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ffmpeg / libavdevice / alsa-audio-enc.c @ 813dbb44

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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 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: output
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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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 *
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 * This avdevice encoder allows to play audio to an ALSA (Advanced Linux
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 * 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 playback period is set to the lower value available for the device,
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 * which gives a low latency suitable for real-time playback.
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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_write_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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    unsigned int sample_rate;
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    enum CodecID codec_id;
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    int res;
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    st = s1->streams[0];
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    sample_rate = st->codec->sample_rate;
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    codec_id    = st->codec->codec_id;
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    res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
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        st->codec->channels, &codec_id);
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    if (sample_rate != st->codec->sample_rate) {
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        av_log(s1, AV_LOG_ERROR,
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               "sample rate %d not available, nearest is %d\n",
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               st->codec->sample_rate, sample_rate);
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        goto fail;
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    }
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    return res;
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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_write_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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    int size     = pkt->size;
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    uint8_t *buf = pkt->data;
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    size /= s->frame_size;
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    if (s->reorder_func) {
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        if (size > s->reorder_buf_size)
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            if (ff_alsa_extend_reorder_buf(s, size))
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                return AVERROR(ENOMEM);
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        s->reorder_func(buf, s->reorder_buf, size);
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        buf = s->reorder_buf;
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    }
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    while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {
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        if (res == -EAGAIN) {
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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 write error: %s\n",
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                   snd_strerror(res));
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            return AVERROR(EIO);
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        }
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    }
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    return 0;
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}
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AVOutputFormat ff_alsa_muxer = {
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    "alsa",
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    NULL_IF_CONFIG_SMALL("ALSA audio output"),
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    "",
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    "",
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    sizeof(AlsaData),
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    DEFAULT_CODEC_ID,
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    CODEC_ID_NONE,
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    audio_write_header,
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    audio_write_packet,
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    ff_alsa_close,
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    .flags = AVFMT_NOFILE,
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};