[alsa-devel] [ASoC] Sample format non available
Liam Girdwood
lg at opensource.wolfsonmicro.com
Thu Apr 26 15:46:03 CEST 2007
On Thu, 2007-04-26 at 15:10 +0200, Markus Korber wrote:
> Dear list,
>
> I'm currently implementing a ASoC solution for a custom chip and I've
> written a codec, I2S, PCM, and machine driver. However, when using
> aplay I get the following error (requested format was: 2):
>
> root:~# aplay -M -D hw:0,0 test.wav
> Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
> aplay: set_params:906: Sample format non available
>
> However, I've told each piece involved, that it is capable of playing
> S16_LE. So I don't know what I am missing here?
>
Is your target CPU ARM based ?
If so, there is a gcc optimisation bug that causes a refinement error
(like above).
http://gcc.gnu.org/bugzilla/show_bug.cgi?id=27363
Fwiw, Openembedded builds a working toolchain for ARM and ALSA.
> Should I provide more debugging output like RULES_DEBUG output from
> sound/core/pcm_native.c?
>
Yes please. Can you also set SOC_DEBUG to 1 in soc-core.c
> ,----[ CODEC DAI ]
> | #define CS4265_RATES (SNDRV_PCM_RATE_8000_192000)
> | #define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
> |
> | struct snd_soc_codec_dai cs4265_dai = {
> | .name = "CS4265",
> | .playback = {
> | .stream_name = "Playback",
> | .channels_min = 1,
> | .channels_max = 2,
> | .rates = CS4265_RATES,
> | .formats = CS4265_FORMATS,},
> | .capture = {
> | .stream_name = "Capture",
> | .channels_min = 1,
> | .channels_max = 2,
> | .rates = CS4265_RATES,
> | .formats = CS4265_FORMATS,},
> | [...]
> | };
> `----
>
> ,----[ CPU DAI ]
> | #define CHIP_I2S_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
> | SNDRV_PCM_RATE_48000)
> | #define CHIP_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE)
> |
> | struct snd_soc_cpu_dai chip_i2s_dai = {
> | /* DAI description */
> | [...]
> | /* DAI callbacks */
> | [...]
> | /* DAI capabilities */
> | .playback = {
> | .channels_min = 2,
> | .channels_max = 2,
> | .rates = CHIP_I2S_RATES,
> | .formats = CHIP_I2S_FORMATS,},
> | .capture = {
> | .channels_min = 2,
> | .channels_max = 2,
> | .rates = CHIP_I2S_RATES,
> | .formats = CHIP_I2S_FORMATS,},
> | /* ops */
> | [...]
> `----
>
> ,----[ HW DAI ]
> | static const struct snd_pcm_hardware chip_pcm_hardware = {
> | .info = SNDRV_PCM_INFO_MMAP |
> | SNDRV_PCM_INFO_MMAP_VALID | /* For OSS emulation */
> | SNDRV_PCM_INFO_BLOCK_TRANSFER | /* For OSS emulation */
> | SNDRV_PCM_INFO_NONINTERLEAVED,
> | .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
> | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE |
> | .channels_min = 2, /* Stereo */
> | .channels_max = 2,
> | .period_bytes_min = 32,
> | .period_bytes_max = 8192,
> | .periods_min = 2,
> | .periods_max = 1024, /* PAGE_SIZE/sizeof(chip_dma_desc), */
> | .buffer_bytes_max = 32 * 1024,
> | .fifo_size = 0, /* Still unused in ALSA? */
> | };
> `----
>
This looks fine. All show S16_LE and 44100 rate stereo.
> root:~# uname -a
> Linux 2.6.21
> root:~# cat /proc/asound/version
> Advanced Linux Sound Architecture Driver Version 1.0.14rc3 (Wed Mar 14 07:25:50 2007 UTC).
> root:~# aplay --version
> aplay: version 1.0.14rc2 by Jaroslav Kysela <perex at suse.cz>
>
> root:~# aplay -M test.wav
> chip-pcm: Entered chip_pcm_open
> board_cs4265: Entered board_cs4265_startup
> Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
> ALSA lib pcm_plug.c:840:(snd_pcm_plug_hw_refine_cchange) Unable to find an usable client format
Can you send the debug output with the debug turned on (as above) and
with aplay writing directly to the hardware i.e.
aplay -Dhw:0,0 test.wav
Thanks
Liam
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