Enable setting the maximum bitrate limit in RtpSender.

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.
  */
 
 #include <string>
 #include <utility>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/api/audiotrack.h"
 #include "webrtc/api/mediastream.h"
 #include "webrtc/api/remoteaudiosource.h"
 #include "webrtc/api/rtpreceiver.h"
 #include "webrtc/api/rtpsender.h"
 #include "webrtc/api/streamcollection.h"
 #include "webrtc/api/videosource.h"
 #include "webrtc/api/videotrack.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/media/base/fakevideocapturer.h"
 #include "webrtc/media/base/mediachannel.h"
 
 using ::testing::_;
 using ::testing::Exactly;
+using ::testing::Return;
 
 static const char kStreamLabel1[] = "local_stream_1";
 static const char kVideoTrackId[] = "video_1";
 static const char kAudioTrackId[] = "audio_1";
 static const uint32_t kVideoSsrc = 98;
 static const uint32_t kVideoSsrc2 = 100;
 static const uint32_t kAudioSsrc = 99;
 static const uint32_t kAudioSsrc2 = 101;
 
 namespace webrtc {
 
 // Helper class to test RtpSender/RtpReceiver.
 class MockAudioProvider : public AudioProviderInterface {
  public:
   ~MockAudioProvider() override {}
 
   MOCK_METHOD2(SetAudioPlayout,
                void(uint32_t ssrc,
                     bool enable));
   MOCK_METHOD4(SetAudioSend,
                void(uint32_t ssrc,
                     bool enable,
                     const cricket::AudioOptions& options,
                     cricket::AudioRenderer* renderer));
   MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume));
+  MOCK_METHOD1(GetAudioRtpParameters, RTCRtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetAudioRtpParameters,
+               bool(uint32_t ssrc, const RTCRtpParameters&));
 
   void SetRawAudioSink(uint32_t,
                        rtc::scoped_ptr<AudioSinkInterface> sink) override {
     sink_ = std::move(sink);
   }
 
  private:
   rtc::scoped_ptr<AudioSinkInterface> sink_;
 };
 
 // Helper class to test RtpSender/RtpReceiver.
 class MockVideoProvider : public VideoProviderInterface {
  public:
   virtual ~MockVideoProvider() {}
   MOCK_METHOD2(SetCaptureDevice,
                bool(uint32_t ssrc, cricket::VideoCapturer* camera));
   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_METHOD1(GetVideoRtpParameters, RTCRtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetVideoRtpParameters,
+               bool(uint32_t ssrc, const RTCRtpParameters&));
 };
 
 class FakeVideoSource : public Notifier<VideoSourceInterface> {
  public:
   static rtc::scoped_refptr<FakeVideoSource> Create(bool remote) {
     return new rtc::RefCountedObject<FakeVideoSource>(remote);
   }
   virtual cricket::VideoCapturer* GetVideoCapturer() { return &fake_capturer_; }
   virtual void Stop() {}
   virtual void Restart() {}
@@ skipping 399 matching lines @@
               SetCaptureDevice(kVideoSsrc2,
                                video_track_->GetSource()->GetVideoCapturer()));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, true, _));
   sender->SetSsrc(kVideoSsrc2);
 
   // Calls expected from destructor.
   EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc2, nullptr)).Times(1);
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1);
 }
 
+TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) {
+  CreateAudioRtpSender();
+
+  EXPECT_CALL(audio_provider_, GetAudioRtpParameters(kAudioSsrc))
+      .WillOnce(Return(RTCRtpParameters()));
+  EXPECT_CALL(audio_provider_, SetAudioRtpParameters(kAudioSsrc, _))
+      .WillOnce(Return(true));
+  RTCRtpParameters params = audio_rtp_sender_->GetParameters();
+  EXPECT_TRUE(audio_rtp_sender_->SetParameters(params));
+
+  DestroyAudioRtpSender();
+}
+
+TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
+  CreateVideoRtpSender();
+
+  EXPECT_CALL(video_provider_, GetVideoRtpParameters(kVideoSsrc))
+      .WillOnce(Return(RTCRtpParameters()));
+  EXPECT_CALL(video_provider_, SetVideoRtpParameters(kVideoSsrc, _))
+      .WillOnce(Return(true));
+  RTCRtpParameters params = video_rtp_sender_->GetParameters();
+  EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
+
+  DestroyVideoRtpSender();
+}
+
 }  // namespace webrtc