static av_cold int audio_write_header(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
AVStream *st;
unsigned int sample_rate;
enum CodecID codec_id;
int res;
st = s1->streams[0];
sample_rate = st->codec->sample_rate;
codec_id = st->codec->codec_id;
res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
st->codec->channels, &codec_id);
if (sample_rate != st->codec->sample_rate) {
av_log(s1, AV_LOG_ERROR,
"sample rate %d not available, nearest is %d\n",
st->codec->sample_rate, sample_rate);
goto fail;
}
av_set_pts_info(st, 64, 1, sample_rate);
return res;
fail:
snd_pcm_close(s->h);
return AVERROR(EIO);
}
static av_cold int audio_write_header(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
AVStream *st = NULL;
unsigned int sample_rate;
enum AVCodecID codec_id;
int res;
if (s1->nb_streams != 1 || s1->streams[0]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) {
av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
return AVERROR(EINVAL);
}
st = s1->streams[0];
sample_rate = st->codecpar->sample_rate;
codec_id = st->codecpar->codec_id;
res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
st->codecpar->channels, &codec_id);
if (sample_rate != st->codecpar->sample_rate) {
av_log(s1, AV_LOG_ERROR,
"sample rate %d not available, nearest is %d\n",
st->codecpar->sample_rate, sample_rate);
goto fail;
}
avpriv_set_pts_info(st, 64, 1, sample_rate);
return res;
fail:
snd_pcm_close(s->h);
return AVERROR(EIO);
}
static av_cold int audio_read_header(AVFormatContext *s1,
AVFormatParameters *ap)
{
AlsaData *s = s1->priv_data;
AVStream *st;
int ret;
enum CodecID codec_id;
snd_pcm_sw_params_t *sw_params;
st = av_new_stream(s1, 0);
if (!st) {
av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
return AVERROR(ENOMEM);
}
codec_id = s1->audio_codec_id;
ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
&codec_id);
if (ret < 0) {
return AVERROR(EIO);
}
if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
av_log(s1, AV_LOG_WARNING,
"capture with some ALSA plugins, especially dsnoop, "
"may hang.\n");
ret = snd_pcm_sw_params_malloc(&sw_params);
if (ret < 0) {
av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
snd_strerror(ret));
goto fail;
}
snd_pcm_sw_params_current(s->h, sw_params);
snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
ret = snd_pcm_sw_params(s->h, sw_params);
snd_pcm_sw_params_free(sw_params);
if (ret < 0) {
av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
snd_strerror(ret));
goto fail;
}
/* take real parameters */
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = codec_id;
st->codec->sample_rate = s->sample_rate;
st->codec->channels = s->channels;
av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
fail:
snd_pcm_close(s->h);
return AVERROR(EIO);
}
static av_cold int audio_read_header(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
AVStream *st;
int ret;
enum AVCodecID codec_id;
st = avformat_new_stream(s1, NULL);
if (!st) {
av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
return AVERROR(ENOMEM);
}
codec_id = s1->audio_codec_id;
ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
&codec_id);
if (ret < 0) {
return AVERROR(EIO);
}
/* take real parameters */
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = codec_id;
st->codecpar->sample_rate = s->sample_rate;
st->codecpar->channels = s->channels;
st->codecpar->frame_size = s->frame_size;
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
/* microseconds instead of seconds, MHz instead of Hz */
s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
s->period_size, 1.5E-6);
if (!s->timefilter)
goto fail;
return 0;
fail:
snd_pcm_close(s->h);
return AVERROR(EIO);
}
