Restarting channel when swapping AudioReceiveStreams in WebrtcVoE.

webrtc/media/engine/webrtcvoiceengine_unittest.cc

Index: webrtc/media/engine/webrtcvoiceengine_unittest.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index e40bc773b6ffb9e75c6dace5202dd0908487803d..36a138a7133b2a5bbb620a0939fad5ce08e06665 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -3340,6 +3340,20 @@ TEST_F(WebRtcVoiceEngineTestFake, OnReadyToSendSignalsNetworkState) {
             call_.GetNetworkState(webrtc::MediaType::VIDEO));
 }
 
+// Test that playout is still started after changing parameters
+TEST_F(WebRtcVoiceEngineTestFake, ParametersShouldPreservePlayout) {

[the sun, 2016/10/03 12:01:49]

nit: Name is confusing - I don't see how the parameters (whoever they are) could
preserve playout. May I suggest "PreservePlayoutWhenRecreateRecvStream"... or if
you have a better suggestion?

[aleloi, 2016/10/03 12:07:58]

I agree. I went with your proposal.

+  SetupRecvStream();
+  channel_->SetPlayout(true);

[the sun, 2016/10/03 12:01:49]

Add
EXPECT_TRUE(GetRecvStream(kSsrc1).started());

[aleloi, 2016/10/03 12:07:57]

Done.

+
+  // Changing RTP header extensions will recreate the AudioReceiveStream.
+  cricket::AudioRecvParameters parameters;
+  parameters.extensions.push_back(
+      webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12));
+  channel_->SetRecvParameters(parameters);
+
+  EXPECT_TRUE(GetRecvStream(kSsrc1).started());
+}
+
 // Tests that the library initializes and shuts down properly.
 TEST(WebRtcVoiceEngineTest, StartupShutdown) {
   // If the VoiceEngine wants to gather available codecs early, that's fine but