Set rtcp_send_transport for AudioReceiveStreams. This was forgotten in https://codereview.webrtc.or…

webrtc/media/engine/webrtcvoiceengine_unittest.cc

 /*
  *  Copyright (c) 2008 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 476 matching lines @@
   FakeAudioSource fake_source_;
  private:
   webrtc::test::ScopedFieldTrials override_field_trials_;
 };
 
 // Tests that we can create and destroy a channel.
 TEST_F(WebRtcVoiceEngineTestFake, CreateChannel) {
   EXPECT_TRUE(SetupChannel());
 }
 
+// Test that we can add a send stream and that it has the correct defaults.
+TEST_F(WebRtcVoiceEngineTestFake, CreateSendStream) {
+  EXPECT_TRUE(SetupChannel());
+  EXPECT_TRUE(
+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc1)));
+  const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrc1);
+  EXPECT_EQ(kSsrc1, config.rtp.ssrc);
+  EXPECT_EQ("", config.rtp.c_name);
+  EXPECT_EQ(0u, config.rtp.extensions.size());
+  EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_),
+            config.send_transport);
+}
+
+// Test that we can add a receive stream and that it has the correct defaults.
+TEST_F(WebRtcVoiceEngineTestFake, CreateRecvStream) {
+  EXPECT_TRUE(SetupChannel());
+  EXPECT_TRUE(
+      channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc1)));
+  const webrtc::AudioReceiveStream::Config& config =
+      GetRecvStreamConfig(kSsrc1);
+  EXPECT_EQ(kSsrc1, config.rtp.remote_ssrc);
+  EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc);
+  EXPECT_FALSE(config.rtp.transport_cc);
+  EXPECT_EQ(0u, config.rtp.extensions.size());
+  EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_),
+            config.rtcp_send_transport);
+  EXPECT_EQ("", config.sync_group);
+}
+
 // Tests that the list of supported codecs is created properly and ordered
 // correctly (such that opus appears first).
 TEST_F(WebRtcVoiceEngineTestFake, CodecOrder) {
   const std::vector<cricket::AudioCodec>& codecs = engine_->codecs();
   ASSERT_FALSE(codecs.empty());
   EXPECT_STRCASEEQ("opus", codecs[0].name.c_str());
   EXPECT_EQ(48000, codecs[0].clockrate);
   EXPECT_EQ(2, codecs[0].channels);
   EXPECT_EQ(64000, codecs[0].bitrate);
 }
@@ skipping 3109 matching lines @@
 TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
   cricket::WebRtcVoiceEngine engine(nullptr);
   std::unique_ptr<webrtc::Call> call(
       webrtc::Call::Create(webrtc::Call::Config()));
   cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
                                            cricket::AudioOptions(), call.get());
   cricket::AudioRecvParameters parameters;
   parameters.codecs = engine.codecs();
   EXPECT_TRUE(channel.SetRecvParameters(parameters));
 }