libfdk-aacdec: Always decode into an intermediate buffer

[ffmpeg] / libavformat / audiointerleave.c

diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c
index 31e59cfe1f3bff3c191041dce8cc2b3be8dd4248..fd07893b5cdb74c81cddd465fa402a946ce5c837 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"
@@ -32,8 +33,8 @@ 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)
-            av_fifo_free(&aic->fifo);
+        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) {
@@ -61,25 +62,28 @@ int ff_audio_interleave_init(AVFormatContext *s,
             aic->samples = aic->samples_per_frame;
             aic->time_base = time_base;
 
-            av_fifo_init(&aic->fifo, 100 * *aic->samples);
+            aic->fifo_size = 100* *aic->samples;
+            aic->fifo= av_fifo_alloc(100 * *aic->samples);
         }
     }
 
     return 0;
 }
 
-static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
-                                   int stream_index, int flush)
+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 size = FFMIN(av_fifo_size(&aic->fifo), *aic->samples * aic->sample_size);
-    if (!size || (!flush && size == av_fifo_size(&aic->fifo)))
+    int ret;
+    int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
+    if (!size || (!flush && size == av_fifo_size(aic->fifo)))
         return 0;
 
-    av_new_packet(pkt, size);
-    av_fifo_read(&aic->fifo, pkt->data, size);
+    ret = av_new_packet(pkt, size);
+    if (ret < 0)
+        return ret;
+    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);
@@ -93,34 +97,42 @@ static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
     return size;
 }
 
-int ff_audio_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
+int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
                         int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
                         int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
 {
-    int i;
+    int i, ret;
 
     if (pkt) {
         AVStream *st = s->streams[pkt->stream_index];
         AudioInterleaveContext *aic = st->priv_data;
-        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
-            av_fifo_generic_write(&aic->fifo, pkt->data, pkt->size, NULL);
+        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)
+                    return -1;
+                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;
-            ff_interleave_add_packet(s, pkt, compare_ts);
+            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->codec->codec_type == CODEC_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);
+        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
+            AVPacket new_pkt = { 0 };
+            while (interleave_new_audio_packet(s, &new_pkt, i, flush))
+                if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
+                    return ret;
         }
     }
 
-    return get_packet(s, out, pkt, flush);
+    return get_packet(s, out, NULL, flush);
 }