[SCM] libav/experimental: Reindent after r21741.

[siretart at users.alioth.debian.org]

The following commit has been merged in the experimental branch:
commit 7515ed0c1d82f76029eca00b78ad018498c46d83
Author: Ronald S. Bultje <rsbultje at gmail.com>
Date:   Wed Feb 10 18:31:47 2010 +0000

    Reindent after r21741.
    
    Originally committed as revision 21742 to svn://svn.ffmpeg.org/ffmpeg/trunk

diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 171d707..8bc940d 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -133,36 +133,36 @@ static int sdp_parse_rtpmap(AVFormatContext *s,
     else
         c_name = "(null)";
 
-        get_word_sep(buf, sizeof(buf), "/", &p);
-        i = atoi(buf);
-        switch (codec->codec_type) {
-        case CODEC_TYPE_AUDIO:
-            av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
-            codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
-            codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
-            if (i > 0) {
-                codec->sample_rate = i;
-                get_word_sep(buf, sizeof(buf), "/", &p);
-                i = atoi(buf);
-                if (i > 0)
-                    codec->channels = i;
-                // TODO: there is a bug here; if it is a mono stream, and
-                // less than 22000Hz, faad upconverts to stereo and twice
-                // the frequency.  No problem, but the sample rate is being
-                // set here by the sdp line. Patch on its way. (rdm)
-            }
-            av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
-                   codec->sample_rate);
-            av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
-                   codec->channels);
-            break;
-        case CODEC_TYPE_VIDEO:
-            av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
-            break;
-        default:
-            break;
+    get_word_sep(buf, sizeof(buf), "/", &p);
+    i = atoi(buf);
+    switch (codec->codec_type) {
+    case CODEC_TYPE_AUDIO:
+        av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
+        codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
+        codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
+        if (i > 0) {
+            codec->sample_rate = i;
+            get_word_sep(buf, sizeof(buf), "/", &p);
+            i = atoi(buf);
+            if (i > 0)
+                codec->channels = i;
+            // TODO: there is a bug here; if it is a mono stream, and
+            // less than 22000Hz, faad upconverts to stereo and twice
+            // the frequency.  No problem, but the sample rate is being
+            // set here by the sdp line. Patch on its way. (rdm)
         }
-        return 0;
+        av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
+               codec->sample_rate);
+        av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
+               codec->channels);
+        break;
+    case CODEC_TYPE_VIDEO:
+        av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
+        break;
+    default:
+        break;
+    }
+    return 0;
 }
 
 /* return the length and optionally the data */

-- 
Libav/FFmpeg packaging