完善rtsp sdp匹配

webrtc/Sdp.cpp

@@ -4,7 +4,9 @@
 
 #include "Sdp.h"
 #include "Common/Parser.h"
+#include "Rtsp/Rtsp.h"
 #include <inttypes.h>
+using namespace mediakit;
 
 using onCreateSdpItem = function<SdpItem::Ptr(const string &key, const string &value)>;
 static map<string, onCreateSdpItem, StrCaseCompare> sdpItemCreator;
@@ -596,11 +598,11 @@ void SdpAttrFmtp::parse(const string &str)  {
     auto vec = split(str.substr(pos + 1), ";");
     for (auto &item : vec) {
         trim(item);
-        auto pr_vec = split(item, "=");
+        auto pos = item.find('=');
-        if (pr_vec.size() != 2) {
+        if(pos == string::npos){
             SDP_THROW();
         }
-        arr.emplace_back(std::make_pair(pr_vec[0], pr_vec[1]));
+        arr.emplace_back(std::make_pair(item.substr(0, pos), item.substr(pos + 1)));
     }
     if (arr.empty()) {
         SDP_THROW();
@@ -746,7 +748,7 @@ string SdpAttrCandidate::toString() const  {
     return SdpItem::toString();
 }
 
-void RtcSession::loadFrom(const string &str) {
+void RtcSession::loadFrom(const string &str, bool check) {
     RtcSessionSdp sdp;
     sdp.parse(str);
 
@@ -887,12 +889,17 @@ void RtcSession::loadFrom(const string &str) {
             auto &plan = rtc_media.plan.back();
             auto rtpmap_it = rtpmap_map.find(pt);
             if (rtpmap_it == rtpmap_map.end()) {
-                throw std::invalid_argument(StrPrinter << "该pt不存在相对于的a=rtpmap:" << pt);
+                plan.pt = pt;
+                plan.codec = RtpPayload::getCodecId(pt);
+                plan.sample_rate = RtpPayload::getClockRate(pt);
+                plan.channel = RtpPayload::getAudioChannel(pt);
+            } else {
+                plan.pt = rtpmap_it->second.pt;
+                plan.codec = rtpmap_it->second.codec;
+                plan.sample_rate = rtpmap_it->second.sample_rate;
+                plan.channel = rtpmap_it->second.channel;
             }
-            plan.pt = rtpmap_it->second.pt;
+
-            plan.codec = rtpmap_it->second.codec;
-            plan.sample_rate = rtpmap_it->second.sample_rate;
-            plan.channel = rtpmap_it->second.channel;
             auto fmtp_it = fmtp_map.find(pt);
             if (fmtp_it != fmtp_map.end()) {
                 plan.fmtp = fmtp_it->second.arr;
@@ -905,7 +912,9 @@ void RtcSession::loadFrom(const string &str) {
     }
 
     group = sdp.getItemClass<SdpAttrGroup>('a', "group");
-    checkValid();
+    if (check) {
+        checkValid();
+    }
 }
 
 std::shared_ptr<SdpItem> wrapSdpAttr(SdpItem::Ptr item){

webrtc/Sdp.h

@@ -649,7 +649,7 @@ public:
     vector<RtcMedia> media;
     SdpAttrGroup group;
 
-    void loadFrom(const string &sdp);
+    void loadFrom(const string &sdp, bool check = true);
     void checkValid() const;
     string toString() const;
     RtcMedia *getMedia(TrackType type);

webrtc/WebRtcTransport.cpp

@@ -109,7 +109,6 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer){
     fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
     RtcConfigure configure;
     configure.setDefaultSetting(_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::recvonly, fingerprint);
-    configure.addCandidate(*getIceCandidate());
     onRtcConfigure(configure);
 
     //// 生成answer sdp ////
@@ -217,6 +216,7 @@ void WebRtcTransportImp::onStartWebRTC() {
 }
 
 void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
+    //需要修改pt
     InfoL << flush;
 }
 
@@ -237,6 +237,36 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{
     }
 }
 
+void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
+    WebRtcTransport::onRtcConfigure(configure);
+
+    RtcSession sdp;
+    sdp.loadFrom(_src->getSdp(), false);
+
+    configure.audio.enable = false;
+    configure.video.enable = false;
+
+    for (auto &m : sdp.media) {
+        switch (m.type) {
+            case TrackVideo: {
+                configure.video.enable = true;
+                configure.video.preferred_codec = {getCodecId(m.plan[0].codec)};
+                break;
+            }
+            case TrackAudio: {
+                configure.audio.enable = true;
+                configure.audio.preferred_codec = {getCodecId(m.plan[0].codec)};
+                break;
+            }
+            default:
+                break;
+        }
+    }
+
+    configure.addCandidate(*getIceCandidate());
+}
+
+
 void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush) {
     auto ptr = BufferRaw::create();
     ptr->assign(buf, len);

webrtc/WebRtcTransport.h

@@ -74,8 +74,6 @@ protected:
     virtual void onStartWebRTC() = 0;
     virtual void onRtcConfigure(RtcConfigure &configure) const {}
     virtual void onCheckSdp(SdpType type, RtcSession &sdp) const;
-
-    virtual SdpAttrCandidate::Ptr getIceCandidate() const = 0;
     virtual void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) = 0;
 
     virtual void onRtp(const char *buf, size_t len) = 0;
@@ -116,8 +114,8 @@ protected:
     void onStartWebRTC() override;
     void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) override;
     void onCheckSdp(SdpType type, RtcSession &sdp) const override;
+    void onRtcConfigure(RtcConfigure &configure) const override;
 
-    SdpAttrCandidate::Ptr getIceCandidate() const override;
     void onRtp(const char *buf, size_t len) override;
     void onRtcp(const char *buf, size_t len) override;
 
@@ -125,6 +123,7 @@ private:
     WebRtcTransportImp(const EventPoller::Ptr &poller);
     void onDestory() override;
     void onSendRtp(const RtpPacket::Ptr &rtp, bool flush);
+    SdpAttrCandidate::Ptr getIceCandidate() const;
 
 private:
     Socket::Ptr _socket;