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

webrtc/audio/audio_receive_stream_unittest.cc

Index: webrtc/audio/audio_receive_stream_unittest.cc
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 8703d6ed324819a0a041e823a88bf90db6f9e7ca..4da503fb539f22221a0360ae99a6d8d3e25d6947 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -120,6 +120,7 @@ struct ConfigHelper {
   AudioReceiveStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoiceEngine& voice_engine() { return voice_engine_; }
+  MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
 
   void SetupMockForBweFeedback(bool send_side_bwe) {
     EXPECT_CALL(congestion_controller_,
@@ -254,6 +255,11 @@ TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) {
                              rtp_packet.size() - kExpectedHeaderLength,
                              VerifyHeaderExtension(expected_extension), false))
       .Times(1);
+  EXPECT_CALL(*helper.channel_proxy(),
+              ReceivedRTPPacket(&rtp_packet[0],
+                                rtp_packet.size(),
+                                _))
+      .WillOnce(Return(true));
   EXPECT_TRUE(
       recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
 }