[alsa-devel] Using 8bit mono at 8000Hz with arecord and aplay - Full duplex operation

[Shilpa Kedar Walvekar]

The driver and the hardware supports it as we have played audio file of this format using raw open/read/write calls (without using ALSA).
Since we want Full duplex so we switched to alsa.

We tried to use plug as well as follows:
pcm_slave.sl3 {
 pcm
 format U8
 channels 1
 rate 8000
}

pcm.complex_convert {
 type plug
 slave sl3
}

Then still the same error, " Sample format non available"

Now we are trying to record and play .wav audio file in full duplex mode: We have two threads, one is playing the file continuously (S16_LE, 48000) and other thread is trying to record to a wav file (here using S16_LE, 44100). Now we get error "Broken pipe" for each snd_pcm_readi call.
We are stuck here and cannot move forward.

Code used for recording is:

/* open sound device */
       if ((err = snd_pcm_open(&handle, "hw:0,0", SND_PCM_STREAM_CAPTURE, 0)) < 0) {
                 printf("Capture open error: %s\n", snd_strerror(err));
                 exit(EXIT_FAILURE);
         }


        if ((err = snd_pcm_set_params(handle,
                                       SND_PCM_FORMAT_S16_LE,
                                       SND_PCM_ACCESS_RW_INTERLEAVED,
                                       2, //channels
                                       44100,
                                       1,
                                            0)) < 0) {

                 printf("set_param_Capture open error: %s\n", snd_strerror(err));
                 exit(EXIT_FAILURE);
         }


/* open file for recording */
        write_fd = open(writefile,O_WRONLY | O_CREAT , 0777);
        if(write_fd < 0)
        {
                fprintf(stderr,"Can't open %s\ns",writefile);
                exit(1);
        }


        while(1)
        {
                if ((err = snd_pcm_prepare (handle)) < 0) {
                        fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
                                snd_strerror (err));

                }


                         if ((err = snd_pcm_readi (handle, buf_mic, mic_samples)) != mic_samples) {
                                fprintf (stderr, "read from audio interface failed (%s)\n",
                                         snd_strerror (err));
                        }

        }

Note: recording works successfully using arecord.

Please help us to resolve:
1. Recording and Playing 8bit mono audio at 8000Hz
2. Full duplex functionality using alsa apis


Thanks & Regards,
--Shilpa.

-----Original Message-----
From: Stefan Schoenleitner [mailto:dev.c0debabe at gmail.com]
Sent: Tuesday, August 18, 2009 3:30 PM
To: Shilpa Kedar Walvekar
Cc: alsa-devel at alsa-project.org
Subject: Re: [alsa-devel] Using 8bit mono at 8000Hz with arecord and aplay

Shilpa Kedar Walvekar wrote:
> Hello,
>
> We tried to record and play the recorded file using arecord and aplay as follows:
> arecord -f cd /home/test.wav
> aplay -f cd /home/test.wav
>
> The file gets recorded and played correctly.
>
> But if I try to record file of format S8 or U8, I get error:
>
> # arecord -f U8 /home/test_alsa_record_2
> Recording WAVE '/home/test_alsa_record_2' : Unsigned 8 bit, Rate 8000 Hz, Mono
> arecord: set_params:979: Sample format non available
>
> # arecord -f S8 /home/alsa_record_2
> Recording WAVE '/home/alsa_record_2' : Signed 8 bit, Rate 8000 Hz, Mono
> arecord: set_params:979: Sample format non available
>
> Is there any other way to set the format as S8 or U8? Or only cd/cdr/dat formats can be played using alsa?

Basically it should be possible to play these formats.
If you look at the aplay help output, it also lists the recognized
sample formats (which include U8 and S8).

However, it also says "Some of these may not be available on selected
hardware".

For this reason I assume that your hardware (or at least the driver)
does not support these formats.

A workaround is to use one of the ALSA format conversion plugins.
(see here: http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html)

Specifically have a look at the rate or plug plugin.

The idea is that you can play arbitrary formats and the plugin
automatically converts the PCM streams to a fixed format that your
hardware can deal with (e.g. 16 bit at 48 kHz).

cheers,
stefan