| /* |
| * Audio Interleaving functions |
| * |
| * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com> |
| * |
| * 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 |
| */ |
| |
| #include "libavutil/fifo.h" |
| #include "libavutil/mathematics.h" |
| #include "avformat.h" |
| #include "audiointerleave.h" |
| #include "internal.h" |
| |
| void ff_audio_interleave_close(AVFormatContext *s) |
| { |
| int i; |
| for (i = 0; i < s->nb_streams; i++) { |
| AVStream *st = s->streams[i]; |
| AudioInterleaveContext *aic = st->priv_data; |
| |
| if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) |
| av_fifo_freep(&aic->fifo); |
| } |
| } |
| |
| int ff_audio_interleave_init(AVFormatContext *s, |
| const int *samples_per_frame, |
| AVRational time_base) |
| { |
| int i; |
| |
| if (!samples_per_frame) |
| return AVERROR(EINVAL); |
| |
| if (!time_base.num) { |
| av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n"); |
| return AVERROR(EINVAL); |
| } |
| for (i = 0; i < s->nb_streams; i++) { |
| AVStream *st = s->streams[i]; |
| AudioInterleaveContext *aic = st->priv_data; |
| |
| if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { |
| aic->sample_size = (st->codecpar->channels * |
| av_get_bits_per_sample(st->codecpar->codec_id)) / 8; |
| if (!aic->sample_size) { |
| av_log(s, AV_LOG_ERROR, "could not compute sample size\n"); |
| return AVERROR(EINVAL); |
| } |
| aic->samples_per_frame = samples_per_frame; |
| aic->samples = aic->samples_per_frame; |
| aic->time_base = time_base; |
| |
| aic->fifo_size = 100* *aic->samples; |
| if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples))) |
| return AVERROR(ENOMEM); |
| } |
| } |
| |
| return 0; |
| } |
| |
| static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, |
| int stream_index, int flush) |
| { |
| AVStream *st = s->streams[stream_index]; |
| AudioInterleaveContext *aic = st->priv_data; |
| int ret; |
| int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size); |
| if (!size || (!flush && size == av_fifo_size(aic->fifo))) |
| return 0; |
| |
| ret = av_new_packet(pkt, size); |
| if (ret < 0) |
| return ret; |
| av_fifo_generic_read(aic->fifo, pkt->data, size, NULL); |
| |
| pkt->dts = pkt->pts = aic->dts; |
| pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base); |
| pkt->stream_index = stream_index; |
| aic->dts += pkt->duration; |
| |
| aic->samples++; |
| if (!*aic->samples) |
| aic->samples = aic->samples_per_frame; |
| |
| return size; |
| } |
| |
| int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, |
| int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), |
| int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *)) |
| { |
| int i, ret; |
| |
| if (pkt) { |
| AVStream *st = s->streams[pkt->stream_index]; |
| AudioInterleaveContext *aic = st->priv_data; |
| if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { |
| unsigned new_size = av_fifo_size(aic->fifo) + pkt->size; |
| if (new_size > aic->fifo_size) { |
| if (av_fifo_realloc2(aic->fifo, new_size) < 0) |
| return AVERROR(ENOMEM); |
| aic->fifo_size = new_size; |
| } |
| av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL); |
| } else { |
| // rewrite pts and dts to be decoded time line position |
| pkt->pts = pkt->dts = aic->dts; |
| aic->dts += pkt->duration; |
| if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0) |
| return ret; |
| } |
| pkt = NULL; |
| } |
| |
| for (i = 0; i < s->nb_streams; i++) { |
| AVStream *st = s->streams[i]; |
| if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { |
| AVPacket new_pkt = { 0 }; |
| while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) { |
| if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0) |
| return ret; |
| } |
| if (ret < 0) |
| return ret; |
| } |
| } |
| |
| return get_packet(s, out, NULL, flush); |
| } |