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libavformat/audiointerleave.c

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00001 /*
00002  * Audio Interleaving functions
00003  *
00004  * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
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 
00023 #include "libavutil/fifo.h"
00024 #include "avformat.h"
00025 #include "audiointerleave.h"
00026 #include "internal.h"
00027 
00028 void ff_audio_interleave_close(AVFormatContext *s)
00029 {
00030     int i;
00031     for (i = 0; i < s->nb_streams; i++) {
00032         AVStream *st = s->streams[i];
00033         AudioInterleaveContext *aic = st->priv_data;
00034 
00035         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
00036             av_fifo_free(aic->fifo);
00037     }
00038 }
00039 
00040 int ff_audio_interleave_init(AVFormatContext *s,
00041                              const int *samples_per_frame,
00042                              AVRational time_base)
00043 {
00044     int i;
00045 
00046     if (!samples_per_frame)
00047         return -1;
00048 
00049     for (i = 0; i < s->nb_streams; i++) {
00050         AVStream *st = s->streams[i];
00051         AudioInterleaveContext *aic = st->priv_data;
00052 
00053         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
00054             aic->sample_size = (st->codec->channels *
00055                                 av_get_bits_per_sample(st->codec->codec_id)) / 8;
00056             if (!aic->sample_size) {
00057                 av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
00058                 return -1;
00059             }
00060             aic->samples_per_frame = samples_per_frame;
00061             aic->samples = aic->samples_per_frame;
00062             aic->time_base = time_base;
00063 
00064             aic->fifo_size = 100* *aic->samples;
00065             aic->fifo= av_fifo_alloc(100 * *aic->samples);
00066         }
00067     }
00068 
00069     return 0;
00070 }
00071 
00072 static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
00073                                    int stream_index, int flush)
00074 {
00075     AVStream *st = s->streams[stream_index];
00076     AudioInterleaveContext *aic = st->priv_data;
00077 
00078     int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
00079     if (!size || (!flush && size == av_fifo_size(aic->fifo)))
00080         return 0;
00081 
00082     av_new_packet(pkt, size);
00083     av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
00084 
00085     pkt->dts = pkt->pts = aic->dts;
00086     pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
00087     pkt->stream_index = stream_index;
00088     aic->dts += pkt->duration;
00089 
00090     aic->samples++;
00091     if (!*aic->samples)
00092         aic->samples = aic->samples_per_frame;
00093 
00094     return size;
00095 }
00096 
00097 int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
00098                         int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
00099                         int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
00100 {
00101     int i;
00102 
00103     if (pkt) {
00104         AVStream *st = s->streams[pkt->stream_index];
00105         AudioInterleaveContext *aic = st->priv_data;
00106         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
00107             unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
00108             if (new_size > aic->fifo_size) {
00109                 if (av_fifo_realloc2(aic->fifo, new_size) < 0)
00110                     return -1;
00111                 aic->fifo_size = new_size;
00112             }
00113             av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
00114         } else {
00115             // rewrite pts and dts to be decoded time line position
00116             pkt->pts = pkt->dts = aic->dts;
00117             aic->dts += pkt->duration;
00118             ff_interleave_add_packet(s, pkt, compare_ts);
00119         }
00120         pkt = NULL;
00121     }
00122 
00123     for (i = 0; i < s->nb_streams; i++) {
00124         AVStream *st = s->streams[i];
00125         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
00126             AVPacket new_pkt;
00127             while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
00128                 ff_interleave_add_packet(s, &new_pkt, compare_ts);
00129         }
00130     }
00131 
00132     return get_packet(s, out, pkt, flush);
00133 }

Generated on Fri Sep 16 2011 17:17:47 for FFmpeg by  doxygen 1.7.1