Parse out ssrcs in REMB message (needed for ViCE) .

Review URL: https://webrtc-codereview.appspot.com/486003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2061 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/rtp_rtcp/source/rtcp_receiver.cc

@@ -1082,10 +1082,14 @@ RTCPReceiver::HandlePsfbApp(RTCPUtility::RTCPParserV2& rtcpParser,
                             RTCPPacketInformation& rtcpPacketInformation)
 {
     RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
-    if (pktType == RTCPUtility::kRtcpPsfbRembItemCode)
+    if (pktType == RTCPUtility::kRtcpPsfbRembCode)
     {
-        HandleREMBItem(rtcpParser, rtcpPacketInformation);
+        pktType = rtcpParser.Iterate();
-        rtcpParser.Iterate();
+        if (pktType == RTCPUtility::kRtcpPsfbRembItemCode)
+        {
+            HandleREMBItem(rtcpParser, rtcpPacketInformation);
+            rtcpParser.Iterate();
+        }
     }
 }
 
@@ -1117,12 +1121,11 @@ void
 RTCPReceiver::HandleREMBItem(RTCPUtility::RTCPParserV2& rtcpParser,
                              RTCPPacketInformation& rtcpPacketInformation)
 {
-    rtcpParser.Iterate();
     const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
 
     rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRemb;
-    rtcpPacketInformation.receiverEstimatedMaxBitrate = rtcpPacket.REMB.BitRate;
+    rtcpPacketInformation.receiverEstimatedMaxBitrate =
-    // TODO(pwestin) send up SSRCs and do a sanity check
+        rtcpPacket.REMBItem.BitRate;
 }
 
 // no need for critsect we have _criticalSectionRTCPReceiver

src/modules/rtp_rtcp/source/rtcp_utility.cc

@@ -14,7 +14,6 @@
 #include <cmath>   // ceil
 #include <cassert>
 
-
 namespace webrtc {
 // RTCPParserV2 : currently read only
 
@@ -1095,7 +1094,9 @@ RTCPUtility::RTCPParserV2::ParseFBCommon(const RTCPCommonHeader& header)
             _state = State_PSFB_FIRItem;
             return true;
         case 15:
-            _packetType            = kRtcpPsfbAppCode;
+            _packetType                = kRtcpPsfbAppCode;
+            _packet.PSFBAPP.SenderSSRC = senderSSRC;
+            _packet.PSFBAPP.MediaSSRC  = mediaSSRC;
 
             _state = State_PSFB_AppItem;
             return true;
@@ -1223,11 +1224,11 @@ RTCPUtility::RTCPParserV2::ParsePsfbAppItem()
         EndCurrentBlock();
         return false;
     }
-    _packetType = kRtcpPsfbRembItemCode;
+    _packetType = kRtcpPsfbRembCode;
     _state = State_PSFB_REMBItem;
     return true;
 }
- 
+
 bool
 RTCPUtility::RTCPParserV2::ParsePsfbREMBItem()
 {
@@ -1241,7 +1242,7 @@ RTCPUtility::RTCPParserV2::ParsePsfbREMBItem()
         return false;
     }
 
-    const WebRtc_UWord8 numSSRC = *_ptrRTCPData++;
+    _packet.REMBItem.NumberOfSSRCs = *_ptrRTCPData++;
     const WebRtc_UWord8 brExp = (_ptrRTCPData[0] >> 2) & 0x3F;
 
     WebRtc_UWord32 brMantissa = (_ptrRTCPData[0] & 0x03) << 16;
@@ -1249,9 +1250,26 @@ RTCPUtility::RTCPParserV2::ParsePsfbREMBItem()
     brMantissa += (_ptrRTCPData[2]);
 
     _ptrRTCPData += 3; // Fwd read data
-    _packet.REMB.BitRate = (brMantissa << brExp);
+    _packet.REMBItem.BitRate = (brMantissa << brExp);
 
-    _ptrRTCPData += 4 * numSSRC; // Ignore the SSRCs for now
+    const ptrdiff_t length_ssrcs = _ptrRTCPBlockEnd - _ptrRTCPData;
+    if (length_ssrcs < 4 * _packet.REMBItem.NumberOfSSRCs)
+    {
+        _state = State_TopLevel;
+
+        EndCurrentBlock();
+        return false;
+    }
+
+    _packetType = kRtcpPsfbRembItemCode;
+
+    for (int i = 0; i < _packet.REMBItem.NumberOfSSRCs; i++)
+    {
+        _packet.REMBItem.SSRCs[i] = *_ptrRTCPData++ << 24;
+        _packet.REMBItem.SSRCs[i] += *_ptrRTCPData++ << 16;
+        _packet.REMBItem.SSRCs[i] += *_ptrRTCPData++ << 8;
+        _packet.REMBItem.SSRCs[i] += *_ptrRTCPData++;
+    }
     return true;
 }
 

src/modules/rtp_rtcp/source/rtcp_utility.h

@@ -179,9 +179,16 @@ namespace RTCPUtility {
         WebRtc_UWord16 NumberOfValidBits;
         WebRtc_UWord8  NativeBitString[RTCP_RPSI_DATA_SIZE];
     };
-    struct RTCPPacketPSFBREMB
+    struct RTCPPacketPSFBAPP
+    {
+        WebRtc_UWord32 SenderSSRC;
+        WebRtc_UWord32 MediaSSRC;
+    };
+    struct RTCPPacketPSFBREMBItem
     {
         WebRtc_UWord32 BitRate;
+        WebRtc_UWord8 NumberOfSSRCs;
+        WebRtc_UWord32 SSRCs[MAX_NUMBER_OF_REMB_FEEDBACK_SSRCS];
     };
     // generic name APP
     struct RTCPPacketAPP
@@ -210,7 +217,8 @@ namespace RTCPUtility {
         RTCPPacketPSFBSLI         SLI;
         RTCPPacketPSFBSLIItem     SLIItem;
         RTCPPacketPSFBRPSI        RPSI;
-        RTCPPacketPSFBREMB        REMB;
+        RTCPPacketPSFBAPP         PSFBAPP;
+        RTCPPacketPSFBREMBItem    REMBItem;
 
         RTCPPacketRTPFBTMMBR      TMMBR;
         RTCPPacketRTPFBTMMBRItem  TMMBRItem;
@@ -251,6 +259,7 @@ namespace RTCPUtility {
         kRtcpPsfbSliCode,
         kRtcpPsfbSliItemCode,
         kRtcpPsfbAppCode,
+        kRtcpPsfbRembCode,
         kRtcpPsfbRembItemCode,
 
         // RFC5104

src/modules/rtp_rtcp/source/rtp_rtcp_config.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -31,6 +31,7 @@ enum { RTCP_NUMBER_OF_SR            = 60 };
 
 enum { MAX_NUMBER_OF_TEMPORAL_ID    = 8 };          // RFC
 enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID  = 128 };// RFC
+enum { MAX_NUMBER_OF_REMB_FEEDBACK_SSRCS = 255 };
 
 enum { BW_HISTORY_SIZE          = 35};
 

src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -86,6 +86,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const WebRtc_Word32 id,
   _rtpReceiver(id, audio, clock, this),
   _rtcpSender(id, audio, clock, this),
   _rtcpReceiver(id, clock, this),
+  _bandwidthManagement(id),
   _owns_clock(false),
   _clock(*clock),
   _id(id),
@@ -104,7 +105,6 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const WebRtc_Word32 id,
   _deadOrAliveActive(false),
   _deadOrAliveTimeoutMS(0),
   _deadOrAliveLastTimer(0),
-  _bandwidthManagement(id),
   _receivedNTPsecsAudio(0),
   _receivedNTPfracAudio(0),
   _RTCPArrivalTimeSecsAudio(0),

src/modules/rtp_rtcp/source/rtp_rtcp_impl.h

@@ -480,7 +480,7 @@ public:
         WebRtc_UWord32* available_bandwidth) const;
 
     virtual void SetRemoteSSRC(const WebRtc_UWord32 SSRC);
-    
+
     virtual WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 sendReport);
 
     virtual RateControlRegion OnOverUseStateUpdate(const RateControlInput& rateControlInput);
@@ -546,6 +546,8 @@ protected:
     RTCPSender                _rtcpSender;
     RTCPReceiver              _rtcpReceiver;
 
+    BandwidthManagement       _bandwidthManagement;
+
     bool                      _owns_clock;
     RtpRtcpClock&             _clock;
 private:
@@ -573,8 +575,6 @@ private:
     WebRtc_UWord32        _deadOrAliveLastTimer;
 
     // receive side
-    BandwidthManagement   _bandwidthManagement;
-
     WebRtc_UWord32        _receivedNTPsecsAudio;
     WebRtc_UWord32        _receivedNTPfracAudio;
     WebRtc_UWord32        _RTCPArrivalTimeSecsAudio;