Remove usage of VoENetwork from VoEWrapper and FakeWebRtcVoiceEngine.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 2250 matching lines @@
 void WebRtcVoiceMediaChannel::OnPacketReceived(
     rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
 
   const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                               packet_time.not_before);
   webrtc::PacketReceiver::DeliveryStatus delivery_result =
       call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
                                        packet->cdata(), packet->size(),
                                        webrtc_packet_time);
-

[ossu, 2016/04/29 09:12:59]

I'm gonna be a bit nit-picky and question the removal of this line in an
otherwise untouched file.

[the sun, 2016/04/29 11:12:49]

You're right, but I'm not uploading a new patch set because of this.

   if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
     return;
   }
 
   // Create a default receive stream for this unsignalled and previously not
   // received ssrc. If there already is a default receive stream, delete it.
   // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
   uint32_t ssrc = 0;
   if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
     return;
@@ skipping 275 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE