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authorAnton Khirnov <anton@khirnov.net>2012-08-05 11:11:04 +0200
committerAnton Khirnov <anton@khirnov.net>2012-08-07 16:00:24 +0200
commit36ef5369ee9b336febc2c270f8718cec4476cb85 (patch)
treed186adbb488e7f002aa894743b1ce0e8925520e6 /libavformat/rtpdec.c
parent104e10fb426f903ba9157fdbfe30292d0e4c3d72 (diff)
Replace all CODEC_ID_* with AV_CODEC_ID_*
Diffstat (limited to 'libavformat/rtpdec.c')
-rw-r--r--libavformat/rtpdec.c30
1 files changed, 15 insertions, 15 deletions
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 2c5e6c8176..8f9f60ac54 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -46,7 +46,7 @@
static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
.enc_name = "X-MP3-draft-00",
.codec_type = AVMEDIA_TYPE_AUDIO,
- .codec_id = CODEC_ID_MP3ADU,
+ .codec_id = AV_CODEC_ID_MP3ADU,
};
/* statistics functions */
@@ -364,7 +364,7 @@ void ff_rtp_send_punch_packets(URLContext* rtp_handle)
/**
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
* MPEG2TS streams to indicate that they should be demuxed inside the
- * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
+ * rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
*/
RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
{
@@ -388,19 +388,19 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext
}
} else if (st) {
switch(st->codec->codec_id) {
- case CODEC_ID_MPEG1VIDEO:
- case CODEC_ID_MPEG2VIDEO:
- case CODEC_ID_MP2:
- case CODEC_ID_MP3:
- case CODEC_ID_MPEG4:
- case CODEC_ID_H263:
- case CODEC_ID_H264:
+ case AV_CODEC_ID_MPEG1VIDEO:
+ case AV_CODEC_ID_MPEG2VIDEO:
+ case AV_CODEC_ID_MP2:
+ case AV_CODEC_ID_MP3:
+ case AV_CODEC_ID_MPEG4:
+ case AV_CODEC_ID_H263:
+ case AV_CODEC_ID_H264:
st->need_parsing = AVSTREAM_PARSE_FULL;
break;
- case CODEC_ID_VORBIS:
+ case AV_CODEC_ID_VORBIS:
st->need_parsing = AVSTREAM_PARSE_HEADERS;
break;
- case CODEC_ID_ADPCM_G722:
+ case AV_CODEC_ID_ADPCM_G722:
/* According to RFC 3551, the stream clock rate is 8000
* even if the sample rate is 16000. */
if (st->codec->sample_rate == 8000)
@@ -537,8 +537,8 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
} else {
// at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
switch(st->codec->codec_id) {
- case CODEC_ID_MP2:
- case CODEC_ID_MP3:
+ case AV_CODEC_ID_MP2:
+ case AV_CODEC_ID_MP3:
/* better than nothing: skip mpeg audio RTP header */
if (len <= 4)
return -1;
@@ -548,8 +548,8 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
- case CODEC_ID_MPEG1VIDEO:
- case CODEC_ID_MPEG2VIDEO:
+ case AV_CODEC_ID_MPEG1VIDEO:
+ case AV_CODEC_ID_MPEG2VIDEO:
/* better than nothing: skip mpeg video RTP header */
if (len <= 4)
return -1;