完善fci相关代码

src/Rtcp/Rtcp.cpp

@@ -12,6 +12,7 @@
 #include <assert.h>
 #include "Rtcp.h"
 #include "Util/logger.h"
+#include "RtcpFCI.h"
 
 namespace mediakit {
 
@@ -493,18 +494,65 @@ string RtcpFB::dumpString() const {
     printer << RtcpHeader::dumpHeader();
     printer << "ssrc:" << ssrc << "\r\n";
     printer << "ssrc_media:" << ssrc_media << "\r\n";
-    auto fci = (uint8_t *)&ssrc_media + sizeof(ssrc_media);
+    auto fci_data = (uint8_t *)&ssrc_media + sizeof(ssrc_media);
     auto fci_len = getSize() - sizeof(RtcpFB);
-    if (fci_len) {
     switch ((RtcpType) pt) {
         case RtcpType::RTCP_PSFB : {
+            switch ((PSFBType) report_count) {
+                case PSFBType::RTCP_PSFB_SLI : {
+                    FCI_SLI *fci = (FCI_SLI *) fci_data;
+                    fci->check(fci_len);
+                    printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << fci->dumpString();
+                    break;
+                }
+                case PSFBType::RTCP_PSFB_PLI : {
+                    CHECK(fci_len == 0);
+                    printer << "fci:" << psfbTypeToStr((PSFBType) report_count);
+                    break;
+                }
+
+                case PSFBType::RTCP_PSFB_FIR : {
+                    FCI_FIR *fci = (FCI_FIR *) fci_data;
+                    fci->check(fci_len);
+                    printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << fci->dumpString();
+                    break;
+                }
+
+                case PSFBType::RTCP_PSFB_REMB : {
+                    FCI_REMB *fci = (FCI_REMB *) fci_data;
+                    fci->check(fci_len);
+                    printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << fci->dumpString();
+                    break;
+                }
+                default:{
+                    printer << "fci:" << psfbTypeToStr((PSFBType) report_count) << " " << hexdump(fci_data, fci_len);
+                    break;
+                }
+            }
             break;
         }
         case RtcpType::RTCP_RTPFB : {
+            switch ((RTPFBType) report_count) {
+                case RTPFBType::RTCP_RTPFB_NACK : {
+                    FCI_NACK *fci = (FCI_NACK *) fci_data;
+                    fci->check(fci_len);
+                    printer << "fci:" << rtpfbTypeToStr((RTPFBType) report_count) << " " << fci->dumpString();
+                    break;
+                }
+                case RTPFBType::RTCP_RTPFB_TWCC : {
+                    FCI_TWCC *fci = (FCI_TWCC *) fci_data;
+                    fci->check(fci_len);
+                    printer << "fci:" << rtpfbTypeToStr((RTPFBType) report_count) << " " << fci->dumpString(fci_len);
+                    break;
+                }
+                default: {
+                    printer << "fci:" << rtpfbTypeToStr((RTPFBType) report_count) << " " << hexdump(fci_data, fci_len);
                     break;
                 }
             }
-        printer << "fci:" << hexdump(fci, fci_len);
+            break;
+        }
+        default: /*不可达*/ assert(0); break;
     }
     return std::move(printer);
 }

src/Rtcp/Rtcp.h

@@ -85,7 +85,7 @@ namespace mediakit {
         XX(RTCP_PSFB_TSTR, 5)\
         XX(RTCP_PSFB_TSTN, 6)\
         XX(RTCP_PSFB_VBCM, 7) \
-        XX(RTCP_PSFB_AFB, 15)
+        XX(RTCP_PSFB_REMB, 15)
 
 //https://tools.ietf.org/html/rfc4585#section-6.2
 //6.2.   Transport Layer Feedback Messages

src/Rtcp/RtcpFCI.cpp

@@ -453,7 +453,7 @@ map<uint16_t, std::pair<SymbolStatus, uint32_t/*stamp*/> > FCI_TWCC::getPacketCh
 string FCI_TWCC::dumpString(size_t total_size) const {
     _StrPrinter printer;
     auto map = getPacketChunkList(total_size);
-    printer << "twcc fci, base_seq:" << getBaseSeq() << ",pkt_status_count:" << getPacketCount() << ", ref time:" << getReferenceTime() << ", fb count:" << (int)fb_pkt_count << "\n";
+    printer << "twcc fci, base_seq:" << getBaseSeq() << ", pkt_status_count:" << getPacketCount() << ", ref time:" << getReferenceTime() << ", fb count:" << (int)fb_pkt_count << "\n";
     for (auto &pr : map) {
         printer << "rtp seq:" << pr.first <<", packet status:" << (int)(pr.second.first) << ", delta:" << pr.second.second << "\n";
     }

webrtc/Sdp.cpp

@@ -1257,18 +1257,29 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
             //此处调整偏好的编码格式优先级
             preferred_codec = {CodecAAC, CodecG711U, CodecG711A, CodecOpus};
             rtcp_fb = {"transport-cc"};
-            extmap = {"1 urn:ietf:params:rtp-hdrext:ssrc-audio-level"};
+            extmap = {"1 urn:ietf:params:rtp-hdrext:ssrc-audio-level",
+                      "2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
+                      "3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
+                      "4 urn:ietf:params:rtp-hdrext:sdes:mid",
+                      "5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id",
+                      "6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"};
             break;
         }
         case TrackVideo: {
             //此处调整偏好的编码格式优先级
             preferred_codec = {CodecH264, CodecH265};
             rtcp_fb = {"nack", "ccm fir", "nack pli", "goog-remb", "transport-cc"};
-            extmap = {"2 urn:ietf:params:rtp-hdrext:toffset",
+            extmap = {"14 urn:ietf:params:rtp-hdrext:toffset",
-                      "3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
+                      "2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
-                      "4 urn:3gpp:video-orientation",
+                      "13 urn:3gpp:video-orientation",
-                      "5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
+                      "3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
-                      "6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"};
+                      "12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay",
+                      "11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type",
+                      "7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing",
+                      "8 http://www.webrtc.org/experiments/rtp-hdrext/color-space",
+                      "4 urn:ietf:params:rtp-hdrext:sdes:mid",
+                      "5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id",
+                      "6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"};
             break;
         }
         case TrackApplication: {

webrtc/WebRtcTransport.cpp

@@ -544,7 +544,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
             }
             case RtcpType::RTCP_PSFB:
             case RtcpType::RTCP_RTPFB: {
-                InfoL << rtcp->dumpString();
+                InfoL << "\n" << rtcp->dumpString();
                 break;
             }
             default: break;

webrtc/readme.md

@@ -1,37 +1,11 @@
-# 致谢与声明
+14 urn:ietf:params:rtp-hdrext:toffset
-本文件夹下部分文件提取自[MediaSoup](https://github.com/versatica/mediasoup) ，分别为：
+2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
-
+13 urn:3gpp:video-orientation
-- ice相关功能：
+3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
-  - IceServer.cpp
+12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
-  - IceServer.hpp
+11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
-  - StunPacket.cpp
+7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
-  - StunPacket.hpp
+8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
-  - Utils.hpp
+4 urn:ietf:params:rtp-hdrext:sdes:mid
-  
+5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
-- dtls相关功能：
+6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
-   - DtlsTransport.cpp
-   - DtlsTransport.hpp
-   
-- srtp相关功能：
-   - SrtpSession.cpp
-   - SrtpSession.hpp
-   
-
-以上源码有一定的修改和裁剪，感谢MediaSoup开源项目及作者，
-用户在使用本项目的同时，应该同时遵循MediaSoup的开源协议。
-
-同时,在此也感谢开源项目[easy_webrtc_server](https://github.com/Mihawk086/easy_webrtc_server) 及作者，
-在集成MediaSoup相关代码前期，主要参考这个项目。
-
-另外，感谢[big panda](<2381267071@qq.com>) 开发并贡献的webrtc js测试客户端(www/webrtc目录下文件)，
-其开源项目地址为：https://gitee.com/xiongguangjie/zlmrtcclient.js
-
-# 现状与规划
-ZLMediaKit的WebRTC相关功能目前仅供测试与开发，现在还不成熟，后续主要工作有：
-
-- 1、完善webrtc rtcp相关功能，包括丢包重传、带宽检测等功能。
-- 2、实现rtp重传等相关功能。
-- 3、实现simulecast相关功能。
-- 4、fec、rtp扩展等其他功能。
-- 5、如果精力允许，逐步替换MediaSoup相关代码，改用自有版权代码。
-