diff options
| author | Tim Redfern <tim@eclectronics.org> | 2013-08-26 15:10:18 +0100 |
|---|---|---|
| committer | Tim Redfern <tim@eclectronics.org> | 2013-08-26 15:10:18 +0100 |
| commit | 150c9823e71a161e97003849cf8b2f55b21520bd (patch) | |
| tree | 3559c840cf403d1386708b2591d58f928c7b160d /ffmpeg1/libavdevice/alsa-audio-enc.c | |
| parent | b4b1e2630c95d5e6014463f7608d59dc2322a3b8 (diff) | |
adding ffmpeg specific version
Diffstat (limited to 'ffmpeg1/libavdevice/alsa-audio-enc.c')
| -rw-r--r-- | ffmpeg1/libavdevice/alsa-audio-enc.c | 129 |
1 files changed, 129 insertions, 0 deletions
diff --git a/ffmpeg1/libavdevice/alsa-audio-enc.c b/ffmpeg1/libavdevice/alsa-audio-enc.c new file mode 100644 index 0000000..0f4e4a2 --- /dev/null +++ b/ffmpeg1/libavdevice/alsa-audio-enc.c @@ -0,0 +1,129 @@ +/* + * ALSA input and output + * Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) + * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * ALSA input and output: output + * @author Luca Abeni ( lucabe72 email it ) + * @author Benoit Fouet ( benoit fouet free fr ) + * + * This avdevice encoder allows to play audio to an ALSA (Advanced Linux + * Sound Architecture) device. + * + * The filename parameter is the name of an ALSA PCM device capable of + * capture, for example "default" or "plughw:1"; see the ALSA documentation + * for naming conventions. The empty string is equivalent to "default". + * + * The playback period is set to the lower value available for the device, + * which gives a low latency suitable for real-time playback. + */ + +#include <alsa/asoundlib.h> + +#include "libavutil/time.h" +#include "libavformat/internal.h" +#include "avdevice.h" +#include "alsa-audio.h" + +static av_cold int audio_write_header(AVFormatContext *s1) +{ + AlsaData *s = s1->priv_data; + AVStream *st; + unsigned int sample_rate; + enum AVCodecID codec_id; + int res; + + st = s1->streams[0]; + sample_rate = st->codec->sample_rate; + codec_id = st->codec->codec_id; + res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate, + st->codec->channels, &codec_id); + if (sample_rate != st->codec->sample_rate) { + av_log(s1, AV_LOG_ERROR, + "sample rate %d not available, nearest is %d\n", + st->codec->sample_rate, sample_rate); + goto fail; + } + avpriv_set_pts_info(st, 64, 1, sample_rate); + + return res; + +fail: + snd_pcm_close(s->h); + return AVERROR(EIO); +} + +static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) +{ + AlsaData *s = s1->priv_data; + int res; + int size = pkt->size; + uint8_t *buf = pkt->data; + + size /= s->frame_size; + if (s->reorder_func) { + if (size > s->reorder_buf_size) + if (ff_alsa_extend_reorder_buf(s, size)) + return AVERROR(ENOMEM); + s->reorder_func(buf, s->reorder_buf, size); + buf = s->reorder_buf; + } + while ((res = snd_pcm_writei(s->h, buf, size)) < 0) { + if (res == -EAGAIN) { + + return AVERROR(EAGAIN); + } + + if (ff_alsa_xrun_recover(s1, res) < 0) { + av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n", + snd_strerror(res)); + + return AVERROR(EIO); + } + } + + return 0; +} + +static void +audio_get_output_timestamp(AVFormatContext *s1, int stream, + int64_t *dts, int64_t *wall) +{ + AlsaData *s = s1->priv_data; + snd_pcm_sframes_t delay = 0; + *wall = av_gettime(); + snd_pcm_delay(s->h, &delay); + *dts = s1->streams[0]->cur_dts - delay; +} + +AVOutputFormat ff_alsa_muxer = { + .name = "alsa", + .long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"), + .priv_data_size = sizeof(AlsaData), + .audio_codec = DEFAULT_CODEC_ID, + .video_codec = AV_CODEC_ID_NONE, + .write_header = audio_write_header, + .write_packet = audio_write_packet, + .write_trailer = ff_alsa_close, + .get_output_timestamp = audio_get_output_timestamp, + .flags = AVFMT_NOFILE, +}; |
