avformat/audiointerleave: only keep the retime functionality of the audio interleaver

And rename it to retimeinterleave, use the pcm_rechunk bitstream filter for
rechunking.

By seperating the two functions we hopefully get cleaner code.

Signed-off-by: Marton Balint <cus@passwd.hu>

1 files changed, 0 insertions, 148 deletions

diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c
deleted file mode 100644
index 36a3288242..0000000000
--- a/libavformat/audiointerleave.c
+++ /dev/null
@@ -1,148 +0,0 @@
-/*
- * Audio Interleaving functions
- *
- * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
- *
- * 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/fifo.h"
-#include "libavutil/mathematics.h"
-#include "avformat.h"
-#include "audiointerleave.h"
-#include "internal.h"
-
-void ff_audio_interleave_close(AVFormatContext *s)
-{
-    int i;
-    for (i = 0; i < s->nb_streams; i++) {
-        AVStream *st = s->streams[i];
-        AudioInterleaveContext *aic = st->priv_data;
-
-        if (aic && st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO)
-            av_fifo_freep(&aic->fifo);
-    }
-}
-
-int ff_audio_interleave_init(AVFormatContext *s,
-                             const int samples_per_frame,
-                             AVRational time_base)
-{
-    int i;
-
-    if (!time_base.num) {
-        av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
-        return AVERROR(EINVAL);
-    }
-    for (i = 0; i < s->nb_streams; i++) {
-        AVStream *st = s->streams[i];
-        AudioInterleaveContext *aic = st->priv_data;
-
-        if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
-            int max_samples = samples_per_frame ? samples_per_frame :
-                              av_rescale_rnd(st->codecpar->sample_rate, time_base.num, time_base.den, AV_ROUND_UP);
-            aic->sample_size = (st->codecpar->channels *
-                                av_get_bits_per_sample(st->codecpar->codec_id)) / 8;
-            if (!aic->sample_size) {
-                av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
-                return AVERROR(EINVAL);
-            }
-            aic->samples_per_frame = samples_per_frame;
-            aic->time_base = time_base;
-
-            if (!(aic->fifo = av_fifo_alloc_array(100, max_samples)))
-                return AVERROR(ENOMEM);
-            aic->fifo_size = 100 * max_samples;
-        }
-    }
-
-    return 0;
-}
-
-static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
-                                       int stream_index, int flush)
-{
-    AVStream *st = s->streams[stream_index];
-    AudioInterleaveContext *aic = st->priv_data;
-    int ret;
-    int nb_samples = aic->samples_per_frame ? aic->samples_per_frame :
-                     (av_rescale_q(aic->n + 1, av_make_q(st->codecpar->sample_rate, 1), av_inv_q(aic->time_base)) - aic->nb_samples);
-    int frame_size = nb_samples * aic->sample_size;
-    int size = FFMIN(av_fifo_size(aic->fifo), frame_size);
-    if (!size || (!flush && size == av_fifo_size(aic->fifo)))
-        return 0;
-
-    ret = av_new_packet(pkt, frame_size);
-    if (ret < 0)
-        return ret;
-    av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
-
-    if (size < pkt->size)
-        memset(pkt->data + size, 0, pkt->size - size);
-
-    pkt->dts = pkt->pts = aic->dts;
-    pkt->duration = av_rescale_q(nb_samples, st->time_base, aic->time_base);
-    pkt->stream_index = stream_index;
-    aic->dts += pkt->duration;
-    aic->nb_samples += nb_samples;
-    aic->n++;
-
-    return pkt->size;
-}
-
-int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
-                        int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
-                        int (*compare_ts)(AVFormatContext *, const AVPacket *, const AVPacket *))
-{
-    int i, ret;
-
-    if (pkt) {
-        AVStream *st = s->streams[pkt->stream_index];
-        AudioInterleaveContext *aic = st->priv_data;
-        if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
-            unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
-            if (new_size > aic->fifo_size) {
-                if (av_fifo_realloc2(aic->fifo, new_size) < 0)
-                    return AVERROR(ENOMEM);
-                aic->fifo_size = new_size;
-            }
-            av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
-        } else {
-            // rewrite pts and dts to be decoded time line position
-            pkt->pts = pkt->dts = aic->dts;
-            aic->dts += pkt->duration;
-            if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0)
-                return ret;
-        }
-        pkt = NULL;
-    }
-
-    for (i = 0; i < s->nb_streams; i++) {
-        AVStream *st = s->streams[i];
-        if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
-            AVPacket new_pkt;
-            while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
-                if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
-                    return ret;
-            }
-            if (ret < 0)
-                return ret;
-        }
-    }
-
-    return get_packet(s, out, NULL, flush);
-}