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| author | Anton Khirnov <anton@khirnov.net> | 2012-08-05 11:11:04 +0200 |
|---|---|---|
| committer | Anton Khirnov <anton@khirnov.net> | 2012-08-07 16:00:24 +0200 |
| commit | 36ef5369ee9b336febc2c270f8718cec4476cb85 (patch) | |
| tree | d186adbb488e7f002aa894743b1ce0e8925520e6 /libavformat/rtpdec.c | |
| parent | 104e10fb426f903ba9157fdbfe30292d0e4c3d72 (diff) | |
| download | ffmpeg-36ef5369ee9b336febc2c270f8718cec4476cb85.tar.gz | |
Replace all CODEC_ID_* with AV_CODEC_ID_*
Diffstat (limited to 'libavformat/rtpdec.c')
| -rw-r--r-- | libavformat/rtpdec.c | 30 |
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; |