mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-22 10:40:05 +08:00
完善rtp ext解析代码
This commit is contained in:
xia-chu
parent
f263dd5d9d
commit
0b8eb6a76f
@ -166,10 +166,13 @@ map<RtpExtType/*type*/, RtpExt/*data*/> RtpExt::getExtValue(const RtpHeader *hea
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XX(sdes_repaired_rtp_stream_id, "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id") \
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XX(video_timing, "http://www.webrtc.org/experiments/rtp-hdrext/video-timing") \
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XX(color_space, "http://www.webrtc.org/experiments/rtp-hdrext/color-space") \
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XX(csrc_audio_level, "urn:ietf:params:rtp-hdrext:csrc-audio-level") \
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XX(framemarking, "http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07") \
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XX(video_content_type, "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type") \
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XX(playout_delay, "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay") \
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XX(video_orientation, "urn:3gpp:video-orientation") \
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XX(toffset, "urn:ietf:params:rtp-hdrext:toffset")
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XX(toffset, "urn:ietf:params:rtp-hdrext:toffset") \
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XX(encrypt, "urn:ietf:params:rtp-hdrext:encrypt")
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#define XX(type, url) {RtpExtType::type , url},
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static unordered_map<RtpExtType/*id*/, string/*ext*/> s_type_to_url = {RTP_EXT_MAP(XX)};
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@ -207,6 +210,291 @@ const char *RtpExt::getExtName(RtpExtType type) {
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string RtpExt::dumpString() const {
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_StrPrinter printer;
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printer << getExtName(_type) << ", id:" << (int)_id << " " << hexdump(data(), size());
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switch (_type) {
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case RtpExtType::ssrc_audio_level : {
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bool vad;
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printer << "audio level:" << (int) getAudioLevel(&vad) << ", vad:" << vad;
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break;
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}
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case RtpExtType::abs_send_time : {
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printer << "abs send time:" << getAbsSendTime();
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break;
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}
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case RtpExtType::transport_cc : {
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printer << "twcc seq:" << getTransportCCSeq();
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break;
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}
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case RtpExtType::sdes_mid : {
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printer << "sdes mid:" << getSdesMid();
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break;
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}
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case RtpExtType::sdes_rtp_stream_id : {
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printer << "rtp stream id:" << getRtpStreamId();
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break;
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}
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case RtpExtType::sdes_repaired_rtp_stream_id : {
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printer << "rtp repaired stream id:" << getRepairedRtpStreamId();
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break;
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}
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case RtpExtType::video_timing : {
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uint8_t flags;
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uint16_t encode_start, encode_finish, packetization_complete, last_pkt_left_pacer, reserved_net0, reserved_net1;
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getVideoTiming(flags, encode_start, encode_finish, packetization_complete, last_pkt_left_pacer,
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reserved_net0, reserved_net1);
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printer << "video timing, flags:" << (int) flags
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<< ",encode:" << encode_start << "-" << encode_finish
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<< ",packetization_complete:" << packetization_complete
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<< ",last_pkt_left_pacer:" << last_pkt_left_pacer
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<< ",reserved_net0:" << reserved_net0
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<< ",reserved_net1:" << reserved_net1;
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break;
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}
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case RtpExtType::video_content_type : {
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printer << "video content type:" << (int)getVideoContentType();
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break;
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}
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case RtpExtType::video_orientation : {
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bool camera_bit, flip_bit, first_rotation, second_rotation;
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getVideoOrientation(camera_bit, flip_bit, first_rotation, second_rotation);
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printer << "video orientation:" << camera_bit << "-" << flip_bit << "-" << first_rotation << "-" << second_rotation;
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break;
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}
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case RtpExtType::playout_delay : {
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uint16_t min_delay, max_delay;
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getPlayoutDelay(min_delay, max_delay);
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printer << "playout delay:" << min_delay << "-" << max_delay;
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break;
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}
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case RtpExtType::toffset : {
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printer << "toffset:" << getTransmissionOffset();
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break;
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}
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case RtpExtType::framemarking : {
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printer << "framemarking tid:" << getFramemarkingTID();
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break;
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}
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default: {
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printer << getExtName(_type) << ", id:" << (int) _id << ", ";
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printer << "hex:" << hexdump(data(), size());
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break;
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}
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}
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return std::move(printer);
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}
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//https://tools.ietf.org/html/rfc6464
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// 0 1
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID | len=0 |V| level |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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//
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// Figure 1: Sample Audio Level Encoding Using the
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// One-Byte Header Format
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//
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//
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// 0 1 2 3
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID | len=1 |V| level | 0 (pad) |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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//
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// Figure 2: Sample Audio Level Encoding Using the
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// Two-Byte Header Format
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uint8_t RtpExt::getAudioLevel(bool *vad) const{
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CHECK(_type == RtpExtType::ssrc_audio_level && size() >= 1);
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auto &byte = (*this)[0];
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if (vad) {
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*vad = byte & 0x80;
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}
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return byte & 0x7F;
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}
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//http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
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//Wire format: 1-byte extension, 3 bytes of data. total 4 bytes extra per packet (plus shared 4 bytes for all extensions present: 2 byte magic word 0xBEDE, 2 byte # of extensions). Will in practice replace the “toffset” extension so we should see no long term increase in traffic as a result.
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//
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//Encoding: Timestamp is in seconds, 24 bit 6.18 fixed point, yielding 64s wraparound and 3.8us resolution (one increment for each 477 bytes going out on a 1Gbps interface).
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//
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//Relation to NTP timestamps: abs_send_time_24 = (ntp_timestamp_64 >> 14) & 0x00ffffff ; NTP timestamp is 32 bits for whole seconds, 32 bits fraction of second.
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//
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//Notes: Packets are time stamped when going out, preferably close to metal. Intermediate RTP relays (entities possibly altering the stream) should remove the extension or set its own timestamp.
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uint32_t RtpExt::getAbsSendTime() const {
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CHECK(_type == RtpExtType::abs_send_time && size() >= 3);
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uint32_t ret = 0;
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ret |= (*this)[0] << 16;
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ret |= (*this)[1] << 8;
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ret |= (*this)[2];
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return ret;
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}
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//https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
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// 0 1 2 3
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | 0xBE | 0xDE | length=1 |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID | L=1 |transport-wide sequence number | zero padding |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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uint16_t RtpExt::getTransportCCSeq() const {
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CHECK(_type == RtpExtType::transport_cc && size() >= 2);
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uint16_t ret;
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ret |= (*this)[0] << 8;
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ret |= (*this)[1];
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return ret;
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}
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//https://tools.ietf.org/html/draft-ietf-avtext-sdes-hdr-ext-07
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// 0 1 2 3
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID | len | SDES Item text value ... |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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const string &RtpExt::getSdesMid() const {
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CHECK(_type == RtpExtType::sdes_mid && size() >= 1);
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return *this;
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}
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//https://tools.ietf.org/html/draft-ietf-avtext-rid-06
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//用于simulecast
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//3.1. RTCP 'RtpStreamId' SDES Extension
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//
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// |RtpStreamId=TBD| length | RtpStreamId ...
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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//
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//
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// The RtpStreamId payload is UTF-8 encoded and is not null-terminated.
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//
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// RFC EDITOR NOTE: Please replace TBD with the assigned SDES
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// identifier value.
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//3.2. RTCP 'RepairedRtpStreamId' SDES Extension
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//
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// |Repaired...=TBD| length | RepairRtpStreamId ...
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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//
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//
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// The RepairedRtpStreamId payload is UTF-8 encoded and is not null-
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// terminated.
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//
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// RFC EDITOR NOTE: Please replace TBD with the assigned SDES
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// identifier value.
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string RtpExt::getRtpStreamId() const {
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CHECK(_type == RtpExtType::sdes_rtp_stream_id && size() >= 1);
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return *this;
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}
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string RtpExt::getRepairedRtpStreamId() const {
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CHECK(_type == RtpExtType::sdes_repaired_rtp_stream_id && size() >= 1);
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return *this;
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}
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//http://www.webrtc.org/experiments/rtp-hdrext/video-timing
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//Wire format: 1-byte extension, 13 bytes of data. Total 14 bytes extra per packet (plus 1-3 padding byte in some cases, plus shared 4 bytes for all extensions present: 2 byte magic word 0xBEDE, 2 byte # of extensions).
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//
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//First byte is a flags field. Defined flags:
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//
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//0x01 - extension is set due to timer.
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//0x02 - extension is set because the frame is larger than usual.
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//Both flags may be set at the same time. All remaining 6 bits are reserved and should be ignored.
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//
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//Next, 6 timestamps are stored as 16-bit values in big-endian order, representing delta from the capture time of a packet in ms. Timestamps are, in order:
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//
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//Encode start.
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//Encode finish.
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//Packetization complete.
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//Last packet left the pacer.
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//Reserved for network.
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//Reserved for network (2).
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void RtpExt::getVideoTiming(uint8_t &flags,
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uint16_t &encode_start,
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uint16_t &encode_finish,
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uint16_t &packetization_complete,
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uint16_t &last_pkt_left_pacer,
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uint16_t &reserved_net0,
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uint16_t &reserved_net1) const {
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CHECK(_type == RtpExtType::video_timing && size() >= 13);
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flags = (*this)[0];
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encode_start = (*this)[1] << 8 | (*this)[2];
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encode_finish = (*this)[3] << 8 | (*this)[4];
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packetization_complete = (*this)[5] << 8 | (*this)[6];
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last_pkt_left_pacer = (*this)[7] << 8 | (*this)[8];
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reserved_net0 = (*this)[9] << 8 | (*this)[10];
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reserved_net1 = (*this)[11] << 8 | (*this)[12];
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}
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//http://www.webrtc.org/experiments/rtp-hdrext/color-space
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// 0 1 2 3
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID | L = 3 | primaries | transfer | matrix |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// |range+chr.sit. |
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// +-+-+-+-+-+-+-+-+
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//http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
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//Values:
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//0x00: Unspecified. Default value. Treated the same as an absence of an extension.
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//0x01: Screenshare. Video stream is of a screenshare type.
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//0x02: 摄像头？
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//Notes: Extension shoud be present only in the last packet of key-frames.
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// If attached to other packets it should be ignored.
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// If extension is absent, Unspecified value is assumed.
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uint8_t RtpExt::getVideoContentType() const {
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CHECK(_type == RtpExtType::video_content_type && size() >= 1);
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return (*this)[0];
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}
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//http://www.3gpp.org/ftp/Specs/html-info/26114.htm
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void RtpExt::getVideoOrientation(bool &camera_bit, bool &flip_bit, bool &first_rotation, bool &second_rotation) const {
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CHECK(_type == RtpExtType::video_orientation && size() >= 1);
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uint8_t byte = (*this)[0];
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camera_bit = (byte & 0x08) >> 3;
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flip_bit = (byte & 0x04) >> 2;
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first_rotation = (byte & 0x02) >> 1;
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second_rotation = byte & 0x01;
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}
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//http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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//+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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//| ID | len=2 | MIN delay | MAX delay |
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//+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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void RtpExt::getPlayoutDelay(uint16_t &min_delay, uint16_t &max_delay) const {
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CHECK(_type == RtpExtType::playout_delay && size() >= 3);
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uint32_t bytes = (*this)[0] << 16 | (*this)[1] << 8 | (*this)[2];
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min_delay = (bytes & 0x00FFF000) >> 12;
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max_delay = bytes & 0x00000FFF;
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}
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//urn:ietf:params:rtp-hdrext:toffset
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//https://tools.ietf.org/html/rfc5450
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// 0 1 2 3
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID | len=2 | transmission offset |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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uint32_t RtpExt::getTransmissionOffset() const {
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CHECK(_type == RtpExtType::toffset && size() >= 3);
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return (*this)[0] << 16 | (*this)[1] << 8 | (*this)[2];
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}
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//http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
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// 0 1 2 3
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | ID=? | L=2 |S|E|I|D|B| TID | LID | TL0PICIDX |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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uint8_t RtpExt::getFramemarkingTID() const {
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CHECK(_type == RtpExtType::framemarking && size() >= 3);
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return (*this)[0] & 0x07;
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}
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@ -30,11 +30,17 @@ enum class RtpExtType : uint8_t {
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sdes_repaired_rtp_stream_id = 6,
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video_timing = 7,
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color_space = 8,
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//for firefox
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csrc_audio_level = 9,
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//svc ?
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framemarking = 10,
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video_content_type = 11,
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playout_delay = 12,
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video_orientation = 13,
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toffset = 14,
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reserved = 15,
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// e2e ?
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encrypt
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};
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class RtcMedia;
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@ -51,6 +57,34 @@ public:
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string dumpString() const;
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uint8_t getAudioLevel(bool *vad) const;
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uint32_t getAbsSendTime() const;
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uint16_t getTransportCCSeq() const;
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const string& getSdesMid() const;
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string getRtpStreamId() const;
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string getRepairedRtpStreamId() const;
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void getVideoTiming(uint8_t &flags,
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uint16_t &encode_start,
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uint16_t &encode_finish,
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uint16_t &packetization_complete,
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uint16_t &last_pkt_left_pacer,
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uint16_t &reserved_net0,
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uint16_t &reserved_net1) const;
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uint8_t getVideoContentType() const;
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void getVideoOrientation(bool &camera_bit,
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bool &flip_bit,
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bool &first_rotation,
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bool &second_rotation) const;
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void getPlayoutDelay(uint16_t &min_delay, uint16_t &max_delay) const;
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uint32_t getTransmissionOffset() const;
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uint8_t getFramemarkingTID() const;
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private:
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RtpExtType _type;
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uint8_t _id;
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@ -1297,6 +1297,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
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rtcp_fb = {SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb};
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extmap = {
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RtpExtType::ssrc_audio_level,
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RtpExtType::csrc_audio_level,
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RtpExtType::abs_send_time,
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RtpExtType::transport_cc,
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RtpExtType::sdes_mid,
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@ -1320,7 +1321,8 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
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RtpExtType::video_content_type,
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RtpExtType::playout_delay,
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RtpExtType::video_orientation,
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RtpExtType::toffset
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RtpExtType::toffset,
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RtpExtType::framemarking
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};
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break;
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}
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