audio_frame_queue.c

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/*
 * Audio Frame Queue
 * Copyright (c) 2012 Justin Ruggles
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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.
 *
 * FFmpeg 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 FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 
#include "libavutil/attributes.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mem.h"
#include "audio_frame_queue.h"
#include "encode.h"
#include "libavutil/avassert.h"
 
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
{
    afq->avctx = avctx;
    afq->remaining_delay   = avctx->initial_padding;
    afq->remaining_samples = avctx->initial_padding;
    afq->output_delay      = avctx->initial_padding;
    afq->frame_count       = 0;
}
 
av_cold void ff_af_queue_close(AudioFrameQueue *afq)
{
    if(afq->frame_count)
        av_log(afq->avctx, AV_LOG_WARNING, "%d frames left in the queue on closing\n", afq->frame_count);
    av_freep(&afq->frames);
    memset(afq, 0, sizeof(*afq));
}
 
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
{
    AudioFrame *new = av_fast_realloc(afq->frames, &afq->frame_alloc, sizeof(*afq->frames)*(afq->frame_count+1));
    const AVFrameSideData *sd;
    int nb_samples = f->nb_samples;
 
    if(!new)
        return AVERROR(ENOMEM);
    afq->frames = new;
    new += afq->frame_count;
 
    sd = av_frame_side_data_get(f->side_data, f->nb_side_data, AV_FRAME_DATA_SKIP_SAMPLES);
    if (sd && sd->size >= 10) {
        int discard_padding = AV_RL32(sd->data + 4);
        if (discard_padding > 0 && discard_padding < nb_samples)
            nb_samples -= discard_padding;
    }
 
    /* get frame parameters */
    new->duration = nb_samples;
    new->duration += afq->remaining_delay;
    if (f->pts != AV_NOPTS_VALUE) {
        new->pts = av_rescale_q(f->pts,
                                      afq->avctx->time_base,
                                      (AVRational){ 1, afq->avctx->sample_rate });
        new->pts -= afq->remaining_delay;
        if(afq->frame_count && new[-1].pts >= new->pts)
            av_log(afq->avctx, AV_LOG_WARNING, "Queue input is backward in time\n");
    } else {
        new->pts = AV_NOPTS_VALUE;
    }
    afq->remaining_delay = 0;
 
    /* add frame sample count */
    afq->remaining_samples += nb_samples;
 
    afq->frame_count++;
 
    return 0;
}
 
int ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, AVPacket *pkt)
{
    int64_t out_pts = AV_NOPTS_VALUE;
    int frame_size = nb_samples;
    int removed_samples = 0;
    int i;
 
    if (afq->frame_count || afq->frame_alloc) {
        if (afq->frames->pts != AV_NOPTS_VALUE)
            out_pts = afq->frames->pts;
    }
    if(!afq->frame_count)
        av_log(afq->avctx, AV_LOG_WARNING, "Trying to remove %d samples, but the queue is empty\n", nb_samples);
    if (pkt)
        pkt->pts = ff_samples_to_time_base(afq->avctx, out_pts);
 
    for(i=0; nb_samples && i<afq->frame_count; i++){
        int n= FFMIN(afq->frames[i].duration, nb_samples);
        afq->frames[i].duration -= n;
        nb_samples              -= n;
        removed_samples         += n;
        if(afq->frames[i].pts != AV_NOPTS_VALUE)
            afq->frames[i].pts      += n;
    }
    afq->remaining_samples -= removed_samples;
    i -= i && afq->frames[i-1].duration;
    memmove(afq->frames, afq->frames + i, sizeof(*afq->frames) * (afq->frame_count - i));
    afq->frame_count -= i;
 
    if(nb_samples){
        av_assert0(!afq->frame_count);
        av_assert0(afq->remaining_samples == afq->remaining_delay);
        if(afq->frames && afq->frames[0].pts != AV_NOPTS_VALUE)
            afq->frames[0].pts += nb_samples;
        av_log(afq->avctx, AV_LOG_DEBUG, "Trying to remove %d more samples than there are in the queue\n", nb_samples);
    }
    if (pkt) {
        int discard_padding = frame_size - removed_samples;
        pkt->duration = ff_samples_to_time_base(afq->avctx, removed_samples);
        if (afq->output_delay > 0 || discard_padding > 0) {
            uint8_t *side_data =
                av_packet_new_side_data(pkt, AV_PKT_DATA_SKIP_SAMPLES, 10);
            if (!side_data)
                return AVERROR(ENOMEM);
            if (afq->output_delay) {
                AV_WL32(side_data, FFMIN(afq->output_delay, removed_samples));
                afq->output_delay -= removed_samples;
                afq->output_delay = FFMAX(afq->output_delay, 0);
            }
            AV_WL32(side_data + 4, discard_padding);
        }
    }
 
    return 0;
}