* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
* @author Benoit Fouet ( benoit fouet free fr )
* @author Nicolas George ( nicolas george normalesup org )
*
- * This avdevice decoder allows to capture audio from an ALSA (Advanced
+ * This avdevice decoder can capture audio from an ALSA (Advanced
* Linux Sound Architecture) device.
*
* The filename parameter is the name of an ALSA PCM device capable of
#include <alsa/asoundlib.h>
#include "libavutil/internal.h"
+#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
+#include "libavutil/time.h"
-#include "libavformat/avformat.h"
#include "libavformat/internal.h"
+#include "avdevice.h"
#include "alsa.h"
static av_cold int audio_read_header(AVFormatContext *s1)
AVStream *st;
int ret;
enum AVCodecID codec_id;
- snd_pcm_sw_params_t *sw_params;
st = avformat_new_stream(s1, NULL);
if (!st) {
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->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;
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_packet_unref(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->codecpar->sample_rate
- - (int64_t)ts_delay * 1000000000LL + st->codecpar->sample_rate * 500LL)
- / (st->codecpar->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, s->last_period);
+ s->last_period = res;
pkt->size = res * s->frame_size;
return 0;
}
+static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
+{
+ return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
+}
+
static const AVOption options[] = {
{ "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
+ .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
};
AVInputFormat ff_alsa_demuxer = {
.read_header = audio_read_header,
.read_packet = audio_read_packet,
.read_close = ff_alsa_close,
+ .get_device_list = audio_get_device_list,
.flags = AVFMT_NOFILE,
.priv_class = &alsa_demuxer_class,
};
projects / ffmpeg / blobdiff