libavdevice/alsa-audio-enc.c
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00001 /*
00002  * ALSA input and output
00003  * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
00004  * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
00005  *
00006  * This file is part of FFmpeg.
00007  *
00008  * FFmpeg is free software; you can redistribute it and/or
00009  * modify it under the terms of the GNU Lesser General Public
00010  * License as published by the Free Software Foundation; either
00011  * version 2.1 of the License, or (at your option) any later version.
00012  *
00013  * FFmpeg is distributed in the hope that it will be useful,
00014  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00015  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00016  * Lesser General Public License for more details.
00017  *
00018  * You should have received a copy of the GNU Lesser General Public
00019  * License along with FFmpeg; if not, write to the Free Software
00020  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00021  */
00022 
00040 #include <alsa/asoundlib.h>
00041 
00042 #include "libavformat/internal.h"
00043 #include "avdevice.h"
00044 #include "alsa-audio.h"
00045 
00046 static av_cold int audio_write_header(AVFormatContext *s1)
00047 {
00048     AlsaData *s = s1->priv_data;
00049     AVStream *st;
00050     unsigned int sample_rate;
00051     enum CodecID codec_id;
00052     int res;
00053 
00054     st = s1->streams[0];
00055     sample_rate = st->codec->sample_rate;
00056     codec_id    = st->codec->codec_id;
00057     res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
00058         st->codec->channels, &codec_id);
00059     if (sample_rate != st->codec->sample_rate) {
00060         av_log(s1, AV_LOG_ERROR,
00061                "sample rate %d not available, nearest is %d\n",
00062                st->codec->sample_rate, sample_rate);
00063         goto fail;
00064     }
00065     avpriv_set_pts_info(st, 64, 1, sample_rate);
00066 
00067     return res;
00068 
00069 fail:
00070     snd_pcm_close(s->h);
00071     return AVERROR(EIO);
00072 }
00073 
00074 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
00075 {
00076     AlsaData *s = s1->priv_data;
00077     int res;
00078     int size     = pkt->size;
00079     uint8_t *buf = pkt->data;
00080 
00081     size /= s->frame_size;
00082     if (s->reorder_func) {
00083         if (size > s->reorder_buf_size)
00084             if (ff_alsa_extend_reorder_buf(s, size))
00085                 return AVERROR(ENOMEM);
00086         s->reorder_func(buf, s->reorder_buf, size);
00087         buf = s->reorder_buf;
00088     }
00089     while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {
00090         if (res == -EAGAIN) {
00091 
00092             return AVERROR(EAGAIN);
00093         }
00094 
00095         if (ff_alsa_xrun_recover(s1, res) < 0) {
00096             av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
00097                    snd_strerror(res));
00098 
00099             return AVERROR(EIO);
00100         }
00101     }
00102 
00103     return 0;
00104 }
00105 
00106 static void
00107 audio_get_output_timestamp(AVFormatContext *s1, int stream,
00108     int64_t *dts, int64_t *wall)
00109 {
00110     AlsaData *s  = s1->priv_data;
00111     snd_pcm_sframes_t delay = 0;
00112     *wall = av_gettime();
00113     snd_pcm_delay(s->h, &delay);
00114     *dts = s1->streams[0]->cur_dts - delay;
00115 }
00116 
00117 AVOutputFormat ff_alsa_muxer = {
00118     .name           = "alsa",
00119     .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio output"),
00120     .priv_data_size = sizeof(AlsaData),
00121     .audio_codec    = DEFAULT_CODEC_ID,
00122     .video_codec    = CODEC_ID_NONE,
00123     .write_header   = audio_write_header,
00124     .write_packet   = audio_write_packet,
00125     .write_trailer  = ff_alsa_close,
00126     .get_output_timestamp = audio_get_output_timestamp,
00127     .flags          = AVFMT_NOFILE,
00128 };