*/
#include <alsa/asoundlib.h>
-#include "libavformat/avformat.h"
+#include "libavutil/opt.h"
+#include "libavutil/mathematics.h"
+#include "avdevice.h"
#include "alsa-audio.h"
static av_cold int audio_read_header(AVFormatContext *s1,
AlsaData *s = s1->priv_data;
AVStream *st;
int ret;
- unsigned int sample_rate;
enum CodecID codec_id;
- snd_pcm_sw_params_t *sw_params;
-
- if (ap->sample_rate <= 0) {
- av_log(s1, AV_LOG_ERROR, "Bad sample rate %d\n", ap->sample_rate);
-
- return AVERROR(EIO);
- }
-
- if (ap->channels <= 0) {
- av_log(s1, AV_LOG_ERROR, "Bad channels number %d\n", ap->channels);
-
- return AVERROR(EIO);
- }
+ double o;
st = av_new_stream(s1, 0);
if (!st) {
return AVERROR(ENOMEM);
}
- sample_rate = ap->sample_rate;
codec_id = s1->audio_codec_id;
- ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &sample_rate, ap->channels,
+ 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 = sample_rate;
- st->codec->channels = ap->channels;
+ 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 */
+ o = 2 * M_PI * s->period_size / s->sample_rate * 1.5; // bandwidth: 1.5Hz
+ s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
+ sqrt(2 * o), o * o);
+ if (!s->timefilter)
+ goto fail;
return 0;
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
AlsaData *s = s1->priv_data;
- AVStream *st = s1->streams[0];
int res;
- snd_htimestamp_t timestamp;
- snd_pcm_uframes_t ts_delay;
+ int64_t dts;
+ snd_pcm_sframes_t delay = 0;
- if (av_new_packet(pkt, s->period_size) < 0) {
+ if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
return AVERROR(EIO);
}
- while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
+ while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
if (res == -EAGAIN) {
av_free_packet(pkt);
return AVERROR(EIO);
}
+ ff_timefilter_reset(s->timefilter);
}
- snd_pcm_htimestamp(s->h, &ts_delay, ×tamp);
- ts_delay += res;
- pkt->pts = timestamp.tv_sec * 1000000LL
- + (timestamp.tv_nsec * st->codec->sample_rate
- - ts_delay * 1000000000LL + st->codec->sample_rate * 500LL)
- / (st->codec->sample_rate * 1000LL);
+ dts = av_gettime();
+ snd_pcm_delay(s->h, &delay);
+ dts -= av_rescale(delay + res, 1000000, s->sample_rate);
+ pkt->pts = ff_timefilter_update(s->timefilter, dts, res);
pkt->size = res * s->frame_size;
return 0;
}
+static const AVOption options[] = {
+ { "sample_rate", "", offsetof(AlsaData, sample_rate), FF_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+ { "channels", "", offsetof(AlsaData, channels), FF_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+ { NULL },
+};
+
+static const AVClass alsa_demuxer_class = {
+ .class_name = "ALSA demuxer",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
AVInputFormat ff_alsa_demuxer = {
"alsa",
NULL_IF_CONFIG_SMALL("ALSA audio input"),
audio_read_packet,
ff_alsa_close,
.flags = AVFMT_NOFILE,
+ .priv_class = &alsa_demuxer_class,
};