yading@11: /*
yading@11:  * Audio Interleaving functions
yading@11:  *
yading@11:  * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
yading@11:  *
yading@11:  * This file is part of FFmpeg.
yading@11:  *
yading@11:  * FFmpeg is free software; you can redistribute it and/or
yading@11:  * modify it under the terms of the GNU Lesser General Public
yading@11:  * License as published by the Free Software Foundation; either
yading@11:  * version 2.1 of the License, or (at your option) any later version.
yading@11:  *
yading@11:  * FFmpeg is distributed in the hope that it will be useful,
yading@11:  * but WITHOUT ANY WARRANTY; without even the implied warranty of
yading@11:  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
yading@11:  * Lesser General Public License for more details.
yading@11:  *
yading@11:  * You should have received a copy of the GNU Lesser General Public
yading@11:  * License along with FFmpeg; if not, write to the Free Software
yading@11:  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
yading@11:  */
yading@11: 
yading@11: #include "libavutil/fifo.h"
yading@11: #include "libavutil/mathematics.h"
yading@11: #include "avformat.h"
yading@11: #include "audiointerleave.h"
yading@11: #include "internal.h"
yading@11: 
yading@11: void ff_audio_interleave_close(AVFormatContext *s)
yading@11: {
yading@11:     int i;
yading@11:     for (i = 0; i < s->nb_streams; i++) {
yading@11:         AVStream *st = s->streams[i];
yading@11:         AudioInterleaveContext *aic = st->priv_data;
yading@11: 
yading@11:         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
yading@11:             av_fifo_free(aic->fifo);
yading@11:     }
yading@11: }
yading@11: 
yading@11: int ff_audio_interleave_init(AVFormatContext *s,
yading@11:                              const int *samples_per_frame,
yading@11:                              AVRational time_base)
yading@11: {
yading@11:     int i;
yading@11: 
yading@11:     if (!samples_per_frame)
yading@11:         return -1;
yading@11: 
yading@11:     if (!time_base.num) {
yading@11:         av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
yading@11:         return -1;
yading@11:     }
yading@11:     for (i = 0; i < s->nb_streams; i++) {
yading@11:         AVStream *st = s->streams[i];
yading@11:         AudioInterleaveContext *aic = st->priv_data;
yading@11: 
yading@11:         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
yading@11:             aic->sample_size = (st->codec->channels *
yading@11:                                 av_get_bits_per_sample(st->codec->codec_id)) / 8;
yading@11:             if (!aic->sample_size) {
yading@11:                 av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
yading@11:                 return -1;
yading@11:             }
yading@11:             aic->samples_per_frame = samples_per_frame;
yading@11:             aic->samples = aic->samples_per_frame;
yading@11:             aic->time_base = time_base;
yading@11: 
yading@11:             aic->fifo_size = 100* *aic->samples;
yading@11:             aic->fifo= av_fifo_alloc(100 * *aic->samples);
yading@11:         }
yading@11:     }
yading@11: 
yading@11:     return 0;
yading@11: }
yading@11: 
yading@11: static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
yading@11:                                    int stream_index, int flush)
yading@11: {
yading@11:     AVStream *st = s->streams[stream_index];
yading@11:     AudioInterleaveContext *aic = st->priv_data;
yading@11: 
yading@11:     int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
yading@11:     if (!size || (!flush && size == av_fifo_size(aic->fifo)))
yading@11:         return 0;
yading@11: 
yading@11:     if (av_new_packet(pkt, size) < 0)
yading@11:         return AVERROR(ENOMEM);
yading@11:     av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
yading@11: 
yading@11:     pkt->dts = pkt->pts = aic->dts;
yading@11:     pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
yading@11:     pkt->stream_index = stream_index;
yading@11:     aic->dts += pkt->duration;
yading@11: 
yading@11:     aic->samples++;
yading@11:     if (!*aic->samples)
yading@11:         aic->samples = aic->samples_per_frame;
yading@11: 
yading@11:     return size;
yading@11: }
yading@11: 
yading@11: int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
yading@11:                         int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
yading@11:                         int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
yading@11: {
yading@11:     int i;
yading@11: 
yading@11:     if (pkt) {
yading@11:         AVStream *st = s->streams[pkt->stream_index];
yading@11:         AudioInterleaveContext *aic = st->priv_data;
yading@11:         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
yading@11:             unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
yading@11:             if (new_size > aic->fifo_size) {
yading@11:                 if (av_fifo_realloc2(aic->fifo, new_size) < 0)
yading@11:                     return -1;
yading@11:                 aic->fifo_size = new_size;
yading@11:             }
yading@11:             av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
yading@11:         } else {
yading@11:             int ret;
yading@11:             // rewrite pts and dts to be decoded time line position
yading@11:             pkt->pts = pkt->dts = aic->dts;
yading@11:             aic->dts += pkt->duration;
yading@11:             ret = ff_interleave_add_packet(s, pkt, compare_ts);
yading@11:             if (ret < 0)
yading@11:                 return ret;
yading@11:         }
yading@11:         pkt = NULL;
yading@11:     }
yading@11: 
yading@11:     for (i = 0; i < s->nb_streams; i++) {
yading@11:         AVStream *st = s->streams[i];
yading@11:         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
yading@11:             AVPacket new_pkt;
yading@11:             int ret;
yading@11:             while ((ret = ff_interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
yading@11:                 ret = ff_interleave_add_packet(s, &new_pkt, compare_ts);
yading@11:                 if (ret < 0)
yading@11:                     return ret;
yading@11:             }
yading@11:             if (ret < 0)
yading@11:                 return ret;
yading@11:         }
yading@11:     }
yading@11: 
yading@11:     return get_packet(s, out, NULL, flush);
yading@11: }