Fix bug causing audio to stop being sent when AudioSendStreams are recreated.

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 2158 matching lines @@
   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
   EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, nullptr, nullptr));
   channel_->SetSend(true);
   EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
   EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, nullptr, &fake_source_));
   EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
   EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, nullptr, nullptr));
   EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
 }
 
+// Test that SetSendParameters() does not alter a stream's send state.
+TEST_F(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) {
+  EXPECT_TRUE(SetupSendStream());
+  EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
+
+  // Turn on sending.
+  channel_->SetSend(true);
+  EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
+
+  // Changing RTP header extensions will recreate the AudioSendStream.
+  send_parameters_.extensions.push_back(
+      cricket::RtpHeaderExtension(kRtpAudioLevelHeaderExtension, 12));
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
+  EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
+
+  // Turn off sending.
+  channel_->SetSend(false);
+  EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
+
+  // Changing RTP header extensions will recreate the AudioSendStream.
+  send_parameters_.extensions.clear();
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
+  EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
+}
+
 // Test that we can create a channel and start playing out on it.
 TEST_F(WebRtcVoiceEngineTestFake, Playout) {
   EXPECT_TRUE(SetupRecvStream());
   int channel_num = voe_.GetLastChannel();
   EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
   EXPECT_TRUE(channel_->SetPlayout(true));
   EXPECT_TRUE(voe_.GetPlayout(channel_num));
   EXPECT_TRUE(channel_->SetPlayout(false));
   EXPECT_FALSE(voe_.GetPlayout(channel_num));
 }
@@ skipping 1304 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));
 }