mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-25 12:11:36 +08:00
整理并注释代码
This commit is contained in:
xiongziliang
parent
cfda7d8ba6
commit
130a06897f
@ -164,12 +164,16 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *
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if (is_rtp(buf)) {
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if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) {
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onRtp(buf, len);
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} else {
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WarnL;
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}
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return;
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}
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if (is_rtcp(buf)) {
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if (_srtp_session_recv->DecryptSrtcp((uint8_t *) buf, &len)) {
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onRtcp(buf, len);
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} else {
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WarnL;
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}
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return;
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}
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@ -230,6 +234,18 @@ void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sock
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_socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr), flush);
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}
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///////////////////////////////////////////////////////////////////
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bool WebRtcTransportImp::canSendRtp() const{
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auto &sdp = getSdp(SdpType::answer);
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return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly;
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}
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bool WebRtcTransportImp::canRecvRtp() const{
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auto &sdp = getSdp(SdpType::answer);
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return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly;
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}
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void WebRtcTransportImp::onStartWebRTC() {
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if (canRecvRtp()) {
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_push_src = std::make_shared<RtspMediaSourceImp>(DEFAULT_VHOST, "live", "push");
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@ -242,6 +258,7 @@ void WebRtcTransportImp::onStartWebRTC() {
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if (!hit_pan) {
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continue;
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}
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//获取offer端rtp的ssrc和pt相关信息
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auto &ref = _rtp_receiver[plan.pt];
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_ssrc_info[m.rtp_ssrc.ssrc] = &ref;
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ref.plan = &plan;
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@ -273,22 +290,13 @@ void WebRtcTransportImp::onStartWebRTC() {
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}
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}
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bool WebRtcTransportImp::canSendRtp() const{
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auto &sdp = getSdp(SdpType::answer);
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return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly;
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}
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bool WebRtcTransportImp::canRecvRtp() const{
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auto &sdp = getSdp(SdpType::answer);
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return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly;
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}
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void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{
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WebRtcTransport::onCheckSdp(type, sdp);
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if (type != SdpType::answer || !canSendRtp()) {
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return;
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}
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//添加answer sdp的ssrc信息，并且记录发送rtp的pt
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for (auto &m : sdp.media) {
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if (m.type == TrackApplication) {
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continue;
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@ -311,18 +319,14 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
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WebRtcTransport::onRtcConfigure(configure);
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_rtsp_send_sdp.loadFrom(_src->getSdp(), false);
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configure.audio.enable = false;
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configure.video.enable = false;
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//根据rtsp流的相关信息，设置rtc最佳编码
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for (auto &m : _rtsp_send_sdp.media) {
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switch (m.type) {
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case TrackVideo: {
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configure.video.enable = true;
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configure.video.preferred_codec.insert(configure.video.preferred_codec.begin(), getCodecId(m.plan[0].codec));
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break;
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}
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case TrackAudio: {
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configure.audio.enable = true;
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configure.audio.preferred_codec.insert(configure.audio.preferred_codec.begin(),getCodecId(m.plan[0].codec));
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break;
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}
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@ -331,21 +335,27 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
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}
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}
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//添加接收端口candidate信息
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configure.addCandidate(*getIceCandidate());
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}
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SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{
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auto candidate = std::make_shared<SdpAttrCandidate>();
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candidate->foundation = "udpcandidate";
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//rtp端口
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candidate->component = 1;
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candidate->transport = "udp";
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//优先级，单candidate时随便
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candidate->priority = 100;
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//todo 此处修改为配置文件
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candidate->address = SockUtil::get_local_ip();
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candidate->port = _socket->get_local_port();
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candidate->type = "host";
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return candidate;
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}
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///////////////////////////////////////////////////////////////////
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class RtpReceiverImp : public RtpReceiver {
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public:
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RtpReceiverImp( function<void(RtpPacket::Ptr rtp)> cb, function<void(const RtpPacket::Ptr &rtp)> cb_before = nullptr){
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@ -402,11 +412,28 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
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}
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break;
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}
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case RtcpType::RTCP_BYE : {
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//todo 此处应该销毁对象
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break;
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}
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default: break;
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}
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}
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}
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void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
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RtpHeader *rtp = (RtpHeader *) buf;
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//根据接收到的rtp的pt信息，找到该流的信息
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auto it = _rtp_receiver.find(rtp->pt);
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if (it == _rtp_receiver.end()) {
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WarnL;
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return;
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}
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auto &info = it->second;
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//解析并排序rtp
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info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len);
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}
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int makeRtcpPli(char *packet, int len) {
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if (packet == NULL || len != 12)
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return -1;
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@ -419,24 +446,16 @@ int makeRtcpPli(char *packet, int len) {
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return 12;
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}
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void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
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RtpHeader *rtp = (RtpHeader *) buf;
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auto it = _rtp_receiver.find(rtp->pt);
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if (it == _rtp_receiver.end()) {
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WarnL;
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return;
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}
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auto &info = it->second;
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info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len);
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}
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///////////////////////////////////////////////////////////////////
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void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) {
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if(!info.is_common_rtp){
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//todo rtx/red/ulpfec类型的rtp先未处理
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WarnL;
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return;
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}
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if (_pli_ticker.elapsedTime() > 2000) {
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//todo 发送pli
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//todo 定期发送pli
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_pli_ticker.resetTime();
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char rtcpbuf[12];
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makeRtcpPli(rtcpbuf, 12);
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@ -447,7 +466,7 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr
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}
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void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) {
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//todo rtcp相关
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//统计rtp收到的情况，好做rr汇报
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info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
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}
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@ -460,11 +479,8 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
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//设置pt
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rtp->getHeader()->pt = _send_rtp_pt[rtp->type];
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sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush);
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//统计rtp发送情况，好做sr汇报
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_rtp_receiver[_send_rtp_pt[rtp->type]].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
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//还原pt
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rtp->getHeader()->pt = tmp;
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}
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///////////////////////////////////////////////////////////////////
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