[alsa-devel] Setting format to SND_PCM_FORMAT_MU_LAW does not let me apply hardware parameters
Mitul Sen (misen)
misen at cisco.com
Thu Jul 3 20:55:38 CEST 2008
> -----Original Message-----
> From: stan [mailto:ghjeold_i_mwee at cox.net]
> Sent: Tuesday, July 01, 2008 7:20 PM
> To: Mitul Sen (misen)
> Cc: alsa-devel at alsa-project.org
> Subject: Re: [alsa-devel] Setting format to
> SND_PCM_FORMAT_MU_LAW does not let me apply hardware parameters
>
> 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.
Point noted!
>
> 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.shtm
> l> (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.
Thanks! You are right. I don't get the error when I only specify mu law and let the library convert it to 16 bit signed form. I can now hear "some" audio but it sounds very faint. I have been playing around with the frame size because I seem to be able to hear the audio only when I use a small frame size. I need to get a better understanding of the code and my sound device to improve the audio quality. For now, at least I am able to set the hardware parms with mulaw.
> 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
>
>
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