From 12be56493e48710f6f5c4e647e7ced991016cab8 Mon Sep 17 00:00:00 2001 From: ziyue <1213642868@qq.com> Date: Wed, 12 May 2021 15:40:40 +0800 Subject: [PATCH] =?UTF-8?q?=E4=BC=98=E5=8C=96=E4=BB=A3=E7=A0=81?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- webrtc/WebRtcTransport.cpp | 76 ++++++++++++++++++-------------------- webrtc/WebRtcTransport.h | 9 +++-- 2 files changed, 40 insertions(+), 45 deletions(-) diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index 36738ebb..50cf0c64 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -414,27 +414,26 @@ void WebRtcTransportImp::onStartWebRTC() { } auto m_with_ssrc = getSdpWithSSRC().getMedia(m.type); //获取offer端rtp的ssrc和pt相关信息 - auto &ref = _rtp_info_pt[plan.pt]; - ref.plan = &plan; - ref.media = m_with_ssrc; - ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid; - if (ref.is_common_rtp) { + auto info = std::make_shared(); + _rtp_info_pt.emplace(plan.pt, info); + info->plan = &plan; + info->media = m_with_ssrc; + info->is_common_rtp = getCodecId(plan.codec) != CodecInvalid; + if (info->is_common_rtp) { //rtp - _rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref; + _rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = info; } else { //rtx auto apt = atoi(plan.getFmtp("apt").data()); - ref.plan_apt = m_with_ssrc->getPlan(apt); + info->plan_apt = m.getPlan(apt); } - ref.rtcp_context_recv = std::make_shared(ref.plan->sample_rate, true); - ref.rtcp_context_send = std::make_shared(ref.plan->sample_rate, false); - ref.receiver = std::make_shared([&ref, this](RtpPacket::Ptr rtp) mutable{ - onSortedRtp(ref, std::move(rtp)); - }, [&ref, this](const RtpPacket::Ptr &rtp) mutable { - onBeforeSortedRtp(ref, rtp); + info->rtcp_context_recv = std::make_shared(info->plan->sample_rate, true); + info->rtcp_context_send = std::make_shared(info->plan->sample_rate, false); + info->receiver = std::make_shared([info, this](RtpPacket::Ptr rtp) mutable { + onSortedRtp(*info, std::move(rtp)); }); - ref.nack_ctx.setOnNack([&ref, this](const FCI_NACK &nack) mutable{ - onNack(ref, nack); + info->nack_ctx.setOnNack([info, this](const FCI_NACK &nack) mutable { + onNack(*info, nack); }); } if (m.type != TrackApplication) { @@ -475,7 +474,7 @@ void WebRtcTransportImp::onStartWebRTC() { auto it = _rtp_info_pt.find(m.plan[0].pt); CHECK(it != _rtp_info_pt.end()); //记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc - _send_rtp_info[m.type] = &it->second; + _send_rtp_info[m.type] = it->second; } } } @@ -564,9 +563,8 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{ class RtpReceiverImp : public RtpReceiver { public: - RtpReceiverImp( function cb, function cb_before = nullptr){ + RtpReceiverImp( function cb){ _on_sort = std::move(cb); - _on_before_sort = std::move(cb_before); } ~RtpReceiverImp() override = default; @@ -580,15 +578,8 @@ protected: _on_sort(std::move(rtp)); } - void onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index) override { - if (_on_before_sort) { - _on_before_sort(rtp); - } - } - private: function _on_sort; - function _on_before_sort; }; void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { @@ -695,8 +686,11 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) { } void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { - _bytes_usage += len; - _alive_ticker.resetTime(); + if (!rtx) { + _bytes_usage += len; + _alive_ticker.resetTime(); + } + RtpHeader *rtp = (RtpHeader *) buf; //根据接收到的rtp的pt信息，找到该流的信息 auto it = _rtp_info_pt.find(rtp->pt); @@ -705,11 +699,11 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { return; } auto &info = it->second; - if (info.is_common_rtp) { + if (info->is_common_rtp) { //这是普通的rtp数据 auto seq = ntohs(rtp->seq); #if 0 - if (!rtx && info.media->type == TrackVideo && seq % 100 == 0) { + if (!rtx && info->media->type == TrackVideo && seq % 100 == 0) { //此处模拟接受丢包 DebugL << "recv dropped:" << seq; return; @@ -717,10 +711,16 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { #endif if (!rtx) { //统计rtp接受情况，便于生成nack rtcp包 - info.nack_ctx.received(seq); + info->nack_ctx.received(seq); + //修改ext id至统一 + changeRtpExtId(rtp, _rtp_ext_id_to_type); + //时间戳转换成毫秒 + auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan->sample_rate; + //统计rtp收到的情况，好做rr汇报 + info->rtcp_context_recv->onRtp(seq, stamp_ms, len); } //解析并排序rtp - info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len); + info->receiver->inputRtp(info->media->type, info->plan->sample_rate, (uint8_t *) buf, len); return; } @@ -735,8 +735,8 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { auto origin_seq = payload[0] << 8 | payload[1]; InfoL << "received rtx rtp: " << origin_seq; rtp->seq = htons(origin_seq); - rtp->ssrc = htonl(info.media->rtp_rtx_ssrc[0].ssrc); - rtp->pt = info.plan_apt->pt; + rtp->ssrc = htonl(info->media->rtp_rtx_ssrc[0].ssrc); + rtp->pt = info->plan_apt->pt; memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf); buf += 2; len -= 2; @@ -752,12 +752,6 @@ void WebRtcTransportImp::onNack(RtpPayloadInfo &info, const FCI_NACK &nack) { /////////////////////////////////////////////////////////////////// -void WebRtcTransportImp::onBeforeSortedRtp(RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) { - changeRtpExtId(rtp->getHeader(), _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); -} - void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp) { 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 @@ -779,7 +773,7 @@ void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp) { /////////////////////////////////////////////////////////////////// void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx){ - auto info = _send_rtp_info[rtp->type]; + auto &info = _send_rtp_info[rtp->type]; if (!info) { //忽略，对方不支持该编码类型 return; @@ -798,7 +792,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r } else { WarnL << "send rtx rtp:" << rtp->getSeq(); } - sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info); + sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info.get()); _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize; } diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index 35046f30..8fc1ddca 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -342,6 +342,8 @@ private: class RtpPayloadInfo { public: + using Ptr = std::shared_ptr; + bool is_common_rtp; const RtcCodecPlan *plan; const RtcCodecPlan *plan_apt; @@ -354,7 +356,6 @@ private: }; void onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp); - void onBeforeSortedRtp(RtpPayloadInfo &info, const RtpPacket::Ptr &rtp); void onNack(RtpPayloadInfo &info, const FCI_NACK &nack); private: @@ -371,7 +372,7 @@ private: //pli rtcp计时器 Ticker _pli_ticker; //记录协商的发送rtp的pt和ssrc - RtpPayloadInfo* _send_rtp_info[2] = {nullptr, nullptr}; + RtpPayloadInfo::Ptr _send_rtp_info[2]; //复合udp端口，接收一切rtp与rtcp Socket::Ptr _socket; //推流的rtsp源 @@ -381,9 +382,9 @@ private: //播放rtsp源的reader对象 RtspMediaSource::RingType::RingReader::Ptr _reader; //根据rtp的pt获取相关信息 - unordered_map _rtp_info_pt; + unordered_map _rtp_info_pt; //根据rtcp的ssrc获取相关信息 - unordered_map _rtp_info_ssrc; + unordered_map _rtp_info_ssrc; //发送rtp时需要修改rtp ext id map _rtp_ext_type_to_id; //接收rtp时需要修改rtp ext id