Fixes a bug which incorrectly logs incoming RTCP as outgoing.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 660 matching lines @@
           event_log_->LogRtcpPacket(true, media_type, packet, length);
       }
     }
   }
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     ReadLockScoped read_lock(*send_crit_);
     for (VideoSendStream* stream : video_send_streams_) {
       if (stream->DeliverRtcp(packet, length)) {
         rtcp_delivered = true;
         if (event_log_)
-          event_log_->LogRtcpPacket(false, media_type, packet, length);
+          event_log_->LogRtcpPacket(true, media_type, packet, length);

[pbos-webrtc, 2016/01/11 12:16:56]

Can we change this into an enum that has kIncoming and kOutgoing to make this
mistake harder?

[terelius, 2016/01/11 12:22:52]

I suppose we could do that, though I think the origin of this bug is that the
RTCP packet is passed to a SendStream, which on first glance looks as if it is
outgoing.

[the sun, 2016/01/11 12:36:16]

+1

[terelius, 2016/01/11 13:07:20]

There is an enum RTPDirections in common_types.h. Do you think I should reuse it
(for both RTP and RTCP), or should I create a new enum in rtc_event_log.h?

[terelius, 2016/01/13 17:08:18]

Added an enum in rtc_event_log.h

       }
     }
   }
   return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 const uint8_t* packet,
                                                 size_t length,
                                                 const PacketTime& packet_time) {
@@ skipping 45 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc