[alsa-devel] ASoC and a codec that can't be controlled

[Timur Tabi]

Liam Girdwood wrote:

>> On big-endian platforms like mine, CS4270_FORMATS will be SNDRV_PCM_FORMAT_S24_BE and on 
>> little-endian platforms it will be SNDRV_PCM_FORMAT_S24_LE.  Assuming the I2S registers 
>> are the same endian as the platform, will this work?
>>
> 
> I think this probably depends on how your audio FIFO's align the data
> received from memory/DMA.  They should do any bit reordering based on
> the audio format, and size etc set in the control register. (Although,
> ymmv.)

Ok, I'm a little confused.  I was going to specify the SNDRV_PCM_FORMAT_S24_BE for my I2C 
device because the actual 32-bit memory-mapped registers are big-endian.  The structure 
definition even uses __be32 for each field.

My goal was to specify a single value (i.e. SNDRV_PCM_FORMAT_S24_BE) such that ALSA would 
give me a 32-bit quantity that exactly matches what my device expects.

I wish asound.h had more documentation.  I don't know what any of these macros *really* mean.

>> #define CS4270_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE)
>>
>> That would tell ALSA that the CS4270 supports both formats, which isn't technically true, 
>> but it wouldn't matter because the I2S interface is what determines the actual 
>> "endianness" of the serial data.
> 
> This looks fine as a workaround atm, I'll try and have a look at this
> bug tomorrow so we don't need to do this.

Ok, I'll do this for now.



-- 
Timur Tabi
Linux Kernel Developer @ Freescale