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

webrtc/call/call_perf_tests.cc

Index: webrtc/call/call_perf_tests.cc
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index 9aa50d0ad9720c0f86fb3c6bd36c5c1ecaba1815..63c61b2f352db377fcea89ba25efc958cb0148c1 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -149,33 +149,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
   const char* kSyncGroup = "av_sync";
   const uint32_t kAudioSendSsrc = 1234;
   const uint32_t kAudioRecvSsrc = 5678;
-  class AudioPacketReceiver : public PacketReceiver {
-   public:
-    AudioPacketReceiver(int channel, VoENetwork* voe_network)
-        : channel_(channel),
-          voe_network_(voe_network),
-          parser_(RtpHeaderParser::Create()) {}
-    DeliveryStatus DeliverPacket(MediaType media_type,
-                                 const uint8_t* packet,
-                                 size_t length,
-                                 const PacketTime& packet_time) override {
-      EXPECT_TRUE(media_type == MediaType::ANY ||
-                  media_type == MediaType::AUDIO);
-      int ret;
-      if (parser_->IsRtcp(packet, length)) {
-        ret = voe_network_->ReceivedRTCPPacket(channel_, packet, length);
-      } else {
-        ret = voe_network_->ReceivedRTPPacket(channel_, packet, length,
-                                              PacketTime());
-      }
-      return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
-    }
-
-   private:
-    int channel_;
-    VoENetwork* voe_network_;
-    std::unique_ptr<RtpHeaderParser> parser_;
-  };
 
   test::ClearHistograms();
   VoiceEngine* voice_engine = VoiceEngine::Create();
@@ -201,44 +174,54 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
   receiver_config.audio_state = sender_config.audio_state;
   CreateCalls(sender_config, receiver_config);
 
-  AudioPacketReceiver voe_send_packet_receiver(send_channel_id, voe_network);
-  AudioPacketReceiver voe_recv_packet_receiver(recv_channel_id, voe_network);
 
   VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock());
 
-  FakeNetworkPipe::Config net_config;
-  net_config.queue_delay_ms = 500;
-  net_config.loss_percent = 5;
-  test::PacketTransport audio_send_transport(
-      nullptr, &observer, test::PacketTransport::kSender, net_config);
-  audio_send_transport.SetReceiver(&voe_recv_packet_receiver);
-  test::PacketTransport audio_receive_transport(
-      nullptr, &observer, test::PacketTransport::kReceiver, net_config);
-  audio_receive_transport.SetReceiver(&voe_send_packet_receiver);
-
-  internal::TransportAdapter send_transport_adapter(&audio_send_transport);
-  send_transport_adapter.Enable();
-  EXPECT_EQ(0, voe_network->RegisterExternalTransport(send_channel_id,
-                                                      send_transport_adapter));
-
-  internal::TransportAdapter recv_transport_adapter(&audio_receive_transport);
-  recv_transport_adapter.Enable();
-  EXPECT_EQ(0, voe_network->RegisterExternalTransport(recv_channel_id,
-                                                      recv_transport_adapter));
-
-  test::PacketTransport sync_send_transport(sender_call_.get(), &observer,
-                                            test::PacketTransport::kSender,
-                                            FakeNetworkPipe::Config());
-  sync_send_transport.SetReceiver(receiver_call_->Receiver());
-  test::PacketTransport sync_receive_transport(receiver_call_.get(), &observer,
-                                               test::PacketTransport::kReceiver,
-                                               FakeNetworkPipe::Config());
-  sync_receive_transport.SetReceiver(sender_call_->Receiver());
+  // Helper class to ensure we deliver correct media_type to the receiving call.
+  class MediaTypePacketReceiver : public PacketReceiver {
+   public:
+    MediaTypePacketReceiver(PacketReceiver* packet_receiver,
+                            MediaType media_type)
+        : packet_receiver_(packet_receiver), media_type_(media_type) {}
+
+    DeliveryStatus DeliverPacket(MediaType media_type,
+                                 const uint8_t* packet,
+                                 size_t length,
+                                 const PacketTime& packet_time) override {
+      return packet_receiver_->DeliverPacket(media_type_, packet, length,
+                                             packet_time);
+    }
+   private:
+    PacketReceiver* packet_receiver_;
+    const MediaType media_type_;
+  };
+
+  FakeNetworkPipe::Config audio_net_config;
+  audio_net_config.queue_delay_ms = 500;
+  audio_net_config.loss_percent = 5;
+  test::PacketTransport audio_send_transport(sender_call_.get(), &observer,
+                                             test::PacketTransport::kSender,
+                                             audio_net_config);
+  MediaTypePacketReceiver audio_receiver(receiver_call_->Receiver(),
+                                         MediaType::AUDIO);
+  audio_send_transport.SetReceiver(&audio_receiver);
+
+  test::PacketTransport video_send_transport(sender_call_.get(), &observer,
+                                             test::PacketTransport::kSender,
+                                             FakeNetworkPipe::Config());
+  MediaTypePacketReceiver video_receiver(receiver_call_->Receiver(),
+                                         MediaType::VIDEO);
+  video_send_transport.SetReceiver(&video_receiver);
+
+  test::PacketTransport receive_transport(
+      receiver_call_.get(), &observer, test::PacketTransport::kReceiver,
+      FakeNetworkPipe::Config());
+  receive_transport.SetReceiver(sender_call_->Receiver());
 
   test::FakeDecoder fake_decoder;
 
-  CreateSendConfig(1, 0, &sync_send_transport);
-  CreateMatchingReceiveConfigs(&sync_receive_transport);
+  CreateSendConfig(1, 0, &video_send_transport);
+  CreateMatchingReceiveConfigs(&receive_transport);
 
   AudioSendStream::Config audio_send_config(&audio_send_transport);

[stefan-webrtc, 2016/04/22 09:16:19]

I'm not sure this actually works in practice. All packets go to the same pacer
which will send them out on the same transport. Have you verified that the
streams aren't in sync already at the beginning?

[mflodman, 2016/04/22 09:41:48]

If that is the case, our API is wrong if we can set one transport per send
stream.

The test starts by printing:
"RESULT stream_offset: synchronization= -553 ms"

to slowly go down to:
"RESULT stream_offset: synchronization= -14 ms"

I'll look at the code though.

[stefan-webrtc, 2016/04/22 10:35:04]

You're right, it does go over separate transports, my mistake.

   audio_send_config.voe_channel_id = send_channel_id;
@@ -298,10 +281,9 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
   fake_audio_device.Stop();
 
   Stop();
-  sync_send_transport.StopSending();
-  sync_receive_transport.StopSending();
+  video_send_transport.StopSending();
   audio_send_transport.StopSending();
-  audio_receive_transport.StopSending();
+  receive_transport.StopSending();
 
   DestroyStreams();