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rtp: Replace hardcoded RTCP packet types with defines

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Patch by Josh Allmann, joshua dot allmann at gmail

Originally committed as revision 24912 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Josh Allmann 2010-08-25 09:15:31 +00:00 committed by Martin Storsjö
parent d21caa952c
commit 7f3468d392
5 changed files with 17 additions and 8 deletions
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9
libavformat/rtp.h
View File

@ -82,4 +82,13 @@ enum CodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type);
*/
#define RTP_XIPH_IDENT 0xfecdba
/* RTCP packet types */
enum RTCPType {
RTCP_SR = 200,
RTCP_RR, // 201
RTCP_SDES, // 202
RTCP_BYE, // 203
RTCP_APP // 204
};
#endif /* AVFORMAT_RTP_H */

10
libavformat/rtpdec.c
View File

@ -74,7 +74,7 @@ void av_register_rtp_dynamic_payload_handlers(void)
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
{
if (buf[1] != 200)
if (buf[1] != RTCP_SR)
return -1;
s->last_rtcp_ntp_time = AV_RB64(buf + 8);
if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
@ -209,7 +209,7 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
// Receiver Report
put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
put_byte(pb, 201);
put_byte(pb, RTCP_RR);
put_be16(pb, 7); /* length in words - 1 */
// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
put_be32(pb, s->ssrc + 1);
@ -248,7 +248,7 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
// CNAME
put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
put_byte(pb, 202);
put_byte(pb, RTCP_SDES);
len = strlen(s->hostname);
put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
put_be32(pb, s->ssrc);
@ -299,7 +299,7 @@ void rtp_send_punch_packets(URLContext* rtp_handle)
return;
put_byte(pb, (RTP_VERSION << 6));
put_byte(pb, 201); /* receiver report */
put_byte(pb, RTCP_RR); /* receiver report */
put_be16(pb, 1); /* length in words - 1 */
put_be32(pb, 0); /* our own SSRC */
@ -434,7 +434,7 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
return -1;
if (buf[1] >= 200 && buf[1] <= 204) {
if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
rtcp_parse_packet(s, buf, len);
return -1;
}

2
libavformat/rtpenc.c
View File

@ -191,7 +191,7 @@ static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
s1->streams[0]->time_base) + s->base_timestamp;
put_byte(s1->pb, (RTP_VERSION << 6));
put_byte(s1->pb, 200);
put_byte(s1->pb, RTCP_SR);
put_be16(s1->pb, 6); /* length in words - 1 */
put_be32(s1->pb, s->ssrc);
put_be32(s1->pb, ntp_time / 1000000);

2
libavformat/rtpproto.c
View File

@ -285,7 +285,7 @@ static int rtp_write(URLContext *h, const uint8_t *buf, int size)
int ret;
URLContext *hd;
if (buf[1] >= 200 && buf[1] <= 204) {
if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
/* RTCP payload type */
hd = s->rtcp_hd;
} else {

2
libavformat/rtspenc.c
View File

@ -85,7 +85,7 @@ static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
size -= 4;
if (packet_len > size || packet_len < 2)
break;
if (ptr[1] >= 200 && ptr[1] <= 204)
if (ptr[1] >= RTCP_SR && ptr[1] <= RTCP_APP)
id = rtsp_st->interleaved_max; /* RTCP */
else
id = rtsp_st->interleaved_min; /* RTP */