mirror of
https://gitee.com/openharmony/third_party_ffmpeg
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130 lines
3.9 KiB
C
130 lines
3.9 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: 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 "libavutil/time.h"
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#include "libavformat/internal.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_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 AVCodecID 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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avpriv_set_pts_info(st, 64, 1, sample_rate);
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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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static void
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audio_get_output_timestamp(AVFormatContext *s1, int stream,
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int64_t *dts, int64_t *wall)
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{
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AlsaData *s = s1->priv_data;
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snd_pcm_sframes_t delay = 0;
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*wall = av_gettime();
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snd_pcm_delay(s->h, &delay);
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*dts = s1->streams[0]->cur_dts - delay;
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}
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AVOutputFormat ff_alsa_muxer = {
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.name = "alsa",
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.long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"),
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.priv_data_size = sizeof(AlsaData),
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.audio_codec = DEFAULT_CODEC_ID,
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.video_codec = AV_CODEC_ID_NONE,
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.write_header = audio_write_header,
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.write_packet = audio_write_packet,
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.write_trailer = ff_alsa_close,
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.get_output_timestamp = audio_get_output_timestamp,
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.flags = AVFMT_NOFILE,
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};
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