asm: Consistently uppercase SECTION markers

[ffmpeg] / libavformat / audiointerleave.c

diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c
index 5df0bb0b40e968d373c36b48c389a5422b430389..aa379f675ea182137fb02e1ce5c1356153766061 100644 (file)
--- a/libavformat/audiointerleave.c
+++ b/libavformat/audiointerleave.c
@@ -33,7 +33,7 @@ void ff_audio_interleave_close(AVFormatContext *s)
         AVStream *st = s->streams[i];
         AudioInterleaveContext *aic = st->priv_data;
 
-        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
+        if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO)
             av_fifo_free(aic->fifo);
     }
 }
@@ -51,9 +51,9 @@ int ff_audio_interleave_init(AVFormatContext *s,
         AVStream *st = s->streams[i];
         AudioInterleaveContext *aic = st->priv_data;
 
-        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 (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
+            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 -1;
@@ -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;
@@ -99,12 +101,12 @@ 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];
         AudioInterleaveContext *aic = st->priv_data;
-        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
+        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)
@@ -116,17 +118,19 @@ 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;
     }
 
     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);
+        if (st->codecpar->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;
         }
     }