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4349 Serge 1 
/*
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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 allows to 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
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#include "libavformat/internal.h"
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#include "libavutil/opt.h"
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#include "libavutil/mathematics.h"
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#include "libavutil/time.h"
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#include "avdevice.h"
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#include "alsa-audio.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->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
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    st->codec->codec_id    = codec_id;
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    st->codec->sample_rate = s->sample_rate;
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    st->codec->channels    = s->channels;
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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_free_packet(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_free_packet(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 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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};
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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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    .flags          = AVFMT_NOFILE,
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    .priv_class     = &alsa_demuxer_class,
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};