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avcodec: add code for a frame queue for use by audio encoders with delay

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This simplifies matching of timestamps between input frames and output
packets.
This commit is contained in:
Justin Ruggles 2012-02-22 21:52:34 -05:00
parent c9594fe0fb
commit 4bf64961a9
2 changed files with 252 additions and 0 deletions
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162
libavcodec/audio_frame_queue.c Normal file
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/*
* Audio Frame Queue
* Copyright (c) 2012 Justin Ruggles
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/mathematics.h"
#include "internal.h"
#include "audio_frame_queue.h"
void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
{
afq->avctx = avctx;
afq->next_pts = AV_NOPTS_VALUE;
afq->remaining_delay = avctx->delay;
afq->remaining_samples = avctx->delay;
afq->frame_queue = NULL;
}
static void delete_next_frame(AudioFrameQueue *afq)
{
AudioFrame *f = afq->frame_queue;
if (f) {
afq->frame_queue = f->next;
f->next = NULL;
av_freep(&f);
}
}
void ff_af_queue_close(AudioFrameQueue *afq)
{
/* remove/free any remaining frames */
while (afq->frame_queue)
delete_next_frame(afq);
memset(afq, 0, sizeof(*afq));
}
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
{
AudioFrame *new_frame;
AudioFrame *queue_end = afq->frame_queue;
/* find the end of the queue */
while (queue_end && queue_end->next)
queue_end = queue_end->next;
/* allocate new frame queue entry */
if (!(new_frame = av_malloc(sizeof(*new_frame))))
return AVERROR(ENOMEM);
/* get frame parameters */
new_frame->next = NULL;
new_frame->duration = f->nb_samples;
if (f->pts != AV_NOPTS_VALUE) {
new_frame->pts = av_rescale_q(f->pts,
afq->avctx->time_base,
(AVRational){ 1, afq->avctx->sample_rate });
afq->next_pts = new_frame->pts + new_frame->duration;
} else {
new_frame->pts = AV_NOPTS_VALUE;
afq->next_pts = AV_NOPTS_VALUE;
}
/* add new frame to the end of the queue */
if (!queue_end)
afq->frame_queue = new_frame;
else
queue_end->next = new_frame;
/* add frame sample count */
afq->remaining_samples += f->nb_samples;
#ifdef DEBUG
ff_af_queue_log_state(afq);
#endif
return 0;
}
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts,
int *duration)
{
int64_t out_pts = AV_NOPTS_VALUE;
int removed_samples = 0;
#ifdef DEBUG
ff_af_queue_log_state(afq);
#endif
/* get output pts from the next frame or generated pts */
if (afq->frame_queue) {
if (afq->frame_queue->pts != AV_NOPTS_VALUE)
out_pts = afq->frame_queue->pts - afq->remaining_delay;
} else {
if (afq->next_pts != AV_NOPTS_VALUE)
out_pts = afq->next_pts - afq->remaining_delay;
}
if (pts) {
if (out_pts != AV_NOPTS_VALUE)
*pts = ff_samples_to_time_base(afq->avctx, out_pts);
else
*pts = AV_NOPTS_VALUE;
}
/* if the delay is larger than the packet duration, we use up delay samples
for the output packet and leave all frames in the queue */
if (afq->remaining_delay >= nb_samples) {
removed_samples += nb_samples;
afq->remaining_delay -= nb_samples;
}
/* remove frames from the queue until we have enough to cover the
requested number of samples or until the queue is empty */
while (removed_samples < nb_samples && afq->frame_queue) {
removed_samples += afq->frame_queue->duration;
delete_next_frame(afq);
}
afq->remaining_samples -= removed_samples;
/* if there are no frames left and we have room for more samples, use
any remaining delay samples */
if (removed_samples < nb_samples && afq->remaining_samples > 0) {
int add_samples = FFMIN(afq->remaining_samples,
nb_samples - removed_samples);
removed_samples += add_samples;
afq->remaining_samples -= add_samples;
}
if (removed_samples > nb_samples)
av_log(afq->avctx, AV_LOG_WARNING, "frame_size is too large\n");
if (duration)
*duration = ff_samples_to_time_base(afq->avctx, removed_samples);
}
void ff_af_queue_log_state(AudioFrameQueue *afq)
{
AudioFrame *f;
av_log(afq->avctx, AV_LOG_DEBUG, "remaining delay = %d\n",
afq->remaining_delay);
av_log(afq->avctx, AV_LOG_DEBUG, "remaining samples = %d\n",
afq->remaining_samples);
av_log(afq->avctx, AV_LOG_DEBUG, "frames:\n");
f = afq->frame_queue;
while (f) {
av_log(afq->avctx, AV_LOG_DEBUG, " [ pts=%9"PRId64" duration=%d ]\n",
f->pts, f->duration);
f = f->next;
}
}

90
libavcodec/audio_frame_queue.h Normal file
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/*
* Audio Frame Queue
* Copyright (c) 2012 Justin Ruggles
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AUDIO_FRAME_QUEUE_H
#define AVCODEC_AUDIO_FRAME_QUEUE_H
#include "avcodec.h"
typedef struct AudioFrame {
int64_t pts;
int duration;
struct AudioFrame *next;
} AudioFrame;
typedef struct AudioFrameQueue {
AVCodecContext *avctx;
int64_t next_pts;
int remaining_delay;
int remaining_samples;
AudioFrame *frame_queue;
} AudioFrameQueue;
/**
* Initialize AudioFrameQueue.
*
* @param avctx context to use for time_base and av_log
* @param afq queue context
*/
void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq);
/**
* Close AudioFrameQueue.
*
* Frees memory if needed.
*
* @param afq queue context
*/
void ff_af_queue_close(AudioFrameQueue *afq);
/**
* Add a frame to the queue.
*
* @param afq queue context
* @param f frame to add to the queue
*/
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f);
/**
* Remove frame(s) from the queue.
*
* Retrieves the pts of the next available frame, or a generated pts based on
* the last frame duration if there are no frames left in the queue. The number
* of requested samples should be the full number of samples represented by the
* packet that will be output by the encoder. If fewer samples are available
* in the queue, a smaller value will be used for the output duration.
*
* @param afq queue context
* @param nb_samples number of samples to remove from the queue
* @param[out] pts output packet pts
* @param[out] duration output packet duration
*/
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts,
int *duration);
/**
* Log the current state of the queue.
*
* @param afq queue context
*/
void ff_af_queue_log_state(AudioFrameQueue *afq);
#endif /* AVCODEC_AUDIO_FRAME_QUEUE_H */