From 4bf64961a99f36b72b69e66310fa828525564166 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Wed, 22 Feb 2012 21:52:34 -0500 Subject: 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. --- libavcodec/audio_frame_queue.c | 162 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 162 insertions(+) create mode 100644 libavcodec/audio_frame_queue.c (limited to 'libavcodec/audio_frame_queue.c') diff --git a/libavcodec/audio_frame_queue.c b/libavcodec/audio_frame_queue.c new file mode 100644 index 0000000000..156c3a109b --- /dev/null +++ b/libavcodec/audio_frame_queue.c @@ -0,0 +1,162 @@ +/* + * 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; + } +} -- cgit v1.2.3