实现rtc转rtsp

src/Rtcp/Rtcp.cpp

@@ -82,6 +82,11 @@ string RtcpHeader::dumpString() const {
             RtcpSdes *rtcp = (RtcpSdes *)this;
             return rtcp->dumpString();
         }
+
+        case RtcpType::RTCP_PSFB: {
+            RtcpPli *rtcp = (RtcpPli *)this;
+            return rtcp->dumpString();
+        }
         default: return StrPrinter << dumpHeader() << hexdump((char *)this + sizeof(*this), length << 2);
     }
 }
@@ -105,6 +110,12 @@ void RtcpHeader::net2Host(size_t len){
             sdes->net2Host(len);
             break;
         }
+
+        case RtcpType::RTCP_PSFB: {
+            RtcpPli *pli = (RtcpPli *)this;
+            pli->net2Host(len);
+            break;
+        }
         default: throw std::runtime_error(StrPrinter << "未处理的rtcp包:" << rtcpTypeToStr((RtcpType) this->pt));
     }
 }
@@ -121,7 +132,7 @@ vector<RtcpHeader *> RtcpHeader::loadFromBytes(char *data, size_t len){
             ret.emplace_back(rtcp);
         } catch (std::exception &ex) {
             //不能处理的rtcp包，或者无法解析的rtcp包，忽略掉
-            WarnL << ex.what();
+            WarnL << ex.what() << ",长度为:" << rtcp_len;
         }
         ptr += rtcp_len;
         remain -= rtcp_len;
@@ -402,4 +413,31 @@ vector<SdesItem *> RtcpSdes::getItemList() {
     return ret;
 }
 
+////////////////////////////////////////////////////////////////////
+
+std::shared_ptr<RtcpPli> RtcpPli::create() {
+    auto bytes = alignSize(sizeof(RtcpPli));
+    auto ptr = (RtcpRR *) new char[bytes];
+    setupHeader(ptr, RtcpType::RTCP_PSFB, 1, bytes);
+    return std::shared_ptr<RtcpPli>((RtcpPli *) ptr, [](RtcpPli *ptr) {
+        delete[] (char *) ptr;
+    });
+}
+
+string RtcpPli::dumpString() const {
+    _StrPrinter printer;
+    printer << RtcpHeader::dumpHeader();
+    printer << "ssrc:" << ssrc << "\r\n";
+    printer << "ssrc_media:" << ssrc_media;
+    return std::move(printer);
+}
+
+void RtcpPli::net2Host(size_t size) {
+    static const size_t kMinSize = sizeof(RtcpPli);
+    CHECK_MIN_SIZE(size, kMinSize);
+    RtcpHeader::net2Host();
+    ssrc = ntohl(ssrc);
+    ssrc_media = ntohl(ssrc_media);
+}
+
 }//namespace mediakit

src/Rtcp/Rtcp.h

@@ -456,6 +456,35 @@ private:
     void net2Host(size_t size);
 } PACKED;
 
+//PLI
+class RtcpPli : public RtcpHeader {
+public:
+    friend class RtcpHeader;
+    uint32_t ssrc;
+    uint32_t ssrc_media;
+
+public:
+    /**
+     * 创建SDES包，只赋值了RtcpHeader以及SdesItem对象的length和text部分
+     * @param item_text SdesItem列表，只赋值length和text部分
+     * @return SDES包
+     */
+    static std::shared_ptr<RtcpPli> create();
+
+private:
+    /**
+    * 打印字段详情
+    * 使用net2Host转换成主机字节序后才可使用此函数
+    */
+    string dumpString() const;
+
+    /**
+     * 网络字节序转换为主机字节序
+     * @param size 字节长度，防止内存越界
+     */
+    void net2Host(size_t size);
+} PACKED;
+
 #if defined(_WIN32)
 #pragma pack(pop)
 #endif // defined(_WIN32)

webrtc/WebRtcTransport.cpp

@@ -253,6 +253,9 @@ void WebRtcTransportImp::onStartWebRTC() {
         _push_src->setSdp(rtsp_sdp);
 
         for (auto &m : getSdp(SdpType::offer).media) {
+            if (m.type == TrackVideo) {
+                _recv_video_ssrc = m.rtp_ssrc.ssrc;
+            }
             for (auto &plan : m.plan) {
                 auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt);
                 if (!hit_pan) {
@@ -416,6 +419,10 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
                 //todo 此处应该销毁对象
                 break;
             }
+            case RtcpType::RTCP_PSFB: {
+//                InfoL << rtcp->dumpString();
+                break;
+            }
             default: break;
         }
     }
@@ -434,18 +441,6 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
     info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len);
 }
 
-int makeRtcpPli(char *packet, int len) {
-    if (packet == NULL || len != 12)
-        return -1;
-    memset(packet, 0, len);
-    RtcpHeader *rtcp = (RtcpHeader *) packet;
-    rtcp->version = 2;
-    rtcp->pt = (uint8_t) RtcpType::RTCP_PSFB;
-    rtcp->report_count = 1;
-    rtcp->length = htons((len / 4) - 1);
-    return 12;
-}
-
 ///////////////////////////////////////////////////////////////////
 
 void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) {
@@ -457,9 +452,10 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr
     if (_pli_ticker.elapsedTime() > 2000) {
         //todo 定期发送pli
         _pli_ticker.resetTime();
-        char rtcpbuf[12];
-        makeRtcpPli(rtcpbuf, 12);
-        sendRtcpPacket(rtcpbuf, 12, true);
+        auto pli = RtcpPli::create();
+        pli->ssrc = htonl(0);
+        pli->ssrc_media = htonl(_recv_video_ssrc);
+        sendRtcpPacket((char *) pli.get(), sizeof(RtcpPli), true);
         InfoL << "send pli";
     }
     _push_src->onWrite(std::move(rtp), false);

webrtc/WebRtcTransport.h

@@ -157,6 +157,7 @@ private:
     RtspMediaSourceImp::Ptr _push_src;
     unordered_map<uint8_t, RtpPayloadInfo> _rtp_receiver;
     unordered_map<uint32_t, RtpPayloadInfo*> _ssrc_info;
+    uint32_t _recv_video_ssrc;
     Ticker _pli_ticker;
 };