Commit 756b6a4c by xiongziliang

Merge branch 'master' of https://github.com/xia-chu/ZLMediaKit into dev

parents 986e9511 ea39ffe5
...@@ -192,23 +192,6 @@ map<uint8_t/*id*/, RtpExt/*data*/> RtpExt::getExtValue(const RtpHeader *header) ...@@ -192,23 +192,6 @@ map<uint8_t/*id*/, RtpExt/*data*/> RtpExt::getExtValue(const RtpHeader *header)
return ret; return ret;
} }
#define RTP_EXT_MAP(XX) \
XX(ssrc_audio_level, "urn:ietf:params:rtp-hdrext:ssrc-audio-level") \
XX(abs_send_time, "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time") \
XX(transport_cc, "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01") \
XX(sdes_mid, "urn:ietf:params:rtp-hdrext:sdes:mid") \
XX(sdes_rtp_stream_id, "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id") \
XX(sdes_repaired_rtp_stream_id, "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id") \
XX(video_timing, "http://www.webrtc.org/experiments/rtp-hdrext/video-timing") \
XX(color_space, "http://www.webrtc.org/experiments/rtp-hdrext/color-space") \
XX(csrc_audio_level, "urn:ietf:params:rtp-hdrext:csrc-audio-level") \
XX(framemarking, "http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07") \
XX(video_content_type, "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type") \
XX(playout_delay, "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay") \
XX(video_orientation, "urn:3gpp:video-orientation") \
XX(toffset, "urn:ietf:params:rtp-hdrext:toffset") \
XX(encrypt, "urn:ietf:params:rtp-hdrext:encrypt")
#define XX(type, url) {RtpExtType::type , url}, #define XX(type, url) {RtpExtType::type , url},
static map<RtpExtType/*id*/, string/*ext*/> s_type_to_url = {RTP_EXT_MAP(XX)}; static map<RtpExtType/*id*/, string/*ext*/> s_type_to_url = {RTP_EXT_MAP(XX)};
#undef XX #undef XX
......
...@@ -20,27 +20,29 @@ ...@@ -20,27 +20,29 @@
using namespace std; using namespace std;
using namespace mediakit; using namespace mediakit;
#define RTP_EXT_MAP(XX) \
XX(ssrc_audio_level, "urn:ietf:params:rtp-hdrext:ssrc-audio-level") \
XX(abs_send_time, "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time") \
XX(transport_cc, "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01") \
XX(sdes_mid, "urn:ietf:params:rtp-hdrext:sdes:mid") \
XX(sdes_rtp_stream_id, "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id") \
XX(sdes_repaired_rtp_stream_id, "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id") \
XX(video_timing, "http://www.webrtc.org/experiments/rtp-hdrext/video-timing") \
XX(color_space, "http://www.webrtc.org/experiments/rtp-hdrext/color-space") \
XX(csrc_audio_level, "urn:ietf:params:rtp-hdrext:csrc-audio-level") \
XX(framemarking, "http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07") \
XX(video_content_type, "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type") \
XX(playout_delay, "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay") \
XX(video_orientation, "urn:3gpp:video-orientation") \
XX(toffset, "urn:ietf:params:rtp-hdrext:toffset") \
XX(encrypt, "urn:ietf:params:rtp-hdrext:encrypt")
enum class RtpExtType : uint8_t { enum class RtpExtType : uint8_t {
padding = 0, padding = 0,
ssrc_audio_level = 1, #define XX(type, uri) type,
abs_send_time = 2, RTP_EXT_MAP(XX)
transport_cc = 3, #undef XX
sdes_mid = 4, reserved = encrypt,
sdes_rtp_stream_id = 5,
sdes_repaired_rtp_stream_id = 6,
video_timing = 7,
color_space = 8,
//for firefox
csrc_audio_level = 9,
//svc ?
framemarking = 10,
video_content_type = 11,
playout_delay = 12,
video_orientation = 13,
toffset = 14,
reserved = 15,
// e2e ?
encrypt = reserved
}; };
class RtcMedia; class RtcMedia;
......
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