增加 RtcpContextForSend/RtcpContextForRecv作为RtcpContext子类

c98e85ca · Johnny · 290b3f37 · c98e85ca · c98e85ca · c98e85ca
Commit c98e85ca authored Oct 13, 2021 by Johnny
--- a/src/Rtcp/RtcpContext.cpp
+++ b/src/Rtcp/RtcpContext.cpp
@@ -14,12 +14,16 @@ using namespace toolkit;
 namespace mediakit {
-RtcpContext::RtcpContext(bool is_receiver) {
-    _is_receiver = is_receiver;
-}
 void RtcpContext::onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes) {
-    if (_is_receiver) {
+    ++_packets;
+    _bytes += bytes;
+    _last_rtp_stamp = stamp;
+    _last_ntp_stamp_ms = ntp_stamp_ms;
+}
+void RtcpContextForRecv::onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes) {
+    {
        //接收者才做复杂的统计运算
        auto sys_stamp = getCurrentMillisecond();
        if (_last_rtp_sys_stamp) {
@@ -57,11 +61,7 @@ void RtcpContext::onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uin
        _last_rtp_seq = seq;
        _last_rtp_sys_stamp = sys_stamp;
    }
+    RtcpContext::onRtp(seq, stamp, ntp_stamp_ms, sample_rate, bytes);
-    ++_packets;
-    _bytes += bytes;
-    _last_rtp_stamp = stamp;
-    _last_ntp_stamp_ms = ntp_stamp_ms;
 }
 void RtcpContext::onRtcp(RtcpHeader *rtcp) {
@@ -115,9 +115,9 @@ uint32_t RtcpContext::getRtt(uint32_t ssrc) const {
 }
 size_t RtcpContext::getExpectedPackets() const {
-    if (!_is_receiver) {
+    throw std::runtime_error("没有实现, rtp发送者无法统计应收包数");
-        throw std::runtime_error("rtp发送者无法统计应收包数");
+}
-    }
+size_t RtcpContextForRecv::getExpectedPackets() const {
    return (_seq_cycles << 16) + _seq_max - _seq_base + 1;
 }
@@ -129,9 +129,10 @@ size_t RtcpContext::getExpectedPacketsInterval() {
 }
 size_t RtcpContext::getLost() {
-    if (!_is_receiver) {
+    throw std::runtime_error("没有实现, rtp发送者无法统计丢包率");
-        throw std::runtime_error("rtp发送者无法统计丢包率");
+}
-    }
+size_t RtcpContextForRecv::getLost() {
    return getExpectedPackets() - _packets;
 }
@@ -143,9 +144,10 @@ size_t RtcpContext::geLostInterval() {
 }
 Buffer::Ptr RtcpContext::createRtcpSR(uint32_t rtcp_ssrc) {
-    if (_is_receiver) {
+    throw std::runtime_error("没有实现, rtp接收者尝试发送sr包");
-        throw std::runtime_error("rtp接收者尝试发送sr包");
+}
-    }
+Buffer::Ptr RtcpContextForSend::createRtcpSR(uint32_t rtcp_ssrc) {
    auto rtcp = RtcpSR::create(0);
    rtcp->setNtpStamp(_last_ntp_stamp_ms);
    rtcp->rtpts = htonl(_last_rtp_stamp);
@@ -165,9 +167,11 @@ Buffer::Ptr RtcpContext::createRtcpSR(uint32_t rtcp_ssrc) {
 }
 Buffer::Ptr RtcpContext::createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc) {
-    if (!_is_receiver) {
+    throw std::runtime_error("没有实现, rtp发送者尝试发送rr包");
-        throw std::runtime_error("rtp发送者尝试发送rr包");
+}
-    }
+Buffer::Ptr RtcpContextForRecv::createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc) {
    auto rtcp = RtcpRR::create(1);
    rtcp->ssrc = htonl(rtcp_ssrc);

--- a/src/Rtcp/RtcpContext.h
+++ b/src/Rtcp/RtcpContext.h
@@ -20,11 +20,7 @@ namespace mediakit {
 class RtcpContext {
 public:
    using Ptr = std::shared_ptr<RtcpContext>;
-    /**
+    virtual ~RtcpContext() = default;
-     * 创建rtcp上下文
-     * @param is_receiver 是否为rtp接收者，接收者更消耗性能
-     */
-    RtcpContext(bool is_receiver);
    /**
     * 输出或输入rtp时调用
@@ -34,7 +30,7 @@ public:
     * @param rtp rtp时间戳采样率，视频一般为90000，音频一般为采样率
     * @param bytes rtp数据长度
     */
-    void onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes);
+    virtual void onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes);
    /**
     * 输入sr rtcp包
@@ -45,19 +41,19 @@ public:
    /**
     * 计算总丢包数
     */
-    size_t getLost();
+    virtual size_t getLost();
    /**
     * 返回理应收到的rtp数
     */
-    size_t getExpectedPackets() const;
+    virtual size_t getExpectedPackets() const;
    /**
     * 创建SR rtcp包
     * @param rtcp_ssrc rtcp的ssrc
     * @return rtcp包
     */
-    Buffer::Ptr createRtcpSR(uint32_t rtcp_ssrc);
+    virtual Buffer::Ptr createRtcpSR(uint32_t rtcp_ssrc);
    /**
     * 创建RR rtcp包
@@ -65,7 +61,7 @@ public:
     * @param rtp_ssrc rtp的ssrc
     * @return rtcp包
     */
-    Buffer::Ptr createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc);
+    virtual Buffer::Ptr createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc);
    /**
     * 获取rtt
@@ -84,9 +80,7 @@ public:
     */
    size_t geLostInterval();
-private:
+protected:
-    //是否为接收者
-    bool _is_receiver;
    //时间戳抖动值
    double _jitter = 0;
    //收到或发送的rtp的字节数
@@ -120,5 +114,17 @@ private:
    map<uint32_t/*last_sr_lsr*/, uint64_t/*ntp stamp*/> _sender_report_ntp;
 };
+class RtcpContextForSend : public RtcpContext {
+public:
+    Buffer::Ptr createRtcpSR(uint32_t rtcp_ssrc) override;
+};
+class RtcpContextForRecv : public RtcpContext {
+public:
+    void onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes) override;
+    Buffer::Ptr createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc) override;
+    size_t getExpectedPackets() const override;
+    size_t getLost() override;
+};
 }//namespace mediakit
 #endif //ZLMEDIAKIT_RTCPCONTEXT_H
--- a/src/Rtp/RtpServer.cpp
+++ b/src/Rtp/RtpServer.cpp
@@ -23,11 +23,11 @@ RtpServer::~RtpServer() {
    }
 }
-class RtcpHelper : public RtcpContext, public std::enable_shared_from_this<RtcpHelper> {
+class RtcpHelper : public RtcpContextForRecv, public std::enable_shared_from_this<RtcpHelper> {
 public:
    using Ptr = std::shared_ptr<RtcpHelper>;
-    RtcpHelper(Socket::Ptr rtcp_sock, uint32_t sample_rate) : RtcpContext(true){
+    RtcpHelper(Socket::Ptr rtcp_sock, uint32_t sample_rate) {
        _rtcp_sock = std::move(rtcp_sock);
        _sample_rate = sample_rate;
    }

--- a/src/Rtsp/RtspPlayer.cpp
+++ b/src/Rtsp/RtspPlayer.cpp
@@ -206,7 +206,7 @@ void RtspPlayer::handleResDESCRIBE(const Parser& parser) {
    }
    _rtcp_context.clear();
    for (auto &track : _sdp_track) {
-        _rtcp_context.emplace_back(std::make_shared<RtcpContext>(true));
+        _rtcp_context.emplace_back(std::make_shared<RtcpContextForRecv>());
    }
    sendSetup(0);
 }

--- a/src/Rtsp/RtspPusher.cpp
+++ b/src/Rtsp/RtspPusher.cpp
@@ -179,7 +179,7 @@ void RtspPusher::sendAnnounce() {
    }
    _rtcp_context.clear();
    for (auto &track : _track_vec) {
-        _rtcp_context.emplace_back(std::make_shared<RtcpContext>(false));
+        _rtcp_context.emplace_back(std::make_shared<RtcpContextForSend>());
    }
    _on_res_func = std::bind(&RtspPusher::handleResAnnounce, this, placeholders::_1);
    sendRtspRequest("ANNOUNCE", _url, {}, src->getSdp());

--- a/src/Rtsp/RtspSession.cpp
+++ b/src/Rtsp/RtspSession.cpp
@@ -257,7 +257,7 @@ void RtspSession::handleReq_ANNOUNCE(const Parser &parser) {
        }
        _rtcp_context.clear();
        for (auto &track : _sdp_track) {
-            _rtcp_context.emplace_back(std::make_shared<RtcpContext>(true));
+            _rtcp_context.emplace_back(std::make_shared<RtcpContextForRecv>());
        }
        _push_src = std::make_shared<RtspMediaSourceImp>(_media_info._vhost, _media_info._app, _media_info._streamid);
        _push_src->setListener(dynamic_pointer_cast<MediaSourceEvent>(shared_from_this()));
@@ -418,7 +418,7 @@ void RtspSession::onAuthSuccess() {
        }
        strongSelf->_rtcp_context.clear();
        for (auto &track : strongSelf->_sdp_track) {
-            strongSelf->_rtcp_context.emplace_back(std::make_shared<RtcpContext>(false));
+            strongSelf->_rtcp_context.emplace_back(std::make_shared<RtcpContextForSend>());
        }
        strongSelf->_sessionid = makeRandStr(12);
        strongSelf->_play_src = rtsp_src;

--- a/webrtc/WebRtcTransport.cpp
+++ b/webrtc/WebRtcTransport.cpp
@@ -420,7 +420,7 @@ void WebRtcTransportImp::onStartWebRTC() {
        track->offer_ssrc_rtx = m_offer->getRtxSSRC();
        track->plan_rtp = &m_answer.plan[0];;
        track->plan_rtx = m_answer.getRelatedRtxPlan(track->plan_rtp->pt);
-        track->rtcp_context_send = std::make_shared<RtcpContext>(false);
+        track->rtcp_context_send = std::make_shared<RtcpContextForSend>();
        //send ssrc --> MediaTrack
        _ssrc_to_track[track->answer_ssrc_rtp] = track;
@@ -656,7 +656,7 @@ private:
 private:
    NackContext _nack_ctx;
-    RtcpContext _rtcp_context{true};
+    RtcpContextForRecv _rtcp_context;
    EventPoller::Ptr _poller;
    DelayTask::Ptr _delay_task;
    function<void(const FCI_NACK &nack)> _on_nack;