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![Anton Khirnov](/assets/img/avatar_default.png)
Currently, the amount of padding inserted at the beginning by some audio encoders, is exported through AVCodecContext.delay. However - the term 'delay' is heavily overloaded and can have multiple different meanings even in the case of audio encoding. - this field has entirely different meanings, depending on whether the codec context is used for encoding or decoding (and has yet another different meaning for video), preventing generic handling of the codec context. Therefore, add a new field -- AVCodecContext.initial_padding. It could conceivably be used for decoding as well at a later point.
165 lines
5.2 KiB
C
165 lines
5.2 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 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/attributes.h"
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#include "libavutil/common.h"
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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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av_cold 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->initial_padding;
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afq->remaining_samples = avctx->initial_padding;
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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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#ifdef DEBUG
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static void af_queue_log_state(AudioFrameQueue *afq)
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{
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AudioFrame *f;
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av_dlog(afq->avctx, "remaining delay = %d\n", afq->remaining_delay);
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av_dlog(afq->avctx, "remaining samples = %d\n", afq->remaining_samples);
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av_dlog(afq->avctx, "frames:\n");
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f = afq->frame_queue;
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while (f) {
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av_dlog(afq->avctx, " [ 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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#endif /* DEBUG */
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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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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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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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