ZeroCodec: Flip output

[ffmpeg] / libavformat / audiointerleave.c

diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c
index a4396f5bfa09ac8e7ff6049b58b97cfdf823df86..5df0bb0b40e968d373c36b48c389a5422b430389 100644 (file)
--- a/libavformat/audiointerleave.c
+++ b/libavformat/audiointerleave.c
@@ -3,24 +3,25 @@
  *
  * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * 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.
  *
- * FFmpeg is distributed in the hope that it will be useful,
+ * 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 FFmpeg; if not, write to the Free Software
+ * 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/fifo.h"
+#include "libavutil/mathematics.h"
 #include "avformat.h"
 #include "audiointerleave.h"
 #include "internal.h"
@@ -32,7 +33,7 @@ void ff_audio_interleave_close(AVFormatContext *s)
         AVStream *st = s->streams[i];
         AudioInterleaveContext *aic = st->priv_data;
 
-        if (st->codec->codec_type == CODEC_TYPE_AUDIO)
+        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
             av_fifo_free(aic->fifo);
     }
 }
@@ -50,7 +51,7 @@ int ff_audio_interleave_init(AVFormatContext *s,
         AVStream *st = s->streams[i];
         AudioInterleaveContext *aic = st->priv_data;
 
-        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
             aic->sample_size = (st->codec->channels *
                                 av_get_bits_per_sample(st->codec->codec_id)) / 8;
             if (!aic->sample_size) {
@@ -80,7 +81,7 @@ static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
         return 0;
 
     av_new_packet(pkt, size);
-    av_fifo_read(aic->fifo, pkt->data, size);
+    av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
 
     pkt->dts = pkt->pts = aic->dts;
     pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
@@ -103,7 +104,7 @@ int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt
     if (pkt) {
         AVStream *st = s->streams[pkt->stream_index];
         AudioInterleaveContext *aic = st->priv_data;
-        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+        if (st->codec->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)
@@ -122,12 +123,12 @@ int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt
 
     for (i = 0; i < s->nb_streams; i++) {
         AVStream *st = s->streams[i];
-        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
             AVPacket new_pkt;
             while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
                 ff_interleave_add_packet(s, &new_pkt, compare_ts);
         }
     }
 
-    return get_packet(s, out, pkt, flush);
+    return get_packet(s, out, NULL, flush);
 }