mirror of https://github.com/FFmpeg/FFmpeg.git
And rename it to retimeinterleave, use the pcm_rechunk bitstream filter for rechunking. By seperating the two functions we hopefully get cleaner code. Signed-off-by: Marton Balint <cus@passwd.hu>pull/338/head
Marton Balint
5 years ago
parent
2035620b7c
commit
c5324d92c5
7 changed files with 87 additions and 188 deletions
@ -1,148 +0,0 @@ |
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/*
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* Audio Interleaving functions |
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* |
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* Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com> |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#include "libavutil/fifo.h" |
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#include "libavutil/mathematics.h" |
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#include "avformat.h" |
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#include "audiointerleave.h" |
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#include "internal.h" |
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void ff_audio_interleave_close(AVFormatContext *s) |
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{ |
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int i; |
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for (i = 0; i < s->nb_streams; i++) { |
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AVStream *st = s->streams[i]; |
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AudioInterleaveContext *aic = st->priv_data; |
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if (aic && st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) |
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av_fifo_freep(&aic->fifo); |
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} |
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} |
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int ff_audio_interleave_init(AVFormatContext *s, |
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const int samples_per_frame, |
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AVRational time_base) |
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{ |
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int i; |
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if (!time_base.num) { |
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av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n"); |
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return AVERROR(EINVAL); |
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} |
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for (i = 0; i < s->nb_streams; i++) { |
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AVStream *st = s->streams[i]; |
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AudioInterleaveContext *aic = st->priv_data; |
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if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { |
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int max_samples = samples_per_frame ? samples_per_frame : |
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av_rescale_rnd(st->codecpar->sample_rate, time_base.num, time_base.den, AV_ROUND_UP); |
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aic->sample_size = (st->codecpar->channels * |
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av_get_bits_per_sample(st->codecpar->codec_id)) / 8; |
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if (!aic->sample_size) { |
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av_log(s, AV_LOG_ERROR, "could not compute sample size\n"); |
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return AVERROR(EINVAL); |
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} |
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aic->samples_per_frame = samples_per_frame; |
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aic->time_base = time_base; |
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if (!(aic->fifo = av_fifo_alloc_array(100, max_samples))) |
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return AVERROR(ENOMEM); |
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aic->fifo_size = 100 * max_samples; |
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} |
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} |
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return 0; |
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} |
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static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, |
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int stream_index, int flush) |
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{ |
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AVStream *st = s->streams[stream_index]; |
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AudioInterleaveContext *aic = st->priv_data; |
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int ret; |
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int nb_samples = aic->samples_per_frame ? aic->samples_per_frame : |
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(av_rescale_q(aic->n + 1, av_make_q(st->codecpar->sample_rate, 1), av_inv_q(aic->time_base)) - aic->nb_samples); |
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int frame_size = nb_samples * aic->sample_size; |
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int size = FFMIN(av_fifo_size(aic->fifo), frame_size); |
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if (!size || (!flush && size == av_fifo_size(aic->fifo))) |
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return 0; |
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ret = av_new_packet(pkt, frame_size); |
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if (ret < 0) |
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return ret; |
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av_fifo_generic_read(aic->fifo, pkt->data, size, NULL); |
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if (size < pkt->size) |
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memset(pkt->data + size, 0, pkt->size - size); |
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pkt->dts = pkt->pts = aic->dts; |
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pkt->duration = av_rescale_q(nb_samples, st->time_base, aic->time_base); |
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pkt->stream_index = stream_index; |
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aic->dts += pkt->duration; |
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aic->nb_samples += nb_samples; |
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aic->n++; |
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return pkt->size; |
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} |
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int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, |
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int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), |
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int (*compare_ts)(AVFormatContext *, const AVPacket *, const AVPacket *)) |
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{ |
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int i, ret; |
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if (pkt) { |
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AVStream *st = s->streams[pkt->stream_index]; |
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AudioInterleaveContext *aic = st->priv_data; |
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if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { |
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unsigned new_size = av_fifo_size(aic->fifo) + pkt->size; |
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if (new_size > aic->fifo_size) { |
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if (av_fifo_realloc2(aic->fifo, new_size) < 0) |
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return AVERROR(ENOMEM); |
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aic->fifo_size = new_size; |
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} |
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av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL); |
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} else { |
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// rewrite pts and dts to be decoded time line position
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pkt->pts = pkt->dts = aic->dts; |
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aic->dts += pkt->duration; |
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if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0) |
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return ret; |
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} |
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pkt = NULL; |
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} |
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for (i = 0; i < s->nb_streams; i++) { |
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AVStream *st = s->streams[i]; |
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if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { |
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AVPacket new_pkt; |
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while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) { |
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if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0) |
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return ret; |
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} |
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if (ret < 0) |
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return ret; |
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} |
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} |
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return get_packet(s, out, NULL, flush); |
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} |
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@ -0,0 +1,51 @@ |
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/*
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* Retime Interleaving functions |
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* |
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* Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com> |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#include "libavutil/mathematics.h" |
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#include "avformat.h" |
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#include "retimeinterleave.h" |
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#include "internal.h" |
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void ff_retime_interleave_init(RetimeInterleaveContext *aic, AVRational time_base) |
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{ |
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aic->time_base = time_base; |
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} |
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int ff_retime_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, |
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int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), |
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int (*compare_ts)(AVFormatContext *, const AVPacket *, const AVPacket *)) |
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{ |
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int ret; |
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if (pkt) { |
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AVStream *st = s->streams[pkt->stream_index]; |
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RetimeInterleaveContext *aic = st->priv_data; |
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pkt->duration = av_rescale_q(pkt->duration, st->time_base, aic->time_base); |
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// rewrite pts and dts to be decoded time line position
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pkt->pts = pkt->dts = aic->dts; |
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aic->dts += pkt->duration; |
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if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0) |
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return ret; |
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} |
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return get_packet(s, out, NULL, flush); |
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} |
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