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

webrtc/audio/audio_receive_stream_unittest.cc

 /*
  *  Copyright (c) 2015 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 102 matching lines @@
 
   MockCongestionController* congestion_controller() {
     return &congestion_controller_;
   }
   MockRemoteBitrateEstimator* remote_bitrate_estimator() {
     return &remote_bitrate_estimator_;
   }
   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_,
                 GetRemoteBitrateEstimator(send_side_bwe))
         .WillOnce(Return(&remote_bitrate_estimator_));
     EXPECT_CALL(remote_bitrate_estimator_,
                 RemoveStream(stream_config_.rtp.remote_ssrc));
   }
 
   void SetupMockForGetStats() {
@@ skipping 95 matching lines @@
   return arg.extension.hasAbsoluteSendTime ==
              expected_extension.hasAbsoluteSendTime &&
          arg.extension.absoluteSendTime ==
              expected_extension.absoluteSendTime &&
          arg.extension.hasTransportSequenceNumber ==
              expected_extension.hasTransportSequenceNumber &&
          arg.extension.transportSequenceNumber ==
              expected_extension.transportSequenceNumber;
 }
 
 TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) {

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

Rename test case to "ReceiveRtpPacket"

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

Done.

   ConfigHelper helper;
   helper.config().rtp.transport_cc = true;
   helper.SetupMockForBweFeedback(true);
   internal::AudioReceiveStream recv_stream(
       helper.congestion_controller(), helper.config(), helper.audio_state());
   const int kTransportSequenceNumberValue = 1234;
   std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
       kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
   PacketTime packet_time(5678000, 0);
   const size_t kExpectedHeaderLength = 20;
   RTPHeaderExtension expected_extension;
   expected_extension.hasTransportSequenceNumber = true;
   expected_extension.transportSequenceNumber = kTransportSequenceNumberValue;
   EXPECT_CALL(*helper.remote_bitrate_estimator(),
               IncomingPacket(packet_time.timestamp / 1000,
                              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));
 }
 

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

Add simple test case for "ReceiveRtcpPacket". Yes, it is just a call-through
right now, but it won't always be like that. With the infrastructure in place it
is more likely the test won't be forgotten down the line.

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

Done. And caught a stupid mistake done when changing the signature of channel
proxy DeliverRTCPPacket.

 TEST(AudioReceiveStreamTest, GetStats) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.congestion_controller(), helper.config(), helper.audio_state());
   helper.SetupMockForGetStats();
   AudioReceiveStream::Stats stats = recv_stream.GetStats();
   EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
   EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd);
   EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
             stats.packets_rcvd);
@@ skipping 23 matching lines @@
   EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
   EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc);
   EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
   EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
             stats.capture_start_ntp_time_ms);
 }
 }  // namespace test
 }  // namespace webrtc