libfdk-aacdec: Always decode into an intermediate buffer

[ffmpeg] / libavformat / audiointerleave.c

diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c
index e48f826e1466eaf163b2986e7d2e855cb86ade20..fd07893b5cdb74c81cddd465fa402a946ce5c837 100644 (file)
--- a/libavformat/audiointerleave.c
+++ b/libavformat/audiointerleave.c
@@ -70,17 +70,19 @@ int ff_audio_interleave_init(AVFormatContext *s,
     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 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);
+    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;
@@ -99,7 +101,7 @@ int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt
                         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];
@@ -116,7 +118,8 @@ int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt
             // 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;
     }
@@ -124,11 +127,12 @@ 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 == 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);
+            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);
 }