mirror of https://github.com/FFmpeg/FFmpeg.git
This simplifies matching of timestamps between input frames and output packets.pull/4/head
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c9594fe0fb
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4bf64961a9
2 changed files with 252 additions and 0 deletions
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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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} |
@ -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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