Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.

Clean some logs and add asserts in the way.

BUG=3153
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/video_coding/main/test/mt_rx_tx_test.cc

@@ -157,7 +157,7 @@ int MTRxTxTest(CmdArgs& args)
     configuration.outgoing_transport = outgoingTransport;
     RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(configuration);
     scoped_ptr<RTPPayloadRegistry> registry(new RTPPayloadRegistry(
-        -1, RTPPayloadStrategy::CreateStrategy(false)));
+        RTPPayloadStrategy::CreateStrategy(false)));
     scoped_ptr<RtpReceiver> rtp_receiver(
         RtpReceiver::CreateVideoReceiver(-1, Clock::GetRealTimeClock(),
                                          &dataCallback, NULL, registry.get()));

webrtc/modules/video_coding/main/test/rtp_player.cc

@@ -273,7 +273,7 @@ class SsrcHandlers {
             LostPackets* lost_packets)
         : rtp_header_parser_(RtpHeaderParser::Create()),
           rtp_payload_registry_(new RTPPayloadRegistry(
-              0, RTPPayloadStrategy::CreateStrategy(false))),
+              RTPPayloadStrategy::CreateStrategy(false))),
           rtp_module_(),
           payload_sink_(),
           ssrc_(ssrc),