Adding getParameters/setParameters APIs to RtpReceiver.

webrtc/api/rtpsenderreceiver_unittest.cc

 /*
  *  Copyright 2012 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 41 matching lines @@
 
   MOCK_METHOD2(SetAudioPlayout,
                void(uint32_t ssrc,
                     bool enable));
   MOCK_METHOD4(SetAudioSend,
                void(uint32_t ssrc,
                     bool enable,
                     const cricket::AudioOptions& options,
                     cricket::AudioSource* source));
   MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume));
-  MOCK_CONST_METHOD1(GetAudioRtpParameters, RtpParameters(uint32_t ssrc));
-  MOCK_METHOD2(SetAudioRtpParameters,
+  MOCK_CONST_METHOD1(GetAudioRtpSendParameters, RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetAudioRtpSendParameters,
+               bool(uint32_t ssrc, const RtpParameters&));
+  MOCK_CONST_METHOD1(GetAudioRtpReceiveParameters,
+                     RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetAudioRtpReceiveParameters,
                bool(uint32_t ssrc, const RtpParameters&));
 
   void SetRawAudioSink(
       uint32_t, std::unique_ptr<AudioSinkInterface> sink) /* override */ {
     sink_ = std::move(sink);
   }
 
  private:
   std::unique_ptr<AudioSinkInterface> sink_;
 };
 
 // Helper class to test RtpSender/RtpReceiver.
 class MockVideoProvider : public VideoProviderInterface {
  public:
   virtual ~MockVideoProvider() {}
   MOCK_METHOD2(SetSource,
                bool(uint32_t ssrc,
                     rtc::VideoSourceInterface<cricket::VideoFrame>* source));
   MOCK_METHOD3(SetVideoPlayout,
                void(uint32_t ssrc,
                     bool enable,
                     rtc::VideoSinkInterface<cricket::VideoFrame>* sink));
   MOCK_METHOD3(SetVideoSend,
                void(uint32_t ssrc,
                     bool enable,
                     const cricket::VideoOptions* options));
 
-  MOCK_CONST_METHOD1(GetVideoRtpParameters, RtpParameters(uint32_t ssrc));
-  MOCK_METHOD2(SetVideoRtpParameters,
+  MOCK_CONST_METHOD1(GetVideoRtpSendParameters, RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetVideoRtpSendParameters,
+               bool(uint32_t ssrc, const RtpParameters&));
+  MOCK_CONST_METHOD1(GetVideoRtpReceiveParameters,
+                     RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetVideoRtpReceiveParameters,
                bool(uint32_t ssrc, const RtpParameters&));
 };
 
 class RtpSenderReceiverTest : public testing::Test {
  public:
   virtual void SetUp() {
     stream_ = MediaStream::Create(kStreamLabel1);
   }
 
   void AddVideoTrack() {
@@ skipping 394 matching lines @@
   sender->SetSsrc(kVideoSsrc2);
 
   // Calls expected from destructor.
   EXPECT_CALL(video_provider_, SetSource(kVideoSsrc2, nullptr)).Times(1);
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1);
 }
 
 TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) {
   CreateAudioRtpSender();
 
-  EXPECT_CALL(audio_provider_, GetAudioRtpParameters(kAudioSsrc))
+  EXPECT_CALL(audio_provider_, GetAudioRtpSendParameters(kAudioSsrc))
       .WillOnce(Return(RtpParameters()));
-  EXPECT_CALL(audio_provider_, SetAudioRtpParameters(kAudioSsrc, _))
+  EXPECT_CALL(audio_provider_, SetAudioRtpSendParameters(kAudioSsrc, _))
       .WillOnce(Return(true));
   RtpParameters params = audio_rtp_sender_->GetParameters();
   EXPECT_TRUE(audio_rtp_sender_->SetParameters(params));
 
   DestroyAudioRtpSender();
 }
 
 TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
   CreateVideoRtpSender();
 
-  EXPECT_CALL(video_provider_, GetVideoRtpParameters(kVideoSsrc))
+  EXPECT_CALL(video_provider_, GetVideoRtpSendParameters(kVideoSsrc))
       .WillOnce(Return(RtpParameters()));
-  EXPECT_CALL(video_provider_, SetVideoRtpParameters(kVideoSsrc, _))
+  EXPECT_CALL(video_provider_, SetVideoRtpSendParameters(kVideoSsrc, _))
       .WillOnce(Return(true));
   RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
 
   DestroyVideoRtpSender();
 }
 
+TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) {
+  CreateAudioRtpReceiver();
+
+  EXPECT_CALL(audio_provider_, GetAudioRtpReceiveParameters(kAudioSsrc))
+      .WillOnce(Return(RtpParameters()));
+  EXPECT_CALL(audio_provider_, SetAudioRtpReceiveParameters(kAudioSsrc, _))
+      .WillOnce(Return(true));
+  RtpParameters params = audio_rtp_receiver_->GetParameters();
+  EXPECT_TRUE(audio_rtp_receiver_->SetParameters(params));
+
+  DestroyAudioRtpReceiver();
+}
+
+TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetParameters) {
+  CreateVideoRtpReceiver();
+
+  EXPECT_CALL(video_provider_, GetVideoRtpReceiveParameters(kVideoSsrc))
+      .WillOnce(Return(RtpParameters()));
+  EXPECT_CALL(video_provider_, SetVideoRtpReceiveParameters(kVideoSsrc, _))
+      .WillOnce(Return(true));
+  RtpParameters params = video_rtp_receiver_->GetParameters();
+  EXPECT_TRUE(video_rtp_receiver_->SetParameters(params));
+
+  DestroyVideoRtpReceiver();
+}
+
 }  // namespace webrtc