mirror of https://git.ffmpeg.org/ffmpeg.git
avcodec: add code for a frame queue for use by audio encoders with delay
This simplifies matching of timestamps between input frames and output packets.
This commit is contained in:
Justin Ruggles
parent
c9594fe0fb
commit
4bf64961a9
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/*
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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 Libav.
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*
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* Libav 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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* Libav 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 Libav; 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 "internal.h"
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#include "audio_frame_queue.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->next_pts = AV_NOPTS_VALUE;
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afq->remaining_delay = avctx->delay;
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afq->remaining_samples = avctx->delay;
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afq->frame_queue = NULL;
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}
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static void delete_next_frame(AudioFrameQueue *afq)
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{
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AudioFrame *f = afq->frame_queue;
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if (f) {
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afq->frame_queue = f->next;
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f->next = NULL;
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av_freep(&f);
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}
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}
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void ff_af_queue_close(AudioFrameQueue *afq)
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{
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/* remove/free any remaining frames */
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while (afq->frame_queue)
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delete_next_frame(afq);
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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_frame;
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AudioFrame *queue_end = afq->frame_queue;
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/* find the end of the queue */
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while (queue_end && queue_end->next)
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queue_end = queue_end->next;
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/* allocate new frame queue entry */
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if (!(new_frame = av_malloc(sizeof(*new_frame))))
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return AVERROR(ENOMEM);
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/* get frame parameters */
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new_frame->next = NULL;
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new_frame->duration = f->nb_samples;
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if (f->pts != AV_NOPTS_VALUE) {
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new_frame->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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afq->next_pts = new_frame->pts + new_frame->duration;
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} else {
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new_frame->pts = AV_NOPTS_VALUE;
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afq->next_pts = AV_NOPTS_VALUE;
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}
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/* add new frame to the end of the queue */
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if (!queue_end)
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afq->frame_queue = new_frame;
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else
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queue_end->next = new_frame;
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/* add frame sample count */
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afq->remaining_samples += f->nb_samples;
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#ifdef DEBUG
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ff_af_queue_log_state(afq);
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#endif
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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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#ifdef DEBUG
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ff_af_queue_log_state(afq);
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#endif
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/* get output pts from the next frame or generated pts */
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if (afq->frame_queue) {
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if (afq->frame_queue->pts != AV_NOPTS_VALUE)
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out_pts = afq->frame_queue->pts - afq->remaining_delay;
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} else {
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if (afq->next_pts != AV_NOPTS_VALUE)
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out_pts = afq->next_pts - afq->remaining_delay;
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}
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if (pts) {
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if (out_pts != AV_NOPTS_VALUE)
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*pts = ff_samples_to_time_base(afq->avctx, out_pts);
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else
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*pts = AV_NOPTS_VALUE;
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}
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/* if the delay is larger than the packet duration, we use up delay samples
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for the output packet and leave all frames in the queue */
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if (afq->remaining_delay >= nb_samples) {
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removed_samples += nb_samples;
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afq->remaining_delay -= nb_samples;
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}
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/* remove frames from the queue until we have enough to cover the
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requested number of samples or until the queue is empty */
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while (removed_samples < nb_samples && afq->frame_queue) {
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removed_samples += afq->frame_queue->duration;
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delete_next_frame(afq);
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}
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afq->remaining_samples -= removed_samples;
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/* if there are no frames left and we have room for more samples, use
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any remaining delay samples */
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if (removed_samples < nb_samples && afq->remaining_samples > 0) {
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int add_samples = FFMIN(afq->remaining_samples,
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nb_samples - removed_samples);
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removed_samples += add_samples;
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afq->remaining_samples -= add_samples;
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}
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if (removed_samples > nb_samples)
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av_log(afq->avctx, AV_LOG_WARNING, "frame_size is too large\n");
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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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void ff_af_queue_log_state(AudioFrameQueue *afq)
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{
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AudioFrame *f;
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av_log(afq->avctx, AV_LOG_DEBUG, "remaining delay = %d\n",
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afq->remaining_delay);
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av_log(afq->avctx, AV_LOG_DEBUG, "remaining samples = %d\n",
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afq->remaining_samples);
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av_log(afq->avctx, AV_LOG_DEBUG, "frames:\n");
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f = afq->frame_queue;
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while (f) {
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av_log(afq->avctx, AV_LOG_DEBUG, " [ pts=%9"PRId64" duration=%d ]\n",
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f->pts, f->duration);
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f = f->next;
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}
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}
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@ -0,0 +1,90 @@
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/*
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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 Libav.
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*
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* Libav 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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* Libav 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 Libav; 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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#ifndef AVCODEC_AUDIO_FRAME_QUEUE_H
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#define AVCODEC_AUDIO_FRAME_QUEUE_H
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#include "avcodec.h"
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typedef struct AudioFrame {
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int64_t pts;
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int duration;
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struct AudioFrame *next;
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} AudioFrame;
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typedef struct AudioFrameQueue {
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AVCodecContext *avctx;
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int64_t next_pts;
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int remaining_delay;
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int remaining_samples;
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AudioFrame *frame_queue;
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} AudioFrameQueue;
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/**
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* Initialize AudioFrameQueue.
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*
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* @param avctx context to use for time_base and av_log
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* @param afq queue context
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*/
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void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq);
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/**
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* Close AudioFrameQueue.
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*
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* Frees memory if needed.
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*
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* @param afq queue context
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*/
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void ff_af_queue_close(AudioFrameQueue *afq);
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/**
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* Add a frame to the queue.
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*
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* @param afq queue context
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* @param f frame to add to the queue
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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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* Remove frame(s) from the queue.
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*
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* Retrieves the pts of the next available frame, or a generated pts based on
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* the last frame duration if there are no frames left in the queue. The number
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* of requested samples should be the full number of samples represented by the
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* packet that will be output by the encoder. If fewer samples are available
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* in the queue, a smaller value will be used for the output duration.
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*
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* @param afq queue context
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* @param nb_samples number of samples to remove from the queue
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* @param[out] pts output packet pts
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* @param[out] duration output packet duration
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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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* Log the current state of the queue.
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*
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* @param afq queue context
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*/
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void ff_af_queue_log_state(AudioFrameQueue *afq);
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#endif /* AVCODEC_AUDIO_FRAME_QUEUE_H */
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