mirror of https://github.com/mpv-player/mpv
new v4l capture patch by Jindrich Makovicka <makovick@kmlinux.fjfi.cvut.cz>:
- multithreaded audio/video buffering (I know mplayer crew hates threads but it seems to me as the only way of doing reliable a/v capture) - a/v timebase synchronization (sample count vs. gettimeofday) - "immediate" mode support for mplayer - fixed colorspace stuff - RGB?? and YUY2 modes now work as expected - native ALSA audio capture - separated audio input layer git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@7061 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
arpi
parent
5a92702245
commit
6c724895d6
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#include "config.h"
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#ifdef HAVE_ALSA9
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#include <alsa/asoundlib.h>
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#include "audio_in.h"
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#include "mp_msg.h"
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int ai_alsa_setup(audio_in_t *ai)
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{
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snd_pcm_hw_params_t *params;
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snd_pcm_sw_params_t *swparams;
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size_t buffer_size;
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int err;
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size_t n;
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unsigned int rate;
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snd_pcm_uframes_t start_threshold, stop_threshold;
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snd_pcm_hw_params_alloca(¶ms);
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snd_pcm_sw_params_alloca(&swparams);
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err = snd_pcm_hw_params_any(ai->alsa.handle, params);
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Broken configuration for this PCM: no configurations available\n");
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return -1;
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}
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err = snd_pcm_hw_params_set_access(ai->alsa.handle, params,
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SND_PCM_ACCESS_RW_INTERLEAVED);
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Access type not available\n");
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return -1;
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}
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err = snd_pcm_hw_params_set_format(ai->alsa.handle, params, SND_PCM_FORMAT_S16_LE);
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Sample format not available\n");
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return -1;
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}
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err = snd_pcm_hw_params_set_channels(ai->alsa.handle, params, ai->req_channels);
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if (err < 0) {
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ai->channels = snd_pcm_hw_params_get_channels(params);
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mp_msg(MSGT_TV, MSGL_ERR, "Channel count not available - reverting to default: %d\n",
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ai->channels);
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} else {
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ai->channels = ai->req_channels;
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}
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err = snd_pcm_hw_params_set_rate_near(ai->alsa.handle, params, ai->req_samplerate, 0);
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assert(err >= 0);
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rate = err;
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ai->samplerate = rate;
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ai->alsa.buffer_time = 1000000;
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ai->alsa.buffer_time = snd_pcm_hw_params_set_buffer_time_near(ai->alsa.handle, params,
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ai->alsa.buffer_time, 0);
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assert(ai->alsa.buffer_time >= 0);
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ai->alsa.period_time = ai->alsa.buffer_time / 4;
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ai->alsa.period_time = snd_pcm_hw_params_set_period_time_near(ai->alsa.handle, params,
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ai->alsa.period_time, 0);
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assert(ai->alsa.period_time >= 0);
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err = snd_pcm_hw_params(ai->alsa.handle, params);
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Unable to install hw params:");
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snd_pcm_hw_params_dump(params, ai->alsa.log);
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return -1;
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}
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ai->alsa.chunk_size = snd_pcm_hw_params_get_period_size(params, 0);
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buffer_size = snd_pcm_hw_params_get_buffer_size(params);
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if (ai->alsa.chunk_size == buffer_size) {
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mp_msg(MSGT_TV, MSGL_ERR, "Can't use period equal to buffer size (%u == %lu)\n", ai->alsa.chunk_size, (long)buffer_size);
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return -1;
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}
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snd_pcm_sw_params_current(ai->alsa.handle, swparams);
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err = snd_pcm_sw_params_set_sleep_min(ai->alsa.handle, swparams,0);
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assert(err >= 0);
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err = snd_pcm_sw_params_set_avail_min(ai->alsa.handle, swparams, ai->alsa.chunk_size);
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assert(err >= 0);
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err = snd_pcm_sw_params_set_start_threshold(ai->alsa.handle, swparams, 0);
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assert(err >= 0);
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err = snd_pcm_sw_params_set_stop_threshold(ai->alsa.handle, swparams, buffer_size);
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assert(err >= 0);
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assert(err >= 0);
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if (snd_pcm_sw_params(ai->alsa.handle, swparams) < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "unable to install sw params:\n");
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snd_pcm_sw_params_dump(swparams, ai->alsa.log);
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return -1;
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}
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if (mp_msg_test(MSGT_TV, MSGL_V)) {
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snd_pcm_dump(ai->alsa.handle, ai->alsa.log);
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}
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ai->alsa.bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
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ai->alsa.bits_per_frame = ai->alsa.bits_per_sample * ai->channels;
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ai->blocksize = ai->alsa.chunk_size * ai->alsa.bits_per_frame / 8;
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ai->samplesize = ai->alsa.bits_per_sample;
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ai->bytes_per_sample = ai->alsa.bits_per_sample/8;
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return 0;
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}
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int ai_alsa_init(audio_in_t *ai)
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{
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int err;
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err = snd_pcm_open(&ai->alsa.handle, ai->alsa.device, SND_PCM_STREAM_CAPTURE, 0);
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Error opening audio");
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return -1;
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}
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err = snd_output_stdio_attach(&ai->alsa.log, stderr, 0);
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if (err < 0) {
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return -1;
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}
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err = ai_alsa_setup(ai);
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return err;
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}
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#endif /* HAVE_ALSA9 */
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#include "config.h"
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#ifdef HAVE_ALSA9
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#include <alsa/asoundlib.h>
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#include "audio_in.h"
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#include "mp_msg.h"
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int ai_alsa_setup(audio_in_t *ai)
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{
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snd_pcm_hw_params_t *params;
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snd_pcm_sw_params_t *swparams;
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size_t buffer_size;
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int err;
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size_t n;
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unsigned int rate;
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snd_pcm_uframes_t start_threshold, stop_threshold;
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snd_pcm_hw_params_alloca(¶ms);
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snd_pcm_sw_params_alloca(&swparams);
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err = snd_pcm_hw_params_any(ai->alsa.handle, params);
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Broken configuration for this PCM: no configurations available\n");
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return -1;
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}
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err = snd_pcm_hw_params_set_access(ai->alsa.handle, params,
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SND_PCM_ACCESS_RW_INTERLEAVED);
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Access type not available\n");
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return -1;
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}
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err = snd_pcm_hw_params_set_format(ai->alsa.handle, params, SND_PCM_FORMAT_S16_LE);
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Sample format not available\n");
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return -1;
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}
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err = snd_pcm_hw_params_set_channels(ai->alsa.handle, params, ai->req_channels);
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if (err < 0) {
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ai->channels = snd_pcm_hw_params_get_channels(params);
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mp_msg(MSGT_TV, MSGL_ERR, "Channel count not available - reverting to default: %d\n",
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ai->channels);
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} else {
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ai->channels = ai->req_channels;
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}
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err = snd_pcm_hw_params_set_rate_near(ai->alsa.handle, params, ai->req_samplerate, 0);
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assert(err >= 0);
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rate = err;
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ai->samplerate = rate;
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ai->alsa.buffer_time = 1000000;
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ai->alsa.buffer_time = snd_pcm_hw_params_set_buffer_time_near(ai->alsa.handle, params,
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ai->alsa.buffer_time, 0);
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assert(ai->alsa.buffer_time >= 0);
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ai->alsa.period_time = ai->alsa.buffer_time / 4;
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ai->alsa.period_time = snd_pcm_hw_params_set_period_time_near(ai->alsa.handle, params,
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ai->alsa.period_time, 0);
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assert(ai->alsa.period_time >= 0);
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err = snd_pcm_hw_params(ai->alsa.handle, params);
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Unable to install hw params:");
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snd_pcm_hw_params_dump(params, ai->alsa.log);
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return -1;
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}
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ai->alsa.chunk_size = snd_pcm_hw_params_get_period_size(params, 0);
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buffer_size = snd_pcm_hw_params_get_buffer_size(params);
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if (ai->alsa.chunk_size == buffer_size) {
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mp_msg(MSGT_TV, MSGL_ERR, "Can't use period equal to buffer size (%u == %lu)\n", ai->alsa.chunk_size, (long)buffer_size);
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return -1;
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}
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snd_pcm_sw_params_current(ai->alsa.handle, swparams);
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err = snd_pcm_sw_params_set_sleep_min(ai->alsa.handle, swparams,0);
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assert(err >= 0);
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err = snd_pcm_sw_params_set_avail_min(ai->alsa.handle, swparams, ai->alsa.chunk_size);
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assert(err >= 0);
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err = snd_pcm_sw_params_set_start_threshold(ai->alsa.handle, swparams, 0);
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assert(err >= 0);
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err = snd_pcm_sw_params_set_stop_threshold(ai->alsa.handle, swparams, buffer_size);
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assert(err >= 0);
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assert(err >= 0);
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if (snd_pcm_sw_params(ai->alsa.handle, swparams) < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "unable to install sw params:\n");
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snd_pcm_sw_params_dump(swparams, ai->alsa.log);
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return -1;
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}
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if (mp_msg_test(MSGT_TV, MSGL_V)) {
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snd_pcm_dump(ai->alsa.handle, ai->alsa.log);
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}
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ai->alsa.bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
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ai->alsa.bits_per_frame = ai->alsa.bits_per_sample * ai->channels;
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ai->blocksize = ai->alsa.chunk_size * ai->alsa.bits_per_frame / 8;
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ai->samplesize = ai->alsa.bits_per_sample;
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ai->bytes_per_sample = ai->alsa.bits_per_sample/8;
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return 0;
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}
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int ai_alsa_init(audio_in_t *ai)
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{
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int err;
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err = snd_pcm_open(&ai->alsa.handle, ai->alsa.device, SND_PCM_STREAM_CAPTURE, 0);
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Error opening audio");
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return -1;
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}
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err = snd_output_stdio_attach(&ai->alsa.log, stderr, 0);
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if (err < 0) {
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return -1;
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}
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err = ai_alsa_setup(ai);
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return err;
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}
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#endif /* HAVE_ALSA9 */
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#include "config.h"
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#include <linux/soundcard.h>
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#include <fcntl.h>
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#include <errno.h>
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#include "audio_in.h"
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#include "mp_msg.h"
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int ai_oss_set_samplerate(audio_in_t *ai)
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{
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int tmp = ai->req_samplerate;
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if (ioctl(ai->oss.audio_fd, SNDCTL_DSP_SPEED, &tmp) == -1) return -1;
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ai->samplerate = ai->req_samplerate;
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return 0;
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}
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int ai_oss_set_channels(audio_in_t *ai)
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{
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int err;
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int ioctl_param;
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if (ai->req_channels > 2)
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{
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ioctl_param = ai->req_channels;
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mp_msg(MSGT_TV, MSGL_V, "ioctl dsp channels: %d\n",
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err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_CHANNELS, &ioctl_param));
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Unable to set channel count: %d\n",
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ai->req_channels);
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return -1;
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}
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}
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else
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{
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ioctl_param = (ai->req_channels == 2);
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mp_msg(MSGT_TV, MSGL_V, "ioctl dsp stereo: %d (req: %d)\n",
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err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_STEREO, &ioctl_param),
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ioctl_param);
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Unable to set stereo: %d\n",
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ai->req_channels == 2);
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return -1;
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}
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}
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ai->channels = ai->req_channels;
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return 0;
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}
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int ai_oss_init(audio_in_t *ai)
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{
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int err;
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int ioctl_param;
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ai->oss.audio_fd = open(ai->oss.device, O_RDONLY);
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if (ai->oss.audio_fd < 0)
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{
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mp_msg(MSGT_TV, MSGL_ERR, "unable to open '%s': %s\n",
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ai->oss.device, strerror(errno));
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return -1;
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}
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ioctl_param = 0 ;
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mp_msg(MSGT_TV, MSGL_V, "ioctl dsp getfmt: %d\n",
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ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETFMTS, &ioctl_param));
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mp_msg(MSGT_TV, MSGL_V, "Supported formats: %x\n", ioctl_param);
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if (!(ioctl_param & AFMT_S16_LE))
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mp_msg(MSGT_TV, MSGL_ERR, "notsupported format\n");
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ioctl_param = AFMT_S16_LE;
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mp_msg(MSGT_TV, MSGL_V, "ioctl dsp setfmt: %d\n",
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err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETFMT, &ioctl_param));
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Unable to set audio format.");
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return -1;
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}
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if (ai_oss_set_channels(ai) < 0) return -1;
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ioctl_param = ai->req_samplerate;
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mp_msg(MSGT_TV, MSGL_V, "ioctl dsp speed: %d\n",
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err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SPEED, &ioctl_param));
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Unable to set samplerate: %d\n",
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ai->req_samplerate);
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return -1;
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}
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ai->samplerate = ai->req_samplerate;
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mp_msg(MSGT_TV, MSGL_V, "ioctl dsp trigger: %d\n",
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ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETTRIGGER, &ioctl_param));
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mp_msg(MSGT_TV, MSGL_V, "trigger: %x\n", ioctl_param);
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ioctl_param = PCM_ENABLE_INPUT;
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mp_msg(MSGT_TV, MSGL_V, "ioctl dsp trigger: %d\n",
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err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETTRIGGER, &ioctl_param));
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Unable to set trigger: %d\n",
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PCM_ENABLE_INPUT);
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return -1;
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}
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ai->blocksize = 0;
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mp_msg(MSGT_TV, MSGL_V, "ioctl dsp getblocksize: %d\n",
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err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETBLKSIZE, &ai->blocksize));
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if (err < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "Unable to get block size!\n");
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}
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mp_msg(MSGT_TV, MSGL_V, "blocksize: %d\n", ai->blocksize);
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// correct the blocksize to a reasonable value
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if (ai->blocksize <= 0) {
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ai->blocksize = 4096*ai->channels*2;
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mp_msg(MSGT_TV, MSGL_ERR, "audio block size is zero, setting to %d!\n", ai->blocksize);
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} else if (ai->blocksize < 4096*ai->channels*2) {
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ai->blocksize *= 4096*ai->channels*2/ai->blocksize;
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mp_msg(MSGT_TV, MSGL_ERR, "audio block size too low, setting to %d!\n", ai->blocksize);
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}
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ai->samplesize = 16;
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ai->bytes_per_sample = 2;
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return 0;
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}
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#include "config.h"
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#include "audio_in.h"
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#include "mp_msg.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <errno.h>
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// sanitizes ai structure before calling other functions
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int audio_in_init(audio_in_t *ai, int type)
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{
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ai->type = type;
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ai->setup = 0;
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ai->channels = -1;
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ai->samplerate = -1;
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ai->blocksize = -1;
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ai->bytes_per_sample = -1;
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ai->samplesize = -1;
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switch (ai->type) {
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#ifdef HAVE_ALSA9
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case AUDIO_IN_ALSA:
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ai->alsa.handle = NULL;
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ai->alsa.log = NULL;
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ai->alsa.device = strdup("default");
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return 0;
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#endif
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case AUDIO_IN_OSS:
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ai->oss.audio_fd = -1;
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ai->oss.device = strdup("/dev/dsp");
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return 0;
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default:
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return -1;
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}
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}
|
||||
|
||||
int audio_in_setup(audio_in_t *ai)
|
||||
{
|
||||
int err;
|
||||
|
||||
switch (ai->type) {
|
||||
#ifdef HAVE_ALSA9
|
||||
case AUDIO_IN_ALSA:
|
||||
if (ai_alsa_init(ai) < 0) return -1;
|
||||
ai->setup = 1;
|
||||
return 0;
|
||||
#endif
|
||||
case AUDIO_IN_OSS:
|
||||
if (ai_oss_init(ai) < 0) return -1;
|
||||
ai->setup = 1;
|
||||
return 0;
|
||||
default:
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
|
||||
int audio_in_set_samplerate(audio_in_t *ai, int rate)
|
||||
{
|
||||
switch (ai->type) {
|
||||
#ifdef HAVE_ALSA9
|
||||
case AUDIO_IN_ALSA:
|
||||
ai->req_samplerate = rate;
|
||||
if (!ai->setup) return 0;
|
||||
if (ai_alsa_setup(ai) < 0) return -1;
|
||||
return ai->samplerate;
|
||||
#endif
|
||||
case AUDIO_IN_OSS:
|
||||
ai->req_samplerate = rate;
|
||||
if (!ai->setup) return 0;
|
||||
if (ai_oss_set_samplerate(ai) < 0) return -1;
|
||||
return ai->samplerate;
|
||||
default:
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
|
||||
int audio_in_set_channels(audio_in_t *ai, int channels)
|
||||
{
|
||||
switch (ai->type) {
|
||||
#ifdef HAVE_ALSA9
|
||||
case AUDIO_IN_ALSA:
|
||||
ai->req_channels = channels;
|
||||
if (!ai->setup) return 0;
|
||||
if (ai_alsa_setup(ai) < 0) return -1;
|
||||
return ai->channels;
|
||||
#endif
|
||||
case AUDIO_IN_OSS:
|
||||
ai->req_channels = channels;
|
||||
if (!ai->setup) return 0;
|
||||
if (ai_oss_set_channels(ai) < 0) return -1;
|
||||
return ai->channels;
|
||||
default:
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
|
||||
int audio_in_set_device(audio_in_t *ai, char *device)
|
||||
{
|
||||
int i;
|
||||
if (ai->setup) return -1;
|
||||
switch (ai->type) {
|
||||
#ifdef HAVE_ALSA9
|
||||
case AUDIO_IN_ALSA:
|
||||
if (ai->alsa.device) free(ai->alsa.device);
|
||||
ai->alsa.device = strdup(device);
|
||||
/* mplayer cannot handle colons in arguments */
|
||||
for (i = 0; i < strlen(ai->alsa.device); i++) {
|
||||
if (ai->alsa.device[i] == ',') ai->alsa.device[i] = ':';
|
||||
}
|
||||
return 0;
|
||||
#endif
|
||||
case AUDIO_IN_OSS:
|
||||
if (ai->oss.device) free(ai->oss.device);
|
||||
ai->oss.device = strdup(device);
|
||||
return 0;
|
||||
default:
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
|
||||
int audio_in_uninit(audio_in_t *ai)
|
||||
{
|
||||
if (ai->setup) {
|
||||
switch (ai->type) {
|
||||
#ifdef HAVE_ALSA9
|
||||
case AUDIO_IN_ALSA:
|
||||
if (ai->alsa.log)
|
||||
snd_output_close(ai->alsa.log);
|
||||
if (ai->alsa.handle) {
|
||||
snd_pcm_close(ai->alsa.handle);
|
||||
}
|
||||
ai->setup = 0;
|
||||
return 0;
|
||||
#endif
|
||||
case AUDIO_IN_OSS:
|
||||
close(ai->oss.audio_fd);
|
||||
ai->setup = 0;
|
||||
return 0;
|
||||
default:
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
int audio_in_start_capture(audio_in_t *ai)
|
||||
{
|
||||
switch (ai->type) {
|
||||
#ifdef HAVE_ALSA9
|
||||
case AUDIO_IN_ALSA:
|
||||
return snd_pcm_start(ai->alsa.handle);
|
||||
#endif
|
||||
case AUDIO_IN_OSS:
|
||||
return 0;
|
||||
default:
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
|
||||
int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer)
|
||||
{
|
||||
int ret;
|
||||
|
||||
switch (ai->type) {
|
||||
#ifdef HAVE_ALSA9
|
||||
case AUDIO_IN_ALSA:
|
||||
ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size);
|
||||
if (ret != ai->alsa.chunk_size) {
|
||||
if (ret < 0) {
|
||||
mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", snd_strerror(ret));
|
||||
} else {
|
||||
mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n");
|
||||
}
|
||||
return -1;
|
||||
}
|
||||
return ret;
|
||||
#endif
|
||||
case AUDIO_IN_OSS:
|
||||
ret = read(ai->oss.audio_fd, buffer, ai->blocksize);
|
||||
if (ret != ai->blocksize) {
|
||||
if (ret < 0) {
|
||||
mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", strerror(errno));
|
||||
} else {
|
||||
mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n");
|
||||
}
|
||||
return -1;
|
||||
}
|
||||
return ret;
|
||||
default:
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
|
|
@ -0,0 +1,68 @@
|
|||
#ifndef _audio_in_h
|
||||
#define _audio_in_h
|
||||
|
||||
#define AUDIO_IN_ALSA 1
|
||||
#define AUDIO_IN_OSS 2
|
||||
|
||||
#include "config.h"
|
||||
|
||||
#ifdef HAVE_ALSA9
|
||||
#include <alsa/asoundlib.h>
|
||||
|
||||
typedef struct {
|
||||
char *device;
|
||||
|
||||
snd_pcm_t *handle;
|
||||
snd_output_t *log;
|
||||
int buffer_time, period_time, chunk_size;
|
||||
size_t bits_per_sample, bits_per_frame;
|
||||
} ai_alsa_t;
|
||||
#endif
|
||||
|
||||
typedef struct {
|
||||
char *device;
|
||||
|
||||
int audio_fd;
|
||||
} ai_oss_t;
|
||||
|
||||
typedef struct
|
||||
{
|
||||
int type;
|
||||
int setup;
|
||||
|
||||
/* requested values */
|
||||
int req_channels;
|
||||
int req_samplerate;
|
||||
|
||||
/* real values read-only */
|
||||
int channels;
|
||||
int samplerate;
|
||||
int blocksize;
|
||||
int bytes_per_sample;
|
||||
int samplesize;
|
||||
|
||||
#ifdef HAVE_ALSA9
|
||||
ai_alsa_t alsa;
|
||||
#endif
|
||||
ai_oss_t oss;
|
||||
} audio_in_t;
|
||||
|
||||
int audio_in_init(audio_in_t *ai, int type);
|
||||
int audio_in_setup(audio_in_t *ai);
|
||||
int audio_in_set_device(audio_in_t *ai, char *device);
|
||||
int audio_in_set_samplerate(audio_in_t *ai, int rate);
|
||||
int audio_in_set_channels(audio_in_t *ai, int channels);
|
||||
int audio_in_uninit(audio_in_t *ai);
|
||||
int audio_in_start_capture(audio_in_t *ai);
|
||||
int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer);
|
||||
|
||||
#ifdef HAVE_ALSA9
|
||||
int ai_alsa_setup(audio_in_t *ai);
|
||||
int ai_alsa_init(audio_in_t *ai);
|
||||
#endif
|
||||
|
||||
int ai_oss_set_samplerate(audio_in_t *ai);
|
||||
int ai_oss_set_channels(audio_in_t *ai);
|
||||
int ai_oss_init(audio_in_t *ai);
|
||||
|
||||
#endif /* _audio_in_h */
|
||||
Loading…
Reference in New Issue