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

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This commit is contained in:
xiongziliang 2021-07-06 23:12:52 +08:00
commit 756b6a4cff
2 changed files with 21 additions and 36 deletions
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17
webrtc/RtpExt.cpp
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@ -192,23 +192,6 @@ map<uint8_t/*id*/, RtpExt/*data*/> RtpExt::getExtValue(const RtpHeader *header)
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},
static map<RtpExtType/*id*/, string/*ext*/> s_type_to_url = {RTP_EXT_MAP(XX)};
#undef XX

40
webrtc/RtpExt.h
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@ -20,27 +20,29 @@
using namespace std;
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 {
padding = 0,
ssrc_audio_level = 1,
abs_send_time = 2,
transport_cc = 3,
sdes_mid = 4,
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
#define XX(type, uri) type,
RTP_EXT_MAP(XX)
#undef XX
reserved = encrypt,
};
class RtcMedia;