Bug 1343691 - fix missing rtcp stats;r=jib

Omitting the RTT when it is not available breaks a lot of tests (as jesup warned).
I am going to fix the RTT behavior and the tests in bug 1344970, for now RTT will
be zero when unavailable.

MozReview-Commit-ID: 9x3eQfbM3ZT

--HG--
extra : rebase_source : f8d46d7232455a3038fd99ffb6cc14111c44a794

media/webrtc/signaling/src/media-conduit/AudioConduit.cpp

@@ -205,10 +205,13 @@ bool WebrtcAudioConduit::GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
                                                     fractionLost,
                                                     *cumulativeLost,
                                                     *rttMs);
-  if (result) {
-    *timestamp = NTPtoDOMHighResTimeStamp(ntpHigh, ntpLow);
+  if (!result) {
+    return false;
   }
-  return result;
+  // Note: timestamp is not correct per the spec... should be time the rtcp
+  // was received (remote) or sent (local)
+  *timestamp = webrtc::Clock::GetRealTimeClock()->TimeInMilliseconds();
+  return true;
 }
 
 bool WebrtcAudioConduit::GetRTCPSenderReport(DOMHighResTimeStamp* timestamp,

media/webrtc/signaling/src/media-conduit/VideoConduit.cpp

@@ -821,24 +821,42 @@ bool WebrtcVideoConduit::GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
   {
     CSFLogVerbose(logTag, "%s for VideoConduit:%p", __FUNCTION__, this);
     MutexAutoLock lock(mCodecMutex);
-    if (!mRecvStream) {
+    if (!mSendStream) {
       return false;
     }
-
-    const webrtc::VideoReceiveStream::Stats &stats = mRecvStream->GetStats();
-    *jitterMs = stats.rtcp_stats.jitter;
-    *cumulativeLost = stats.rtcp_stats.cumulative_lost;
-    *bytesReceived = stats.rtp_stats.MediaPayloadBytes();
-    *packetsReceived = stats.rtp_stats.transmitted.packets;
-    // Note: timestamp is not correct per the spec... should be time the rtcp
-    // was received (remote) or sent (local)
-    *timestamp = webrtc::Clock::GetRealTimeClock()->TimeInMilliseconds();
-    int64_t rtt = mRecvStream->GetRtt();
+    const webrtc::VideoSendStream::Stats& sendStats = mSendStream->GetStats();
+    if (sendStats.substreams.size() == 0
+        || mSendStreamConfig.rtp.ssrcs.size() == 0) {
+      return false;
+    }
+    uint32_t ssrc = mSendStreamConfig.rtp.ssrcs.front();
+    auto ind = sendStats.substreams.find(ssrc);
+    if (ind == sendStats.substreams.end()) {
+      CSFLogError(logTag,
+        "%s for VideoConduit:%p ssrc not found in SendStream stats.",
+        __FUNCTION__, this);
+      return false;
+    }
+    *jitterMs = ind->second.rtcp_stats.jitter;
+    *cumulativeLost = ind->second.rtcp_stats.cumulative_lost;
+    *bytesReceived = ind->second.rtp_stats.MediaPayloadBytes();
+    *packetsReceived = ind->second.rtp_stats.transmitted.packets;
+    int64_t rtt = mSendStream->GetRtt(); // TODO: BUG 1241066, mozRtt is 0 or 1
     if (rtt >= 0) {
       *rttMs = rtt;
     } else {
       *rttMs = 0;
     }
+#ifdef DEBUG
+    if (rtt > INT32_MAX) {
+      CSFLogError(logTag,
+        "%s for VideoConduit:%p mRecvStream->GetRtt() is larger than the"
+        " maximum size of an RTCP RTT.", __FUNCTION__, this);
+    }
+#endif
+    // Note: timestamp is not correct per the spec... should be time the rtcp
+    // was received (remote) or sent (local)
+    *timestamp = webrtc::Clock::GetRealTimeClock()->TimeInMilliseconds();
   }
   return true;
 }
@@ -849,22 +867,16 @@ WebrtcVideoConduit::GetRTCPSenderReport(DOMHighResTimeStamp* timestamp,
                                         uint64_t* bytesSent)
 {
   CSFLogVerbose(logTag, "%s for VideoConduit:%p", __FUNCTION__, this);
-  MutexAutoLock lock(mCodecMutex);
-  if (!mSendStream) {
-    return false;
+  webrtc::RTCPSenderInfo senderInfo;
+  {
+    MutexAutoLock lock(mCodecMutex);
+    if (!mRecvStream || !mRecvStream->GetRemoteRTCPSenderInfo(&senderInfo)) {
+      return false;
+    }
   }
-
-  const webrtc::VideoSendStream::Stats& stats = mSendStream->GetStats();
-  *packetsSent = 0;
-  for (auto entry: stats.substreams){
-    *packetsSent += entry.second.rtp_stats.transmitted.packets;
-    // NG -- per https://www.w3.org/TR/webrtc-stats/ this is only payload bytes
-    *bytesSent += entry.second.rtp_stats.MediaPayloadBytes();
-  }
-
-  // Note: timestamp is not correct per the spec... should be time the rtcp
-  // was received (remote) or sent (local)
   *timestamp = webrtc::Clock::GetRealTimeClock()->TimeInMilliseconds();
+  *packetsSent = senderInfo.sendPacketCount;
+  *bytesSent = senderInfo.sendOctetCount;
   return true;
 }
 

media/webrtc/trunk/webrtc/video/video_receive_stream.cc

@@ -407,5 +407,10 @@ int64_t VideoReceiveStream::GetRtt() const {
   return -1;
 }
 
+bool
+VideoReceiveStream::GetRemoteRTCPSenderInfo(RTCPSenderInfo* sender_info) const {
+  return -1 != vie_channel_->GetRemoteRTCPSenderInfo(sender_info);
+}
+
 }  // namespace internal
 }  // namespace webrtc

media/webrtc/trunk/webrtc/video/video_receive_stream.h

@@ -76,6 +76,7 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
 
   void SetSyncChannel(VoiceEngine* voice_engine, int audio_channel_id) override;
   int64_t GetRtt() const override;
+  bool GetRemoteRTCPSenderInfo(RTCPSenderInfo* sender_info) const override;
  private:
   TransportAdapter transport_adapter_;
   EncodedFrameCallbackAdapter encoded_frame_proxy_;

media/webrtc/trunk/webrtc/video/video_send_stream.h

@@ -70,7 +70,7 @@ class VideoSendStream : public webrtc::VideoSendStream,
   typedef std::map<uint32_t, RtpState> RtpStateMap;
   RtpStateMap GetRtpStates() const;
 
-  int64_t GetRtt() const;
+  int64_t GetRtt() const override;
   int GetPaddingNeededBps() const;
 
  private:

media/webrtc/trunk/webrtc/video/vie_channel.cc

@@ -891,21 +891,15 @@ int32_t ViEChannel::GetRemoteRTCPReceiverInfo(uint32_t& NTPHigh,
   return 0;
 }
 
-//->@@NG // int32_t ViEChannel::GetRemoteRTCPSenderInfo(RTCPSenderInfo* sender_info) const {
-//->@@NG //   // Get the sender info from the latest received RTCP Sender Report.
-//->@@NG //   RTCPSenderInfo rtcp_sender_info;
-//->@@NG //   if (rtp_rtcp_->RemoteRTCPStat(&rtcp_sender_info) != 0) {
-//->@@NG //     LOG_F(LS_ERROR) << "failed to read RTCP SR sender info";
-//->@@NG //     return -1;
-//->@@NG //   }
-//->@@NG //
-//->@@NG //   sender_info->NTP_timestamp_high = rtcp_sender_info.NTPseconds;
-//->@@NG //   sender_info->NTP_timestamp_low = rtcp_sender_info.NTPfraction;
-//->@@NG //   sender_info->RTP_timestamp = rtcp_sender_info.RTPtimeStamp;
-//->@@NG //   sender_info->sender_packet_count = rtcp_sender_info.sendPacketCount;
-//->@@NG //   sender_info->sender_octet_count = rtcp_sender_info.sendOctetCount;
-//->@@NG //   return 0;
-//->@@NG // }
+int32_t ViEChannel::GetRemoteRTCPSenderInfo(RTCPSenderInfo* sender_info) const {
+  // Get the sender info from the latest received RTCP Sender Report.
+  if (rtp_rtcp_modules_[0] &&
+    rtp_rtcp_modules_[0]->RemoteRTCPStat(sender_info) != 0) {
+    LOG_F(LS_ERROR) << "failed to read RTCP SR sender info";
+    return -1;
+  }
+  return 0;
+}
 
 void ViEChannel::RegisterSendChannelRtcpStatisticsCallback(
     RtcpStatisticsCallback* callback) {

media/webrtc/trunk/webrtc/video_receive_stream.h

@@ -179,6 +179,10 @@ class VideoReceiveStream : public ReceiveStream {
   // TODO(pbos): Add info on currently-received codec to Stats.
   virtual Stats GetStats() const = 0;
   virtual int64_t GetRtt() const = 0;
+
+  virtual bool
+  GetRemoteRTCPSenderInfo(RTCPSenderInfo* sender_info) const = 0;
+
   virtual void SetSyncChannel(VoiceEngine* voice_engine, int audio_channel_id) = 0;
 };
 

media/webrtc/trunk/webrtc/video_send_stream.h

@@ -187,6 +187,8 @@ class VideoSendStream : public SendStream {
   virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
 
   virtual Stats GetStats() = 0;
+
+  virtual int64_t GetRtt() const = 0;
 };
 
 }  // namespace webrtc