[alsa-devel] some problem about alsa lib usage on arm926

[Akio]

Dear All:

          right now, I'm using alsa lib interface to work with voip application.
          Our hardware platform is an ARM926 266MHz box with Linux kernel
          2.6.19.2. We got some problems. would you like to do us favors?

          1. what is the different between "plughw:0,0", "defalut", "hw:0.0"?
              after reading the document in 
              http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html
              about "PCM naming conventions", we still not very sure the
relationship
              between those device conventions. is there any more detail documents
              talk about on this? right now, we are using alsa lib 1.0.13 and
the device
              is "plughw:0,0".

         2. Is there any documents detail discuss the setting about
             sample rate, buffer size, period size. because, I have two
different voice
             codec to handle, one is narrow band(8k Hz), the other is wide
band(16K Hz).
             right now, I using the following functions and setting to make alsa
work.
             snd_pcm_hw_params_set_rate_near sets sample rate, 8k or 16k.
             snd_pcm_hw_params_set_period_size_near sets period size, current is
128.
             snd_pcm_hw_params_set_buffer_size_near sets buffer size, 8k is 2048, 
            16k is 4096. the unit seems be samples not byte. 

         3. the code was based on the following URL.
             http://www.suse.de/~mana/alsa090_howto.html
             http://www.equalarea.com/paul/alsa-audio.html
             open capture and playback at one function, and provide one read and
one write
             interface to up layer. here comes three problems:
             a. the channel opened is just only one, but If I set the 
                 snd_pcm_hw_params_set_access as 
                 SND_PCM_ACCESS_RW_NONINTERLEAVED.
                 using snd_pcm_readn and snd_pcm_writen the code will not work.
what is
                 the wrong? 
                 but if set snd_pcm_hw_params_set_access as
                 SND_PCM_ACCESS_RW_INTERLEAVED
                 and using snd_pcm_readi and snd_pcm_writei. the code work very
well.
                 one channel also have data interleave problem? I don't
understand why?

             b. in snd_pcm_open the block flag is set as 0. what is the mode of
the  
                 operation? blocking or non-blocking? 

             c. call the xrun_recovery() timing. currently the xrun_recovery
only do on EPIPE.
                 the following is the handling core.
                 err = snd_pcm_prepare(handle);
                 if(err == -ESTRPIPE) {
                     while ((err = snd_pcm_resume(handle)) == -EAGAIN) {
                               nanosleep(&retryTime, &tmpTime);
                               LOG("audio device resume fail, %s\n",
snd_strerror(err));
                     }
                     if(err < 0) {
                             err = snd_pcm_prepare(handle);
                             if(err < 0) LOG("audio data IO fail, %s\n",
snd_strerror(err));
                     }
                 } else {
                     if(err < 0) LOG("audio device prepare fail, %s\n",
snd_strerror(err));
                 }

                Does the code have any missing point to handle? If any missing,
what should
                I to add? 

         4. BTW, I call xrun_recovery both in capture and playback direction.
After long 
             time test. the test procedure is as the following steps:
                1. capture voice data as PCM data.
                2. encoded as amr-wb bit stream.
                3. decoded as PCM data from encoded bit stream.
                4. playback PCM data.
             after 16 hours continue tests, there are some delay, about 800 mili
seconds.
             is there any way to reduce the delay after long time test. we want
make the
             delay about < 400 mili seconds after one week continue test. BTW,
is there
             any way to prevent xrun_recovery calling? is there any index to
monitor ALSA
             device buffer information? 

          5. If we want using salsa lib to replace the usage of alsa lib, what
do we have  to
              take care? how about the latency and performance between salsa and
alsa?
              how about the stability of salsa?

          There are so many problems after many tests. If you found any documents we
          need to study more detail. please tell us. Thanks in advance. Thank you.

Best Regards,
Akio