Switch voice transport to use Call and Stream instead of VoENetwork.

webrtc/media/engine/fakewebrtccall.cc

Index: webrtc/media/engine/fakewebrtccall.cc
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index 8eff0ebcf8e29b258dd36437e7771606019847be..32bd2710226d41059d1a22f76ef08cdcd7aca383 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -67,8 +67,21 @@ void FakeAudioReceiveStream::SetStats(
   stats_ = stats;
 }
 
-void FakeAudioReceiveStream::IncrementReceivedPackets() {
+bool FakeAudioReceiveStream::VerifyLastPacket(const void* data,
+                                              size_t length) const {
+  return last_packet_ == std::string(static_cast<const char*>(data), length);

[the sun, 2016/04/22 12:40:31]

CppReference for std::string::operator==() say it shall point at a
null-terminated sequence for char* strings:
http://www.cplusplus.com/reference/string/string/operators/

To me that indicates that the comparison may not actually compare the whole
contents of the string.

I suggest you use rtc::Buffer here and for last_packet_, as that has the
required functionality, and more accurately conveys the intent.

[tommi, 2016/04/23 16:00:09]

+1.  string is a misleading type for binary data and in some implementations
comes with an extra inline allocated buffer which isn't necessary for this
class.

Creating a copy of the data just for comparison is also something I'd prefer to
never do in our code base (since code can get copy pasted).

[mflodman, 2016/04/27 13:42:17]

Done. This was just copied from elsewhere in these tests.

And good suggestion, I'm still in the mental mode of not using stuff in the base
folder :(

+}
+
+bool FakeAudioReceiveStream::DeliverRtp(const uint8_t* packet,
+                                        size_t length,
+                                        const webrtc::PacketTime& packet_time) {
   received_packets_++;

[tommi, 2016/04/23 16:00:09]

++foo

[mflodman, 2016/04/27 13:42:17]

Done.

+  last_packet_ = std::string(reinterpret_cast<const char*>(packet), length);

[tommi, 2016/04/23 16:00:09]

What about using a movable type such as rtc::Buffer instead?

[mflodman, 2016/04/27 13:42:17]

Done.

+  return true;
+}
+
+bool FakeAudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  return true;
 }
 
 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
@@ -409,7 +422,7 @@ FakeCall::DeliveryStatus FakeCall::DeliverPacket(
       media_type == webrtc::MediaType::AUDIO) {
     for (auto receiver : audio_receive_streams_) {
       if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
-        receiver->IncrementReceivedPackets();
+        receiver->DeliverRtp(packet, length, packet_time);
         return DELIVERY_OK;
       }
     }