "asoc: failed to remove %s: %d\n", + codec->name, err); + } + + /* Make sure all DAPM widgets are freed */ + snd_soc_dapm_free(&codec->dapm); + + soc_cleanup_codec_debugfs(codec); + codec->probed = 0; + list_del(&codec->card_list); + module_put(codec->dev->driver->owner); +} + static void soc_remove_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; @@ -1338,6 +1359,7 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) /* unregister the rtd device */ if (rtd->dev_registered) { device_remove_file(&rtd->dev, &dev_attr_pmdown_time); + device_remove_file(&rtd->dev, &dev_attr_codec_reg); device_unregister(&rtd->dev); rtd->dev_registered = 0; } @@ -1366,22 +1388,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the CODEC */ - if (codec && codec->probed) { - if (codec->driver->remove) { - err = codec->driver->remove(codec); - if (err < 0) - printk(KERN_ERR "asoc: failed to remove %s\n", codec->name); - } - - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&codec->dapm); - - soc_cleanup_codec_debugfs(codec); - device_remove_file(&rtd->dev, &dev_attr_codec_reg); - codec->probed = 0; - list_del(&codec->card_list); - module_put(codec->dev->driver->owner); - } + if (codec && codec->probed) + soc_remove_codec(codec); /* remove the cpu_dai */ if (cpu_dai && cpu_dai->probed) { @@ -1413,8 +1421,105 @@ static void soc_set_name_prefix(struct snd_soc_card *card, } } +static int soc_probe_codec(struct snd_soc_card *card, + struct snd_soc_codec *codec) +{ + int ret = 0; + + codec->card = card; + codec->dapm.card = card; + soc_set_name_prefix(card, codec); + + if (codec->driver->probe) { + ret = codec->driver->probe(codec); + if (ret < 0) { + dev_err(codec->dev, + "asoc: failed to probe CODEC %s: %d\n", + codec->name, ret); + return ret; + } + } + + soc_init_codec_debugfs(codec); + + /* mark codec as probed and add to card codec list */ + codec->probed = 1; + list_add(&codec->card_list, &card->codec_dev_list); + + return ret; +} + static void rtd_release(struct device *dev) {} +static int soc_post_component_init(struct snd_soc_card *card, + struct snd_soc_codec *codec, + int num, int dailess) +{ + struct snd_soc_dai_link *dai_link = NULL; + struct snd_soc_aux_dev *aux_dev = NULL; + struct snd_soc_pcm_runtime *rtd; + const char *temp, *name; + int ret = 0; + + if (!dailess) { + dai_link = &card->dai_link[num]; + rtd = &card->rtd[num]; + name = dai_link->name; + } else { + aux_dev = &card->aux_dev[num]; + rtd = &card->rtd_aux[num]; + name = aux_dev->name; + } + + /* machine controls, routes and widgets are not prefixed */ + temp = codec->name_prefix; + codec->name_prefix = NULL; + + /* do machine specific initialization */ + if (!dailess && dai_link->init) + ret = dai_link->init(rtd); + else if (dailess && aux_dev->init) + ret = aux_dev->init(&codec->dapm); + if (ret < 0) { + dev_err(card->dev, "asoc: failed to init %s: %d\n", name, ret); + return ret; + } + codec->name_prefix = temp; + + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(&codec->dapm); + snd_soc_dapm_sync(&codec->dapm); + + /* register the rtd device */ + rtd->codec = codec; + rtd->card = card; + rtd->dev.parent = card->dev; + rtd->dev.release = rtd_release; + rtd->dev.init_name = name; + ret = device_register(&rtd->dev); + if (ret < 0) { + dev_err(card->dev, + "asoc: failed to register runtime device: %d\n", ret); + return ret; + } + rtd->dev_registered = 1; + + /* add DAPM sysfs entries for this codec */ + ret = snd_soc_dapm_sys_add(&rtd->dev); + if (ret < 0) + dev_err(codec->dev, + "asoc: failed to add codec dapm sysfs entries: %d\n", + ret); + + /* add codec sysfs entries */ + ret = device_create_file(&rtd->dev, &dev_attr_codec_reg); + if (ret < 0) + dev_err(codec->dev, + "asoc: failed to add codec sysfs files: %d\n", ret); + + return 0; +} + static int soc_probe_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; @@ -1422,17 +1527,13 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; - const char *temp; int ret; dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num); /* config components */ codec_dai->codec = codec; - codec->card = card; cpu_dai->platform = platform; - rtd->card = card; - rtd->dev.parent = card->dev; codec_dai->card = card; cpu_dai->card = card; @@ -1456,22 +1557,9 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) /* probe the CODEC */ if (!codec->probed) { - codec->dapm.card = card; - soc_set_name_prefix(card, codec); - if (codec->driver->probe) { - ret = codec->driver->probe(codec); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to probe CODEC %s\n", - codec->name); - return ret; - } - } - - soc_init_codec_debugfs(codec); - - /* mark codec as probed and add to card codec list */ - codec->probed = 1; - list_add(&codec->card_list, &card->codec_dev_list); + ret = soc_probe_codec(card, codec); + if (ret < 0) + return ret; } /* probe the platform */ @@ -1508,47 +1596,14 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); - /* now that all clients have probed, initialise the DAI link */ - if (dai_link->init) { - /* machine controls, routes and widgets are not prefixed */ - temp = rtd->codec->name_prefix; - rtd->codec->name_prefix = NULL; - ret = dai_link->init(rtd); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to init %s\n", dai_link->stream_name); - return ret; - } - rtd->codec->name_prefix = temp; - } - - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(&codec->dapm); - snd_soc_dapm_sync(&codec->dapm); - - /* register the rtd device */ - rtd->dev.release = rtd_release; - rtd->dev.init_name = dai_link->name; - ret = device_register(&rtd->dev); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to register DAI runtime device %d\n", ret); + ret = soc_post_component_init(card, codec, num, 0); + if (ret) return ret; - } - rtd->dev_registered = 1; ret = device_create_file(&rtd->dev, &dev_attr_pmdown_time); if (ret < 0) printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(&rtd->dev); - if (ret < 0) - printk(KERN_WARNING "asoc: failed to add codec dapm sysfs entries\n"); - - /* add codec sysfs entries */ - ret = device_create_file(&rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); - /* create the pcm */ ret = soc_new_pcm(rtd, num); if (ret < 0) { @@ -1606,9 +1661,7 @@ static void soc_unregister_ac97_dai_link(struct snd_soc_codec *codec) static int soc_probe_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_codec *codec; - const char *temp; int ret = -ENODEV; /* find CODEC from registered CODECs*/ @@ -1631,67 +1684,11 @@ found: if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; - codec->card = card; - codec->dapm.card = card; - - soc_set_name_prefix(card, codec); - if (codec->driver->probe) { - ret = codec->driver->probe(codec); - if (ret < 0) { - dev_err(codec->dev, "asoc: failed to probe CODEC"); - return ret; - } - } - - soc_init_codec_debugfs(codec); - - /* mark codec as probed and add to card codec list */ - codec->probed = 1; - list_add(&codec->card_list, &card->codec_dev_list); - - /* now that all clients have probed, initialise the DAI link */ - if (aux_dev->init) { - /* machine controls, routes and widgets are not prefixed */ - temp = codec->name_prefix; - codec->name_prefix = NULL; - ret = aux_dev->init(&codec->dapm); - if (ret < 0) { - dev_err(codec->dev, - "asoc: failed to init %s\n", aux_dev->name); - return ret; - } - codec->name_prefix = temp; - } - - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(&codec->dapm); - snd_soc_dapm_sync(&codec->dapm); - - /* register the rtd device */ - rtd->codec = codec; - rtd->card = card; - rtd->dev.parent = card->dev; - rtd->dev.release = rtd_release; - rtd->dev.init_name = aux_dev->name; - ret = device_register(&rtd->dev); - if (ret < 0) { - dev_err(codec->dev, - "asoc: failed to register aux runtime device %d\n", - ret); - return ret; - } - rtd->dev_registered = 1; - - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(&rtd->dev); + ret = soc_probe_codec(card, codec); if (ret < 0) - dev_err(codec->dev, - "asoc: failed to add codec dapm sysfs entries\n"); + return ret; - /* add codec sysfs entries */ - ret = device_create_file(&rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(codec->dev, "asoc: failed to add codec sysfs files\n"); + ret = soc_post_component_init(card, codec, num, 1); out: return ret; @@ -1701,33 +1698,16 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_codec *codec = rtd->codec; - int err; /* unregister the rtd device */ if (rtd->dev_registered) { + device_remove_file(&rtd->dev, &dev_attr_codec_reg); device_unregister(&rtd->dev); rtd->dev_registered = 0; } - /* remove the CODEC */ - if (codec && codec->probed) { - if (codec->driver->remove) { - err = codec->driver->remove(codec); - if (err < 0) - dev_err(codec->dev, - "asoc: failed to remove %s\n", - codec->name); - } - - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&codec->dapm); - - soc_cleanup_codec_debugfs(codec); - device_remove_file(&rtd->dev, &dev_attr_codec_reg); - codec->probed = 0; - list_del(&codec->card_list); - module_put(codec->dev->driver->owner); - } + if (codec && codec->probed) + soc_remove_codec(codec); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: Remove check for WM8994 AIF clock rate
by Mark Brown 06 Dec '10

06 Dec '10

We do not need the individual AIF clocks to be over 256fs, we only have some minimum requirements on SYSCLK. The SYSCLK rate requirements depend on the DAC and ADC oversampling rates. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d475858..5c3dccc 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1773,10 +1773,6 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif) aif + 1, rate); } - if (rate && rate < 3000000) - dev_warn(codec->dev, "AIF%dCLK is %dHz, should be >=3MHz for optimal performance\n", - aif + 1, rate); - wm8994->aifclk[aif] = rate; snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1 + offset, -- 1.7.2.3

1 0

Re: [alsa-devel] sound: Fixed line limit issue in sound/ac97_bus.c
by Takashi Iwai 06 Dec '10

06 Dec '10

At Mon, 6 Dec 2010 20:37:24 +0530, Jeffrin Jose wrote: > > hello. > a small patch for a format issue is attached with this > mail. Thanks for the patch. I applied it now, but had to fix a few things manually: - A blank line is necessary between the patch subject and the changelog text; otherwise git-am will be confused - You don't have to indent the change text - Align the closing comment "*/" Takashi > -- > software engineer. > department of computer science > rajagiri school of engineering and technology. > [2 0001-sound-Fixed-line-limit-issue-in-sound-ac97_bus.c.patch <text/x-diff; us-ascii (7bit)>] > >From a2c2867876c246420a199a0fb4c36ad2965a42cc Mon Sep 17 00:00:00 2001 > From: Jeffrin Jose <ahiliation(a)yahoo.co.in> > Date: Mon, 6 Dec 2010 19:27:53 +0530 > Subject: [PATCH] sound: Fixed line limit issue in sound/ac97_bus.c > This is a patch to the sound/ac97_bus.c file that fixes up a 80 character > line limit issue found by the checkpatch.pl tool. > Signed-off-by: Jeffrin Jose <ahiliation(a)yahoo.co.in> > > --- > sound/ac97_bus.c | 6 +++--- > 1 files changed, 3 insertions(+), 3 deletions(-) > > diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c > index a351dd0..c93251a 100644 > --- a/sound/ac97_bus.c > +++ b/sound/ac97_bus.c > @@ -19,9 +19,9 @@ > > /* > * Let drivers decide whether they want to support given codec from their > - * probe method. Drivers have direct access to the struct snd_ac97 structure and may > - * decide based on the id field amongst other things. > - */ > + * probe method. Drivers have direct access to the struct snd_ac97 > + * structure and may decide based on the id field amongst other things. > +*/ > static int ac97_bus_match(struct device *dev, struct device_driver *drv) > { > return 1; > -- > 1.7.1 >

1 0

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 06 Dec '10

06 Dec '10

The following changes since commit 28216bf48bbe4314a7a02b7c9822dbb26aca530c: Merge branch 'for-2.6.37' into for-2.6.38 (2010-12-03 17:28:29 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Axel Lin (1): ASoC: Hold client_mutex while calling snd_soc_instantiate_cards() Dimitris Papastamos (1): ASoC: soc-cache: Add optional cache name member to snd_soc_cache_ops Mark Brown (3): ASoC: Add post-CODEC bias level callback for machine driver ASoC: Add trace events for jack detection Merge branch 'for-2.6.37' into for-2.6.38 Seungwhan Youn (2): ASoC: SAMSUNG: Fix initial return value ASoC: SAMSUNG: Remove duplicated snd_card on smdk_spdif Uk Kim (2): ASoC: Fix off by one error in WM8994 EQ register bank size ASoC: Fix swap of left and right channels for WM8993/4 speaker boost gain include/linux/mfd/wm8994/pdata.h | 2 +- include/sound/soc.h | 3 ++ include/trace/events/asoc.h | 59 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/samsung/smdk_spdif.c | 4 +-- sound/soc/soc-cache.c | 18 ++++++++++- sound/soc/soc-core.c | 6 ++-- sound/soc/soc-dapm.c | 4 ++ sound/soc/soc-jack.c | 7 ++++ 9 files changed, 95 insertions(+), 10 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.37
by Mark Brown 06 Dec '10

06 Dec '10

The following changes since commit af745bd724e304cc94a88a2fb43d53baffbd63a5: Merge branch 'for-2.6.37' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into fix/asoc (2010-12-02 17:31:18 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.37 More small localised fixes. Axel Lin (1): ASoC: Hold client_mutex while calling snd_soc_instantiate_cards() Uk Kim (2): ASoC: Fix off by one error in WM8994 EQ register bank size ASoC: Fix swap of left and right channels for WM8993/4 speaker boost gain include/linux/mfd/wm8994/pdata.h | 2 +- sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/soc-core.c | 2 ++ 3 files changed, 4 insertions(+), 2 deletions(-)

2 3

[alsa-devel] [PATCH] ASoC: Hold client_mutex while calling snd_soc_instantiate_cards()
by Axel Lin 06 Dec '10

06 Dec '10

As the comments of snd_soc_instantiate_cards() said, snd_soc_instantiate_cards() must be called with client_mutex. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index eb950f7..0d69b91 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3299,7 +3299,9 @@ int snd_soc_register_dais(struct device *dev, pr_debug("Registered DAI '%s'\n", dai->name); } + mutex_lock(&client_mutex); snd_soc_instantiate_cards(); + mutex_unlock(&client_mutex); return 0; err: -- 1.7.2

3 2

[alsa-devel] [PATCH] ASoC: soc-cache: Add optional cache name member to snd_soc_cache_ops
by Dimitris Papastamos 06 Dec '10

06 Dec '10

Added an optional name member to snd_soc_cache_ops to enable more sensible diagnostic messages during cache init, exit and sync. Remove redundant newline in source code. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-cache.c | 18 ++++++++++++++++-- sound/soc/soc-core.c | 3 --- 3 files changed, 17 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0eea08e..cabc450 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -437,6 +437,7 @@ struct snd_soc_ops { /* SoC cache ops */ struct snd_soc_cache_ops { + const char *name; enum snd_soc_compress_type id; int (*init)(struct snd_soc_codec *codec); int (*exit)(struct snd_soc_codec *codec); diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index ff2bc8b..678fd75 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1533,6 +1533,7 @@ static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) static const struct snd_soc_cache_ops cache_types[] = { { .id = SND_SOC_FLAT_COMPRESSION, + .name = "flat", .init = snd_soc_flat_cache_init, .exit = snd_soc_flat_cache_exit, .read = snd_soc_flat_cache_read, @@ -1541,6 +1542,7 @@ static const struct snd_soc_cache_ops cache_types[] = { }, { .id = SND_SOC_LZO_COMPRESSION, + .name = "LZO", .init = snd_soc_lzo_cache_init, .exit = snd_soc_lzo_cache_exit, .read = snd_soc_lzo_cache_read, @@ -1549,6 +1551,7 @@ static const struct snd_soc_cache_ops cache_types[] = { }, { .id = SND_SOC_RBTREE_COMPRESSION, + .name = "rbtree", .init = snd_soc_rbtree_cache_init, .exit = snd_soc_rbtree_cache_exit, .read = snd_soc_rbtree_cache_read, @@ -1573,8 +1576,12 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) mutex_init(&codec->cache_rw_mutex); codec->cache_ops = &cache_types[i]; - if (codec->cache_ops->init) + if (codec->cache_ops->init) { + if (codec->cache_ops->name) + dev_dbg(codec->dev, "Initializing %s cache for %s codec\n", + codec->cache_ops->name, codec->name); return codec->cache_ops->init(codec); + } return -EINVAL; } @@ -1584,8 +1591,12 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) */ int snd_soc_cache_exit(struct snd_soc_codec *codec) { - if (codec->cache_ops && codec->cache_ops->exit) + if (codec->cache_ops && codec->cache_ops->exit) { + if (codec->cache_ops->name) + dev_dbg(codec->dev, "Destroying %s cache for %s codec\n", + codec->cache_ops->name, codec->name); return codec->cache_ops->exit(codec); + } return -EINVAL; } @@ -1657,6 +1668,9 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) } if (codec->cache_ops && codec->cache_ops->sync) { + if (codec->cache_ops->name) + dev_dbg(codec->dev, "Syncing %s cache for %s codec\n", + codec->cache_ops->name, codec->name); ret = codec->cache_ops->sync(codec); if (!ret) codec->cache_sync = 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 78a0101..880ef4b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1741,8 +1741,6 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, /* override the compress_type if necessary */ if (compress_type && codec->compress_type != compress_type) codec->compress_type = compress_type; - dev_dbg(codec->dev, "Cache compress_type for %s is %d\n", - codec->name, codec->compress_type); ret = snd_soc_cache_init(codec); if (ret < 0) { dev_err(codec->dev, "Failed to set cache compression type: %d\n", @@ -1753,7 +1751,6 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, return 0; } - static void snd_soc_instantiate_card(struct snd_soc_card *card) { struct platform_device *pdev = to_platform_device(card->dev); -- 1.7.3.2

3 2

[alsa-devel] [PATCH 1/2] ASoC: SAMSUNG: Fix initial return value
by Seungwhan Youn 06 Dec '10

06 Dec '10

This patch fixed intial return value to be a '0' as asuccess on set_audio_clock_heirachy(). This avoids unintended error on initialize. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/samsung/smdk_spdif.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index ca779ad..56d058d 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -28,7 +28,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev) { struct clk *fout_epll, *mout_epll, *sclk_audio0, *sclk_spdif; - int ret; + int ret = 0; fout_epll = clk_get(NULL, "fout_epll"); if (IS_ERR(fout_epll)) { -- 1.6.2.5

4 3

[alsa-devel] [PATCH 1/2] ASoC: Fix off by one error in WM8994 EQ register bank size
by Mark Brown 06 Dec '10

06 Dec '10

From: Uk Kim <w0806.kim(a)samsung.com> Signed-off-by: Uk Kim <w0806.kim(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/pdata.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 5c51f36..add8a1b 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -29,7 +29,7 @@ struct wm8994_ldo_pdata { #define WM8994_CONFIGURE_GPIO 0x8000 #define WM8994_DRC_REGS 5 -#define WM8994_EQ_REGS 19 +#define WM8994_EQ_REGS 20 /** * DRC configurations are specified with a label and a set of register -- 1.7.2.3

2 2

[alsa-devel] [PATCH] ASoC: Add trace events for jack detection
by Mark Brown 06 Dec '10

06 Dec '10

As jack detection can trigger DAPM and the latency in debouncing can create confusing windows in operation provide some trace events which will hopefully help in diagnostics. The soc-jack core traces all reports that it gets and the resulting notifications to upper layers. An event for jack IRQs is also provided for instrumentation of debounce, and used in the GPIO jack code. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/trace/events/asoc.h | 58 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-jack.c | 7 +++++ 2 files changed, 65 insertions(+), 0 deletions(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 9978856..4a7ab26 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -170,6 +170,64 @@ DEFINE_EVENT(snd_soc_dapm_widget, snd_soc_dapm_widget_event_done, ); +TRACE_EVENT(snd_soc_jack_irq, + + TP_PROTO(const char *name), + + TP_ARGS(name), + + TP_STRUCT__entry( + __string( name, name ) + ), + + TP_fast_assign( + __assign_str(name, w->name); + ), + + TP_printk("%s", __get_str(name)) +); + +TRACE_EVENT(snd_soc_jack_report, + + TP_PROTO(struct snd_soc_jack *jack, int mask, int val), + + TP_ARGS(jack, mask, val), + + TP_STRUCT__entry( + __string( name, jack->jack->name ) + __field( int, mask ) + __field( int, val ) + ), + + TP_fast_assign( + __assign_str(name, jack->jack->name); + __entry->mask = mask; + __entry->val = val; + ), + + TP_printk("jack=%s %x/%x", __get_str(name), (int)__entry->val, + (int)__entry->mask) +); + +TRACE_EVENT(snd_soc_jack_notify, + + TP_PROTO(struct snd_soc_jack *jack, int val), + + TP_ARGS(jack, val), + + TP_STRUCT__entry( + __string( name, jack->jack->name ) + __field( int, val ) + ), + + TP_fast_assign( + __assign_str(name, jack->jack->name); + __entry->val = val; + ), + + TP_printk("jack=%s %x", __get_str(name), (int)__entry->val) +); + #endif /* _TRACE_ASOC_H */ /* This part must be outside protection */ diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0e9b071..ac5a5bc 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -17,6 +17,7 @@ #include <linux/interrupt.h> #include <linux/workqueue.h> #include <linux/delay.h> +#include <trace/events/asoc.h> /** * snd_soc_jack_new - Create a new jack @@ -64,6 +65,8 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) int enable; int oldstatus; + trace_snd_soc_jack_report(jack, mask, status); + if (!jack) return; @@ -82,6 +85,8 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) if (mask && (jack->status == oldstatus)) goto out; + trace_snd_soc_jack_notify(jack, status); + list_for_each_entry(pin, &jack->pins, list) { enable = pin->mask & jack->status; @@ -210,6 +215,8 @@ static irqreturn_t gpio_handler(int irq, void *data) struct snd_soc_jack_gpio *gpio = data; struct device *dev = gpio->jack->codec->card->dev; + trace_snd_soc_jack_irq(gpio->name); + if (device_may_wakeup(dev)) pm_wakeup_event(dev, gpio->debounce_time + 50); -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: Add post-CODEC bias level callback for machine driver
by Mark Brown 06 Dec '10

06 Dec '10

Currently the machine driver can only do bias level configuration before the CODEC bias level is brought up. This means that the machine cannot do any configuration which depends on the CODEC bias level being maintained. Provide a post-CODEC callback which allows the machine driver to do things like enable the FLL on a CODEC which is brought down to BIAS_OFF when idle. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-dapm.c | 4 ++++ 2 files changed, 6 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0eea08e..4abc2f8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -632,6 +632,8 @@ struct snd_soc_card { /* callbacks */ int (*set_bias_level)(struct snd_soc_card *, enum snd_soc_bias_level level); + int (*set_bias_level_post)(struct snd_soc_card *, + enum snd_soc_bias_level level); long pmdown_time; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6a29d59..9af2d8a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -162,6 +162,10 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_card *card, else dapm->bias_level = level; } + if (ret == 0) { + if (card && card->set_bias_level_post) + ret = card->set_bias_level_post(card, level); + } trace_snd_soc_bias_level_done(card, level); -- 1.7.2.3

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[alsa-devel] [PATCH] ALSA: hda: Use position_fix=1 for Acer Aspire 5538 to enable capture on internal mic
by Daniel Chen 06 Dec '10

06 Dec '10

Hi, Attached is a patch against sound-2.6 master HEAD (c2e685d3) enabling internal microphone capture on the reporter's Acer Aspire 5538. Please apply. Thanks, -Dan

2 1

[alsa-devel] [PATCH 2/2] ASoC: SAMSUNG: Remove duplicated snd_card on smdk_spdif
by Seungwhan Youn 06 Dec '10

06 Dec '10

This patch remove duplicated snd_card defination on smdk_spdif. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/samsung/smdk_spdif.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index 56d058d..cb2f4d0 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -152,8 +152,6 @@ static struct snd_soc_ops smdk_spdif_ops = { .hw_params = smdk_hw_params, }; -static struct snd_soc_card smdk; - static struct snd_soc_dai_link smdk_dai = { .name = "S/PDIF", .stream_name = "S/PDIF PCM Playback", -- 1.6.2.5

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[alsa-devel] [PATCH] ALSA: hda - use generic hdmi parser for ATI R6xx codec
by Anssi Hannula 05 Dec '10

05 Dec '10

Switch to the generic hdmi parser for codec id 1002:aa01 (ATI R6xx HDMI), as the codec appears to work fine with it. Note that the codec is still limited to stereo output only, despite it reportedly being multichannel capable. Some as of yet unknown quirks will be needed to get that working. Testing was done on 2.6.36 by John Ettedgui. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Tested-by: John Ettedgui <john.ettedgui(a)gmail.com> --- As per the "ALSA Multichannel through HDMI" thread, the generic parser seems to work for this. So here's a patch against sound master. sound/pci/hda/patch_hdmi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d3e49aa..f2267cf 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1533,7 +1533,7 @@ static struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi }, -{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi }, +{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_generic_hdmi }, { .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_generic_hdmi }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_generic_hdmi }, { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_generic_hdmi }, -- 1.7.3

2 1

Re: [alsa-devel] poor sound from ALC 850/880 in alsa 1.0.21 and 1.0.23
by Colin Kincaid Williams 05 Dec '10

05 Dec '10

On Fri, Nov 26, 2010 at 8:35 PM, Daniel Chen <seven.steps(a)gmail.com> wrote: > Please use http://www.alsa-project.org/alsa-info.sh >

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Re: [alsa-devel] Building alsa-driver-1.0.23 for sis7012
by JD 04 Dec '10

04 Dec '10

On 12/03/2010 10:32 PM, Daniel Chen wrote: > > No, it's probably because the capture and/or mic mixer levels are > muted. Please see https://wiki.ubuntu.com/Audio/AlsaInfo. > I checked. Nothing is muted. This is a big bummer for me because I cannot use Skype. I uploaded the output of the shell script your refered me to. The output is at http://www.alsa-project.org/db/?f=f4e4e92704c233d60d6b3fe9704b9ee4ea9e9f8f

2 2

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 04 Dec '10

04 Dec '10

The following changes since commit 649e5fb033bbadbfa78aba5e2711f1faf2affaf0: Merge branch 'for-2.6.37' into for-2.6.38 (2010-11-29 18:33:47 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Axel Lin (4): ASoC: Fix missing spin_unlock_irqrestore ASoC: Simplify pm860x_probe error handling ASoC: ak4535: Improve readability for setting mute ASoC: Fix inconsistent meaning of default case while checking alc5623->id Daniel Glöckner (3): s6000-i2s: fix compilation s6000-pcm: fix compilation s6105-ipcam: fix compilation Dimitris Papastamos (7): ASoC: soc-cache: Fix memory overflow in LZO initialization ASoC: WM8731: Fix incorrect mask for bypass path disable ASoC: Add compress_type as a member to snd_soc_codec ASoC: Change the base value of compress_type ASoC: soc-core: Generalize snd_soc_prefix_map and rename to snd_soc_codec_conf ASoC: soc-cache: Use reg_def_copy instead of reg_cache_default ASoC: soc-core: Allow machine drivers to override compress_type Jarkko Nikula (4): ASoC: Add support for optional auxiliary dailess codecs ASoC: Fix build failure in soc-core.c ASoC: omap: N810: Don't select CONFIG_OMAP_MUX but make it as dependency ASoC: Don't oops in soc_probe_aux_dev in case of missing codec Jassi Brar (1): ASoC: WM8580: Debug interface index Kuninori Morimoto (8): ASoC: sh: fsi-ak4642: Add FSI port and ak464x selection ASoC: sh: fsi-ak4642: tidyup unnecessary variables ASoC: sh: fsi-ak4642: midify card name ASoC: sh: fsi: clean up SPDIF defines ASoC: sh: fsi: remove fsi_master_write ASoC: sh: fsi: change fsi->mst_ctrl to master->a/b_mclk ASoC: sh: fsi: remove runtime register check from fsi_reg_xxx ASoC: sh: fsi: remove runtime register check from fsi_master_xxx Mark Brown (14): Merge branch 'for-2.6.37' into for-2.6.38 Merge branch 'for-2.6.37' into for-2.6.38 ASoC: Correct event flags for WM8958 AIF DACs ASoC: Allow user-specified WM8958 multiband compressor configurations ASoC: Provide WM8731 microphone boost TLV information ASoC: Split WM8731 enumeration array into individual enums ASoC: Automatically manage WM8731 deemphasis ASoC: Tune performance of WM8958 revision A ASoC: Apostrophe patrol in soc-core.c ASoC: Make the DAI ops constant in the DAI structure ASoC: Move active copy of CODEC read and write into runtime structure ASoC: Constify struct snd_soc_codec_driver ASoC: When disabling WM8994 FLL force a source selection Merge branch 'for-2.6.37' into for-2.6.38 Takashi Iwai (1): Merge branch 'for-2.6.37' of git://git.kernel.org/.../lrg/asoc-2.6 into fix/asoc arch/arm/mach-shmobile/board-ap4evb.c | 5 + arch/sh/boards/mach-se/7724/setup.c | 5 + include/linux/mfd/wm8994/pdata.h | 17 ++ include/linux/mfd/wm8994/registers.h | 77 +++++++++ include/sound/soc-dai.h | 2 +- include/sound/soc.h | 46 +++++- sound/soc/codecs/88pm860x-codec.c | 12 +- sound/soc/codecs/ak4535.c | 10 +- sound/soc/codecs/alc5623.c | 10 +- sound/soc/codecs/wm8580.c | 8 +- sound/soc/codecs/wm8731.c | 88 ++++++++++- sound/soc/codecs/wm8994.c | 129 ++++++++++++++-- sound/soc/nuc900/nuc900-pcm.c | 9 +- sound/soc/omap/Kconfig | 2 +- sound/soc/s6000/s6000-i2s.c | 2 +- sound/soc/s6000/s6000-pcm.c | 2 +- sound/soc/s6000/s6105-ipcam.c | 2 +- sound/soc/sh/fsi-ak4642.c | 140 +++++++++++++++-- sound/soc/sh/fsi.c | 207 +++++++++++-------------- sound/soc/soc-cache.c | 58 ++++--- sound/soc/soc-core.c | 275 +++++++++++++++++++++++++++++--- 21 files changed, 867 insertions(+), 239 deletions(-)

2 1

[alsa-devel] Building alsa-driver-1.0.23 for sis7012
by JD 04 Dec '10

04 Dec '10

Hi devels, I would like to build the driver for my sis audio chipset 7012, but I do not see it listed in the list of supported cards, which is printed out from ./configure --help So, if I configure --with-cards=sis7012 will it actually support my chipset? Best regards, JD

1 0

[alsa-devel] [PATCH] ASoC: When disabling WM8994 FLL force a source selection
by Mark Brown 03 Dec '10

03 Dec '10

When we disable the WM8994 FLL code path sharing means that we end up writing out a configuration. Currently this is the currently active input and output frequency (which causes snd_soc_update_bits() to suppress actual writes both immediately and in the common case where we reenable the same configuration later) but we allow machine drivers to pass through a source of zero. Since the register values written are one less than the source constants this causes corruption of other bitfields in the register. Fix this by using the most recently configured FLL source when none is provided. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index af3a98a..b37e95c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1677,6 +1677,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, /* Allow no source specification when stopping */ if (freq_out) return -EINVAL; + src = wm8994->fll[id].src; break; case WM8994_FLL_SRC_MCLK1: case WM8994_FLL_SRC_MCLK2: -- 1.7.1

2 1

[alsa-devel] [PATCH 0/3] ASoC: Laying the groundwork for compress_type overriding
by Dimitris Papastamos 03 Dec '10

03 Dec '10

This patch series is to prepare ASoC for allowing the compress_type of the codec to be overriden by the machine driver. Typically this will be done by providing an snd_soc_codec_conf struct which will contain the new compress_type in the machine driver. Dimitris Papastamos (3): ASoC: Add compress_type as a member to snd_soc_codec ASoC: Change the base value of compress_type ASoC: soc-core: Generalize snd_soc_prefix_map and rename to snd_soc_codec_conf

4 6

[alsa-devel] [PATCH] ASoC: Constify struct snd_soc_codec_driver
by Mark Brown 03 Dec '10

03 Dec '10

Allow the CODEC driver structure to be marked const by making all the APIs that use it do so. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 4 ++-- sound/soc/soc-cache.c | 13 +++++++------ sound/soc/soc-core.c | 5 +++-- 3 files changed, 12 insertions(+), 10 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7d53cc4..0eea08e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -265,7 +265,7 @@ int snd_soc_register_platform(struct device *dev, struct snd_soc_platform_driver *platform_drv); void snd_soc_unregister_platform(struct device *dev); int snd_soc_register_codec(struct device *dev, - struct snd_soc_codec_driver *codec_drv, + const struct snd_soc_codec_driver *codec_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_codec(struct device *dev); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); @@ -453,7 +453,7 @@ struct snd_soc_codec { const char *name_prefix; int id; struct device *dev; - struct snd_soc_codec_driver *driver; + const struct snd_soc_codec_driver *driver; struct mutex mutex; struct snd_soc_card *card; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 6c6ced7..c0497e1 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1078,7 +1078,7 @@ static int snd_soc_lzo_decompress_cache_block(struct snd_soc_codec *codec, static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec, unsigned int reg) { - struct snd_soc_codec_driver *codec_drv; + const struct snd_soc_codec_driver *codec_drv; size_t reg_size; codec_drv = codec->driver; @@ -1090,7 +1090,7 @@ static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec, static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, unsigned int reg) { - struct snd_soc_codec_driver *codec_drv; + const struct snd_soc_codec_driver *codec_drv; size_t reg_size; codec_drv = codec->driver; @@ -1101,7 +1101,7 @@ static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) { - struct snd_soc_codec_driver *codec_drv; + const struct snd_soc_codec_driver *codec_drv; size_t reg_size; codec_drv = codec->driver; @@ -1301,10 +1301,11 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) { struct snd_soc_lzo_ctx **lzo_blocks; size_t reg_size, bmp_size; - struct snd_soc_codec_driver *codec_drv; + const struct snd_soc_codec_driver *codec_drv; int ret, tofree, i, blksize, blkcount; const char *p, *end; unsigned long *sync_bmp; + const char *reg_default; ret = 0; codec_drv = codec->driver; @@ -1404,7 +1405,7 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) { int i; int ret; - struct snd_soc_codec_driver *codec_drv; + const struct snd_soc_codec_driver *codec_drv; unsigned int val; codec_drv = codec->driver; @@ -1504,7 +1505,7 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) { - struct snd_soc_codec_driver *codec_drv; + const struct snd_soc_codec_driver *codec_drv; size_t reg_size; codec_drv = codec->driver; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 41fe13d..35cdb8f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3487,8 +3487,9 @@ static void fixup_codec_formats(struct snd_soc_pcm_stream *stream) * @codec: codec to register */ int snd_soc_register_codec(struct device *dev, - struct snd_soc_codec_driver *codec_drv, - struct snd_soc_dai_driver *dai_drv, int num_dai) + const struct snd_soc_codec_driver *codec_drv, + struct snd_soc_dai_driver *dai_drv, + int num_dai) { size_t reg_size; struct snd_soc_codec *codec; -- 1.7.1

2 1

[alsa-devel] [RFC 0/2] ASoC: core: Option to reorder widget power sequence
by Peter Ujfalusi 03 Dec '10

03 Dec '10

Hello, Quite some time ago I have faced similar issue with the sequence of the DAPM register writes [1] For that issue, I used a workaround within the twl4030 codec driver. Now I have again issue with the register write sequence. The dac33 codec emmits a pop noise, when user enable the analog bypass path. The route cause in this codec are the DAPM update power and register write sequence: 1. User enable the bypass 2. DAPM power up sequence takes place 2.1 codec DAPM widgets got powered on, codec is enabled 2.2 External speaker enabled 3. The bit enabling the bypass got changed. Step 3 cause a pop, which propagate to the speaker, since it is already enabled. The first patch implements a new flag for the core to reorder the sequence to: 1. User enable the bypass 2. The bit enabling the bypass got changed. 3. DAPM power up sequence takes place 3.1 codec DAPM widgets got powered on, codec is enabled 3.2 External speaker enabled The default sequence is kept, codec driver must ask to have the reordered sequence to avoid any side effect on other codecs. [1]: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-August/030024.html Peter --- Peter Ujfalusi (2): ASoC: core: Reordered DAPM update power on widgets ASoC: tlv320dac33: Request for reordered update power sequence include/sound/soc-dapm.h | 1 + sound/soc/codecs/tlv320dac33.c | 1 + sound/soc/soc-dapm.c | 27 +++++++++++++++++++-------- 3 files changed, 21 insertions(+), 8 deletions(-) -- 1.7.3.2

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[alsa-devel] netbook: pcspeaker doesn't work since 2.6.35 see attached alsa-info-files
by Halim Sahin 03 Dec '10

03 Dec '10

Hi, The subject sais it :-(. In alsamixer I can't umute or set any pc speaker related settings since 2.6.35. The control for pc speaker is gone :-(. Now I started the last working kernel 2.6.34, run alsa-info. Then repeated same with 2.6.36. can you please help to solve this? For accessibility software a working pc-speaker is really useful :-). Regards Halim

2 3

[alsa-devel] [PATCH] ASoC: Don't oops in soc_probe_aux_dev in case of missing codec
by Jarkko Nikula 03 Dec '10

03 Dec '10

Blind copy of codec finding algorithm from soc_bind_dai_link does not work in soc_probe_aux_dev if matching codec name is not found. In that case the code falls through and tries to start the probing procedure with invalid codec pointer. Fix this and add an error print showing the codec name that cannot be found. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/soc-core.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index eb950f7..1fd5e72 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1609,7 +1609,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_codec *codec; const char *temp; - int ret = 0; + int ret = -ENODEV; /* find CODEC from registered CODECs*/ list_for_each_entry(codec, &codec_list, list) { @@ -1620,10 +1620,14 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) ret = -EBUSY; goto out; } - break; + goto found; } } + /* codec not found */ + dev_err(card->dev, "asoc: codec %s not found", aux_dev->codec_name); + goto out; +found: if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; -- 1.7.2.3

3 2

[alsa-devel] [PATCH] ASoC: Fix inconsistent meaning of default case while checking alc5623->id
by Axel Lin 03 Dec '10

03 Dec '10

In alc5623_i2c_probe(), the default case for checking alc5623->id behaves the same as case 0x23. However, In alc5623_probe() the default case for checking alc5623->id becomes to be the same as case 0x21. This makes the meaning of default case inconsistent. Since we have checked codec id in alc5623_i2c_probe() by comparing vid2 with id->driver_data, it is not possible to run into the default case now. In case we may add more supported devices to alc5623_i2c_table in the future, this patch changes the default case return -EINVAL to let people know that they should not run into this case. They should also add a new case accordingly for the new id. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Or alternatively, we can simply remove the default cases. Axel sound/soc/codecs/alc5623.c | 10 +++++++--- 1 files changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 9783e7e..4f377c9 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -925,7 +925,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) } switch (alc5623->id) { - default: case 0x21: snd_soc_add_controls(codec, rt5621_vol_snd_controls, ARRAY_SIZE(rt5621_vol_snd_controls)); @@ -938,6 +937,8 @@ static int alc5623_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, alc5623_vol_snd_controls, ARRAY_SIZE(alc5623_vol_snd_controls)); break; + default: + return -EINVAL; } snd_soc_add_controls(codec, alc5623_snd_controls, @@ -950,7 +951,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); switch (alc5623->id) { - default: case 0x21: case 0x22: snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets, @@ -962,6 +962,8 @@ static int alc5623_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, intercon_spk, ARRAY_SIZE(intercon_spk)); break; + default: + return -EINVAL; } return ret; @@ -1039,10 +1041,12 @@ static int alc5623_i2c_probe(struct i2c_client *client, case 0x22: alc5623_dai.name = "alc5622-hifi"; break; - default: case 0x23: alc5623_dai.name = "alc5623-hifi"; break; + default: + kfree(alc5623); + return -EINVAL; } i2c_set_clientdata(client, alc5623); -- 1.7.2

4 3

[alsa-devel] [PATCH] ASoC: sh: fsi: clean up SPDIF defines
by Kuninori Morimoto 03 Dec '10

03 Dec '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 17 ++++++++++++++--- 1 files changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 507e709..76ef3f3 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -50,13 +50,20 @@ /* DO_FMT */ /* DI_FMT */ +#define CR_BWS_24 (0x0 << 20) /* FSI2 */ +#define CR_BWS_16 (0x1 << 20) /* FSI2 */ +#define CR_BWS_20 (0x2 << 20) /* FSI2 */ + +#define CR_DTMD_PCM (0x0 << 8) /* FSI2 */ +#define CR_DTMD_SPDIF_PCM (0x1 << 8) /* FSI2 */ +#define CR_DTMD_SPDIF_STREAM (0x2 << 8) /* FSI2 */ + #define CR_MONO (0x0 << 4) #define CR_MONO_D (0x1 << 4) #define CR_PCM (0x2 << 4) #define CR_I2S (0x3 << 4) #define CR_TDM (0x4 << 4) #define CR_TDM_D (0x5 << 4) -#define CR_SPDIF 0x00100120 /* DOFF_CTL */ /* DIFF_CTL */ @@ -93,6 +100,10 @@ #define IR (1 << 4) /* Interrupt Reset */ #define FSISR (1 << 0) /* Software Reset */ +/* OUT_SEL (FSI2) */ +#define DMMD (1 << 4) /* SPDIF output timing 0: Biphase only */ + /* 1: Biphase and serial */ + /* FIFO_SZ */ #define FIFO_SZ_MASK 0x7 @@ -828,10 +839,10 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, dev_err(dai->dev, "This FSI can not use SPDIF\n"); return -EINVAL; } - data = CR_SPDIF; + data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM; io->chan_num = 2; fsi_spdif_clk_ctrl(fsi, 1); - fsi_reg_mask_set(fsi, OUT_SEL, 0x0010, 0x0010); + fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD); break; default: dev_err(dai->dev, "unknown format.\n"); -- 1.7.1

3 2

[alsa-devel] [PATCH 0/4] ASoC: sh: fsi: remove runtime reg check
by Kuninori Morimoto 03 Dec '10

03 Dec '10

Dear Mark, Liam These are "remove runtime reg check" patches from fsi_xx function. Kuninori Morimoto (4): ASoC: sh: fsi: remove fsi_master_write ASoC: sh: fsi: change fsi->mst_ctrl to master->a/b_mclk ASoC: sh: fsi: remove runtime register check from fsi_reg_xxx ASoC: sh: fsi: remove runtime register check from fsi_master_xxx current runtime check is just useless Best regards -- Kuninori Morimoto

3 6

[PATCH 0/2] Fixed issues/defects reported by Coverity tool.
by Sudarshan 03 Dec '10

03 Dec '10

Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. * First patch here is just a cosmetic one as it removes tabs and trailing whitespaces from the code. * Second patch has got fixes for some minor issues/defects reported by Coverity tool and these fixes can prevent memory leaks and crashes in some cases. It also has some whitespaces and tabs removal differences, please ignore them. Sudarshan (1): alsa-lib: Removed trailing whitespaces as per pulseaudio coding guidelines. bisht (1): alsa-lib: Coverity tool has picked some minor issues in alsa-lib source code, fixed them. aserver/aserver.c | 1813 +++++++------- modules/mixer/simple/sbase.c | 837 ++++---- modules/mixer/simple/sbasedl.c | 126 +- src/alisp/alisp.c | 4 +- src/alisp/alisp_snd.c | 1493 ++++++------ src/conf.c | 5117 ++++++++++++++++++++-------------------- src/control/control_hw.c | 597 +++--- src/control/control_shm.c | 1019 ++++---- src/control/setup.c | 3 +- src/hwdep/hwdep.c | 496 ++-- src/mixer/mixer.c | 962 ++++---- src/mixer/simple_abst.c | 1 + src/mixer/simple_none.c | 15 +- src/pcm/pcm.c | 4812 +++++++++++++++++++------------------- src/pcm/pcm_file.c | 1452 ++++++------ src/pcm/pcm_hooks.c | 937 ++++---- src/pcm/pcm_ladspa.c | 2284 +++++++++--------- src/pcm/pcm_local.h | 3 +- src/pcm/pcm_meter.c | 1634 +++++++------- src/pcm/pcm_rate.c | 2166 +++++++++--------- src/pcm/pcm_share.c | 10 +- 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1 0

[alsa-devel] [PATCH 0/1] ThinkPad T410[s] docking station line-out
by John Baboval 03 Dec '10

03 Dec '10

Hopefully I get the protocol for this correct... If not, let me know and I'll re-submit. On the docking station for the Lenovo T410 and T410s, the line-out doesn't work. The trouble seems to be that it generates a plug event, but then doesn't report that the jack is connected. So automute mutes the jack when you plug something into it. The following patch (next message) fixes it... -John

2 1

[alsa-devel] [PATCH] ALSA: hda: Use model=lg quirk for LG P1 Express to enable playback and capture
by Daniel Chen 03 Dec '10

03 Dec '10

Hi, Attached is a patch against current sound-2.6 master HEAD enabling audible playback from the LG P1 Express's internal speakers. Capture and jack sense were also verified working. This patch is also applicable to all stable, supported kernels. Please apply. Best, -Dan

2 1

[alsa-devel] [PATCH 1/2] ASoC: Make the DAI ops constant in the DAI structure
by Mark Brown 03 Dec '10

03 Dec '10

Neither drivers nor the core should be fiddling with the actual ops structure at runtime. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dai.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 0195b7e..1bafe95 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -203,7 +203,7 @@ struct snd_soc_dai_driver { int (*resume)(struct snd_soc_dai *dai); /* ops */ - struct snd_soc_dai_ops *ops; + const struct snd_soc_dai_ops *ops; /* DAI capabilities */ struct snd_soc_pcm_stream capture; -- 1.7.1

2 2

[alsa-devel] [PATCH] ASoC: Apostrophe patrol in soc-core.c
by Mark Brown 03 Dec '10

03 Dec '10

Silly little grammar nit but it bugs the hell out of me. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index eb950f7..20dcc97 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -525,7 +525,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - /* Check that the codec and cpu DAI's are compatible */ + /* Check that the codec and cpu DAIs are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { runtime->hw.rate_min = max(codec_dai_drv->playback.rate_min, @@ -874,7 +874,7 @@ codec_err: } /* - * Free's resources allocated by hw_params, can be called multiple times + * Frees resources allocated by hw_params, can be called multiple times */ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { @@ -898,7 +898,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) if (platform->driver->ops->hw_free) platform->driver->ops->hw_free(substream); - /* now free hw params for the DAI's */ + /* now free hw params for the DAIs */ if (codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); @@ -1005,7 +1005,7 @@ static int soc_suspend(struct device *dev) /* we're going to block userspace touching us until resume completes */ snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot); - /* mute any active DAC's */ + /* mute any active DACs */ for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *dai = card->rtd[i].codec_dai; struct snd_soc_dai_driver *drv = dai->driver; -- 1.7.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: sh: fsi: remove runtime reg check
by Kuninori Morimoto 03 Dec '10

03 Dec '10

Dear Mark, Liam These are "remove runtime reg check" patches from fsi_xx function. Kuninori Morimoto (4): ASoC: sh: fsi: remove fsi_master_write ASoC: sh: fsi: change fsi->mst_ctrl to master->a/b_mclk ASoC: sh: fsi: remove runtime register check from fsi_reg_xxx ASoC: sh: fsi: remove runtime register check from fsi_master_xxx current runtime check is just useless Best regards -- Kuninori Morimoto

1 0

[alsa-devel] Does anyone really have success experience with SPI codecs with new name-based SOC API?
by Alexander 02 Dec '10

02 Dec '10

Does anyone really have success experience with SPI codecs with new name-based SOC API? There are really few SPI codecs in ASoC subsystem, so I wonder if only I cannot bind my SPI codec to machine driver. The codec is registered with name spi0.0 it's the behavior of SPI subsystem, I suppose. So the form is %s%d.%d and not %s.%d as supposed in fmt_single_name() in soc_core.c For me dev_name(dev)==spi0.0 dev->driver->name==cs4271-codec Is this really working code? ----- strncpy(name, dev_name(dev), NAME_SIZE); /* are we a "%s.%d" name (platform and SPI components) */ found = strstr(name, dev->driver->name); if (found) { /* get ID */ if (sscanf(&found[strlen(dev->driver->name)], ".%d", id) == 1) { /* discard ID from name if ID == -1 */ if (*id == -1) found[strlen(dev->driver->name)] = '\0'; } } else { -----

3 4

[alsa-devel] ASoC updates for 2.6.37
by Mark Brown 02 Dec '10

02 Dec '10

The following changes since commit 649e5fb033bbadbfa78aba5e2711f1faf2affaf0: Merge branch 'for-2.6.37' into for-2.6.38 (2010-11-29 18:33:47 +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 All isolated bug fixes in drivers, and in the case of s6000 the fact that it won't build anyway does lower the risk :) Axel Lin (1): ASoC: Fix missing spin_unlock_irqrestore Daniel Glöckner (3): s6000-i2s: fix compilation s6000-pcm: fix compilation s6105-ipcam: fix compilation Dimitris Papastamos (1): ASoC: WM8731: Fix incorrect mask for bypass path disable sound/soc/codecs/wm8731.c | 2 +- sound/soc/nuc900/nuc900-pcm.c | 9 +++++---- sound/soc/s6000/s6000-i2s.c | 2 +- sound/soc/s6000/s6000-pcm.c | 2 +- sound/soc/s6000/s6105-ipcam.c | 2 +- 5 files changed, 9 insertions(+), 8 deletions(-)

3 5

[alsa-devel] [PATCH 1/1] ThinkPad T410[s] docking station line-out
by John Baboval 02 Dec '10

02 Dec '10

Signed-off-by: John Baboval <john.baboval at virtualcomputer.com> --- diff -r e37815d2e813 sound/pci/hda/patch_conexant.c --- a/sound/pci/hda/patch_conexant.c Mon Nov 29 12:56:29 2010 -0500 +++ b/sound/pci/hda/patch_conexant.c Mon Nov 29 17:35:36 2010 -0500 @@ -2060,20 +2060,29 @@ struct conexant_spec *spec = codec->spec; unsigned int pinctl; - snd_printdd("CXT5066: update speaker, hp_present=%d\n", - spec->hp_present); + snd_printdd("CXT5066: update speaker, hp_present=%d, cur_eapd=%d\n", + spec->hp_present, spec->cur_eapd); /* Port A (HP) */ pinctl = ((spec->hp_present& 1)&& spec->cur_eapd) ? PIN_HP : 0; snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); - /* Port D (HP/LO) */ - pinctl = ((spec->hp_present& 2)&& spec->cur_eapd) - ? spec->port_d_mode : 0; - /* Mute if Port A is connected on Thinkpad */ - if (spec->thinkpad&& (spec->hp_present& 1)) - pinctl = 0; + /* Port D (HP/LO */ + if (spec->dell_automute) { + /* DELL AIO Port Rule: PortA> PortD> IntSpk */ + pinctl = (!(spec->hp_present& 1)&& spec->cur_eapd) + ? PIN_OUT : 0; + } else if (spec->thinkpad) { + if (spec->cur_eapd) + pinctl = spec->port_d_mode; + /* Mute dock line-out if Port A (laptop HP) is present */ + if (spec->hp_present& 1) + pinctl = 0; + } else { + pinctl = ((spec->hp_present& 2)&& spec->cur_eapd) + ? spec->port_d_mode : 0; + } snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); @@ -2081,14 +2090,6 @@ pinctl = (!spec->hp_present&& spec->cur_eapd) ? PIN_OUT : 0; snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); - - if (spec->dell_automute) { - /* DELL AIO Port Rule: PortA> PortD> IntSpk */ - pinctl = (!(spec->hp_present& 1)&& spec->cur_eapd) - ? PIN_OUT : 0; - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); - } } /* turn on/off EAPD (+ mute HP) as a master switch */

1 0

[alsa-devel] [PATCH 1/3] ASoC: Provide WM8731 microphone boost TLV information
by Mark Brown 02 Dec '10

02 Dec '10

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8731.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 94e0cf3..c90e025 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -74,6 +74,7 @@ static const struct soc_enum wm8731_enum[] = { static const DECLARE_TLV_DB_SCALE(in_tlv, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 2000, 0); static const struct snd_kcontrol_new wm8731_snd_controls[] = { @@ -86,7 +87,7 @@ SOC_DOUBLE_R_TLV("Capture Volume", WM8731_LINVOL, WM8731_RINVOL, 0, 31, 0, in_tlv), SOC_DOUBLE_R("Line Capture Switch", WM8731_LINVOL, WM8731_RINVOL, 7, 1, 1), -SOC_SINGLE("Mic Boost (+20dB)", WM8731_APANA, 0, 1, 0), +SOC_SINGLE_TLV("Mic Boost Volume", WM8731_APANA, 0, 1, 0, mic_tlv), SOC_SINGLE("Mic Capture Switch", WM8731_APANA, 1, 1, 1), SOC_SINGLE_TLV("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1, -- 1.7.1

2 3

[alsa-devel] [PATCH] ASoC: Remove unused aic3x_i2c_init and aic3x_i2c_exit functions
by Axel Lin 02 Dec '10

02 Dec '10

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 15 --------------- 1 files changed, 0 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index fc5abdf..8cd4cf5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1545,21 +1545,6 @@ static struct i2c_driver aic3x_i2c_driver = { .remove = aic3x_i2c_remove, .id_table = aic3x_i2c_id, }; - -static inline void aic3x_i2c_init(void) -{ - int ret; - - ret = i2c_add_driver(&aic3x_i2c_driver); - if (ret) - printk(KERN_ERR "%s: error regsitering i2c driver, %d\n", - __func__, ret); -} - -static inline void aic3x_i2c_exit(void) -{ - i2c_del_driver(&aic3x_i2c_driver); -} #endif static int __init aic3x_modinit(void) -- 1.7.2

4 3

[alsa-devel] [PATCH] ASoC: omap: N810: Don't select CONFIG_OMAP_MUX but make it as dependency
by Jarkko Nikula 02 Dec '10

02 Dec '10

Not all omap boards use kernel based pin multiplexing so CONFIG_SND_OMAP_SOC_N810 should not select it by default as it can make harm to other boards in multi-board kernels. Therefore put CONFIG_OMAP_MUX as a dependency to N810 ASoC machine driver. Thanks to Tony Lindgren <tony(a)atomide.com> for noticing. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Tony Lindgren <tony(a)atomide.com> --- sound/soc/omap/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index d542ea2..a088db6 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -12,8 +12,8 @@ config SND_OMAP_SOC_MCPDM config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C + depends on OMAP_MUX select SND_OMAP_SOC_MCBSP - select OMAP_MUX select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. -- 1.7.2.3

4 3

[alsa-devel] [PATCH v2] ASoC: tpa6130a2: Replace DAPM code with direct interface
by Peter Ujfalusi 02 Dec '10

02 Dec '10

The use of DAPM widgets, and extra routing can cause ordering problems in the system. Machine drivers should use the exported direct interface with SND_SOC_DAPM_HP's event callback to manage the state of the amplifier. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello, Changes since v1: - DAPM routing removed from the driver - tpa6130a2_stereo_enable takes snd_soc_codec as argument Peter sound/soc/codecs/tpa6130a2.c | 69 +++++++----------------------------------- sound/soc/codecs/tpa6130a2.h | 1 + 2 files changed, 12 insertions(+), 58 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index c97badf..0a99f31 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -317,65 +317,24 @@ static void tpa6130a2_channel_enable(u8 channel, int enable) } } -static int tpa6130a2_pga_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - switch (event) { - case SND_SOC_DAPM_POST_PMU: - tpa6130a2_channel_enable(w->shift, 1); - break; - case SND_SOC_DAPM_POST_PMD: - tpa6130a2_channel_enable(w->shift, 0); - break; - } - return 0; -} - -static int tpa6130a2_supply_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +int tpa6130a2_stereo_enable(struct snd_soc_codec *codec, int enable) { int ret = 0; - - switch (event) { - case SND_SOC_DAPM_POST_PMU: + if (enable) { ret = tpa6130a2_power(1); - break; - case SND_SOC_DAPM_POST_PMD: + if (ret < 0) + return ret; + tpa6130a2_channel_enable(TPA6130A2_HP_EN_R | TPA6130A2_HP_EN_L, + 1); + } else { + tpa6130a2_channel_enable(TPA6130A2_HP_EN_R | TPA6130A2_HP_EN_L, + 0); ret = tpa6130a2_power(0); - break; } + return ret; } - -static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = { - SND_SOC_DAPM_PGA_E("TPA6130A2 Left", SND_SOC_NOPM, - TPA6130A2_HP_EN_L, 0, NULL, 0, tpa6130a2_pga_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_PGA_E("TPA6130A2 Right", SND_SOC_NOPM, - TPA6130A2_HP_EN_R, 0, NULL, 0, tpa6130a2_pga_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_PGA_E("TPA6130A2 Stereo", SND_SOC_NOPM, - TPA6130A2_HP_EN_L | TPA6130A2_HP_EN_R, 0, NULL, 0, - tpa6130a2_pga_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SUPPLY("TPA6130A2 Enable", SND_SOC_NOPM, - 0, 0, tpa6130a2_supply_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - /* Outputs */ - SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Left"), - SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Right"), - SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Stereo"), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - {"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Left"}, - {"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Right"}, - {"TPA6130A2 Headphone Stereo", NULL, "TPA6130A2 Stereo"}, - - {"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Enable"}, - {"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Enable"}, - {"TPA6130A2 Headphone Stereo", NULL, "TPA6130A2 Enable"}, -}; +EXPORT_SYMBOL_GPL(tpa6130a2_stereo_enable); int tpa6130a2_add_controls(struct snd_soc_codec *codec) { @@ -387,18 +346,12 @@ int tpa6130a2_add_controls(struct snd_soc_codec *codec) data = i2c_get_clientdata(tpa6130a2_client); - snd_soc_dapm_new_controls(dapm, tpa6130a2_dapm_widgets, - ARRAY_SIZE(tpa6130a2_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - if (data->id == TPA6140A2) return snd_soc_add_controls(codec, tpa6140a2_controls, ARRAY_SIZE(tpa6140a2_controls)); else return snd_soc_add_controls(codec, tpa6130a2_controls, ARRAY_SIZE(tpa6130a2_controls)); - } EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); diff --git a/sound/soc/codecs/tpa6130a2.h b/sound/soc/codecs/tpa6130a2.h index 57e867f..5df49c8 100644 --- a/sound/soc/codecs/tpa6130a2.h +++ b/sound/soc/codecs/tpa6130a2.h @@ -57,5 +57,6 @@ #define TPA6130A2_VERSION_MASK (0x0f) extern int tpa6130a2_add_controls(struct snd_soc_codec *codec); +extern int tpa6130a2_stereo_enable(struct snd_soc_codec *codec, int enable); #endif /* __TPA6130A2_H__ */ -- 1.7.3.2

3 3

[alsa-devel] asoc device probing and dai question
by Koul, Vinod 02 Dec '10

02 Dec '10

Hi, In our driver, the machine probe creates the soc-audio device. I am adding the codec and platform device in my soc-device. But the probe for the codec doesn't get called unless we add the cpu and codec dais in the machine probe. I saw the soc-core code and looks like it checks if cpu and codec dais are registerd if so then only proceeds and calls the codec probe. So should I continue to register the dai for both cpu and codec in machine probe. I didn't see other driver in soc/ doing this, so did I miss something. Code: In m/c driver... (the m/c driver probe is called by kernel when it finds this device, and I register it as a platform device) /* Audio machine driver */ static struct snd_soc_card snd_soc_intelmid_card = { .name = "intel_mid_sound_card", .platform = &intelmid_soc_platform, .dai_link = intelmid_msic_dai, .num_links = 1, }; /* Audio subsystem */ static struct snd_soc_device intelmid_snd_devdata = { .card = &snd_soc_intelmid_card, .codec_dev = &intel_msic_codec, }; int __devinit snd_intelmid_mc_probe(struct platform_device *pdev) { int ret_val; snd_printk(KERN_DEBUG "snd_intelmad_probe called\n"); ret_val = snd_soc_register_dais(mid_bsp_dai, ARRAY_SIZE(mid_bsp_dai)); ret_val = snd_soc_register_dais(intel_msic_dais, ARRAY_SIZE(intel_msic_dais)); intelmid_snd_device = platform_device_alloc("soc-audio", -1); if (!intelmid_snd_device) { snd_printk(KERN_ERR "Platform device allocation failed\n"); return -ENOMEM; } platform_set_drvdata(intelmid_snd_device, &intelmid_snd_devdata); intelmid_snd_devdata.dev = &intelmid_snd_device->dev; ret_val = platform_device_add(intelmid_snd_device); if (ret_val) { snd_printk(KERN_ERR "Unable to add platform device\n"); platform_device_put(intelmid_snd_device); } return ret_val; } In platform init (note I don't have probe for platform) static int __init intel_mid_soc_platform_init(void) { snd_printk(KERN_DEBUG "intel_mid_soc_platform_init called\n"); return snd_soc_register_platform(&intelmid_soc_platform); } Thanks Vinod

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[alsa-devel] Hardware events
by Rolf Theunissen 02 Dec '10

02 Dec '10

Hello, For some hardware devices (for instance the ThinkPad EC) the volume can be changed by (only) hardware buttons. Changes in the the hardware mixers are notified in the driver via the function "snd_ctl_notify" with the mask SNDRV_CLT_EVENT_MASK_VALUE. For notification of volume changes to the users of the system, an OSD can be used. To implement such a OSD, it must be possible to respond to changes done by the hardware. This raises the following question: is it possible to detect the events originating from the hardware in the functions that handle the callbacks (set via snd_mixer_elem_set_callback/snd_hctl_elem_set_callback)? This would allow to implement an OSD that only responds to the hardware buttons and that would ignore events originating from "software". Thanks

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[alsa-devel] [PATCH 0/5] ASoC: tpa6130a2: pop removal (mostly) changes
by Peter Ujfalusi 02 Dec '10

02 Dec '10

Hello, The following series improves the tpa driver to have better pop filtering capabilities. New stereo DAPM route can be used to reduce left/right power time differences. Machine drivers can opt to not use the built in DAPM routes, and use direct call to enable the stereo path. --- Peter Ujfalusi (5): ASoC: tpa6130a2: Simplify power state management ASoC: tpa6130a2: Defer SW enable from power enable ASoC: tpa6130a2: Use one event handler for PGA_E ASoC: tpa6130a2: Add stereo DAPM path ASoC: tpa6130a2: Make DAPM registration optional, and direct interface sound/soc/codecs/tpa6130a2.c | 77 ++++++++++++++++++++++++------------------ sound/soc/codecs/tpa6130a2.h | 3 +- 2 files changed, 46 insertions(+), 34 deletions(-) -- 1.7.3.2

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[alsa-devel] [PATCH] ALSA: hda: Use "alienware" model quirk for another SSID
by Daniel Chen 02 Dec '10

02 Dec '10

Hi, Attached is a patch against current sound-2.6 master HEAD (also applicable to all supported/maintained stable kernels inclusive of 2.6.32 and newer) that adds a bug reporter's SSID to the appropriate quirk table so that headphones and mics are usable. Please apply. Best, -Dan

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[alsa-devel] SND-SOC - how to test drivers ?
by Vaclav Peroutka 02 Dec '10

02 Dec '10

Hello all, I am trying to develop ALSA driver for audio subsystem for one Android mobile phone. I already have some messages in the kernel log - please see below. It means that I have s3c-i2s connected to MAX9880 (source code was not IMHO tested). And then the output from MAX9880 goes into unknown switching matrix ( for which I even don't have a datasheet - only register list and names of bits in registers - I can guess what purpose they serve to). What I want is to make some testbench driver and an application which can set switching matrix registers through IOCTL. In MAX9880 I want to preset microphone path to output and switch matrix until I hear my voice from microphone in speaker or headphones. Or, is there any better method how to do that ? AFAIK there is no alsamixer application for Android. What do you use for path and gain settings in this case ? And last question so far - where to look to fix this message: "Failed to add route LMICIN->Mic Input" ? All other routes were added so far. Thank you very much in advance, Vaclav <6>[ 2.932154] Advanced Linux Sound Architecture Driver Version 1.0.18a. <4>[ 2.953782] No device for DAI MAX9880 <4>[ 2.962009] No device for DAI s3c-i2s <4>[ 3.003635] I2S off <4>[ 3.019939] I2S off <4>[ 3.027948] Maxim MAX9880 Audio CODEC 0.12 <6>[ 3.054919] max9880: max9880_init <4>[ 3.063481] Entered s3c24xx_pcm_new <4>[ 3.071566] Entered s3c24xx_pcm_preallocate_dma_buffer <4>[ 3.080354] Entered s3c24xx_pcm_preallocate_dma_buffer <6>[ 3.088901] asoc: MAX9880 <-> s3c-i2s mapping ok <6>[ 3.097441] max9880: max9880_dapm_event 769 <6>[ 3.124311] max9880: max9880_dapm_event 0 <6>[ 3.140593] max9880: max9880_add_widgets <3>[ 3.156166] Failed to add route LMICIN->Mic Input <6>[ 3.164204] max9880: max9880_add_controls <6>[ 3.203234] ALSA device list: <6>[ 3.210962] #0: S3C6410 MAX9880 (MAX9880)

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[alsa-devel] poor sound from ALC 850/880 in alsa 1.0.21 and 1.0.23
by Colin Kincaid Williams 02 Dec '10

02 Dec '10

From: Colin Kincaid Williams <discord(a)uw.edu> Date: Wed, Dec 1, 2010 at 3:47 PM Subject: Re: [alsa-devel] poor sound from ALC 850/880 in alsa 1.0.21 and 1.0.23 To: Raymond Yau <superquad.vortex2(a)gmail.com> how do I mute them manually, without hdaanalyser? On Wed, Dec 1, 2010 at 1:35 AM, Raymond Yau <superquad.vortex2(a)gmail.com> wrote: > 2010/12/1 Colin Kincaid Williams <discord(a)uw.edu> > >> I selected node 0x0c in hdaanalyzer and see audio mixer [OxOb] in the >> Connection list, but I cannot select the radio button. >> > > According to the block diagram of alc880 datasheet, it look like two mute > switches for node 0x02 and 0x0b or checkbox in hda_analyzer > >> Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-In vals: [0x00 0x00] [0x00 0x00] > > Unfortunately hda_analyzer seem fail to parse your codec -info > > File "./hda_analyzer.py", line 483, in <module> > sys.exit(main(sys.argv)) > File "./hda_analyzer.py", line 460, in main > if read_nodes(sys.argv[1:]) == 0: > File "./hda_analyzer.py", line 110, in read_nodes > read_nodes3(card, idx, i) > File "./hda_analyzer.py", line 77, in read_nodes3 > c = HDACodecProc(card, codec, proc_file) > File "/home/raymond/hda-analyzer/hda_proc.py", line 474, in __init__ > self.parse(proc_file) > File "/home/raymond/hda-analyzer/hda_proc.py", line 678, in parse > node.add_ampvals(line[15:], HDA_INPUT) > File "/home/raymond/hda-analyzer/hda_proc.py", line 353, in add_ampvals > self.wrongfile('amp vals ]') > File "/home/raymond/hda-analyzer/hda_proc.py", line 143, in wrongfile > raise ValueError, "wrong proc file format (%s)" % msg > ValueError: wrong proc file format (amp vals ]) > > > > >> >> On Sun, Nov 28, 2010 at 11:40 PM, Raymond Yau >> <superquad.vortex2(a)gmail.com> wrote: >> > 2010/11/28 Colin Kincaid Williams <discord(a)uw.edu> >> > >> >> no, the motherboard has 6 plugs, but no digital outputs on the back. >> >> Furthermore, I muted the CD channel, the line in was already muted, >> >> but I still have crap sound compared to my $20 mp3 player. Something >> >> else is wrong! >> >> >> >> >> > Take a look at the block diagram of alc880 datasheet >> > >> > does it help when you use hda-analyzer to mute the amp-in value from node >> > 0x0b in 0x0c ? >> > >> > >> > Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >> > Control: name="Front Playback Volume", index=0, device=0 >> > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 >> > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> > Amp-In vals: [0x00 0x00] [0x00 0x00] >> > Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 >> > Amp-Out vals: [0x33 0x33] >> > Connection: 2 >> > 0x02 0x0b >> > >> > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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[alsa-devel] MAYA44E
by c0 01 Dec '10

01 Dec '10

Hi. My name is Jose. The maya44 (pci) seems to be supported in snd-ice17xx drivers. But the new maya44e (pci express) is "only" recognized in my system as: root@debian:/home/co# lspci -vvv -s 04:00.0 04:00.0 Multimedia audio controller: Device 19fe:7000 (rev 03) Subsystem: Device 0e51:0001 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (40000ns max) Interrupt: pin A routed to IRQ 7 Region 0: Memory at fd9fc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- I would like to help to make this card work. Best regards. Jose.

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[alsa-devel] [PATCH 0/2] ASoC: sh: fsi-ak4642: modify patches
by Kuninori Morimoto 01 Dec '10

01 Dec '10

Dear Mark, Liam These are FSI-AK4642 modify patches. Kuninori Morimoto (2): ASoC: sh: fsi-ak4642: tidyup unnecessary variables ASoC: sh: fsi-ak4642: midify card name BTW, current Mark's for-2.6.38 branch has below issue ? ---------------------- /linux-2.6/sound/soc/soc-core.c: In function 'soc_probe_aux_dev': /linux-2.6/sound/soc/soc-core.c:1647: error: 'struct snd_soc_dapm_context' has no member named 'list' /linux-2.6/sound/soc/soc-core.c:1647: error: 'struct snd_soc_card' has no member named 'dapm_list' make[4]: *** [sound/soc/soc-core.o] Error 1 make[4]: *** Waiting for unfinished jobs.... ---------------------- Best regards -- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: Fix build failure in soc-core.c
by Jarkko Nikula 01 Dec '10

01 Dec '10

Commit 2eea392 "ASoC: Add support for optional auxiliary dailess codecs" causes a build failure in soc-core.c: soc_probe_aux_dev since code tries to access non-existing struct snd_soc_dapm_context and struct snd_soc_card members. Root cause for this was a double accident. Author sent the RFC patch from top of another patch set and the RFC got committed. Fix the build failure by removing the code line that depends on that another patch set. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/soc-core.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a7670d5..eb950f7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1644,7 +1644,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) /* mark codec as probed and add to card codec list */ codec->probed = 1; list_add(&codec->card_list, &card->codec_dev_list); - list_add(&codec->dapm.list, &card->dapm_list); /* now that all clients have probed, initialise the DAI link */ if (aux_dev->init) { -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: WM8731: Fix incorrect mask for bypass path disable
by Dimitris Papastamos 01 Dec '10

01 Dec '10

According to the datasheet the bypass path enable/disable is bit 3 therefore we need 0x8 and not 0x4. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8731.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index a104145..94e0cf3 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -526,7 +526,7 @@ static int wm8731_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0); /* Disable bypass path by default */ - snd_soc_update_bits(codec, WM8731_APANA, 0x4, 0); + snd_soc_update_bits(codec, WM8731_APANA, 0x8, 0); snd_soc_add_controls(codec, wm8731_snd_controls, ARRAY_SIZE(wm8731_snd_controls)); -- 1.7.3.2

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[alsa-devel] [PATCH 1/5] [RFC]intel_hdmi_audio:include header
by ramesh.babu＠intel.com 01 Dec '10

01 Dec '10

From: Ramesh Babu K V <ramesh.babu(a)intel.com> This patch creates intel_mid_hdmi_audio.h file. This file contains Intel MID platform specific HDMI Audio register definitions, enums and structure definitions used in this driver. This also exposes interface function. Signed-off-by: Ramesh Babu K V <ramesh.babu(a)intel.com> Signed-off-by: Sailaja Bandarupalli <sailaja.bandarupalli(a)intel.com> --- .../drivers/intel_mid_hdmi/intel_mid_hdmi_audio.h | 424 ++++++++++++++++++++ 1 files changed, 424 insertions(+), 0 deletions(-) create mode 100644 sound/drivers/intel_mid_hdmi/intel_mid_hdmi_audio.h diff --git a/sound/drivers/intel_mid_hdmi/intel_mid_hdmi_audio.h b/sound/drivers/intel_mid_hdmi/intel_mid_hdmi_audio.h new file mode 100644 index 0000000..19e47ee --- /dev/null +++ b/sound/drivers/intel_mid_hdmi/intel_mid_hdmi_audio.h @@ -0,0 +1,424 @@ +/* + * intel_mid_hdmi_audio.h - Intel HDMI audio driver for MID + * + * Copyright (C) 2010 Intel Corp + * Authors: Sailaja Bandarupalli <sailaja.bandarupalli(a)intel.com> + * Ramesh Babu K V <ramesh.babu(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * ALSA driver for Intel MID HDMI audio controller + */ +#ifndef __INTEL_MID_HDMI_AUDIO_H +#define __INTEL_MID_HDMI_AUDIO_H + +#include <linux/types.h> +#include <sound/initval.h> +#include "../../../drivers/staging/mrst/drv/mdfld_hdmi_audio_if.h" +#include "../../../drivers/staging/mrst/drv/psb_intel_hdmi.h" + +#define HAD_DRIVER_VERSION "0.01.002" +#define HAD_MAX_DEVICES 1 +#define HAD_MIN_CHANNEL 1 +#define HAD_MAX_CHANNEL 8 +#define HAD_NUM_OF_RING_BUFS 4 +#define HAD_MIN_RATE 32000 +#define HAD_MAX_RATE 192000 +#define HAD_MAX_BUFFER (800*1024) +#define HAD_MIN_BUFFER (800*1024) +#define HAD_MAX_PERIODS 4 +#define HAD_MIN_PERIODS 1 +#define HAD_MAX_PERIOD_BYTES HAD_MAX_BUFFER +#define HAD_MIN_PERIOD_BYTES 32 +#define HAD_FIFO_SIZE 0 /* fifo not being used */ + +#define AUD_SAMPLE_RATE_32 HAD_MIN_RATE +#define AUD_SAMPLE_RATE_44_1 44100 +#define AUD_SAMPLE_RATE_48 48000 +#define AUD_SAMPLE_RATE_88_2 88200 +#define AUD_SAMPLE_RATE_96 96000 +#define AUD_SAMPLE_RATE_176_4 176400 +#define AUD_SAMPLE_RATE_192 HAD_MAX_RATE + +#define DRIVER_NAME "intelmid_hdmi_audio" +#define DIS_SAMPLE_RATE_25_2 25200 +#define DIS_SAMPLE_RATE_27 27000 +#define DIS_SAMPLE_RATE_54 54000 +#define DIS_SAMPLE_RATE_74_25 74250 +#define DIS_SAMPLE_RATE_148_5 148500 +#define REG_BIT_0 0x1 +#define REG_BIT_1 0x2 +#define REG_BIT_2 0x4 +#define REG_BIT_3 0x8 +#define REG_BIT_20 0x100000 +#define SET_BIT0 1 +#define CLEAR_BIT0 0 +#define REG_WIDTH 0x08 +#define MAX_HW_BUFS 0x04 +#define MAX_DIP_WORDS 16 +#define MAX_PERIODS 4 +#define INTEL_HAD "INTEL_HAD" + +/** + * enum had_status - Audio stream states + * + * @STREAM_INIT: Stream initialized + * @STREAM_RUNNING: Stream running + * @STREAM_PAUSED: Stream paused + * @STREAM_DROPPED: Stream dropped + * */ +enum had_stream_status { + STREAM_INIT = 0, + STREAM_RUNNING = 1, + STREAM_PAUSED = 2, + STREAM_DROPPED = 3 +}; + +/** + * enum intel_had_aud_buf_type - HDMI controller ring buffer types + * + * @HAD_BUF_TYPE_A: ring buffer A + * @HAD_BUF_TYPE_B: ring buffer B + * @HAD_BUF_TYPE_C: ring buffer C + * @HAD_BUF_TYPE_D: ring buffer D + */ +enum intel_had_aud_buf_type { + HAD_BUF_TYPE_A = 0, + HAD_BUF_TYPE_B = 1, + HAD_BUF_TYPE_C = 2, + HAD_BUF_TYPE_D = 3, +}; + +enum num_aud_ch { + CH_STEREO = 0, + CH_THREE_FOUR = 1, + CH_FIVE_SIX = 2, + CH_SEVEN_EIGHT = 3 +}; + +/*HDMI controller register offsets*/ +enum hdmi_ctrl_reg { + AUD_CONFIG = 0x69000, + AUD_CH_STATUS_0 = 0x69008, + AUD_CH_STATUS_1 = 0x6900C, + AUD_HDMI_CTS = 0x69010, + AUD_N_ENABLE = 0x69014, + AUD_SAMPLE_RATE = 0x69018, + AUD_BUF_CONFIG = 0x69020, + AUD_BUF_CH_SWAP = 0x69024, + AUD_BUF_A_ADDR = 0x69040, + AUD_BUF_A_LENGTH = 0x69044, + AUD_BUF_B_ADDR = 0x69048, + AUD_BUF_B_LENGTH = 0x6904c, + AUD_BUF_C_ADDR = 0x69050, + AUD_BUF_C_LENGTH = 0x69054, + AUD_BUF_D_ADDR = 0x69058, + AUD_BUF_D_LENGTH = 0x6905c, + AUD_CNTL_ST = 0x69060, + AUD_HDMI_STATUS = 0x69068, + AUD_HDMIW_INFOFR = 0x69114, +}; +/** + * union aud_cfg - Audio configuration offset - 69000 + * + * @cfg_regx: individual register bits + * @cfg_regval: full register value + * + * */ +union aud_cfg { + struct { + u32 aud_en:1; + u32 layout:1; + u32 fmt:2; + u32 num_ch:2; + u32 rsvd0:1; + u32 set:1; + u32 flat:1; + u32 val_bit:1; + u32 user_bit:1; + u32 underrun:1; + u32 rsvd1:20; + } cfg_regx; + u32 cfg_regval; +}; +/** + * union aud_ch_status_0 - Audio Channel Status 0 Attributes offset - 0x69008 + * + * @status_0_regx:individual register bits + * @status_0_regval:full register value + * + * */ +union aud_ch_status_0 { + struct { + u32 ch_status:1; + u32 lpcm_id:1; + u32 cp_info:1; + u32 format:3; + u32 mode:2; + u32 ctg_code:8; + u32 src_num:4; + u32 ch_num:4; + u32 samp_freq:4; + u32 clk_acc:2; + u32 rsvd:2; + } status_0_regx; + u32 status_0_regval; +}; + +/** + * union aud_ch_status_1 - Audio Channel Status 1 Attributes offset - 0x6900c + * + * @status_1_regx: individual register bits + * @status_1_regval: full register value + * + **/ +union aud_ch_status_1 { + struct { + u32 max_wrd_len:1; + u32 wrd_len:3; + u32 rsvd:28; + } status_1_regx; + u32 status_1_regval; +}; + +/** + * union aud_hdmi_cts - CTS register offset -0x69010 + * + * @cts_regx: individual register bits + * @cts_regval: full register value + * + * */ +union aud_hdmi_cts { + struct { + u32 cts_val:20; + u32 en_cts_prog:1; + u32 rsvd:11; + } cts_regx; + u32 cts_regval; +}; + +/** + * union aud_hdmi_n_enable - N register offset -0x69014 + * + * @n_regx: individual register bits + * @n_regval: full register value + * +*/ +union aud_hdmi_n_enable { + struct { + u32 n_val:20; + u32 en_n_prog:1; + u32 rsvd:11; + } n_regx; + u32 n_regval; +}; + +/** + * union aud_buf_config - Audio Buffer configurations offset -0x69020 + * + * @buf_cfg_regx: individual register bits + * @buf_cfgval: full register value + * +*/ +union aud_buf_config { + struct { + u32 fifo_width:8; + u32 rsvd0:8; + u32 aud_delay:8; + u32 rsvd1:8; + } buf_cfg_regx; + u32 buf_cfgval; +}; + +/** + * union aud_buf_ch_swap - Audio Sample Swapping offset - 0x69024 + * + * @buf_ch_swap_regx: individual register bits + * @buf_ch_swap_val: full register value + * + * */ +union aud_buf_ch_swap { + struct { + u32 first_0:3; + u32 second_0:3; + u32 first_1:3; + u32 second_1:3; + u32 first_2:3; + u32 second_2:3; + u32 first_3:3; + u32 second_3:3; + u32 rsvd:8; + } buf_ch_swap_regx; + u32 buf_ch_swap_val; +}; + +/** + * union aud_buf_addr - Address for Audio Buffer + * + * @buf_addr_regx: individual register bits + * @buf_addr_val: full register value + * + * */ +union aud_buf_addr { + struct { + u32 valid:1; + u32 intr_en:1; + u32 rsvd:4; + u32 addr:26; + } buf_addr_regx; + u32 buf_addr_val; +}; +/** + * union aud_buf_len - Length of Audio Buffer + * + * @buf_len_regx: individual register bits + * @buf_len_val: full register value + * + * */ +union aud_buf_len { + struct { + u32 buf_len:20; + u32 rsvd:12; + } buf_len_regx; + u32 buf_len_val; +}; + +/** + * union aud_ctrl_st - Audio Control State Register offset 0x69060 + * + * @ctrl_regx: individual register bits + * @ctrl_val: full register value + * + * */ +union aud_ctrl_st { + struct { + u32 ram_addr:4; + u32 eld_ack:1; + u32 eld_addr:4; + u32 eld_buf_size:5; + u32 eld_valid:1; + u32 cp_ready:1; + u32 dip_freq:2; + u32 dip_idx:3; + u32 dip_en_sta:4; + u32 rsvd:7; + } ctrl_regx; + u32 ctrl_val; +}; +/** + * union aud_info_frame1 - Audio HDMI Widget Data Island Packet offset 0x69114 + * + * @fr1_regx: individual register bits + * @fr1_val: full register value + * + * */ +union aud_info_frame1 { + struct { + u32 pkt_type:8; + u32 ver_num:8; + u32 len:5; + u32 rsvd:11; + } fr1_regx; + u32 fr1_val; +}; +/** + * union aud_info_frame2 - DIP frame 2 + * + * @fr2_regx: individual register bits + * @fr2_val: full register value + * + */ +union aud_info_frame2 { + struct { + u32 chksum:8; + u32 chnl_cnt:3; + u32 rsvd0:1; + u32 coding_type:4; + u32 smpl_size:2; + u32 smpl_freq:3; + u32 rsvd1:3; + u32 format:8; + } fr2_regx; + u32 fr2_val; +}; +/** + * union aud_info_frame3 - DIP frame 3 + * + * @fr3_regx: individual register bits + * @fr3_val: full register value + * + */ +union aud_info_frame3 { + struct { + u32 chnl_alloc:8; + u32 rsvd0:3; + u32 lsv:4; + u32 dm_inh:1; + u32 rsvd1:16; + } fr3_regx; + u32 fr3_val; +}; + + +struct pcm_stream_info { + int str_id; + void *had_substream; + void (*period_elapsed) (void *had_substream); + u32 buffer_ptr; + u64 buffer_rendered; + u32 ring_buf_size; + int sfreq; +}; + +struct ring_buf_info { + uint32_t buf_addr; + uint32_t buf_size; + uint8_t is_valid; +}; + +struct had_stream_pvt { + enum had_stream_status stream_status; + int stream_ops; + struct snd_pcm_substream *substream; + ssize_t dbg_cum_bytes; +}; + +struct had_callback_ops { + had_event_call_back intel_had_event_call_back; +}; + +/*HAD Driver context*/ +struct snd_intelhad { + struct snd_card *card; /*holds details of snd_card*/ + int card_index; /*card index in alsa*/ + char *card_id; /*name of card*/ + /*holds regsiters read/write callbacks*/ + struct hdmi_audio_registers_ops *reg_ops; + /*holds caps get/set callbacks*/ + struct hdmi_audio_query_set_ops *query_ops; + int playback_cnt; + struct ring_buf_info buf_info[HAD_NUM_OF_RING_BUFS]; + struct pcm_stream_info stream_info; + hdmi_eeld_t eeld; + enum intel_had_aud_buf_type curr_buf; + int valid_buf_cnt; +}; +int had_event_handler(enum had_event_type event_type, void *data); +extern struct snd_intelhad *intelhaddata; +extern struct snd_pcm_ops snd_intelhad_playback_ops; +extern int hdmi_card_index; +extern char *hdmi_card_id; +#endif -- 1.6.2.5

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[alsa-devel] [PATCH] ASoC: ak4535: Improve readability for setting mute
by Axel Lin 01 Dec '10

01 Dec '10

The mute/unmute is controled by SMUTE (Soft Mute Control bit): 0: Normal Operation (Default) 1: DAC outputs soft-muted I think this change improves readability. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4535.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 7367545..8b38739 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -366,9 +366,9 @@ static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai, static int ak4535_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; + u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); if (!mute) - ak4535_write(codec, AK4535_DAC, mute_reg); + ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20); else ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); return 0; @@ -381,11 +381,11 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; - ak4535_write(codec, AK4535_DAC, mute_reg); + mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); + ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20); break; case SND_SOC_BIAS_PREPARE: - mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; + mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); break; case SND_SOC_BIAS_STANDBY: -- 1.7.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: Correct event flags for WM8958 AIF DACs
by Mark Brown 01 Dec '10

01 Dec '10

We need a post notification as we need to shut down the MBC after the data stops flowing rather than before. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e4aab94..c9e5f38 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1205,10 +1205,10 @@ SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture", 0, WM8994_POWER_MANAGEMENT_4, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture", 0, WM8994_POWER_MANAGEMENT_4, 11, 0), @@ -1216,10 +1216,10 @@ SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture", 0, WM8994_POWER_MANAGEMENT_4, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2R", NULL, 0, WM8994_POWER_MANAGEMENT_5, 10, 0, wm8958_aif_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0, aif1adc1l_mix, ARRAY_SIZE(aif1adc1l_mix)), -- 1.7.1

2 2

[alsa-devel] Unable to open CAPTURE device
by Irfan Shaikh 01 Dec '10

01 Dec '10

Hello, I have connected ipod to my hardware (using USB) and i want to read the audio pcm data from ipod. I am running following section of code to detect connection of ipod. But i am not finding any capture device connected. Am i giving some wrong name of capture device ? If wrong how can i find correct name of Audio Capture device. /***********************************************************************************************************************************************/ const char *capture_device="plughw:1,0"; while((retVal = snd_pcm_open( &pcm_capture, (const char *)capture_device, SND_PCM_STREAM_CAPTURE, 1))!=0) { if (retVal != 0){ printf("Opening Capture device FAILED\n"); printf("Still waiting for Capture device to be connected\n"); usleep(5000); } } /***********************************************************************************************************************************************/ OUTPUT : Opening Capture device FAILED Still waiting for Capture device to be connected ALSA lib pcm_hw.c:1433:(_snd_pcm_hw_open) Invalid value for card Opening Capture device FAILED Still waiting for Capture device to be connected ALSA lib pcm_hw.c:1433:(_snd_pcm_hw_open) Invalid value for card Opening Capture device FAILED Still waiting for Capture device to be connected ALSA lib pcm_hw.c:1433:(_snd_pcm_hw_open) Invalid value for card Opening Capture device FAILED Still waiting for Capture device to be connected ALSA lib pcm_hw.c:1433:(_snd_pcm_hw_open) Invalid value for card Opening Capture device FAILED Still waiting for Capture device to be connected . . . . . . . /*Continues*/ . . . . . . . . . . . . . Regards SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

2 4

[alsa-devel] [PATCH 0/2] ASoC: tlv320dac33: codec enable fixes
by Peter Ujfalusi 30 Nov '10

30 Nov '10

Hello, two small fixes to avoid duplicated codec enable, which reduces i2c access to the chip. --- Peter Ujfalusi (2): ASoC: tlv320dac33: Do not enable the codec in init_chip ASoC: tlv320dac33: Avoid multiple soft power up sound/soc/codecs/tlv320dac33.c | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) -- 1.7.3.2

3 5

[alsa-devel] [RFC don't apply] ASoC: Add support for optional auxiliary dailess codecs
by Jarkko Nikula 30 Nov '10

30 Nov '10

This makes possible to register auxiliary dailess codecs in a machine driver. Term dailess is used here for amplifiers and codecs without DAI or DAI being unused. Dailess auxiliary codecs are kept in struct snd_soc_aux_dev and those codecs are probed after initializing the DAI links. There are no major differences between DAI link codecs and dailess codecs in ASoC core point of view. DAPM handles them equally and sysfs and debugfs directories for dailess codecs are similar except the pmdown_time node is not created. Only suspend and resume functions are modified to traverse all probed codecs instead of DAI link codecs. Example below shows a dailess codec registration. struct snd_soc_aux_dev foo_aux_dev[] = { { .name = "Amp", .codec_name = "codec.2", .init = foo_init2, }, }; static struct snd_soc_card card = { ... .aux_dev = foo_aux_dev, .num_aux_devs = ARRAY_SIZE(foo_aux_dev), }; Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- Idea sharing version, not to be applied. Needs at least cross-device DAPM to be usuful and e.g. soc_probe_dai_link and soc_probe_aux_dev share a lot of same code. I'm not entirely sure of reusing struct snd_soc_pcm_runtime but having some common struct on top of it for registering the sysfs nodes and passing to snd_soc_dapm_sys_add didn't sound clear either. Anyway suspend/resume is working with this version and doesn't need any other modifications to soc_suspend/soc_resume than traversing through the registered codecs instead of doing bunch of rtd->dailess etc hacks there. --- include/sound/soc.h | 17 ++++++ sound/soc/soc-core.c | 155 ++++++++++++++++++++++++++++++++++++++++++++++++-- 2 files changed, 166 insertions(+), 6 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 346c59e..c62225e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -583,6 +583,14 @@ struct snd_soc_prefix_map { const char *name_prefix; }; +struct snd_soc_aux_dev { + const char *name; /* Codec name */ + const char *codec_name; /* for multi-codec */ + + /* codec/machine specific init - e.g. add machine controls */ + int (*init)(struct snd_soc_dapm_context *dapm); +}; + /* SoC card */ struct snd_soc_card { const char *name; @@ -624,6 +632,15 @@ struct snd_soc_card { struct snd_soc_prefix_map *prefix_map; int num_prefixes; + /* + * optional auxiliary devices such as amplifiers or codecs with DAI + * link unused + */ + struct snd_soc_aux_dev *aux_dev; + int num_aux_devs; + struct snd_soc_pcm_runtime *rtd_aux; + int num_aux_rtd; + struct work_struct deferred_resume_work; /* lists of probed devices belonging to this card */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b0e1aea..d7fc3a6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -986,6 +986,7 @@ static int soc_suspend(struct device *dev) { struct platform_device *pdev = to_platform_device(dev); struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; int i; /* If the initialization of this soc device failed, there is no codec @@ -1064,8 +1065,7 @@ static int soc_suspend(struct device *dev) } /* suspend all CODECs */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_codec *codec = card->rtd[i].codec; + list_for_each_entry(codec, &card->codec_dev_list, card_list) { /* If there are paths active then the CODEC will be held with * bias _ON and should not be suspended. */ if (!codec->suspended && codec->driver->suspend) { @@ -1106,6 +1106,7 @@ static void soc_resume_deferred(struct work_struct *work) struct snd_soc_card *card = container_of(work, struct snd_soc_card, deferred_resume_work); struct platform_device *pdev = to_platform_device(card->dev); + struct snd_soc_codec *codec; int i; /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, @@ -1131,8 +1132,7 @@ static void soc_resume_deferred(struct work_struct *work) cpu_dai->driver->resume(cpu_dai); } - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_codec *codec = card->rtd[i].codec; + list_for_each_entry(codec, &card->codec_dev_list, card_list) { /* If the CODEC was idle over suspend then it will have been * left with bias OFF or STANDBY and suspended so we must now * resume. Otherwise the suspend was suppressed. @@ -1604,6 +1604,130 @@ static void soc_unregister_ac97_dai_link(struct snd_soc_codec *codec) } #endif +static int soc_probe_aux_dev(struct snd_soc_card *card, int num) +{ + struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; + struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; + struct snd_soc_codec *codec; + const char *temp; + int ret = 0; + + /* find CODEC from registered CODECs*/ + list_for_each_entry(codec, &codec_list, list) { + if (!strcmp(codec->name, aux_dev->codec_name)) { + if (codec->probed) { + dev_err(codec->dev, + "asoc: codec already probed"); + ret = -EBUSY; + goto out; + } + break; + } + } + + if (!try_module_get(codec->dev->driver->owner)) + return -ENODEV; + + codec->card = card; + codec->dapm.card = card; + + soc_set_name_prefix(card, codec); + if (codec->driver->probe) { + ret = codec->driver->probe(codec); + if (ret < 0) { + dev_err(codec->dev, "asoc: failed to probe CODEC"); + return ret; + } + } + + soc_init_codec_debugfs(codec); + + /* mark codec as probed and add to card codec list */ + codec->probed = 1; + list_add(&codec->card_list, &card->codec_dev_list); + list_add(&codec->dapm.list, &card->dapm_list); + + /* now that all clients have probed, initialise the DAI link */ + if (aux_dev->init) { + /* machine controls, routes and widgets are not prefixed */ + temp = codec->name_prefix; + codec->name_prefix = NULL; + ret = aux_dev->init(&codec->dapm); + if (ret < 0) { + dev_err(codec->dev, + "asoc: failed to init %s\n", aux_dev->name); + return ret; + } + codec->name_prefix = temp; + } + + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(&codec->dapm); + snd_soc_dapm_sync(&codec->dapm); + + /* register the rtd device */ + rtd->codec = codec; + rtd->card = card; + rtd->dev.parent = card->dev; + rtd->dev.release = rtd_release; + rtd->dev.init_name = aux_dev->name; + ret = device_register(&rtd->dev); + if (ret < 0) { + dev_err(codec->dev, + "asoc: failed to register aux runtime device %d\n", + ret); + return ret; + } + rtd->dev_registered = 1; + + /* add DAPM sysfs entries for this codec */ + ret = snd_soc_dapm_sys_add(&rtd->dev); + if (ret < 0) + dev_err(codec->dev, + "asoc: failed to add codec dapm sysfs entries\n"); + + /* add codec sysfs entries */ + ret = device_create_file(&rtd->dev, &dev_attr_codec_reg); + if (ret < 0) + dev_err(codec->dev, "asoc: failed to add codec sysfs files\n"); + +out: + return ret; +} + +static void soc_remove_aux_dev(struct snd_soc_card *card, int num) +{ + struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; + struct snd_soc_codec *codec = rtd->codec; + int err; + + /* unregister the rtd device */ + if (rtd->dev_registered) { + device_unregister(&rtd->dev); + rtd->dev_registered = 0; + } + + /* remove the CODEC */ + if (codec && codec->probed) { + if (codec->driver->remove) { + err = codec->driver->remove(codec); + if (err < 0) + dev_err(codec->dev, + "asoc: failed to remove %s\n", + codec->name); + } + + /* Make sure all DAPM widgets are freed */ + snd_soc_dapm_free(&codec->dapm); + + soc_cleanup_codec_debugfs(codec); + device_remove_file(&rtd->dev, &dev_attr_codec_reg); + codec->probed = 0; + list_del(&codec->card_list); + module_put(codec->dev->driver->owner); + } +} + static void snd_soc_instantiate_card(struct snd_soc_card *card) { struct platform_device *pdev = to_platform_device(card->dev); @@ -1658,6 +1782,15 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } + for (i = 0; i < card->num_aux_devs; i++) { + ret = soc_probe_aux_dev(card, i); + if (ret < 0) { + pr_err("asoc: failed to add auxiliary devices %s: %d\n", + card->name, ret); + goto probe_aux_dev_err; + } + } + snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), @@ -1684,6 +1817,10 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) mutex_unlock(&card->mutex); return; +probe_aux_dev_err: + for (i = 0; i < card->num_aux_devs; i++) + soc_remove_aux_dev(card, i); + probe_dai_err: for (i = 0; i < card->num_links; i++) soc_remove_dai_link(card, i); @@ -1748,6 +1885,10 @@ static int soc_remove(struct platform_device *pdev) run_delayed_work(&rtd->delayed_work); } + /* remove auxiliary devices */ + for (i = 0; i < card->num_aux_devs; i++) + soc_remove_aux_dev(card, i); + /* remove and free each DAI */ for (i = 0; i < card->num_rtd; i++) soc_remove_dai_link(card, i); @@ -2950,10 +3091,12 @@ static int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; - card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * card->num_links, - GFP_KERNEL); + card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * + (card->num_links + card->num_aux_devs), + GFP_KERNEL); if (card->rtd == NULL) return -ENOMEM; + card->rtd_aux = &card->rtd[card->num_links]; for (i = 0; i < card->num_links; i++) card->rtd[i].dai_link = &card->dai_link[i]; -- 1.7.2.3

4 11

[alsa-devel] [PATCH] ASoC: Tune performance of WM8958 revision A
by Mark Brown 30 Nov '10

30 Nov '10

Update some of the default configuration for the device to improve the performance. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/registers.h | 9 ++++++++ sound/soc/codecs/wm8994.c | 35 +++++++++++++++++++++++++-------- 2 files changed, 35 insertions(+), 9 deletions(-) diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index a610c87..4d5a561 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -64,6 +64,7 @@ #define WM8994_LDO_1 0x3B #define WM8994_LDO_2 0x3C #define WM8994_CHARGE_PUMP_1 0x4C +#define WM8958_CHARGE_PUMP_2 0x4D #define WM8994_CLASS_W_1 0x51 #define WM8994_DC_SERVO_1 0x54 #define WM8994_DC_SERVO_2 0x55 @@ -1858,6 +1859,14 @@ #define WM8994_CP_ENA_WIDTH 1 /* CP_ENA */ /* + * R77 (0x4D) - Charge Pump (2) + */ +#define WM8958_CP_DISCH 0x8000 /* CP_DISCH */ +#define WM8958_CP_DISCH_MASK 0x8000 /* CP_DISCH */ +#define WM8958_CP_DISCH_SHIFT 15 /* CP_DISCH */ +#define WM8958_CP_DISCH_WIDTH 1 /* CP_DISCH */ + +/* * R81 (0x51) - Class W (1) */ #define WM8994_CP_DYN_SRC_SEL_MASK 0x0300 /* CP_DYN_SRC_SEL - [9:8] */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bfd4cf4..b4c3cca 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1806,15 +1806,32 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { pm_runtime_get_sync(codec->dev); - /* Tweak DC servo and DSP configuration for - * improved performance. */ - if (control->type == WM8994 && wm8994->revision < 4) { - /* Tweak DC servo and DSP configuration for - * improved performance. */ - snd_soc_write(codec, 0x102, 0x3); - snd_soc_write(codec, 0x56, 0x3); - snd_soc_write(codec, 0x817, 0); - snd_soc_write(codec, 0x102, 0); + switch (control->type) { + case WM8994: + if (wm8994->revision < 4) { + /* Tweak DC servo and DSP + * configuration for improved + * performance. */ + snd_soc_write(codec, 0x102, 0x3); + snd_soc_write(codec, 0x56, 0x3); + snd_soc_write(codec, 0x817, 0); + snd_soc_write(codec, 0x102, 0); + } + break; + + case WM8958: + if (wm8994->revision == 0) { + /* Optimise performance for rev A */ + snd_soc_write(codec, 0x102, 0x3); + snd_soc_write(codec, 0xcb, 0x81); + snd_soc_write(codec, 0x817, 0); + snd_soc_write(codec, 0x102, 0); + + snd_soc_update_bits(codec, + WM8958_CHARGE_PUMP_2, + WM8958_CP_DISCH, + WM8958_CP_DISCH); + } } /* Discharge LINEOUT1 & 2 */ -- 1.7.1

1 0

[alsa-devel] [PATCH] ASoC: soc-cache: Fix memory overflow in LZO initialization
by Dimitris Papastamos 30 Nov '10

30 Nov '10

The bitmap_zero() nbits argument was improperly set to reg_size but the underlying buffer was bmp_size long. This caused the memset to zero past the end of the allocated buffer and into the kernel heap causing strange kernel crashes sometimes by overwriting critical kernel structures. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 9b1ba33..5143984 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1348,7 +1348,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) ret = -ENOMEM; goto err; } - bitmap_zero(sync_bmp, reg_size); + bitmap_zero(sync_bmp, bmp_size); /* allocate the lzo blocks and initialize them */ for (i = 0; i < blkcount; ++i) { -- 1.7.3.2

3 2

[alsa-devel] [PATCH] ASoC: sh: fsi-ak4642: Add FSI port and ak464x selection
by Kuninori Morimoto 30 Nov '10

30 Nov '10

Current FSI-Ak4642 device had niche settings which were FSI2-A-AK4643 and FSI-A-AK4642. This patch add platform_device_id which can control FSI/FSI2, PortA/PortB, AK4642/AK4643 from platform data. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- arch/arm/mach-shmobile/board-ap4evb.c | 5 + arch/sh/boards/mach-se/7724/setup.c | 5 + sound/soc/sh/fsi-ak4642.c | 133 ++++++++++++++++++++++++++++++--- 3 files changed, 131 insertions(+), 12 deletions(-) diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index d326054..82c3204 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -640,6 +640,10 @@ static struct platform_device fsi_device = { }, }; +static struct platform_device fsi_ak4643_device = { + .name = "sh_fsi2_a_ak4643", +}; + static struct sh_mobile_lcdc_info sh_mobile_lcdc1_info = { .clock_source = LCDC_CLK_EXTERNAL, .ch[0] = { @@ -838,6 +842,7 @@ static struct platform_device *ap4evb_devices[] __initdata = { &sdhi1_device, &usb1_host_device, &fsi_device, + &fsi_ak4643_device, &sh_mmcif_device, &lcdc1_device, &lcdc_device, diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index c31d228..dff89d0 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -343,6 +343,10 @@ static struct platform_device fsi_device = { }, }; +static struct platform_device fsi_ak4642_device = { + .name = "sh_fsi_a_ak4642", +}; + /* KEYSC in SoC (Needs SW33-2 set to ON) */ static struct sh_keysc_info keysc_info = { .mode = SH_KEYSC_MODE_1, @@ -615,6 +619,7 @@ static struct platform_device *ms7724se_devices[] __initdata = { &sh7724_usb0_host_device, &sh7724_usb1_gadget_device, &fsi_device, + &fsi_ak4642_device, &sdhi0_cn7_device, &sdhi1_cn8_device, &irda_device, diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index d96602d..4ef279c 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -12,6 +12,13 @@ #include <linux/platform_device.h> #include <sound/sh_fsi.h> +struct fsi_ak4642_data { + const char *name; + const char *cpu_dai; + const char *codec; + const char *platform; +}; + static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->codec_dai; @@ -27,17 +34,12 @@ static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd) } static struct snd_soc_dai_link fsi_dai_link = { - .name = "AK4642", - .stream_name = "AK4642", - .cpu_dai_name = "fsia-dai", /* fsi A */ + /* .name */ + /* .stream_name */ + /* .cpu_dai_name */ .codec_dai_name = "ak4642-hifi", -#ifdef CONFIG_MACH_AP4EVB - .platform_name = "sh_fsi2", - .codec_name = "ak4642-codec.0-0013", -#else - .platform_name = "sh_fsi.0", - .codec_name = "ak4642-codec.0-0012", -#endif + /* .platform_name */ + /* .codec_name */ .init = fsi_ak4642_dai_init, .ops = NULL, }; @@ -50,14 +52,30 @@ static struct snd_soc_card fsi_soc_card = { static struct platform_device *fsi_snd_device; -static int __init fsi_ak4642_init(void) +static int fsi_ak4642_probe(struct platform_device *pdev) { int ret = -ENOMEM; + const struct platform_device_id *id_entry; + struct fsi_ak4642_data *pdata; + + id_entry = pdev->id_entry; + if (!id_entry) { + dev_err(&pdev->dev, "unknown fsi ak4642\n"); + return -ENODEV; + } + + pdata = (struct fsi_ak4642_data *)id_entry->driver_data; fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_A); if (!fsi_snd_device) goto out; + fsi_dai_link.name = pdata->name; + fsi_dai_link.stream_name = pdata->name; + fsi_dai_link.cpu_dai_name = pdata->cpu_dai; + fsi_dai_link.platform_name = pdata->platform; + fsi_dai_link.codec_name = pdata->codec; + platform_set_drvdata(fsi_snd_device, &fsi_soc_card); ret = platform_device_add(fsi_snd_device); @@ -68,9 +86,100 @@ out: return ret; } -static void __exit fsi_ak4642_exit(void) +static int fsi_ak4642_remove(struct platform_device *pdev) { platform_device_unregister(fsi_snd_device); + return 0; +} + +static struct fsi_ak4642_data fsi_a_ak4642 = { + .name = "AK4642", + .cpu_dai = "fsia-dai", + .codec = "ak4642-codec.0-0012", + .platform = "sh_fsi.0", +}; + +static struct fsi_ak4642_data fsi_b_ak4642 = { + .name = "AK4642", + .cpu_dai = "fsib-dai", + .codec = "ak4642-codec.0-0012", + .platform = "sh_fsi.0", +}; + +static struct fsi_ak4642_data fsi_a_ak4643 = { + .name = "AK4643", + .cpu_dai = "fsia-dai", + .codec = "ak4642-codec.0-0013", + .platform = "sh_fsi.0", +}; + +static struct fsi_ak4642_data fsi_b_ak4643 = { + .name = "AK4643", + .cpu_dai = "fsib-dai", + .codec = "ak4642-codec.0-0013", + .platform = "sh_fsi.0", +}; + +static struct fsi_ak4642_data fsi2_a_ak4642 = { + .name = "AK4642", + .cpu_dai = "fsia-dai", + .codec = "ak4642-codec.0-0012", + .platform = "sh_fsi2", +}; + +static struct fsi_ak4642_data fsi2_b_ak4642 = { + .name = "AK4642", + .cpu_dai = "fsib-dai", + .codec = "ak4642-codec.0-0012", + .platform = "sh_fsi2", +}; + +static struct fsi_ak4642_data fsi2_a_ak4643 = { + .name = "AK4643", + .cpu_dai = "fsia-dai", + .codec = "ak4642-codec.0-0013", + .platform = "sh_fsi2", +}; + +static struct fsi_ak4642_data fsi2_b_ak4643 = { + .name = "AK4643", + .cpu_dai = "fsib-dai", + .codec = "ak4642-codec.0-0013", + .platform = "sh_fsi2", +}; + +static struct platform_device_id fsi_id_table[] = { + /* FSI */ + { "sh_fsi_a_ak4642", (kernel_ulong_t)&fsi_a_ak4642 }, + { "sh_fsi_b_ak4642", (kernel_ulong_t)&fsi_b_ak4642 }, + { "sh_fsi_a_ak4643", (kernel_ulong_t)&fsi_a_ak4643 }, + { "sh_fsi_b_ak4643", (kernel_ulong_t)&fsi_b_ak4643 }, + + /* FSI 2 */ + { "sh_fsi2_a_ak4642", (kernel_ulong_t)&fsi2_a_ak4642 }, + { "sh_fsi2_b_ak4642", (kernel_ulong_t)&fsi2_b_ak4642 }, + { "sh_fsi2_a_ak4643", (kernel_ulong_t)&fsi2_a_ak4643 }, + { "sh_fsi2_b_ak4643", (kernel_ulong_t)&fsi2_b_ak4643 }, + {}, +}; + +static struct platform_driver fsi_ak4642 = { + .driver = { + .name = "fsi-ak4642-audio", + }, + .probe = fsi_ak4642_probe, + .remove = fsi_ak4642_remove, + .id_table = fsi_id_table, +}; + +static int __init fsi_ak4642_init(void) +{ + return platform_driver_register(&fsi_ak4642); +} + +static void __exit fsi_ak4642_exit(void) +{ + platform_driver_unregister(&fsi_ak4642); } module_init(fsi_ak4642_init); -- 1.7.0.4

4 3

[alsa-devel] [PATCH] ASoC: Simplify pm860x_probe error handling
by Axel Lin 30 Nov '10

30 Nov '10

Simplify pm860x_probe error handling and return actual error code we got. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/88pm860x-codec.c | 12 +++++------- 1 files changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 7e4d880..08e15de 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1358,7 +1358,7 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x->name[i], pm860x); if (ret < 0) { dev_err(codec->dev, "Failed to request IRQ!\n"); - goto out_irq; + goto out; } } @@ -1369,7 +1369,7 @@ static int pm860x_probe(struct snd_soc_codec *codec) if (ret < 0) { dev_err(codec->dev, "Failed to fill register cache: %d\n", ret); - goto out_codec; + goto out; } snd_soc_add_controls(codec, pm860x_snd_controls, @@ -1379,12 +1379,10 @@ static int pm860x_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; -out_codec: - i = 3; -out_irq: - for (; i >= 0; i--) +out: + while (--i >= 0) free_irq(pm860x->irq[i], pm860x); - return -EINVAL; + return ret; } static int pm860x_remove(struct snd_soc_codec *codec) -- 1.7.2

2 1

[alsa-devel] S6000 ASoC drivers with multi-component
by Mark Brown 30 Nov '10

30 Nov '10

I'm a little concerned that there have been no updates to the s6000 audio drivers since the merge of multi-component ASoC (which is now in mainline and should be released as part of 2.6.37). This was a very big and invasive update, and experience with other architectures has been that there's usually at least some typos that need fixing. It'd be good to get some confirmation that the audio drivers have at least been tested with current mainline - if everything worked fine that's obviously excellent but it'd be good if someone could confirm that this is the case.

4 7

[alsa-devel] [PATCH] ASoC: Fix compile error for s6000-i2s.c by adding a missing '; ' character
by Axel Lin 30 Nov '10

30 Nov '10

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I don't have toolchain to compile s6000, I'm not able to put the error message in the commit log. But I know this patch indeed fixes a compile error. sound/soc/s6000/s6000-i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 8778faa..3052f64 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -434,7 +434,7 @@ static struct snd_soc_dai_driver s6000_i2s_dai = { .rate_max = 1562500, }, .ops = &s6000_i2s_dai_ops, -} +}; static int __devinit s6000_i2s_probe(struct platform_device *pdev) { -- 1.7.2

3 2

[alsa-devel] [PATCH 0/5] Fix compile error and missing spin_unlock_irqrestore for NUC900 SoC audio
by Axel Lin 30 Nov '10

30 Nov '10

This serial fixes compile error and missing spin_unlock_irqrestore for NUC900 SoC audio. Regards, Axel Axel Lin (5): ASoC: Fix compile error for nuc900-ac97.c ASoC: Fix prototype for nuc900_ac97_probe and nuc900_ac97_remove ASoC: Fix compile error for nuc900-pcm.c ASoC: Fix missing spin_unlock_irqrestore ASoC: Remove unneeded !! operations while checking return value of nuc900_checkready sound/soc/nuc900/nuc900-ac97.c | 15 +++++++-------- sound/soc/nuc900/nuc900-audio.h | 2 ++ sound/soc/nuc900/nuc900-pcm.c | 11 ++++++----- 3 files changed, 15 insertions(+), 13 deletions(-) -- 1.7.2

4 9

[alsa-devel] [PATCHES] Integrate minimal systemd support in alsa-utils
by Lennart Poettering 30 Nov '10

30 Nov '10

Heya, Up to this point most distributions have been using different init scripts/udev rules files to save/restore mixer settings at boot/hotplug/shutdown. Please see my two attached patches for alsa-utils which add standardized, simplified support for this. The first patch is stolen from Debian/Ubuntu, which moves asound.state to /var/lib, which is a more appropriate place, since /etc might be read-only. The second patch includes a minimal udev rule (executed synchronously, enabled on all systems, regardless of systemd is used or not) and two systemd service files (executed asynchronously at boot/shutdown, only enabled if built with systemd support). Given that most big distributions are moving to adopt systemd sooner or later it would be great to have support for this in the upstream package to avoid further balkanization how things are done here. Lennart -- Lennart Poettering - Red Hat, Inc.

7 23

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 30 Nov '10

30 Nov '10

The following changes since commit ab4e57446ff2e0a5c154cb703595d574130dc919: Merge branch 'for-2.6.37' into for-2.6.38 (2010-11-24 11:57:43 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Axel Lin (24): ASoC: Call snd_soc_unregister_dais instead of snd_soc_unregister_dai in sh4_soc_dai_remove ASoC: efika-audio-fabric: fix resource leak in efika_fabric_init error path ASoC: pcm030-audio-fabric: fix resource leak in pcm030_fabric_init error path ASoC: snd-soc-afeb9260: remove unneeded platform_device_del in error path ASoC: sam9g20_wm8731: fix resource leak in at91sam9g20ek_init error path ASoC: simone: fix resource leak in simone_init error path ASoC: imx-ssi: fix resource leak ASoC: phycore-ac97: fix resource leak ASoC: davinci-vcif - fix a memory leak ASoC: neo1973_gta02_wm8753: fix wrong parameter for snd_soc_register_dai and snd_soc_unregister_dai ASoC: Fix compile error for smartq_wm8987.c ASoC: smdk_wm9713: fix resource leak in smdk_init error path ASoC: tlv320aic3x - fix variable may be used uninitialized warning ASoC: Fix resource reclaim for osk5912 ASoC: Fix resource leak in goni_wm8994.c ASoC: Fix resource leak in neo1973_gta02_init() error path ASoC: Add missing gpio_free(S3C64XX_GPK(12)) in smartq_exit() ASoC: Fix resource leak in smdk_spdif.c ASoC: Fix compile error for nuc900-ac97.c ASoC: Fix prototype for nuc900_ac97_probe and nuc900_ac97_remove ASoC: Fix compile error for nuc900-pcm.c ASoC: Remove unneeded !! operations while checking return value of nuc900_checkready ASoC: Add missing dev_set_drvdata in mpc8610_hpcd_probe ASoC: Add missing dev_set_drvdata in p1022_ds_probe Dimitris Papastamos (2): ASoC: soc-cache: Add error checking in the *_cache_sync functions ASoC: soc-cache: Ensure consistent cache naming Dmitry Artamonow (1): ASoC: fix SND_PXA2XX_LIB Kconfig warning Mark Brown (18): Merge branch 'for-2.6.37' into for-2.6.38 ASoC: Make SMDK WM8580 driver unloadable ASoC: Update MAINTAINERS for Samsung driver move ASoC: Add WM8737 ALC support ARM: Samsung: Make sure ASoC DMA device is prototyped Merge branch 'for-2.6.37' into for-2.6.38 ASoC: Remove redundant hw_write initialisation in WM8737 mfd: Add initial WM8958 support gpiolib: Add WM8958 support to the WM8994 driver ASoC: Use DC servo startup mode when not doing DCS correction ASoC: Initial WM8958 audio configuration ASoC: Implement support for enhanced AIF3 on WM8958 ASoC: Add WM8958 Multi-band compressor support ASoC: Add WM8958 microphone detection support ASoC: Convert WM8994 to use soc-cache.c cache functions ASoC: Enable rbtree compression for WM8994/58 register cache ASoC: Implement runtime PM for WM8994/58 Merge branch 'for-2.6.37' into for-2.6.38 MAINTAINERS | 2 +- arch/arm/plat-samsung/dev-asocdma.c | 1 + drivers/gpio/wm8994-gpio.c | 24 + drivers/mfd/wm8994-core.c | 93 ++- include/linux/mfd/wm8994/core.h | 8 + include/linux/mfd/wm8994/registers.h | 225 +++++ include/sound/soc.h | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 5 +- sound/soc/atmel/snd-soc-afeb9260.c | 1 - sound/soc/codecs/tlv320aic3x.c | 6 +- sound/soc/codecs/wm8737.c | 43 +- sound/soc/codecs/wm8994-tables.c | 1572 ++++++++++++++++++++++++++++++ sound/soc/codecs/wm8994.c | 728 ++++++++++++-- sound/soc/codecs/wm8994.h | 5 + sound/soc/codecs/wm_hubs.c | 20 +- sound/soc/davinci/davinci-vcif.c | 3 + sound/soc/ep93xx/simone.c | 18 +- sound/soc/fsl/efika-audio-fabric.c | 1 + sound/soc/fsl/mpc8610_hpcd.c | 1 + sound/soc/fsl/p1022_ds.c | 1 + sound/soc/fsl/pcm030-audio-fabric.c | 1 + sound/soc/imx/imx-ssi.c | 15 +- sound/soc/imx/phycore-ac97.c | 28 +- sound/soc/nuc900/nuc900-ac97.c | 15 +- sound/soc/nuc900/nuc900-audio.h | 2 + sound/soc/nuc900/nuc900-pcm.c | 2 +- sound/soc/omap/osk5912.c | 11 +- sound/soc/pxa/Kconfig | 1 + sound/soc/samsung/goni_wm8994.c | 9 +- sound/soc/samsung/neo1973_gta02_wm8753.c | 26 +- sound/soc/samsung/smartq_wm8987.c | 10 +- sound/soc/samsung/smdk_spdif.c | 13 +- sound/soc/samsung/smdk_wm8580.c | 6 + sound/soc/samsung/smdk_wm9713.c | 17 +- sound/soc/sh/ssi.c | 2 +- sound/soc/soc-cache.c | 30 +- 36 files changed, 2742 insertions(+), 205 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.37
by Mark Brown 30 Nov '10

30 Nov '10

The following changes since commit 25436180ee8bed6740f29d92c2030c759885c147: ASoC: Fix resource reclaim for osk5912 (2010-11-25 21:15:49 +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 More small, targetted bugfixes from Axel. Axel Lin (6): ASoC: Fix compile error for nuc900-ac97.c ASoC: Fix prototype for nuc900_ac97_probe and nuc900_ac97_remove ASoC: Fix compile error for nuc900-pcm.c ASoC: Remove unneeded !! operations while checking return value of nuc900_checkready ASoC: Add missing dev_set_drvdata in mpc8610_hpcd_probe ASoC: Add missing dev_set_drvdata in p1022_ds_probe sound/soc/fsl/mpc8610_hpcd.c | 1 + sound/soc/fsl/p1022_ds.c | 1 + sound/soc/nuc900/nuc900-ac97.c | 15 +++++++-------- sound/soc/nuc900/nuc900-audio.h | 2 ++ sound/soc/nuc900/nuc900-pcm.c | 2 +- 5 files changed, 12 insertions(+), 9 deletions(-)

2 1

[alsa-devel] reset of kernel input buffer for /dev/dsp
by Amit Nagal 30 Nov '10

30 Nov '10

Hi , i am doing usb audio streaming using usb audio class interface . our application is OSS based , uses OSS calls to interact with linux audio driver . Our device is capture only device . we want to flush kernel input pcm buffer without using close() system call . does OSS supports in userspace any ioctl call which can flush kernel pcm buffer at any time . right now we have to close audio-fd and open audio-fd again . how we can flush kernel input pcm buffer without using close system call in based OSS application regards Amit Nagal

3 5

[alsa-devel] Help wanted: emachines em350 internal mic
by Eliot Blennerhassett 30 Nov '10

30 Nov '10

Greetings, There is only a single mic on this netbook. However, the alsa device shows up as stereo, and the right channel carries an inverted copy of the left channel. So when the two channels are summed to make a mono (e.g. for telephony, or playback on mono internal speaker) it appears that there is no signal. Is there a way to "cure" this problem in the driver? I can think of * Force internal mic capture to be mono (left only) * Copy left to right capture * Invert right capture but don't know how easy any are to implement. Note more info about this 'card' can be found attached to https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/639846 thanks and regards Eliot

1 0

[alsa-devel] Expected behaviour for two mic jacks
by David Henningsson 30 Nov '10

30 Nov '10

Hi, A while ago I asked for the default/expected behaviour for auto-mute. Here's the same question for mics: If there are two mic jacks, e g one on the front and one on the back, is the standard/default behaviour that we should: 1) Show one mic only and add auto-switch functionality, and if so, what should determine which mic takes precedence? 2) Show both mics and let the user choose between them? 3) Show both mics, two volume control and let the user mix both signals together (when possible)? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

3 5

[alsa-devel] [PATCH 1/2] ASoC: Add missing dev_set_drvdata in mpc8610_hpcd_probe
by Axel Lin 29 Nov '10

29 Nov '10

Otherwise, calling dev_get_drvdata in mpc8610_hpcd_remove returns NULL. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/fsl/mpc8610_hpcd.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 0d7dcf1..7d7847a 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -498,6 +498,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) dev_err(&pdev->dev, "platform device add failed\n"); goto error; } + dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); -- 1.7.2

4 4

[alsa-devel] alsa-driver: compilation fails
by pl bossart 29 Nov '10

29 Nov '10

Somehow I pulled the latest and greatest code alsa-kmirror and alsa-driver, and the compilation now fails. ./gitcompile yells about a problematic patch on pcm_native.c. Any idea on how to refresh this patch ? Thanks, -Pierre copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #2 FAILED at 22. Hunk #3 succeeded at 370 (offset -1 lines). Hunk #4 succeeded at 458 (offset 2 lines). Hunk #5 succeeded at 535 (offset 2 lines). Hunk #6 succeeded at 1587 (offset 2 lines). Hunk #7 succeeded at 2033 (offset 2 lines). Hunk #8 succeeded at 2862 (offset 2 lines). Hunk #9 succeeded at 2915 (offset 2 lines). Hunk #10 succeeded at 2943 (offset 2 lines). Hunk #11 succeeded at 2959 (offset 2 lines). Hunk #12 succeeded at 2985 (offset 2 lines). Hunk #13 succeeded at 3073 (offset 2 lines). Hunk #14 succeeded at 3097 (offset 2 lines). Hunk #15 succeeded at 3111 (offset 2 lines). Hunk #16 succeeded at 3168 (offset 2 lines). Hunk #17 succeeded at 3180 (offset 2 lines). Hunk #18 succeeded at 3194 (offset 2 lines). Hunk #19 succeeded at 3252 (offset 2 lines). Hunk #20 succeeded at 3297 (offset 2 lines). Hunk #21 succeeded at 3333 (offset 2 lines). Hunk #22 succeeded at 3395 (offset 2 lines). Hunk #23 succeeded at 3450 (offset 2 lines). Hunk #24 succeeded at 3513 (offset 2 lines). Hunk #25 succeeded at 3521 (offset 2 lines). Hunk #26 succeeded at 3548 (offset 2 lines). Hunk #27 succeeded at 3581 (offset 2 lines). Hunk #28 succeeded at 3747 (offset 2 lines). 1 out of 28 hunks FAILED -- saving rejects to file pcm_native.c.rej [ume@plb alsa-driver]$ more acore/pcm_native.c.rej --- ../alsa-kernel/core/pcm_native.c 2010-09-16 23:10:16.996220991 +0200 +++ pcm_native.c 2010-09-17 12:23:15.092424934 +0200 @@ -22,9 +24,9 @@ */ #include <linux/mm.h> +#include <linux/smp_lock.h> #include <linux/file.h> #include <linux/slab.h> -#include <linux/smp_lock.h> #include <linux/time.h> #include <linux/pm_qos_params.h> #include <linux/uio.h>

3 3

[alsa-devel] [PATCH] ASoC: soc-cache: Ensure consistent cache naming
by Dimitris Papastamos 29 Nov '10

29 Nov '10

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 +- sound/soc/soc-cache.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 3eb92ef..9e593cf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -256,7 +256,7 @@ enum snd_soc_control_type { }; enum snd_soc_compress_type { - SND_SOC_NO_COMPRESSION, + SND_SOC_FLAT_COMPRESSION, SND_SOC_LZO_COMPRESSION, SND_SOC_RBTREE_COMPRESSION }; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index e5b1888..9b1ba33 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1520,7 +1520,7 @@ static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) /* an array of all supported compression types */ static const struct snd_soc_cache_ops cache_types[] = { { - .id = SND_SOC_NO_COMPRESSION, + .id = SND_SOC_FLAT_COMPRESSION, .init = snd_soc_flat_cache_init, .exit = snd_soc_flat_cache_exit, .read = snd_soc_flat_cache_read, -- 1.7.3.2

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[alsa-devel] [PATCH] ASoC: soc-cache: Add error checking in the *_cache_sync functions
by Dimitris Papastamos 29 Nov '10

29 Nov '10

Ensure that we report any errors encountered during reads/writes in the cache syncing functions. Remove redundant newline in the source code. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 28 +++++++++++++++++++++------- 1 files changed, 21 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 6c0589e..e5b1888 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -792,7 +792,6 @@ static struct snd_soc_rbtree_node *snd_soc_rbtree_lookup( return NULL; } - static int snd_soc_rbtree_insert(struct rb_root *root, struct snd_soc_rbtree_node *rbnode) { @@ -826,14 +825,19 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) struct rb_node *node; struct snd_soc_rbtree_node *rbnode; unsigned int val; + int ret; rbtree_ctx = codec->reg_cache; for (node = rb_first(&rbtree_ctx->root); node; node = rb_next(node)) { rbnode = rb_entry(node, struct snd_soc_rbtree_node, node); if (rbnode->value == rbnode->defval) continue; - snd_soc_cache_read(codec, rbnode->reg, &val); - snd_soc_write(codec, rbnode->reg, val); + ret = snd_soc_cache_read(codec, rbnode->reg, &val); + if (ret) + return ret; + ret = snd_soc_write(codec, rbnode->reg, val); + if (ret) + return ret; dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", rbnode->reg, val); } @@ -1110,11 +1114,16 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) struct snd_soc_lzo_ctx **lzo_blocks; unsigned int val; int i; + int ret; lzo_blocks = codec->reg_cache; for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) { - snd_soc_cache_read(codec, i, &val); - snd_soc_write(codec, i, val); + ret = snd_soc_cache_read(codec, i, &val); + if (ret) + return ret; + ret = snd_soc_write(codec, i, val); + if (ret) + return ret; dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", i, val); } @@ -1390,12 +1399,15 @@ err_tofree: static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) { int i; + int ret; struct snd_soc_codec_driver *codec_drv; unsigned int val; codec_drv = codec->driver; for (i = 0; i < codec_drv->reg_cache_size; ++i) { - snd_soc_cache_read(codec, i, &val); + ret = snd_soc_cache_read(codec, i, &val); + if (ret) + return ret; if (codec_drv->reg_cache_default) { switch (codec_drv->reg_word_size) { case 1: { @@ -1418,7 +1430,9 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) BUG(); } } - snd_soc_write(codec, i, val); + ret = snd_soc_write(codec, i, val); + if (ret) + return ret; dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", i, val); } -- 1.7.3.2

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[alsa-devel] [PATCH 0/4] Fixes resource leaks for Samsung SoCs
by Axel Lin 29 Nov '10

29 Nov '10

This serial of patches fixes resource leaks for Samsung SoCs. Axel Lin (4): ASoC: Fix resource leask in goni_wm8994.c ASoC: Fix resource leask in neo1973_gta02_init() error path ASoC: Add missing gpio_free(S3C64XX_GPK(12)) in smartq_exit() ASoC: Fix resource leask in smdk_spdif.c sound/soc/samsung/goni_wm8994.c | 9 +++++++-- sound/soc/samsung/neo1973_gta02_wm8753.c | 22 +++++++++++----------- sound/soc/samsung/smartq_wm8987.c | 1 + sound/soc/samsung/smdk_spdif.c | 13 +++++++++---- 4 files changed, 28 insertions(+), 17 deletions(-) -- 1.7.2

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Re: [alsa-devel] Blackfin ASoC drivers with multi-component
by Mark Brown 29 Nov '10

29 Nov '10

On Sun, Nov 28, 2010 at 09:27:17PM -0500, Zhang, Sonic wrote: [Reflowed into 80 columns] > We are busy with 2010R1 Linux on blackfin release based on kernel > 2.6.34. There is no activity in ADI on multi component audio drivers in > latest 2.6.37 rc kernel. However, we are switching to the 2011 > development cycle soon, which will cover some audio drivers on top of > the multi-component framework. Please bear in mind that 2.6.37 will be released probably around the end of next month so it's important to get testing and any fixes done before then.

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[alsa-devel] Writing 20ms of AUDIO data
by Irfan Shaikh 29 Nov '10

29 Nov '10

Hello developers, Please can some tell me how can i configure my snd_pcm_writei so that i can write 20ms of audio data each time. Is there any way to find out the time duration interval of PCM frames we write to playback device {So that soem i can write each tie 20 millisec of audio data to device}. Thanks and regards, SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

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[alsa-devel] Is calling snd_pcm_sw_params() mandatory ?
by Raymond Yau 29 Nov '10

29 Nov '10

>>Twas brillig, and Raymond Yau at 24/07/10 04:51 did gyre and gimble: >> 2010/7/15 Colin Guthrie <gmane <at> colin.guthr.ie> >> >> >> Can I remove pulseaudio from mandriva 2000 ? >> >> I just want to test *libcanberra*-alsa alone ( i.e. *libcanberra* without >> pulseaudio ) >> >> when PA server abort, and *libcanberra*-pulse cannot connect to PA server , >> >> Will *libcanberra* use *libcanberra*-alsa connect to PA using alsa-pulse plugin >> ? > Use draksound (or via Mandriva Control Center) to disable pulseaudio and > the ALSA backend for canberra should be used instead. > It's based on the alternatives system and the file > /etc/sound/profiles/alsa/canberra.conf > Col It is strange that the alsa back end of libcanberra (http://git.0pointer.de/?p=libcanberra.git;a=blob;f=src/alsa.c) does not call snd_pcm_sw_params() Does alsa-lib has any check whether the application had called snd_pcm_sw_params() or it provide default values for those software parameters (e.g. start_threshold, stop_threshold) when the application only call snd_pcm_hw_params() and snd_pcm_prepare()

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[alsa-devel] [PATCH] ALSA: hda: Use BIOS auto-parsing instead of existing model quirk for MEDION MD2
by Daniel Chen 29 Nov '10

29 Nov '10

Hi, Attached is a patch against current sound-2.6.git master HEAD that resolves a regression from 2.6.32 for the user's machine. Briefly, the existing quirk entry, ALC883_MEDION_MD2, is no longer necessary for the user's machine; we can use BIOS auto-parsing successfully. Please apply. Thanks, -Dan

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