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https://github.com/FFmpeg/FFmpeg.git
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36583d23bd
Also update libav->ffmpeg as theres pretty much no code left from libav. The new code is faster, requires fewer mallocs and less memory. Its also half the number of lines of code. This code is not 100% identical in behavior to the previous, but the differences appear to be rather limitations of the previous design than intended though i could be wrong of course. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
111 lines
3.7 KiB
C
111 lines
3.7 KiB
C
/*
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* Audio Frame Queue
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* Copyright (c) 2012 Justin Ruggles
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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 "audio_frame_queue.h"
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#include "internal.h"
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#include "libavutil/avassert.h"
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void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
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{
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afq->avctx = avctx;
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afq->remaining_delay = avctx->delay;
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afq->remaining_samples = avctx->delay;
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afq->frame_count = 0;
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}
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void ff_af_queue_close(AudioFrameQueue *afq)
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{
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if(afq->frame_count)
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av_log(afq->avctx, AV_LOG_WARNING, "%d frames left in que on closing\n", afq->frame_count);
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av_freep(&afq->frames);
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memset(afq, 0, sizeof(*afq));
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}
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int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
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{
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AudioFrame *new = av_fast_realloc(afq->frames, &afq->frame_alloc, sizeof(*afq->frames)*(afq->frame_count+1));
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if(!new)
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return AVERROR(ENOMEM);
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afq->frames = new;
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new += afq->frame_count;
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/* get frame parameters */
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new->duration = f->nb_samples;
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new->duration += afq->remaining_delay;
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if (f->pts != AV_NOPTS_VALUE) {
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new->pts = av_rescale_q(f->pts,
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afq->avctx->time_base,
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(AVRational){ 1, afq->avctx->sample_rate });
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new->pts -= afq->remaining_delay;
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if(afq->frame_count && new[-1].pts >= new->pts)
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av_log(afq->avctx, AV_LOG_WARNING, "Que input is backward in time\n");
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} else {
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new->pts = AV_NOPTS_VALUE;
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}
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afq->remaining_delay = 0;
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/* add frame sample count */
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afq->remaining_samples += f->nb_samples;
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afq->frame_count++;
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return 0;
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}
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void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts,
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int *duration)
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{
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int64_t out_pts = AV_NOPTS_VALUE;
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int removed_samples = 0;
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int i;
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if (afq->frame_count || afq->frame_alloc) {
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if (afq->frames->pts != AV_NOPTS_VALUE)
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out_pts = afq->frames->pts;
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}
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if(!afq->frame_count)
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av_log(afq->avctx, AV_LOG_WARNING, "Trying to remove %d samples, but que empty\n", nb_samples);
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if (pts)
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*pts = ff_samples_to_time_base(afq->avctx, out_pts);
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for(i=0; nb_samples && i<afq->frame_count; i++){
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int n= FFMIN(afq->frames[i].duration, nb_samples);
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afq->frames[i].duration -= n;
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nb_samples -= n;
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removed_samples += n;
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if(afq->frames[i].pts != AV_NOPTS_VALUE)
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afq->frames[i].pts += n;
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}
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i -= i && afq->frames[i-1].duration;
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memmove(afq->frames, afq->frames + i, sizeof(*afq->frames) * (afq->frame_count - i));
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afq->frame_count -= i;
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if(nb_samples){
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av_assert0(!afq->frame_count);
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if(afq->frames[0].pts != AV_NOPTS_VALUE)
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afq->frames[0].pts += nb_samples;
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av_log(afq->avctx, AV_LOG_DEBUG, "Trying to remove %d more samples than are in the que\n", nb_samples);
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}
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if (duration)
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*duration = ff_samples_to_time_base(afq->avctx, removed_samples);
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}
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