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![Derek Buitenhuis](/assets/img/avatar_default.png)
* commit '9200514ad8717c63f82101dc394f4378854325bf': lavf: replace AVStream.codec with AVStream.codecpar This has been a HUGE effort from: - Derek Buitenhuis <derek.buitenhuis@gmail.com> - Hendrik Leppkes <h.leppkes@gmail.com> - wm4 <nfxjfg@googlemail.com> - Clément Bœsch <clement@stupeflix.com> - James Almer <jamrial@gmail.com> - Michael Niedermayer <michael@niedermayer.cc> - Rostislav Pehlivanov <atomnuker@gmail.com> Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
169 lines
5.2 KiB
C
169 lines
5.2 KiB
C
/*
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* ALSA input and output
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* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
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* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg 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 GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* ALSA input and output: input
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* @author Luca Abeni ( lucabe72 email it )
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* @author Benoit Fouet ( benoit fouet free fr )
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* @author Nicolas George ( nicolas george normalesup org )
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*
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* This avdevice decoder can capture audio from an ALSA (Advanced
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* Linux Sound Architecture) device.
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*
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* The filename parameter is the name of an ALSA PCM device capable of
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* capture, for example "default" or "plughw:1"; see the ALSA documentation
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* for naming conventions. The empty string is equivalent to "default".
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*
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* The capture period is set to the lower value available for the device,
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* which gives a low latency suitable for real-time capture.
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*
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* The PTS are an Unix time in microsecond.
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*
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* Due to a bug in the ALSA library
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* (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
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* decoder does not work with certain ALSA plugins, especially the dsnoop
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* plugin.
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*/
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#include <alsa/asoundlib.h>
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#include "libavutil/internal.h"
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#include "libavutil/mathematics.h"
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#include "libavutil/opt.h"
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#include "libavutil/time.h"
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#include "libavformat/internal.h"
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#include "avdevice.h"
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#include "alsa.h"
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static av_cold int audio_read_header(AVFormatContext *s1)
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{
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AlsaData *s = s1->priv_data;
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AVStream *st;
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int ret;
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enum AVCodecID codec_id;
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st = avformat_new_stream(s1, NULL);
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if (!st) {
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av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
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return AVERROR(ENOMEM);
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}
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codec_id = s1->audio_codec_id;
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ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
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&codec_id);
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if (ret < 0) {
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return AVERROR(EIO);
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}
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/* take real parameters */
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st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codecpar->codec_id = codec_id;
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st->codecpar->sample_rate = s->sample_rate;
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st->codecpar->channels = s->channels;
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st->codecpar->frame_size = s->frame_size;
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avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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/* microseconds instead of seconds, MHz instead of Hz */
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s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
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s->period_size, 1.5E-6);
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if (!s->timefilter)
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goto fail;
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return 0;
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fail:
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snd_pcm_close(s->h);
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return AVERROR(EIO);
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}
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static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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AlsaData *s = s1->priv_data;
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int res;
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int64_t dts;
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snd_pcm_sframes_t delay = 0;
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if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
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return AVERROR(EIO);
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}
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while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
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if (res == -EAGAIN) {
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av_packet_unref(pkt);
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return AVERROR(EAGAIN);
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}
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if (ff_alsa_xrun_recover(s1, res) < 0) {
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av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
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snd_strerror(res));
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av_packet_unref(pkt);
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return AVERROR(EIO);
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}
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ff_timefilter_reset(s->timefilter);
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}
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dts = av_gettime();
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snd_pcm_delay(s->h, &delay);
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dts -= av_rescale(delay + res, 1000000, s->sample_rate);
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pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
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s->last_period = res;
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pkt->size = res * s->frame_size;
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return 0;
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}
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static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
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{
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return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
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}
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static const AVOption options[] = {
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{ "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ NULL },
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};
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static const AVClass alsa_demuxer_class = {
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.class_name = "ALSA demuxer",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
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};
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AVInputFormat ff_alsa_demuxer = {
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.name = "alsa",
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.long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
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.priv_data_size = sizeof(AlsaData),
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.read_header = audio_read_header,
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.read_packet = audio_read_packet,
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.read_close = ff_alsa_close,
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.get_device_list = audio_get_device_list,
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.flags = AVFMT_NOFILE,
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.priv_class = &alsa_demuxer_class,
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};
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