[alsa-devel] Setting format to SND_PCM_FORMAT_MU_LAW does not let me apply hardware parameters

[stan]

Mitul Sen (misen) wrote:
> Hi Stan,
>
> Thanks for all your help! I have some more questions though...
>
> I downloaded the source code for alsa-lib-1.0.15 
> Based on the code, if the format is SND_PCM_FORMAT_MU_LAW, I am not sure
> why it does a get/put index to SND_PCM_FORMAT_S16 Also, if the stream is
> SND_PCM_STREAM_PLAYBACK, then I would think that it should decode the
> data. Why does it call snd_pcm_mulaw_decode function if the format is
> SND_PCM_FORMAT_MU_LAW and snd_pcm_mulaw_encode otherwise. I have an
> Intel HDA soundcard and according to the specs, it should support PCM
> ulaw format.
>
> All ALSA documentation and examples I have come across use specific
> hw_params (like sample rate of 44100, SND_PCM_FORMAT_S16, 2 channel
> interleaved data). According to the documents, hw_params refer to the
> stream related info so that's the reason I tried to change it to that of
> mu-law (sampling rate of 8000 Hz, SND_PCM_FORMAT_MU_LAW etc). Not sure
> if that's the way to do it though. Based on the code it looks like the
> hardware just seems to support SND_PCM_FORMAT_S16. Any pointers to help
> me better understand the ALSA code would be much appreciated.
>
>   

Hi Misen,

First, a gentle remonstrance.  You probably have noticed that I always 
put my responses after or mixed with your message.  On public mailing 
lists this is considered good form, rather than posting your response at 
the top of the message.  Why?  So that anyone who steps into the 
interaction doesn't have to read the messages out of order and that 
future searchers have an easier time understanding the message.  While 
top posting is the norm in communications between two or a few people 
because the context is familiar to all and it saves time not to have to 
look for the response, on a public mailing list that isn't necessarily true.

Now to the matter at hand.
I had never heard of mu law so I looked it up.  
http://www.digitalpreservation.gov/formats/fdd/fdd000039.shtml
...
Standard companding algorithm used in digital communications systems in 
North America and Japan (telephones, for the most part) to optimize the 
dynamic range of an analog signal (generally a voice) for digitizing, 
i.e., to compress 16 bit LPCM 
<http://www.digitalpreservation.gov/formats/fdd/fdd000011.shtml> (Linear 
Pulse Code Modulated) data down to 8 bits of logarithmic data. See also 
Notes 
<http://www.digitalpreservation.gov/formats/fdd/fdd000039.shtml#notes> 
below. µ-Law is similar to the A-Law 
<http://www.digitalpreservation.gov/formats/fdd/fdd000038.shtml> 
algorithm used in Europe.
...

The code that you extracted below is designed to convert mu law from the 
compressed form back into the 16 bit signed form.  I haven't checked the 
rest of the code myself, but it appears to assume that the sound device 
is incapable of internal conversion.  If that is true, you shouldn't 
have to specify anything else to the library except mu law.  It should 
take care of everything else. i.e.  as soon as you specify mu law, it is 
known that the stream is 8 bit mono that has to be uncompressed to 16 
bit mono.   I presume that is why there is the error when you try to set 
the hardware parms with mu law.  The library should probably be modified 
to use this new capability of sound device internal conversion for mu 
law if it is available on the sound device.  Maybe it already does;  as 
I said I haven't looked at the code, and I'm not really familiar with mu 
law.

So, given my ignorance, my explanation and proposed solution might be 
completely wrong.  :-)  Perhaps a developer familiar with the coding of 
mu law will give a better explanation.

At this point, I really don't have more to offer for your problem.  I 
would have to look at the code to decipher it in order to give an 
answer.  You might as well do that yourself, as you will get a better 
understanding than I could give with an explanation.
> The code that I am referring to is in pcm_mulaw.c and is as follows:-
>
> static int snd_pcm_mulaw_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t *
> params)
> {
>         snd_pcm_mulaw_t *mulaw = pcm->private_data;
>         snd_pcm_format_t format;
>         int err = snd_pcm_hw_params_slave(pcm, params,
>  
> snd_pcm_mulaw_hw_refine_cchange,
>  
> snd_pcm_mulaw_hw_refine_sprepare,
>  
> snd_pcm_mulaw_hw_refine_schange,
>                                           snd_pcm_generic_hw_params);
>         if (err < 0)
>                 return err;
>
>         err = INTERNAL(snd_pcm_hw_params_get_format)(params, &format);
>         if (err < 0)
>                 return err;
>
>         if (pcm->stream == SND_PCM_STREAM_PLAYBACK) {
>                 if (mulaw->sformat == SND_PCM_FORMAT_MU_LAW) {
>                         mulaw->getput_idx =
> snd_pcm_linear_get_index(format, SND_PCM_FORMAT_S16);
>                         mulaw->func = snd_pcm_mulaw_encode;
>                 } else {
>                         mulaw->getput_idx =
> snd_pcm_linear_put_index(SND_PCM_FORMAT_S16, mulaw->sformat);
>                         mulaw->func = snd_pcm_mulaw_decode;
>                 }
>         } else {
>                 if (mulaw->sformat == SND_PCM_FORMAT_MU_LAW) {
>                         mulaw->getput_idx =
> snd_pcm_linear_put_index(SND_PCM_FORMAT_S16, format);
>                         mulaw->func = snd_pcm_mulaw_decode;
>                 } else {
>                         mulaw->getput_idx =
> snd_pcm_linear_get_index(mulaw->sformat, SND_PCM_FORMAT_S16);
>                         mulaw->func = snd_pcm_mulaw_encode;
>                 }
>         }
>         return 0;
> }  
>
> Thanks and regards,
> Mitul
>
> -----Or