Support RtpEncodingParameters::active in voice engine.

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 875 matching lines @@
   webrtc::RtpParameters parameters;
   EXPECT_FALSE(channel_->SetRtpParameters(kSsrc1, parameters));
   // Setting RtpParameters with exactly one encoding should succeed.
   parameters.encodings.push_back(webrtc::RtpEncodingParameters());
   EXPECT_TRUE(channel_->SetRtpParameters(kSsrc1, parameters));
   // Two or more encodings should result in failure.
   parameters.encodings.push_back(webrtc::RtpEncodingParameters());
   EXPECT_FALSE(channel_->SetRtpParameters(kSsrc1, parameters));
 }
 
+// Test that a stream will not be sending if its encoding is made
+// inactive through SetRtpParameters.
+TEST_F(WebRtcVoiceEngineTestFake, SetRtpParametersEncodingsActive) {
+  EXPECT_TRUE(SetupSendStream());
+  SetSend(channel_, true);
+  EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
+  // Get current parameters and change "active" to false.
+  webrtc::RtpParameters parameters = channel_->GetRtpParameters(kSsrc1);
+  ASSERT_EQ(1u, parameters.encodings.size());
+  ASSERT_TRUE(parameters.encodings[0].active);
+  parameters.encodings[0].active = false;
+  EXPECT_TRUE(channel_->SetRtpParameters(kSsrc1, parameters));
+  EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
+
+  // Now change it back to active and verify we resume sending.
+  parameters.encodings[0].active = true;
+  EXPECT_TRUE(channel_->SetRtpParameters(kSsrc1, parameters));
+  EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
+}
+
 // Test that SetRtpParameters configures the correct encoding channel for each
 // SSRC.
 TEST_F(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) {
   SetupForMultiSendStream();
   // Create send streams.
   for (uint32_t ssrc : kSsrcs4) {
     EXPECT_TRUE(
         channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
   }
   // Configure one stream to be limited by the stream config, another to be
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 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));
 }