新增webrtc配置选项nackAudioRtpSize、preferred_tcp

nackAudioRtpSize: 音频nack包中rtp个数，减小此值可以让nack包响应更灵敏
preferred_tcp: 是否优先采用webrtc over tcp模式

conf/config.ini

@@ -412,10 +412,14 @@ nackMaxMS=3000
 nackMaxCount=15
 #nack重传频率，rtt的倍数
 nackIntervalRatio=1.0
-#nack包中rtp个数，减小此值可以让nack包响应更灵敏
+#视频nack包中rtp个数，减小此值可以让nack包响应更灵敏
 nackRtpSize=8
+#音频nack包中rtp个数，减小此值可以让nack包响应更灵敏
+nackAudioRtpSize=4
 #是否尝试过滤 b帧
 bfilter=0
+# 是否优先采用webrtc over tcp模式
+preferred_tcp=0
 
 [srt]
 #srt播放推流、播放超时时间,单位秒

webrtc/IceTransport.cpp

@@ -1175,7 +1175,7 @@ void IceAgent::connectivityCheck(const Pair::Ptr &pair, CandidateTuple& candidat
 }
 
 void IceAgent::tryTriggerredCheck(const Pair::Ptr& pair) {
-    DebugL;
+    // DebugL;
     //FIXME 暂不实现,因为当前实现基本收到candidate就会发起check
 }
 

webrtc/Nack.cpp

@@ -46,6 +46,7 @@ const string kNackIntervalRatio = RTC_FIELD "nackIntervalRatio";
 // nack包中rtp个数，减小此值可以让nack包响应更灵敏  [AUTO-TRANSLATED:12393868]
 // Number of rtp in nack packet, reducing this value can make nack packet response more sensitive
 const string kNackRtpSize = RTC_FIELD "nackRtpSize";
+const string kNackAudioRtpSize = RTC_FIELD "nackAudioRtpSize";
 
 static onceToken token([]() {
     mINI::Instance()[kMaxRtpCacheMS] = 5 * 1000;
@@ -55,6 +56,7 @@ static onceToken token([]() {
     mINI::Instance()[kNackMaxCount] = 15;
     mINI::Instance()[kNackIntervalRatio] = 1.0f;
     mINI::Instance()[kNackRtpSize] = 8;
+    mINI::Instance()[kNackAudioRtpSize] = 4;
 });
 
 } // namespace Rtc
@@ -155,7 +157,8 @@ int64_t NackList::getNtpStamp(uint16_t seq) {
 
 ////////////////////////////////////////////////////////////////////////////////////////////////
 
-NackContext::NackContext() {
+NackContext::NackContext(TrackType type) {
+    _type = type;
     setOnNack(nullptr);
 }
 
@@ -217,7 +220,9 @@ void NackContext::makeNack(uint16_t max_seq, bool flush) {
     // 最多生成5个nack包，防止seq大幅跳跃导致一直循环  [AUTO-TRANSLATED:9cc5da25]
     // Generate at most 5 nack packets to prevent seq from jumping significantly and causing continuous loops
     auto max_nack = 5u;
-    GET_CONFIG(uint32_t, nack_rtpsize, Rtc::kNackRtpSize);
+    GET_CONFIG(uint32_t, nack_video_rtpsize, Rtc::kNackRtpSize);
+    GET_CONFIG(uint32_t, nack_audio_rtpsize, Rtc::kNackAudioRtpSize);
+    auto nack_rtpsize = _type == TrackVideo ? nack_video_rtpsize : nack_audio_rtpsize;
     // kNackRtpSize must between 0 and 16
     nack_rtpsize = std::min<uint32_t>(nack_rtpsize, FCI_NACK::kBitSize);
     while (_nack_seq != max_seq && max_nack--) {

webrtc/Nack.h

@@ -55,7 +55,7 @@ public:
     using Ptr = std::shared_ptr<NackContext>;
     using onNack = std::function<void(const FCI_NACK &nack)>;
 
-    NackContext();
+    NackContext(TrackType type = TrackVideo);
 
     void received(uint16_t seq, bool is_rtx = false);
     void setOnNack(onNack cb);
@@ -71,6 +71,7 @@ private:
 private:
     bool _started = false;
     int _rtt = 50;
+    TrackType _type;
     onNack _cb;
     std::set<uint16_t> _seq;
     // 最新nack包中的rtp seq值  [AUTO-TRANSLATED:6984d95a]

webrtc/WebRtcTransport.cpp

@@ -82,6 +82,7 @@ const string kMinBitrate = RTC_FIELD "min_bitrate";
 // 数据通道设置  [AUTO-TRANSLATED:2dc48bc3]
 // Data channel setting
 const string kDataChannelEcho = RTC_FIELD "datachannel_echo";
+const string kPreferredTcp = RTC_FIELD "preferred_tcp";
 
 static onceToken token([]() {
     mINI::Instance()[kTimeOutSec] = 15;
@@ -105,6 +106,7 @@ static onceToken token([]() {
     mINI::Instance()[kIceTransportPolicy] = 0;  // 默认值：不限制(kAll)
     mINI::Instance()[kIceUfrag] = "ZLMediaKit";
     mINI::Instance()[kIcePwd] = "ZLMediaKit";
+    mINI::Instance()[kPreferredTcp] = 0;
 });
 
 } // namespace Rtc
@@ -1102,7 +1104,7 @@ void WebRtcTransportImp::setIceCandidate(vector<SdpAttrCandidate> cands) {
 
 class RtpChannel : public RtpTrackImp, public std::enable_shared_from_this<RtpChannel> {
 public:
-    RtpChannel(EventPoller::Ptr poller, RtpTrackImp::OnSorted cb, function<void(const FCI_NACK &nack)> on_nack) {
+    RtpChannel(TrackType type, EventPoller::Ptr poller, RtpTrackImp::OnSorted cb, function<void(const FCI_NACK &nack)> on_nack) : _nack_ctx(type){
         _poller = std::move(poller);
         _on_nack = std::move(on_nack);
         setOnSorted(std::move(cb));
@@ -1314,7 +1316,7 @@ void WebRtcTransportImp::createRtpChannel(const string &rid, uint32_t ssrc, Medi
     // rid --> RtpReceiverImp
     auto &ref = track.rtp_channel[rid];
     weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
-    ref = std::make_shared<RtpChannel>(
+    ref = std::make_shared<RtpChannel>(track.media->type,
         getPoller(), [&track, this, rid](RtpPacket::Ptr rtp) mutable { onSortedRtp(track, rid, std::move(rtp)); },
         [&track, weak_self, ssrc](const FCI_NACK &nack) mutable {
             // nack发送可能由定时器异步触发  [AUTO-TRANSLATED:186d6723]
@@ -1631,6 +1633,7 @@ WebRtcPluginManager &WebRtcPluginManager::Instance() {
 }
 
 void WebRtcPluginManager::registerPlugin(const string &type, Plugin cb) {
+    InfoL << "Load webrtc plugin:" << type;
     lock_guard<mutex> lck(_mtx_creator);
     _map_creator[type] = std::move(cb);
 }
@@ -1667,6 +1670,7 @@ void echo_plugin(SocketHelper& sender, const WebRtcArgs &args, const onCreateWeb
     cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
 }
 
+template<typename Type>
 void push_plugin(SocketHelper& sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
     MediaInfo info(args["url"]);
     Broadcast::PublishAuthInvoker invoker = [cb, info](const string &err, const ProtocolOption &option) mutable {
@@ -1711,7 +1715,7 @@ void push_plugin(SocketHelper& sender, const WebRtcArgs &args, const onCreateWeb
             push_src_ownership = push_src->getOwnership();
             push_src->setProtocolOption(option);
         }
-        auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option, 
+        auto rtc = Type::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option,
             WebRtcTransport::Role::PEER, WebRtcTransport::SignalingProtocols::WHEP_WHIP);
         push_src->setListener(rtc);
         cb(*rtc);
@@ -1782,9 +1786,12 @@ static void setWebRtcArgs(const WebRtcArgs &args, WebRtcInterface &rtc) {
         }
     }
 
-    bool preferred_tcp = args["preferred_tcp"];
-    {
+    auto preferred_tcp = args["preferred_tcp"];
+    if (!preferred_tcp.empty()) {
         rtc.setPreferredTcp(preferred_tcp);
+    } else {
+        GET_CONFIG(bool, s_preferred_tcp, Rtc::kPreferredTcp);
+        rtc.setPreferredTcp(s_preferred_tcp);
     }
 
     {
@@ -1832,7 +1839,7 @@ static onceToken s_rtc_auto_register([]() {
     // Enable echo plugin only in debug mode
     WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
 #endif
-    WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
+    WebRtcPluginManager::Instance().registerPlugin("push", push_plugin<WebRtcPusher>);
     WebRtcPluginManager::Instance().registerPlugin("play", play_plugin<WebRtcPlayer>);
     WebRtcPluginManager::Instance().registerPlugin("talk", play_plugin<WebRtcTalk>);
 

webrtc/WebRtcTransport.h

@@ -286,7 +286,7 @@ public:
 
 struct WrappedMediaTrack {
     MediaTrack::Ptr track;
-    explicit WrappedMediaTrack(MediaTrack::Ptr ptr): track(ptr) {}
+    explicit WrappedMediaTrack(MediaTrack::Ptr ptr): track(std::move(ptr)) {}
     virtual ~WrappedMediaTrack() {}
     virtual void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) = 0;
 };