From d5eb486f5445fbdd1cc39de1202dd2af31252b03 Mon Sep 17 00:00:00 2001 From: xia-chu <771730766@qq.com> Date: Sat, 8 May 2021 10:49:09 +0800 Subject: [PATCH] =?UTF-8?q?=E4=BF=AE=E6=94=B9rtp=E7=9A=84ext=20id?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit 优化代码结构 --- webrtc/WebRtcTransport.cpp | 81 +++++++++++++++++++++++++------------- webrtc/WebRtcTransport.h | 8 ++-- 2 files changed, 59 insertions(+), 30 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index eda327a0..74f544dd 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -390,10 +390,8 @@ bool WebRtcTransportImp::canRecvRtp() const{ } void WebRtcTransportImp::onStartWebRTC() { + //获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息 for (auto &m : getSdp(SdpType::offer).media) { - if (m.type == TrackVideo) { - _recv_video_ssrc = m.rtp_ssrc.ssrc; - } for (auto &plan : m.plan) { auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt); if (!hit_pan) { @@ -413,6 +411,14 @@ void WebRtcTransportImp::onStartWebRTC() { onBeforeSortedRtp(ref, rtp); }); } + if (m.type != TrackApplication) { + //记录rtp ext类型与id的关系，方便接收或发送rtp时修改rtp ext id + for (auto &pr : m.extmap) { + auto ext_type = RtpExt::getExtType(pr.second.ext); + _rtp_ext_id_to_type.emplace(pr.second.id, ext_type); + _rtp_ext_type_to_id.emplace(ext_type, pr.second.id); + } + } } if (canRecvRtp()) { @@ -431,6 +437,19 @@ void WebRtcTransportImp::onStartWebRTC() { strongSelf->onSendRtp(rtp, ++i == pkt->size()); }); }); + + RtcSession rtsp_send_sdp; + rtsp_send_sdp.loadFrom(_play_src->getSdp(), false); + for (auto &m : getSdp(SdpType::answer).media) { + if (m.type == TrackApplication) { + continue; + } + auto rtsp_media = rtsp_send_sdp.getMedia(m.type); + if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) { + //记录发送rtp时约定的pt，届时发送rtp时需要修改pt + _send_rtp_pt[m.type] = m.plan[0].pt; + } + } } } @@ -440,6 +459,7 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ return; } + //修改sdp的ip、端口信息 GET_CONFIG(string, extern_ip, RTC::kExternIP); for (auto &m : sdp.media) { m.addr.reset(); @@ -455,9 +475,6 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ return; } - RtcSession rtsp_send_sdp; - rtsp_send_sdp.loadFrom(_play_src->getSdp(), false); - for (auto &m : sdp.media) { if (m.type == TrackApplication) { continue; @@ -470,11 +487,6 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){ m.rtx_ssrc.ssrc = RTX_SSRC_OFFSET + m.rtp_ssrc.ssrc; m.rtx_ssrc.cname = RTX_CNAME; } - auto rtsp_media = rtsp_send_sdp.getMedia(m.type); - if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) { - //记录发送rtp的pt - _send_rtp_pt[m.type] = m.plan[0].pt; - } } } @@ -557,6 +569,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { it->second->rtcp_context_recv->onRtcp(sr); auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->items.ssrc, sr->ssrc); sendRtcpPacket(rr->data(), rr->size(), true); + } else { + WarnL << "未识别的sr rtcp包:" << sr->ssrc; } break; } @@ -568,6 +582,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { if (it != _rtp_info_ssrc.end()) { auto sr = it->second->rtcp_context_send->createRtcpSR(rr->items.ssrc); sendRtcpPacket(sr->data(), sr->size(), true); + } else { + WarnL << "未识别的rr rtcp包:" << rr->ssrc; } break; } @@ -577,6 +593,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { for (auto ssrc : bye->getSSRC()) { auto it = _rtp_info_ssrc.find(*ssrc); if (it == _rtp_info_ssrc.end()) { + WarnL << "未识别的bye rtcp包:" << *ssrc; continue; } _rtp_info_pt.erase(it->second->plan->pt); @@ -587,7 +604,6 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { } case RtcpType::RTCP_PSFB: case RtcpType::RTCP_RTPFB: { - InfoL << "\n" << rtcp->dumpString(); break; } default: break; @@ -617,36 +633,46 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr //todo rtx/red/ulpfec类型的rtp先未处理 return; } - if (_pli_ticker.elapsedTime() > 2000) { + if (info.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) { //定期发送pli请求关键帧，方便非rtc等协议 _pli_ticker.resetTime(); - sendRtcpPli(_recv_video_ssrc); + sendRtcpPli(rtp->getSSRC()); //开启remb，则发送remb包调节比特率 GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate); if (remb_bit_rate && getSdp(SdpType::answer).supportRtcpFb(SdpConst::kRembRtcpFb)) { - sendRtcpRemb(_recv_video_ssrc, remb_bit_rate); + sendRtcpRemb(rtp->getSSRC(), remb_bit_rate); } } - auto header = rtp->getHeader(); - auto ext_map = RtpExt::getExtValue(header, *(info.media)); - for (auto &pr : ext_map) { - if (rtp->type == TrackVideo) { - InfoL << pr.second.dumpString(); - } else { - DebugL << pr.second.dumpString(); - } - //推流时修改ext id为统一的id，播放时再修改为对方设置的ext id - pr.second.setExtId((uint8_t) pr.first); - } - if (_push_src) { _push_src->onWrite(std::move(rtp), false); } } +void setExtType(RtpExt &ext, uint8_t tp) {} +void setExtType(RtpExt &ext, RtpExtType tp) { + ext.setType(tp); +} + +template +static void changeRtpExtId(const RtpPacket::Ptr &rtp, const Type &map) { + auto header = rtp->getHeader(); + auto ext_map = RtpExt::getExtValue(header); + for (auto &pr : ext_map) { + auto it = map.find((Type::key_type) pr.first); + if (it == map.end()) { + WarnL << "未处理的rtp ext, 类型不识别:" << (int) pr.first; + continue; + } + setExtType(pr.second, it->first); + setExtType(pr.second, it->second); + pr.second.setExtId((uint8_t) it->second); + } +} + void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { + changeRtpExtId(rtp, _rtp_ext_id_to_type); //统计rtp收到的情况，好做rr汇报 info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize); } @@ -657,6 +683,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){ //忽略，对方不支持该编码类型 return; } + changeRtpExtId(rtp, _rtp_ext_type_to_id); _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt); //统计rtp发送情况，好做sr汇报 diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 9cc64664..84394534 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -211,8 +211,6 @@ private: Ticker _alive_ticker; //pli rtcp计时器 Ticker _pli_ticker; - //rtc rtp推流的视频ssrc - uint32_t _recv_video_ssrc; //记录协商的rtp的pt类型 uint8_t _send_rtp_pt[2] = {0xFF, 0xFF}; //复合udp端口，接收一切rtp与rtcp @@ -225,8 +223,12 @@ private: RtspMediaSource::RingType::RingReader::Ptr _reader; //根据rtp的pt获取相关信息 unordered_map _rtp_info_pt; - //根据推流端rtp的ssrc获取相关信息 + //根据推流端rtcp的ssrc获取相关信息 unordered_map _rtp_info_ssrc; + //发送rtp时需要修改rtp ext id + unordered_map _rtp_ext_type_to_id; + //接收rtp时需要修改rtp ext id + unordered_map _rtp_ext_id_to_type; };