Bug 1301286: At least in the webrtc49 update, 100Kbps isn't enough for simulcast tests r=abr

MozReview-Commit-ID: kQHNnr7rAg

dom/media/tests/mochitest/test_peerConnection_simulcastOffer.html

@@ -26,7 +26,10 @@
 
   runNetworkTest(() =>
     pushPrefs(['media.peerconnection.simulcast', true],
-              ['media.peerconnection.video.min_bitrate_estimate', 100*1000]).then(() => {
+              // 180Kbps was determined empirically, set well-higher than
+              // the 80Kbps+overhead needed for the two simulcast streams.
+              // 100Kbps was apparently too low.
+              ['media.peerconnection.video.min_bitrate_estimate', 180*1000]).then(() => {
       SimpleTest.requestCompleteLog();
       var helper;
 

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

@@ -1858,6 +1858,12 @@ WebrtcVideoConduit::SendRtp(const uint8_t* packet, size_t length,
   // extension for TransportSequenceNumber is being used, which we don't.
   CSFLogDebug(logTag, "%s : len %lu", __FUNCTION__, (unsigned long)length);
 
+  { static int x = 0;
+    if (++x % 150 == 0) {
+      CSFLogDebug(logTag, "%s Faking packet loss, seq %d ", __FUNCTION__,
+                  ntohs(*((uint16_t*)&packet[2])));
+      return true;
+    }
   ReentrantMonitorAutoEnter enter(mTransportMonitor);
   if (!mTransmitterTransport ||
      NS_FAILED(mTransmitterTransport->SendRtpPacket(packet, length)))

media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc

@@ -727,6 +727,8 @@ rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildNACK(
   packet_type_counter_.nack_requests = nack_stats_.requests();
   packet_type_counter_.unique_nack_requests = nack_stats_.unique_requests();
 
+  LOG(LS_ERROR) << "RTPSender: Sending Nack: " << stringBuilder.GetResult().c_str();
+    
   TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTCPSender::NACK", "nacks",
                        TRACE_STR_COPY(stringBuilder.GetResult().c_str()));