[FFmpeg-devel] [PATCH 1/2] lavf/swfdec: factorize the creation of a new stream.

[Clément Bœsch]

This also makes the changes of a3949fe11 applicable in both cases.
---
 libavformat/swfdec.c | 51 +++++++++++++++++++++++++--------------------------
 1 file changed, 25 insertions(+), 26 deletions(-)

diff --git a/libavformat/swfdec.c b/libavformat/swfdec.c
index 8fb4aeb..192dbec 100644
--- a/libavformat/swfdec.c
+++ b/libavformat/swfdec.c
@@ -138,6 +138,29 @@ static int swf_read_header(AVFormatContext *s)
     return 0;
 }
 
+static AVStream *create_new_audio_stream(AVFormatContext *s, int id, int info)
+{
+    int sample_rate_code;
+    AVStream *ast = avformat_new_stream(s, NULL);
+    if (!ast)
+        return NULL;
+    ast->id = id; /* -1 to avoid clash with video stream ch_id */
+    if (info & 1) {
+        ast->codec->channels       = 2;
+        ast->codec->channel_layout = AV_CH_LAYOUT_STEREO;
+    } else {
+        ast->codec->channels       = 1;
+        ast->codec->channel_layout = AV_CH_LAYOUT_MONO;
+    }
+    ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+    ast->codec->codec_id   = ff_codec_get_id(swf_audio_codec_tags, info>>4 & 15);
+    ast->need_parsing = AVSTREAM_PARSE_FULL;
+    sample_rate_code = info>>2 & 3;
+    ast->codec->sample_rate = 44100 >> (3 - sample_rate_code);
+    avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate);
+    return ast;
+}
+
 static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
 {
     SWFContext *swf = s->priv_data;
@@ -184,7 +207,6 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
             len -= 8;
         } else if (tag == TAG_STREAMHEAD || tag == TAG_STREAMHEAD2) {
             /* streaming found */
-            int sample_rate_code;
 
             for (i=0; i<s->nb_streams; i++) {
                 st = s->streams[i];
@@ -195,27 +217,12 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
             avio_r8(pb);
             v = avio_r8(pb);
             swf->samples_per_frame = avio_rl16(pb);
-            ast = avformat_new_stream(s, NULL);
+            ast = create_new_audio_stream(s, -1, v); /* -1 to avoid clash with video stream ch_id */
             if (!ast)
                 return AVERROR(ENOMEM);
-            ast->id = -1; /* -1 to avoid clash with video stream ch_id */
-            if (v & 1) {
-                ast->codec->channels       = 2;
-                ast->codec->channel_layout = AV_CH_LAYOUT_STEREO;
-            } else {
-                ast->codec->channels       = 1;
-                ast->codec->channel_layout = AV_CH_LAYOUT_MONO;
-            }
-            ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-            ast->codec->codec_id = ff_codec_get_id(swf_audio_codec_tags, (v>>4) & 15);
-            ast->need_parsing = AVSTREAM_PARSE_FULL;
-            sample_rate_code= (v>>2) & 3;
-            ast->codec->sample_rate = 44100 >> (3 - sample_rate_code);
-            avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate);
             len -= 4;
         } else if (tag == TAG_DEFINESOUND) {
             /* audio stream */
-            int sample_rate_code;
             int ch_id = avio_rl16(pb);
 
             for (i=0; i<s->nb_streams; i++) {
@@ -229,17 +236,9 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
             // these are smaller audio streams in DEFINESOUND tags, but it's technically
             // possible they could be huge. Break it up into multiple packets if it's big.
             v = avio_r8(pb);
-            ast = avformat_new_stream(s, NULL);
+            ast = create_new_audio_stream(s, ch_id, v);
             if (!ast)
                 return AVERROR(ENOMEM);
-            ast->id = ch_id;
-            ast->codec->channels = 1 + (v&1);
-            ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-            ast->codec->codec_id = ff_codec_get_id(swf_audio_codec_tags, (v>>4) & 15);
-            ast->need_parsing = AVSTREAM_PARSE_FULL;
-            sample_rate_code= (v>>2) & 3;
-            ast->codec->sample_rate = 44100 >> (3 - sample_rate_code);
-            avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate);
             ast->duration = avio_rl32(pb); // number of samples
             if (((v>>4) & 15) == 2) { // MP3 sound data record
                 ast->skip_samples = avio_rl16(pb);
-- 
1.8.1.4