Merge branch 'master' of github.com:ZLMediaKit/ZLMediaKit

78bcd9e8 · ziyue · 4700ab77 · 78d014cf · 2bb688aa · 78bcd9e8
Commit 78bcd9e8 authored Nov 30, 2021 by ziyue
--- a/ZLToolKit @ 2bb688aa
+++ b/ZLToolKit @ 2bb688aa
-Subproject commit 28c3d20c3a94b6e72bcd2d4e7057d28b8215594a
+Subproject commit 2bb688aa9b29b79f262e3f90613c177ebac109ac
--- a/README.md
+++ b/README.md
@@ -164,8 +164,9 @@ bash build_docker_images.sh
   - [DotNetCore的RESTful客户端](https://github.com/MingZhuLiu/ZLMediaKit.DotNetCore.Sdk)
 - 播放器
+   - [基于wasm支持H265的播放器](https://github.com/numberwolf/h265web.js)
   - [基于MSE的websocket-fmp4播放器](https://github.com/v354412101/wsPlayer) 
 ## 授权协议
 本项目自有代码使用宽松的MIT协议，在保留版权信息的情况下可以自由应用于各自商用、非商业的项目。

--- a/server/main.cpp
+++ b/server/main.cpp
@@ -194,8 +194,6 @@ int start_main(int argc,char *argv[]) {
        string ssl_file = cmd_main["ssl"];
        int threads = cmd_main["threads"];
-        setThreadName("main thread");
        //设置日志
        Logger::Instance().add(std::make_shared<ConsoleChannel>("ConsoleChannel", logLevel));
 #ifndef ANDROID
@@ -248,28 +246,28 @@ int start_main(int argc,char *argv[]) {
        //简单的telnet服务器，可用于服务器调试，但是不能使用23端口，否则telnet上了莫名其妙的现象
        //测试方法:telnet 127.0.0.1 9000
-        TcpServer::Ptr shellSrv = std::make_shared<TcpServer>();
+        auto shellSrv = std::make_shared<TcpServer>();
        //rtsp[s]服务器, 可用于诸如亚马逊echo show这样的设备访问
-        TcpServer::Ptr rtspSrv = std::make_shared<TcpServer>();;
+        auto rtspSrv = std::make_shared<TcpServer>();;
-        TcpServer::Ptr rtspSSLSrv = std::make_shared<TcpServer>();;
+        auto rtspSSLSrv = std::make_shared<TcpServer>();;
        //rtmp[s]服务器
-        TcpServer::Ptr rtmpSrv = std::make_shared<TcpServer>();;
+        auto rtmpSrv = std::make_shared<TcpServer>();;
-        TcpServer::Ptr rtmpsSrv = std::make_shared<TcpServer>();;
+        auto rtmpsSrv = std::make_shared<TcpServer>();;
        //http[s]服务器
-        TcpServer::Ptr httpSrv = std::make_shared<TcpServer>();;
+        auto httpSrv = std::make_shared<TcpServer>();;
-        TcpServer::Ptr httpsSrv = std::make_shared<TcpServer>();;
+        auto httpsSrv = std::make_shared<TcpServer>();;
 #if defined(ENABLE_RTPPROXY)
        //GB28181 rtp推流端口，支持UDP/TCP
-        RtpServer::Ptr rtpServer = std::make_shared<RtpServer>();
+        auto rtpServer = std::make_shared<RtpServer>();
 #endif//defined(ENABLE_RTPPROXY)
 #if defined(ENABLE_WEBRTC)
        //webrtc udp服务器
-        UdpServer::Ptr rtcSrv = std::make_shared<UdpServer>();
+        auto rtcSrv = std::make_shared<UdpServer>();
        rtcSrv->setOnCreateSocket([](const EventPoller::Ptr &poller, const Buffer::Ptr &buf, struct sockaddr *, int) {
            if (!buf) {
                return Socket::createSocket(poller, false);

--- a/src/Extension/H264Rtp.cpp
+++ b/src/Extension/H264Rtp.cpp
@@ -268,10 +268,6 @@ void H264RtpEncoder::packRtpStapA(const char *ptr, size_t len, uint32_t pts, boo
 bool H264RtpEncoder::inputFrame(const Frame::Ptr &frame) {
    auto ptr = frame->data() + frame->prefixSize();
    switch (H264_TYPE(ptr[0])) {
-        case H264Frame::NAL_AUD:
-        case H264Frame::NAL_SEI : {
-            return false;
-        }
        case H264Frame::NAL_SPS: {
            _sps = Frame::getCacheAbleFrame(frame);
            return true;

--- a/webrtc/Sdp.cpp
+++ b/webrtc/Sdp.cpp
@@ -1737,16 +1737,21 @@ RETRY:
 void RtcConfigure::setPlayRtspInfo(const string &sdp){
    RtcSession session;
+    video.direction = RtpDirection::inactive;
+    audio.direction = RtpDirection::inactive;
    session.loadFrom(sdp);
    for (auto &m : session.media) {
        switch (m.type) {
            case TrackVideo : {
+                video.direction = RtpDirection::sendonly;
                _rtsp_video_plan = std::make_shared<RtcCodecPlan>(m.plan[0]);
                video.preferred_codec.clear();
                video.preferred_codec.emplace_back(getCodecId(_rtsp_video_plan->codec));
                break;
            }
            case TrackAudio : {
+                 audio.direction = RtpDirection::sendonly;
                _rtsp_audio_plan = std::make_shared<RtcCodecPlan>(m.plan[0]);
                audio.preferred_codec.clear();
                audio.preferred_codec.emplace_back(getCodecId(_rtsp_audio_plan->codec));

--- a/webrtc/TwccContext.cpp
+++ b/webrtc/TwccContext.cpp
@@ -21,7 +21,7 @@ enum class ExtSeqStatus : int {
 void TwccContext::onRtp(uint32_t ssrc, uint16_t twcc_ext_seq, uint64_t stamp_ms) {
    switch ((ExtSeqStatus) checkSeqStatus(twcc_ext_seq)) {
-        case ExtSeqStatus::jumped: /*回环后，收到回环前的大ext seq包,过滤掉*/ return;
+        case ExtSeqStatus::jumped: /*seq异常,过滤掉*/ return;
        case ExtSeqStatus::looped: /*回环，触发发送twcc rtcp*/ onSendTwcc(ssrc); break;
        case ExtSeqStatus::normal: break;
        default: /*不可达*/assert(0); break;
@@ -56,16 +56,29 @@ int TwccContext::checkSeqStatus(uint16_t twcc_ext_seq) const {
        return (int) ExtSeqStatus::normal;
    }
    auto max = _rtp_recv_status.rbegin()->first;
-    if (max > 0xFF00 && twcc_ext_seq < 0xFF) {
+    auto delta = (int32_t) twcc_ext_seq - (int32_t) max;
-        //发生回环了
+    if (delta > 0 && delta < 0xFFFF / 2) {
+        //正常增长
+        return (int) ExtSeqStatus::normal;
+    }
+    if (delta < -0xFF00) {
+        //回环
        TraceL << "rtp twcc ext seq looped:" << max << " -> " << twcc_ext_seq;
        return (int) ExtSeqStatus::looped;
    }
-    if (twcc_ext_seq - max > 0xFFFF / 2) {
+    if (delta > 0xFF00) {
-        TraceL << "rtp twcc ext seq jumped:" << max << " -> " << twcc_ext_seq;
+        //回环后收到前面大的乱序的包，无法处理，丢弃
+        TraceL << "rtp twcc ext seq jumped after looped:" << max << " -> " << twcc_ext_seq;
        return (int) ExtSeqStatus::jumped;
    }
-    return (int) ExtSeqStatus::normal;
+    auto min = _rtp_recv_status.begin()->first;
+    if (min <= twcc_ext_seq || twcc_ext_seq <= max) {
+        //正常回退
+        return (int) ExtSeqStatus::normal;
+    }
+    //seq莫名的大幅增加或减少，无法处理，丢弃
+    TraceL << "rtp twcc ext seq jumped:" << max << " -> " << twcc_ext_seq;
+    return (int) ExtSeqStatus::jumped;
 }
 void TwccContext::onSendTwcc(uint32_t ssrc) {

--- a/webrtc/WebRtcSession.cpp
+++ b/webrtc/WebRtcSession.cpp
@@ -55,7 +55,7 @@ void WebRtcSession::onRecv(const Buffer::Ptr &buffer) {
        //只允许寻找一次transport
        _find_transport = false;
        auto user_name = getUserName(buffer);
-        _identifier = user_name + '-' + to_string(reinterpret_cast<uint64_t>(this));
+        _identifier = to_string(getSock()->rawFD()) + '-' + user_name;
        auto transport = WebRtcTransportManager::Instance().getItem(user_name);
        CHECK(transport && transport->getPoller()->isCurrentThread());
        transport->setSession(shared_from_this());

--- a/webrtc/WebRtcTransport.cpp
+++ b/webrtc/WebRtcTransport.cpp
@@ -43,9 +43,12 @@ static onceToken token([]() {
 }//namespace RTC
+static atomic<uint64_t> s_key{0};
 WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) {
    _poller = poller;
-    _identifier = to_string(reinterpret_cast<uint64_t>(this));
+    _identifier = "zlm_"+to_string(++s_key);
+    _packet_pool.setSize(64);
 }
 void WebRtcTransport::onCreate(){
@@ -69,7 +72,7 @@ const string &WebRtcTransport::getIdentifier() const {
 //////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
 void WebRtcTransport::OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) {
-    onSendSockData((char *) packet->GetData(), packet->GetSize(), (struct sockaddr_in *) tuple);
+    sendSockData((char *) packet->GetData(), packet->GetSize(), tuple);
 }
 void WebRtcTransport::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) {
@@ -110,7 +113,7 @@ void WebRtcTransport::OnDtlsTransportConnected(
 }
 void WebRtcTransport::OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
-    onSendSockData((char *)data, len);
+    sendSockData((char *)data, len, nullptr);
 }
 void WebRtcTransport::OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) {
@@ -132,10 +135,10 @@ void WebRtcTransport::OnDtlsTransportApplicationDataReceived(const RTC::DtlsTran
 }
 //////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
-void WebRtcTransport::onSendSockData(const char *buf, size_t len, bool flush){
+void WebRtcTransport::sendSockData(const char *buf, size_t len, RTC::TransportTuple *tuple){
-    auto tuple = _ice_server->GetSelectedTuple();
+    auto pkt = _packet_pool.obtain();
-    assert(tuple);
+    pkt->assign(buf, len);
-    onSendSockData(buf, len, (struct sockaddr_in *) tuple, flush);
+    onSendSockData(std::move(pkt), true, tuple ? tuple : _ice_server->GetSelectedTuple());
 }
 RTC::TransportTuple* WebRtcTransport::getSelectedTuple() const{
@@ -266,23 +269,28 @@ void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tup
 void WebRtcTransport::sendRtpPacket(const char *buf, int len, bool flush, void *ctx) {
    if (_srtp_session_send) {
+        auto pkt = _packet_pool.obtain();
        //预留rtx加入的两个字节
-        CHECK((size_t)len + SRTP_MAX_TRAILER_LEN + 2 <= sizeof(_srtp_buf));
+        pkt->setCapacity((size_t) len + SRTP_MAX_TRAILER_LEN + 2);
-        memcpy(_srtp_buf, buf, len);
+        pkt->assign(buf, len);
-        onBeforeEncryptRtp((char *) _srtp_buf, len, ctx);
+        onBeforeEncryptRtp(pkt->data(), len, ctx);
-        if (_srtp_session_send->EncryptRtp(_srtp_buf, &len)) {
+        if (_srtp_session_send->EncryptRtp(reinterpret_cast<uint8_t *>(pkt->data()), &len)) {
-            onSendSockData((char *) _srtp_buf, len, flush);
+            pkt->setSize(len);
+            onSendSockData(std::move(pkt), flush);
        }
    }
 }
-void WebRtcTransport::sendRtcpPacket(const char *buf, int len, bool flush, void *ctx){
+void WebRtcTransport::sendRtcpPacket(const char *buf, int len, bool flush, void *ctx) {
    if (_srtp_session_send) {
-        CHECK((size_t)len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf));
+        auto pkt = _packet_pool.obtain();
-        memcpy(_srtp_buf, buf, len);
+        //预留rtx加入的两个字节
-        onBeforeEncryptRtcp((char *) _srtp_buf, len, ctx);
+        pkt->setCapacity((size_t) len + SRTP_MAX_TRAILER_LEN + 2);
-        if (_srtp_session_send->EncryptRtcp(_srtp_buf, &len)) {
+        pkt->assign(buf, len);
-            onSendSockData((char *) _srtp_buf, len, flush);
+        onBeforeEncryptRtcp(pkt->data(), len, ctx);
+        if (_srtp_session_send->EncryptRtcp(reinterpret_cast<uint8_t *>(pkt->data()), &len)) {
+            pkt->setSize(len);
+            onSendSockData(std::move(pkt), flush);
        }
    }
 }
@@ -313,7 +321,6 @@ void WebRtcTransportImp::onCreate(){
 WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) {
    InfoL << getIdentifier();
-    _packet_pool.setSize(64);
 }
 WebRtcTransportImp::~WebRtcTransportImp() {
@@ -325,16 +332,14 @@ void WebRtcTransportImp::onDestory() {
    unregisterSelf();
 }
-void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush) {
+void WebRtcTransportImp::onSendSockData(Buffer::Ptr buf, bool flush, RTC::TransportTuple *tuple) {
    if (!_session) {
-        WarnL << "send data failed:" << len;
+        WarnL << "send data failed:" << buf->size();
        return;
    }
-    auto ptr = _packet_pool.obtain();
-    ptr->assign(buf, len);
    //一次性发送一帧的rtp数据，提高网络io性能
    _session->setSendFlushFlag(flush);
-    _session->send(std::move(ptr));
+    _session->send(std::move(buf));
 }
 ///////////////////////////////////////////////////////////////////
@@ -807,7 +812,8 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r
        }
 #endif
    } else {
-        WarnL << "send rtx rtp:" << rtp->getSeq();
+        //发送rtx重传包
+        TraceL << "send rtx rtp:" << rtp->getSeq();
    }
    pair<bool/*rtx*/, MediaTrack *> ctx{rtx, track.get()};
    sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, &ctx);

--- a/webrtc/WebRtcTransport.h
+++ b/webrtc/WebRtcTransport.h
@@ -135,7 +135,7 @@ protected:
    virtual void onStartWebRTC() = 0;
    virtual void onRtcConfigure(RtcConfigure &configure) const;
    virtual void onCheckSdp(SdpType type, RtcSession &sdp) = 0;
-    virtual void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) = 0;
+    virtual void onSendSockData(Buffer::Ptr buf, bool flush = true, RTC::TransportTuple *tuple = nullptr) = 0;
    virtual void onRtp(const char *buf, size_t len, uint64_t stamp_ms) = 0;
    virtual void onRtcp(const char *buf, size_t len) = 0;
@@ -149,7 +149,7 @@ protected:
    void sendRtcpPli(uint32_t ssrc);
 private:
-    void onSendSockData(const char *buf, size_t len, bool flush = true);
+    void sendSockData(const char *buf, size_t len, RTC::TransportTuple *tuple);
    void setRemoteDtlsFingerprint(const RtcSession &remote);
 protected:
@@ -157,7 +157,6 @@ protected:
    RtcSession::Ptr _answer_sdp;
 private:
-    uint8_t _srtp_buf[2000];
    string _identifier;
    EventPoller::Ptr _poller;
    std::shared_ptr<RTC::IceServer> _ice_server;
@@ -165,6 +164,8 @@ private:
    std::shared_ptr<RTC::SrtpSession> _srtp_session_send;
    std::shared_ptr<RTC::SrtpSession> _srtp_session_recv;
    Ticker _ticker;
+    //循环池
+    ResourcePool<BufferRaw> _packet_pool;
 };
 class RtpChannel;
@@ -232,7 +233,7 @@ public:
 protected:
    WebRtcTransportImp(const EventPoller::Ptr &poller);
    void onStartWebRTC() override;
-    void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) override;
+    void onSendSockData(Buffer::Ptr buf, bool flush = true, RTC::TransportTuple *tuple = nullptr) override;
    void onCheckSdp(SdpType type, RtcSession &sdp) override;
    void onRtcConfigure(RtcConfigure &configure) const override;
@@ -279,8 +280,6 @@ private:
    unordered_map<uint32_t/*ssrc*/, MediaTrack::Ptr> _ssrc_to_track;
    //根据接收rtp的pt获取相关信息
    unordered_map<uint8_t/*pt*/, std::unique_ptr<WrappedMediaTrack>> _pt_to_track;
-    //循环池
-    ResourcePool<BufferRaw> _packet_pool;
 };
 class WebRtcTransportManager {