rtpenc: Always do the default initialization regardless of codecs

This avoids having to jump to the defaultcase in the switch. Manually
override the stream time base back to 90 kHz for the few audio codecs
that don't use the sample rate as time base (mp2, mp3).

Signed-off-by: Martin Storsjö <martin@martin.st>

diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index ae1d3e04c6a40d80f415782a41bc0e8c190c13fa..9a68c6d698fc914ca013d510b4333b820ff6accf 100644 (file)
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -176,11 +176,17 @@ static int rtp_write_header(AVFormatContext *s1)
         }
     }
 
-    avpriv_set_pts_info(st, 32, 1, 90000);
+    if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
+        avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
+    } else {
+        avpriv_set_pts_info(st, 32, 1, 90000);
+    }
+    s->buf_ptr = s->buf;
     switch(st->codec->codec_id) {
     case AV_CODEC_ID_MP2:
     case AV_CODEC_ID_MP3:
         s->buf_ptr = s->buf + 4;
+        avpriv_set_pts_info(st, 32, 1, 90000);
         break;
     case AV_CODEC_ID_MPEG1VIDEO:
     case AV_CODEC_ID_MPEG2VIDEO:
@@ -224,7 +230,7 @@ static int rtp_write_header(AVFormatContext *s1)
             s->max_frames_per_packet = 15;
         s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
         s->num_frames = 0;
-        goto defaultcase;
+        break;
     case AV_CODEC_ID_ADPCM_G722:
         /* Due to a historical error, the clock rate for G722 in RTP is
          * 8000, even if the sample rate is 16000. See RFC 3551. */
@@ -249,7 +255,7 @@ static int rtp_write_header(AVFormatContext *s1)
             s->max_frames_per_packet = 1;
         s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
                                          s->max_payload_size / st->codec->block_align);
-        goto defaultcase;
+        break;
     case AV_CODEC_ID_AMR_NB:
     case AV_CODEC_ID_AMR_WB:
         if (!s->max_frames_per_packet)
@@ -268,18 +274,13 @@ static int rtp_write_header(AVFormatContext *s1)
             goto fail;
         }
         s->num_frames = 0;
-        goto defaultcase;
+        break;
     case AV_CODEC_ID_AAC:
         s->num_frames = 0;
         if (!s->max_frames_per_packet)
             s->max_frames_per_packet = 5;
-        goto defaultcase;
+        break;
     default:
-defaultcase:
-        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
-            avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
-        }
-        s->buf_ptr = s->buf;
         break;
     }