January 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: Allow drivers to specify how many bits are significant on a DAI
by Mark Brown 17 Jan '12

17 Jan '12

Most devices accept data in formats that don't correspond directly to their internal format. ALSA allows us to set a msbits constraint which tells userspace about this in case it finds it useful (for example, in order to avoid wasting effort dithering bits that will be ignored when raising the sample size of data) so provide a mechanism for drivers to specify the number of bits that are actually significant on a DAI and add the appropriate constraints along with all the others. This is done slightly awkwardly as the constraint is specified per sample size - we loop over every possible sample size, including ones that the device doesn't support and including ones that have fewer bits than are actually used, but this is harmless as the upper layers do the right thing in these cases. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-pcm.c | 36 ++++++++++++++++++++++++++++++++++++ 2 files changed, 37 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0a56767..346a458 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -519,6 +519,7 @@ struct snd_soc_pcm_stream { unsigned int rate_max; /* max rate */ unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ + unsigned int sig_bits; /* number of bits of content */ }; /* SoC audio ops */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cdc860a..8bb1793 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -63,6 +63,39 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, } /* + * List of sample sizes that might go over the bus for parameter + * application. There ought to be a wildcard sample size for things + * like the DAC/ADC resolution to use but there isn't right now. + */ +static int sample_sizes[] = { + 8, 16, 24, 32, +}; + +static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret, i, bits; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + bits = dai->driver->playback.sig_bits; + else + bits = dai->driver->capture.sig_bits; + + if (!bits) + return; + + for (i = 0; i < ARRAY_SIZE(sample_sizes); i++) { + ret = snd_pcm_hw_constraint_msbits(substream->runtime, + 0, sample_sizes[i], + bits); + if (ret != 0) + dev_warn(dai->dev, + "Failed to set MSB %d/%d: %d\n", + bits, sample_sizes[i], ret); + } +} + +/* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls * startup for the cpu DAI, platform, machine and codec DAI. @@ -187,6 +220,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } + soc_pcm_apply_msb(substream, codec_dai); + soc_pcm_apply_msb(substream, cpu_dai); + /* Symmetry only applies if we've already got an active stream. */ if (cpu_dai->active) { ret = soc_pcm_apply_symmetry(substream, cpu_dai); -- 1.7.7.3

3 16

[alsa-devel] sound devices USBPre2 gets 'cannot submit syncpipe for urb 0' error when run in usb2 mode
by alexander 17 Jan '12

17 Jan '12

I've understood that linux infact does have support for uac2 so this device should work in usb2 mode but currently doesn't. I've made an alsa bug report with more info. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5512

1 0

[alsa-devel] [PATCH 1/2] ASoC: Wait for WM8993 FLL to stabilise
by Mark Brown 17 Jan '12

17 Jan '12

Ensure the FLL is locked before we return from set_fll(). 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 36562ed..69f5491 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -542,6 +542,12 @@ static int _wm8993_set_fll(struct snd_soc_codec *codec, int fll_id, int source, /* Enable the FLL */ snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA); + /* Both overestimates */ + if (Fref < 1000000) + msleep(3); + else + msleep(1); + dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout); wm8993->fll_fref = Fref; -- 1.7.7.3

1 1

Re: [alsa-devel] [Alsa-user] Read and write with different output sizes
by Elias Kokkinis 17 Jan '12

17 Jan '12

Dear all, after some further research into the matter, it seems I have to do some asynchronous capture/playback with ALSA. I have searched for examples and found this site: http://alumnos.elo.utfsm.cl/~yanez/alsa-sample-programs/ where two example programs are included. microphone.c and speaker.c which use almost all available input/output modes in ALSA. The interesting thing is that you can run both programs concatenated since they write and read to the standard output and create a simple audio loop-back. I have tested different configurations and the configuration I want works with direct and asynchronous read/write with mmap and read/write with poll. Now I have taken the bits and pieces of those to programs and I have tried to create a single program that will read and write with different buffer sizes. I can't seem to be able to "reroute" the write from the standard output [these programs use write() and read()] to a buffer, or the read from the standard input to a buffer. Any ideas? Comments? I thank everyone for their time, Elias On 16 Jan 2012, at 12:53, Elias Kokkinis wrote: > Hello all, > > I am posting a question I previously posted in alsa-devel, but did not get the expected feedback. > > I am using alsa drivers in an OMAP3530 processor to do some audio processing. I am implementing a typical speech enhancement algorithm with overlap-add. I use a window length of 2048 samples and an overlap of 1024 samples. That means I need to use an input (capture) buffer of 2048 and an output (playback) buffer of 1024. Can this be configured in alsa? Can the playback and capture streams have different buffer sizes, but operate at the same sample rate (e.g. 44100Hz)? > > Best regards, > Elias > ------------------------------------------------------------------------------ > RSA(R) Conference 2012 > Mar 27 - Feb 2 > Save $400 by Jan. 27 > Register now! > http://p.sf.net/sfu/rsa-sfdev2dev2 > _______________________________________________ > Alsa-user mailing list > Alsa-user(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-user

1 0

[alsa-devel] [PATCH 0/2] ASoC: omap-mcbsp: CPU version check changes
by Peter Ujfalusi 17 Jan '12

17 Jan '12

Hello, The following patches changes the way we check CPU versions to enable different features (FIFO, and some register differencies). In this way we are not going to need to change (and bloat) the checks for future versions of OMAP. Regards, Peter --- Peter Ujfalusi (2): ASoC: omap-mcbsp: Simplify checks for McBSP IP with FIFO ASoC: omap-mcbsp: Use future proof OMAP version checks for configuration sound/soc/omap/omap-mcbsp.c | 15 ++++++++------- 1 files changed, 8 insertions(+), 7 deletions(-) -- 1.7.8.3

3 5

[alsa-devel] QasTools 0.17.0 released
by Sebastian H. 17 Jan '12

17 Jan '12

QasTools version 0.17.0 is available. QasTools is a collection of dektop applications for ALSA. Changes: * all: setting names for state save/restore changed/regrouped * all: smart main window sizeHint() to get a useable window size after leaving fullscreen * all: new common info dialog - uses QListView page selection intead of tabs * shared: device selection style tweaks: selected item now emphasized by a thin frame * shared: device selection style tweaks: Rounded selection area * shared: use QSplitter instead of QDockwidget for device selection view. It couldn't be detached anyway and size restoring works more reliable. * qasconfig: settings directory changed to shared ~/.config/qastools/ (was ~/.config/qasconfig/) * qasconfig: new basic command line options (--version, --help, etc.) * qashctl: new basic command line options (--version, --help, etc.) * qasmixer: new application icon Homepage with more information http://xwmw.org/qastools Project page http://sourceforge.net/projects/qastools/ Happy volume changing! -- Sebastian Holtermann

1 0

[alsa-devel] [PATCH] ASoC: Make WM8993 I2C usage unconditional
by Mark Brown 17 Jan '12

17 Jan '12

The WM8993 only supports I2C so don't ifdef the I2C support in the driver. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 6 ------ 1 files changed, 0 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index b64fb91..36562ed 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1654,7 +1654,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8993 = { .set_bias_level = wm8993_set_bias_level, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1758,27 +1757,22 @@ static struct i2c_driver wm8993_i2c_driver = { .remove = __devexit_p(wm8993_i2c_remove), .id_table = wm8993_i2c_id, }; -#endif static int __init wm8993_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8993_i2c_driver); if (ret != 0) { pr_err("WM8993: Unable to register I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8993_modinit); static void __exit wm8993_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8993_i2c_driver); -#endif } module_exit(wm8993_exit); -- 1.7.7.3

1 0

[alsa-devel] Unable to see ADC output data with ADAV801
by Wayne Tams 17 Jan '12

17 Jan '12

Hi, I am attempting to add ADAV801 codec support to my 2.6.35 kernel to be used with an ARM processor. I can see that there is a mainlined adav80x.c driver that appears at kernel version 3.1, I have had a quick look at back-porting the driver but this looks as though it could get quite messy. However, I did come across an older ADAV801 driver which looks a lot simpler, but I cannot find it in previous mainlined kernel versions, http://www.spinics.net/lists/alsa-devel/msg36465.html, so I thought I would use this as a basis for my own driver. _From what I understand from the datasheet and the way the older driver has configured the registers on the ADAV801 I should be able to see some data on the record outputs, however this is not the case. Could anyone confirm whether or not this driver dated August 2010 is OK to use? Or perhaps some advice on areas to look at that may be preventing the CODEC from manipulating and outputting the digital conversion. Best regards Wayne

1 0

[alsa-devel] [PATCH] ASoC: Fix DMA channel leak in imx-pcm-dma-mx2 driver.
by Javier Martin 17 Jan '12

17 Jan '12

"snd_imx_pcm_hw_params" callback can be called several times by the user (i.e. OSS emulation) leading to a DMA channel leak. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/imx-pcm-dma-mx2.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 43fdc24..d2bb1cb 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -88,11 +88,13 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, iprtd->dma_data.dma_request = dma_params->dma; /* Try to grab a DMA channel */ - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); - if (!iprtd->dma_chan) - return -EINVAL; + if (!iprtd->dma_chan) { + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); + if (!iprtd->dma_chan) + return -EINVAL; + } switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: Add WM8962 DAC and ADC L/R swap support
by Mark Brown 16 Jan '12

16 Jan '12

Simple switches since there's no per-channel control. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 11125c9..e35237a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2223,6 +2223,8 @@ SOC_DOUBLE_R_TLV("Sidetone Volume", WM8962_DAC_DSP_MIXING_1, SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8962_LEFT_DAC_VOLUME, WM8962_RIGHT_DAC_VOLUME, 1, 127, 0, digital_tlv), SOC_SINGLE("DAC High Performance Switch", WM8962_ADC_DAC_CONTROL_2, 0, 1, 0), +SOC_SINGLE("DAC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 5, 1, 0), +SOC_SINGLE("ADC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 8, 1, 0), SOC_SINGLE("ADC High Performance Switch", WM8962_ADDITIONAL_CONTROL_1, 5, 1, 0), -- 1.7.7.3

1 0