simpleaudio: alsa back-end
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/*
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* simpleaudio-alsa.c
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*
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* Copyright (C) 2012 Kamal Mostafa <kamal@whence.com>
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*
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* This program is free software: you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation, either version 3 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#if USE_ALSA
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#include <stdio.h>
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#include <malloc.h>
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#include <assert.h>
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#include <alsa/asoundlib.h>
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#include "simpleaudio.h"
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#include "simpleaudio_internal.h"
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/*
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* ALSA backend for simpleaudio
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*/
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static ssize_t
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sa_alsa_read( simpleaudio *sa, float *buf, size_t nframes )
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{
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ssize_t frames_read = 0;
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snd_pcm_t *pcm = (snd_pcm_t *)sa->backend_handle;
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while ( frames_read < nframes ) {
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ssize_t r;
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r = snd_pcm_readi(pcm, buf, nframes-frames_read);
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if (r < 0) {
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/* silently recover from e.g. overruns, and try once more */
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snd_pcm_prepare(pcm);
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r = snd_pcm_readi(pcm, buf, nframes-frames_read);
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}
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if (r < 0) {
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fprintf(stderr, "snd_pcm_readi: %s\n", snd_strerror(r));
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return -1;
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}
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if (r == 0)
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break;
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frames_read += r;
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}
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return frames_read;
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}
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static ssize_t
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sa_alsa_write( simpleaudio *sa, float *buf, size_t nframes )
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{
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ssize_t frames_written;
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snd_pcm_t *pcm = (snd_pcm_t *)sa->backend_handle;
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frames_written = snd_pcm_writei(pcm, buf, nframes);
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if (frames_written < 0) {
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/* silently recover from e.g. underruns, and try once more */
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snd_pcm_recover(pcm, frames_written, 1 /*silent*/);
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frames_written = snd_pcm_writei(pcm, buf, nframes);
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}
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if (frames_written < 0) {
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fprintf(stderr, "E: %s\n", snd_strerror(frames_written));
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return -1;
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}
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return frames_written;
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}
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static void
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sa_alsa_close( simpleaudio *sa )
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{
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snd_pcm_drain(sa->backend_handle);
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snd_pcm_close(sa->backend_handle);
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}
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static const struct simpleaudio_backend simpleaudio_backend_alsa = {
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sa_alsa_read,
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sa_alsa_write,
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sa_alsa_close,
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};
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simpleaudio *
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simpleaudio_open_stream_alsa(
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// unsigned int rate, unsigned int channels,
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int sa_stream_direction,
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char *app_name, char *stream_name )
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{
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unsigned int rate = 48000;
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unsigned int channels = 1;
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snd_pcm_t *pcm;
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int error;
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error = snd_pcm_open(&pcm,
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"plughw:0,0",
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sa_stream_direction == SA_STREAM_RECORD ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
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0 /*mode*/);
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if (error) {
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fprintf(stderr, "E: Cannot create ALSA stream: %s\n", snd_strerror(error));
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return NULL;
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}
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error = snd_pcm_set_params(pcm,
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SND_PCM_FORMAT_FLOAT,
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SND_PCM_ACCESS_RW_INTERLEAVED,
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channels,
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rate,
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1 /* soft_resample (allow) */,
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0 /* latency */);
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if (error) {
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fprintf(stderr, "E: %s\n", snd_strerror(error));
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snd_pcm_close(pcm);
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return NULL;
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}
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simpleaudio *sa = malloc(sizeof(simpleaudio));
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if ( !sa ) {
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perror("malloc");
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snd_pcm_close(pcm);
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return NULL;
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}
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sa->rate = rate;
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sa->channels = channels;
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sa->backend = &simpleaudio_backend_alsa;
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sa->backend_handle = pcm;
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sa->backend_framesize = sizeof(float);
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return sa;
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}
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#endif /* USE_PULSEAUDIO */
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@ -42,6 +42,12 @@ simpleaudio_open_stream_pulseaudio(
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int sa_stream_direction,
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char *app_name, char *stream_name );
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simpleaudio *
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simpleaudio_open_stream_alsa(
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// unsigned int rate, unsigned int channels,
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int sa_stream_direction,
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char *app_name, char *stream_name );
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simpleaudio *
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simpleaudio_open_stream_sndfile(
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int sa_stream_direction,
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