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ffmpeg / libavdevice / alsa-audio-dec.c @ 2912e87a

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/*
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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 Libav.
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 *
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 * Libav 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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 * Libav 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 Libav; 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 <alsa/asoundlib.h>
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#include "libavformat/avformat.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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                                     AVFormatParameters *ap)
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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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    unsigned int sample_rate;
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    enum CodecID codec_id;
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    snd_pcm_sw_params_t *sw_params;
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    if (ap->sample_rate <= 0) {
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        av_log(s1, AV_LOG_ERROR, "Bad sample rate %d\n", ap->sample_rate);
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        return AVERROR(EIO);
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    }
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    if (ap->channels <= 0) {
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        av_log(s1, AV_LOG_ERROR, "Bad channels number %d\n", ap->channels);
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        return AVERROR(EIO);
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    }
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    st = av_new_stream(s1, 0);
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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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    sample_rate = ap->sample_rate;
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    codec_id    = s1->audio_codec_id;
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    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &sample_rate, ap->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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    if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
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        av_log(s1, AV_LOG_WARNING,
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               "capture with some ALSA plugins, especially dsnoop, "
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               "may hang.\n");
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    ret = snd_pcm_sw_params_malloc(&sw_params);
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    if (ret < 0) {
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        av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
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               snd_strerror(ret));
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        goto fail;
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    }
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    snd_pcm_sw_params_current(s->h, sw_params);
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    snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
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    ret = snd_pcm_sw_params(s->h, sw_params);
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    snd_pcm_sw_params_free(sw_params);
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    if (ret < 0) {
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        av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
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               snd_strerror(ret));
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        goto fail;
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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 = sample_rate;
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    st->codec->channels    = ap->channels;
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    av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
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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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    AVStream *st = s1->streams[0];
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    int res;
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    snd_htimestamp_t timestamp;
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    snd_pcm_uframes_t ts_delay;
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    if (av_new_packet(pkt, s->period_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, pkt->size / s->frame_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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    }
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    snd_pcm_htimestamp(s->h, &ts_delay, &timestamp);
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    ts_delay += res;
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    pkt->pts = timestamp.tv_sec * 1000000LL
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               + (timestamp.tv_nsec * st->codec->sample_rate
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                  - ts_delay * 1000000000LL + st->codec->sample_rate * 500LL)
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               / (st->codec->sample_rate * 1000LL);
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    pkt->size = res * s->frame_size;
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    return 0;
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}
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AVInputFormat ff_alsa_demuxer = {
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    "alsa",
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    NULL_IF_CONFIG_SMALL("ALSA audio input"),
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    sizeof(AlsaData),
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    NULL,
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    audio_read_header,
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    audio_read_packet,
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    ff_alsa_close,
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    .flags = AVFMT_NOFILE,
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};