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..3116e7fd6968e11178e778c893a7401713d21e01 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, PlayoutSetParameters) {
+  SetupRecvStream();
+  channel_->SetPlayout(true);
+
+  // 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());
+}
+

[aleloi, 2016/10/03 11:55:30]

I added a new test: There is currently no test that both starts playout and
makes a stream be recreated (tested by checking when Recreate is called and
playout_ is true) and was afraid to increase complexity of the existing tests.
What do you think?

 // 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