Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

This eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.

It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2046173002 .

Cr-Commit-Position: refs/heads/master@{#13285}

webrtc/api/rtpsenderreceiver_unittest.cc

@@ -15,6 +15,8 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/api/audiotrack.h"
+#include "webrtc/api/fakemediacontroller.h"
+#include "webrtc/api/localaudiosource.h"
 #include "webrtc/api/mediastream.h"
 #include "webrtc/api/remoteaudiosource.h"
 #include "webrtc/api/rtpreceiver.h"
@@ -25,6 +27,10 @@
 #include "webrtc/api/videotrack.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/media/base/mediachannel.h"
+#include "webrtc/media/base/fakemediaengine.h"
+#include "webrtc/media/engine/fakewebrtccall.h"
+#include "webrtc/p2p/base/faketransportcontroller.h"
+#include "webrtc/pc/channelmanager.h"
 
 using ::testing::_;
 using ::testing::Exactly;
@@ -41,70 +47,57 @@ static const uint32_t kAudioSsrc2 = 101;
 
 namespace webrtc {
 
-// Helper class to test RtpSender/RtpReceiver.
-class MockAudioProvider : public AudioProviderInterface {
- public:
-  // TODO(nisse): Valid overrides commented out, because the gmock
-  // methods don't use any override declarations, and we want to avoid
-  // warnings from -Winconsistent-missing-override. See
-  // http://crbug.com/428099.
-  ~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::AudioSource* source));
-  MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume));
-  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_METHOD3(SetVideoPlayout,
-               void(uint32_t ssrc,
-                    bool enable,
-                    rtc::VideoSinkInterface<cricket::VideoFrame>* sink));
-  MOCK_METHOD4(SetVideoSend,
-               void(uint32_t ssrc,
-                    bool enable,
-                    const cricket::VideoOptions* options,
-                    rtc::VideoSourceInterface<cricket::VideoFrame>* source));
-
-  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);
+  RtpSenderReceiverTest()
+      :  // Create fake media engine/etc. so we can create channels to use to
+        // test RtpSenders/RtpReceivers.
+        media_engine_(new cricket::FakeMediaEngine()),
+        channel_manager_(media_engine_,
+                         rtc::Thread::Current(),
+                         rtc::Thread::Current()),
+        fake_call_(webrtc::Call::Config()),
+        fake_media_controller_(&channel_manager_, &fake_call_),
+        stream_(MediaStream::Create(kStreamLabel1)) {
+    // Create channels to be used by the RtpSenders and RtpReceivers.
+    channel_manager_.Init();
+    voice_channel_ = channel_manager_.CreateVoiceChannel(
+        &fake_media_controller_, &fake_transport_controller_, cricket::CN_AUDIO,
+        nullptr, false, cricket::AudioOptions());
+    video_channel_ = channel_manager_.CreateVideoChannel(
+        &fake_media_controller_, &fake_transport_controller_, cricket::CN_VIDEO,
+        nullptr, false, cricket::VideoOptions());
+    voice_media_channel_ = media_engine_->GetVoiceChannel(0);
+    video_media_channel_ = media_engine_->GetVideoChannel(0);
+    RTC_CHECK(voice_channel_);
+    RTC_CHECK(video_channel_);
+    RTC_CHECK(voice_media_channel_);
+    RTC_CHECK(video_media_channel_);
+
+    // Create streams for predefined SSRCs. Streams need to exist in order
+    // for the senders and receievers to apply parameters to them.
+    // Normally these would be created by SetLocalDescription and
+    // SetRemoteDescription.
+    voice_media_channel_->AddSendStream(
+        cricket::StreamParams::CreateLegacy(kAudioSsrc));
+    voice_media_channel_->AddRecvStream(
+        cricket::StreamParams::CreateLegacy(kAudioSsrc));
+    voice_media_channel_->AddSendStream(
+        cricket::StreamParams::CreateLegacy(kAudioSsrc2));
+    voice_media_channel_->AddRecvStream(
+        cricket::StreamParams::CreateLegacy(kAudioSsrc2));
+    video_media_channel_->AddSendStream(
+        cricket::StreamParams::CreateLegacy(kVideoSsrc));
+    video_media_channel_->AddRecvStream(
+        cricket::StreamParams::CreateLegacy(kVideoSsrc));
+    video_media_channel_->AddSendStream(
+        cricket::StreamParams::CreateLegacy(kVideoSsrc2));
+    video_media_channel_->AddRecvStream(
+        cricket::StreamParams::CreateLegacy(kVideoSsrc2));
   }
 
+  void TearDown() override { channel_manager_.Terminate(); }
+
   void AddVideoTrack() {
     rtc::scoped_refptr<VideoTrackSourceInterface> source(
         FakeVideoTrackSource::Create());
@@ -112,68 +105,128 @@ class RtpSenderReceiverTest : public testing::Test {
     EXPECT_TRUE(stream_->AddTrack(video_track_));
   }
 
-  void CreateAudioRtpSender() {
-    audio_track_ = AudioTrack::Create(kAudioTrackId, NULL);
+  void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); }
+
+  void CreateAudioRtpSender(rtc::scoped_refptr<LocalAudioSource> source) {
+    audio_track_ = AudioTrack::Create(kAudioTrackId, source);
     EXPECT_TRUE(stream_->AddTrack(audio_track_));
-    EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
     audio_rtp_sender_ =
         new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(),
-                           &audio_provider_, nullptr);
+                           voice_channel_, nullptr);
     audio_rtp_sender_->SetSsrc(kAudioSsrc);
+    VerifyVoiceChannelInput();
   }
 
   void CreateVideoRtpSender() {
     AddVideoTrack();
-    EXPECT_CALL(video_provider_,
-                SetVideoSend(kVideoSsrc, true, _, video_track_.get()));
     video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
-                                           stream_->label(), &video_provider_);
+                                           stream_->label(), video_channel_);
     video_rtp_sender_->SetSsrc(kVideoSsrc);
+    VerifyVideoChannelInput();
   }
 
   void DestroyAudioRtpSender() {
-    EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _))
-        .Times(1);
     audio_rtp_sender_ = nullptr;
+    VerifyVoiceChannelNoInput();
   }
 
   void DestroyVideoRtpSender() {
-    EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr))
-        .Times(1);
     video_rtp_sender_ = nullptr;
+    VerifyVideoChannelNoInput();
   }
 
   void CreateAudioRtpReceiver() {
     audio_track_ = AudioTrack::Create(
         kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL));
     EXPECT_TRUE(stream_->AddTrack(audio_track_));
-    EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true));
     audio_rtp_receiver_ = new AudioRtpReceiver(stream_, kAudioTrackId,
-                                               kAudioSsrc, &audio_provider_);
+                                               kAudioSsrc, voice_channel_);
     audio_track_ = audio_rtp_receiver_->audio_track();
+    VerifyVoiceChannelOutput();
   }
 
   void CreateVideoRtpReceiver() {
-    EXPECT_CALL(video_provider_, SetVideoPlayout(kVideoSsrc, true, _));
     video_rtp_receiver_ =
         new VideoRtpReceiver(stream_, kVideoTrackId, rtc::Thread::Current(),
-                             kVideoSsrc, &video_provider_);
+                             kVideoSsrc, video_channel_);
     video_track_ = video_rtp_receiver_->video_track();
+    VerifyVideoChannelOutput();
   }
 
   void DestroyAudioRtpReceiver() {
-    EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, false));
     audio_rtp_receiver_ = nullptr;
+    VerifyVoiceChannelNoOutput();
   }
 
   void DestroyVideoRtpReceiver() {
-    EXPECT_CALL(video_provider_, SetVideoPlayout(kVideoSsrc, false, NULL));
     video_rtp_receiver_ = nullptr;
+    VerifyVideoChannelNoOutput();
+  }
+
+  void VerifyVoiceChannelInput() { VerifyVoiceChannelInput(kAudioSsrc); }
+
+  void VerifyVoiceChannelInput(uint32_t ssrc) {
+    // Verify that the media channel has an audio source, and the stream isn't
+    // muted.
+    EXPECT_TRUE(voice_media_channel_->HasSource(ssrc));
+    EXPECT_FALSE(voice_media_channel_->IsStreamMuted(ssrc));
+  }
+
+  void VerifyVideoChannelInput() { VerifyVideoChannelInput(kVideoSsrc); }
+
+  void VerifyVideoChannelInput(uint32_t ssrc) {
+    // Verify that the media channel has a video source,
+    EXPECT_TRUE(video_media_channel_->HasSource(ssrc));
+  }
+
+  void VerifyVoiceChannelNoInput() { VerifyVoiceChannelNoInput(kAudioSsrc); }
+
+  void VerifyVoiceChannelNoInput(uint32_t ssrc) {
+    // Verify that the media channel's source is reset.
+    EXPECT_FALSE(voice_media_channel_->HasSource(ssrc));
+  }
+
+  void VerifyVideoChannelNoInput() { VerifyVideoChannelNoInput(kVideoSsrc); }
+
+  void VerifyVideoChannelNoInput(uint32_t ssrc) {
+    // Verify that the media channel's source is reset.
+    EXPECT_FALSE(video_media_channel_->HasSource(ssrc));
+  }
+
+  void VerifyVoiceChannelOutput() {
+    // Verify that the volume is initialized to 1.
+    double volume;
+    EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
+    EXPECT_EQ(1, volume);
+  }
+
+  void VerifyVideoChannelOutput() {
+    // Verify that the media channel has a sink.
+    EXPECT_TRUE(video_media_channel_->HasSink(kVideoSsrc));
+  }
+
+  void VerifyVoiceChannelNoOutput() {
+    // Verify that the volume is reset to 0.
+    double volume;
+    EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
+    EXPECT_EQ(0, volume);
+  }
+
+  void VerifyVideoChannelNoOutput() {
+    // Verify that the media channel's sink is reset.
+    EXPECT_FALSE(video_media_channel_->HasSink(kVideoSsrc));
   }
 
  protected:
-  MockAudioProvider audio_provider_;
-  MockVideoProvider video_provider_;
+  cricket::FakeMediaEngine* media_engine_;
+  cricket::FakeTransportController fake_transport_controller_;
+  cricket::ChannelManager channel_manager_;
+  cricket::FakeCall fake_call_;
+  cricket::FakeMediaController fake_media_controller_;
+  cricket::VoiceChannel* voice_channel_;
+  cricket::VideoChannel* video_channel_;
+  cricket::FakeVoiceMediaChannel* voice_media_channel_;
+  cricket::FakeVideoMediaChannel* video_media_channel_;
   rtc::scoped_refptr<AudioRtpSender> audio_rtp_sender_;
   rtc::scoped_refptr<VideoRtpSender> video_rtp_sender_;
   rtc::scoped_refptr<AudioRtpReceiver> audio_rtp_receiver_;
@@ -183,72 +236,96 @@ class RtpSenderReceiverTest : public testing::Test {
   rtc::scoped_refptr<AudioTrackInterface> audio_track_;
 };
 
-// Test that |audio_provider_| is notified when an audio track is associated
+// Test that |voice_channel_| is updated when an audio track is associated
 // and disassociated with an AudioRtpSender.
 TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpSender) {
   CreateAudioRtpSender();
   DestroyAudioRtpSender();
 }
 
-// Test that |video_provider_| is notified when a video track is associated and
+// Test that |video_channel_| is updated when a video track is associated and
 // disassociated with a VideoRtpSender.
 TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpSender) {
   CreateVideoRtpSender();
   DestroyVideoRtpSender();
 }
 
-// Test that |audio_provider_| is notified when a remote audio and track is
+// Test that |voice_channel_| is updated when a remote audio track is
 // associated and disassociated with an AudioRtpReceiver.
 TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpReceiver) {
   CreateAudioRtpReceiver();
   DestroyAudioRtpReceiver();
 }
 
-// Test that |video_provider_| is notified when a remote
-// video track is associated and disassociated with a VideoRtpReceiver.
+// Test that |video_channel_| is updated when a remote video track is
+// associated and disassociated with a VideoRtpReceiver.
 TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiver) {
   CreateVideoRtpReceiver();
   DestroyVideoRtpReceiver();
 }
 
-TEST_F(RtpSenderReceiverTest, LocalAudioTrackDisable) {
-  CreateAudioRtpSender();
+// Test that the AudioRtpSender applies options from the local audio source.
+TEST_F(RtpSenderReceiverTest, LocalAudioSourceOptionsApplied) {
+  cricket::AudioOptions options;
+  options.echo_cancellation = rtc::Optional<bool>(true);
+  auto source = LocalAudioSource::Create(
+      PeerConnectionFactoryInterface::Options(), &options);
+  CreateAudioRtpSender(source.get());
 
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _));
-  audio_track_->set_enabled(false);
-
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
-  audio_track_->set_enabled(true);
+  EXPECT_EQ(rtc::Optional<bool>(true),
+            voice_media_channel_->options().echo_cancellation);
 
   DestroyAudioRtpSender();
 }
 
+// Test that the stream is muted when the track is disabled, and unmuted when
+// the track is enabled.
+TEST_F(RtpSenderReceiverTest, LocalAudioTrackDisable) {
+  CreateAudioRtpSender();
+
+  audio_track_->set_enabled(false);
+  EXPECT_TRUE(voice_media_channel_->IsStreamMuted(kAudioSsrc));
+
+  audio_track_->set_enabled(true);
+  EXPECT_FALSE(voice_media_channel_->IsStreamMuted(kAudioSsrc));
+
+  DestroyAudioRtpSender();
+}
+
+// Test that the volume is set to 0 when the track is disabled, and back to
+// 1 when the track is enabled.
 TEST_F(RtpSenderReceiverTest, RemoteAudioTrackDisable) {
   CreateAudioRtpReceiver();
 
-  EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, false));
-  audio_track_->set_enabled(false);
+  double volume;
+  EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
+  EXPECT_EQ(1, volume);
+
+  audio_track_->set_enabled(false);
+  EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
+  EXPECT_EQ(0, volume);
 
-  EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true));
   audio_track_->set_enabled(true);
+  EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
+  EXPECT_EQ(1, volume);
 
   DestroyAudioRtpReceiver();
 }
 
+// Currently no action is taken when a remote video track is disabled or
+// enabled, so there's nothing to test here, other than what is normally
+// verified in DestroyVideoRtpSender.
 TEST_F(RtpSenderReceiverTest, LocalVideoTrackDisable) {
   CreateVideoRtpSender();
 
-  EXPECT_CALL(video_provider_,
-              SetVideoSend(kVideoSsrc, false, _, video_track_.get()));
   video_track_->set_enabled(false);
-
-  EXPECT_CALL(video_provider_,
-              SetVideoSend(kVideoSsrc, true, _, video_track_.get()));
   video_track_->set_enabled(true);
 
   DestroyVideoRtpSender();
 }
 
+// Test that the state of the video track created by the VideoRtpReceiver is
+// updated when the receiver is destroyed.
 TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) {
   CreateVideoRtpReceiver();
 
@@ -263,282 +340,268 @@ TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) {
             video_track_->GetSource()->state());
 }
 
+// Currently no action is taken when a remote video track is disabled or
+// enabled, so there's nothing to test here, other than what is normally
+// verified in DestroyVideoRtpReceiver.
 TEST_F(RtpSenderReceiverTest, RemoteVideoTrackDisable) {
   CreateVideoRtpReceiver();
 
   video_track_->set_enabled(false);
-
   video_track_->set_enabled(true);
 
   DestroyVideoRtpReceiver();
 }
 
+// Test that the AudioRtpReceiver applies volume changes from the track source
+// to the media channel.
 TEST_F(RtpSenderReceiverTest, RemoteAudioTrackSetVolume) {
   CreateAudioRtpReceiver();
 
-  double volume = 0.5;
-  EXPECT_CALL(audio_provider_, SetAudioPlayoutVolume(kAudioSsrc, volume));
-  audio_track_->GetSource()->SetVolume(volume);
+  double volume;
+  audio_track_->GetSource()->SetVolume(0.5);
+  EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
+  EXPECT_EQ(0.5, volume);
 
   // Disable the audio track, this should prevent setting the volume.
-  EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, false));
   audio_track_->set_enabled(false);
-  audio_track_->GetSource()->SetVolume(1.0);
+  audio_track_->GetSource()->SetVolume(0.8);
+  EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
+  EXPECT_EQ(0, volume);
 
-  EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true));
+  // When the track is enabled, the previously set volume should take effect.
   audio_track_->set_enabled(true);
+  EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
+  EXPECT_EQ(0.8, volume);
 
-  double new_volume = 0.8;
-  EXPECT_CALL(audio_provider_, SetAudioPlayoutVolume(kAudioSsrc, new_volume));
-  audio_track_->GetSource()->SetVolume(new_volume);
+  // Try changing volume one more time.
+  audio_track_->GetSource()->SetVolume(0.9);
+  EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
+  EXPECT_EQ(0.9, volume);
 
   DestroyAudioRtpReceiver();
 }
 
-// Test that provider methods aren't called without both a track and an SSRC.
+// Test that the media channel isn't enabled for sending if the audio sender
+// doesn't have both a track and SSRC.
 TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) {
-  rtc::scoped_refptr<AudioRtpSender> sender =
-      new AudioRtpSender(&audio_provider_, nullptr);
+  audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr);
   rtc::scoped_refptr<AudioTrackInterface> track =
       AudioTrack::Create(kAudioTrackId, nullptr);
-  EXPECT_TRUE(sender->SetTrack(track));
-  EXPECT_TRUE(sender->SetTrack(nullptr));
-  sender->SetSsrc(kAudioSsrc);
-  sender->SetSsrc(0);
-  // Just let it get destroyed and make sure it doesn't call any methods on the
-  // provider interface.
+
+  // Track but no SSRC.
+  EXPECT_TRUE(audio_rtp_sender_->SetTrack(track));
+  VerifyVoiceChannelNoInput();
+
+  // SSRC but no track.
+  EXPECT_TRUE(audio_rtp_sender_->SetTrack(nullptr));
+  audio_rtp_sender_->SetSsrc(kAudioSsrc);
+  VerifyVoiceChannelNoInput();
 }
 
-// Test that provider methods aren't called without both a track and an SSRC.
+// Test that the media channel isn't enabled for sending if the video sender
+// doesn't have both a track and SSRC.
 TEST_F(RtpSenderReceiverTest, VideoSenderWithoutTrackAndSsrc) {
-  rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(&video_provider_);
-  EXPECT_TRUE(sender->SetTrack(video_track_));
-  EXPECT_TRUE(sender->SetTrack(nullptr));
-  sender->SetSsrc(kVideoSsrc);
-  sender->SetSsrc(0);
-  // Just let it get destroyed and make sure it doesn't call any methods on the
-  // provider interface.
+  video_rtp_sender_ = new VideoRtpSender(video_channel_);
+
+  // Track but no SSRC.
+  EXPECT_TRUE(video_rtp_sender_->SetTrack(video_track_));
+  VerifyVideoChannelNoInput();
+
+  // SSRC but no track.
+  EXPECT_TRUE(video_rtp_sender_->SetTrack(nullptr));
+  video_rtp_sender_->SetSsrc(kVideoSsrc);
+  VerifyVideoChannelNoInput();
 }
 
-// Test that an audio sender calls the expected methods on the provider once
-// it has a track and SSRC, when the SSRC is set first.
+// Test that the media channel is enabled for sending when the audio sender
+// has a track and SSRC, when the SSRC is set first.
 TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupSsrcThenTrack) {
-  rtc::scoped_refptr<AudioRtpSender> sender =
-      new AudioRtpSender(&audio_provider_, nullptr);
+  audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr);
   rtc::scoped_refptr<AudioTrackInterface> track =
       AudioTrack::Create(kAudioTrackId, nullptr);
-  sender->SetSsrc(kAudioSsrc);
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
-  sender->SetTrack(track);
+  audio_rtp_sender_->SetSsrc(kAudioSsrc);
+  audio_rtp_sender_->SetTrack(track);
+  VerifyVoiceChannelInput();
 
-  // Calls expected from destructor.
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
+  DestroyAudioRtpSender();
 }
 
-// Test that an audio sender calls the expected methods on the provider once
-// it has a track and SSRC, when the SSRC is set last.
+// Test that the media channel is enabled for sending when the audio sender
+// has a track and SSRC, when the SSRC is set last.
 TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupTrackThenSsrc) {
-  rtc::scoped_refptr<AudioRtpSender> sender =
-      new AudioRtpSender(&audio_provider_, nullptr);
+  audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr);
   rtc::scoped_refptr<AudioTrackInterface> track =
       AudioTrack::Create(kAudioTrackId, nullptr);
-  sender->SetTrack(track);
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
-  sender->SetSsrc(kAudioSsrc);
+  audio_rtp_sender_->SetTrack(track);
+  audio_rtp_sender_->SetSsrc(kAudioSsrc);
+  VerifyVoiceChannelInput();
 
-  // Calls expected from destructor.
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
+  DestroyAudioRtpSender();
 }
 
-// Test that a video sender calls the expected methods on the provider once
-// it has a track and SSRC, when the SSRC is set first.
+// Test that the media channel is enabled for sending when the video sender
+// has a track and SSRC, when the SSRC is set first.
 TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) {
   AddVideoTrack();
-  rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(&video_provider_);
-  sender->SetSsrc(kVideoSsrc);
-  EXPECT_CALL(video_provider_,
-              SetVideoSend(kVideoSsrc, true, _, video_track_.get()));
-  sender->SetTrack(video_track_);
+  video_rtp_sender_ = new VideoRtpSender(video_channel_);
+  video_rtp_sender_->SetSsrc(kVideoSsrc);
+  video_rtp_sender_->SetTrack(video_track_);
+  VerifyVideoChannelInput();
 
-  // Calls expected from destructor.
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr))
-      .Times(1);
+  DestroyVideoRtpSender();
 }
 
-// Test that a video sender calls the expected methods on the provider once
-// it has a track and SSRC, when the SSRC is set last.
+// Test that the media channel is enabled for sending when the video sender
+// has a track and SSRC, when the SSRC is set last.
 TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) {
   AddVideoTrack();
-  rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(&video_provider_);
-  sender->SetTrack(video_track_);
-  EXPECT_CALL(video_provider_,
-              SetVideoSend(kVideoSsrc, true, _, video_track_.get()));
-  sender->SetSsrc(kVideoSsrc);
+  video_rtp_sender_ = new VideoRtpSender(video_channel_);
+  video_rtp_sender_->SetTrack(video_track_);
+  video_rtp_sender_->SetSsrc(kVideoSsrc);
+  VerifyVideoChannelInput();
 
-  // Calls expected from destructor.
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr))
-      .Times(1);
+  DestroyVideoRtpSender();
 }
 
-// Test that the sender is disconnected from the provider when its SSRC is
-// set to 0.
+// Test that the media channel stops sending when the audio sender's SSRC is set
+// to 0.
 TEST_F(RtpSenderReceiverTest, AudioSenderSsrcSetToZero) {
-  rtc::scoped_refptr<AudioTrackInterface> track =
-      AudioTrack::Create(kAudioTrackId, nullptr);
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
-  rtc::scoped_refptr<AudioRtpSender> sender =
-      new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
-  sender->SetSsrc(kAudioSsrc);
+  CreateAudioRtpSender();
 
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
-  sender->SetSsrc(0);
-
-  // Make sure it's SetSsrc that called methods on the provider, and not the
-  // destructor.
-  EXPECT_CALL(audio_provider_, SetAudioSend(_, _, _, _)).Times(0);
+  audio_rtp_sender_->SetSsrc(0);
+  VerifyVoiceChannelNoInput();
 }
 
-// Test that the sender is disconnected from the provider when its SSRC is
-// set to 0.
+// Test that the media channel stops sending when the video sender's SSRC is set
+// to 0.
 TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) {
-  AddVideoTrack();
-  EXPECT_CALL(video_provider_,
-              SetVideoSend(kVideoSsrc, true, _, video_track_.get()));
-  rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
-  sender->SetSsrc(kVideoSsrc);
+  CreateAudioRtpSender();
 
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr))
-      .Times(1);
-  sender->SetSsrc(0);
-
-  // Make sure it's SetSsrc that called methods on the provider, and not the
-  // destructor.
-  EXPECT_CALL(video_provider_, SetVideoSend(_, _, _, _)).Times(0);
+  audio_rtp_sender_->SetSsrc(0);
+  VerifyVideoChannelNoInput();
 }
 
+// Test that the media channel stops sending when the audio sender's track is
+// set to null.
 TEST_F(RtpSenderReceiverTest, AudioSenderTrackSetToNull) {
-  rtc::scoped_refptr<AudioTrackInterface> track =
-      AudioTrack::Create(kAudioTrackId, nullptr);
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
-  rtc::scoped_refptr<AudioRtpSender> sender =
-      new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
-  sender->SetSsrc(kAudioSsrc);
+  CreateAudioRtpSender();
 
-  // Expect that SetAudioSend will be called before the reference to the track
-  // is released.
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, nullptr))
-      .Times(1)
-      .WillOnce(InvokeWithoutArgs([&track] {
-        EXPECT_LT(2, track->AddRef());
-        track->Release();
-      }));
-  EXPECT_TRUE(sender->SetTrack(nullptr));
-
-  // Make sure it's SetTrack that called methods on the provider, and not the
-  // destructor.
-  EXPECT_CALL(audio_provider_, SetAudioSend(_, _, _, _)).Times(0);
+  EXPECT_TRUE(audio_rtp_sender_->SetTrack(nullptr));
+  VerifyVoiceChannelNoInput();
 }
 
+// Test that the media channel stops sending when the video sender's track is
+// set to null.
 TEST_F(RtpSenderReceiverTest, VideoSenderTrackSetToNull) {
-  rtc::scoped_refptr<VideoTrackSourceInterface> source(
-      FakeVideoTrackSource::Create());
-  rtc::scoped_refptr<VideoTrackInterface> track =
-      VideoTrack::Create(kVideoTrackId, source);
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _, track.get()));
-  rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(track, kStreamLabel1, &video_provider_);
-  sender->SetSsrc(kVideoSsrc);
+  CreateVideoRtpSender();
 
-  // Expect that SetVideoSend will be called before the reference to the track
-  // is released.
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr))
-      .Times(1)
-      .WillOnce(InvokeWithoutArgs([&track] {
-        EXPECT_LT(2, track->AddRef());
-        track->Release();
-      }));
-  EXPECT_TRUE(sender->SetTrack(nullptr));
-
-  // Make sure it's SetTrack that called methods on the provider, and not the
-  // destructor.
-  EXPECT_CALL(video_provider_, SetVideoSend(_, _, _, _)).Times(0);
+  video_rtp_sender_->SetSsrc(0);
+  VerifyVideoChannelNoInput();
 }
 
+// Test that when the audio sender's SSRC is changed, the media channel stops
+// sending with the old SSRC and starts sending with the new one.
 TEST_F(RtpSenderReceiverTest, AudioSenderSsrcChanged) {
-  AddVideoTrack();
-  rtc::scoped_refptr<AudioTrackInterface> track =
-      AudioTrack::Create(kAudioTrackId, nullptr);
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
-  rtc::scoped_refptr<AudioRtpSender> sender =
-      new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
-  sender->SetSsrc(kAudioSsrc);
+  CreateAudioRtpSender();
 
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc2, true, _, _)).Times(1);
-  sender->SetSsrc(kAudioSsrc2);
+  audio_rtp_sender_->SetSsrc(kAudioSsrc2);
+  VerifyVoiceChannelNoInput(kAudioSsrc);
+  VerifyVoiceChannelInput(kAudioSsrc2);
 
-  // Calls expected from destructor.
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc2, false, _, _)).Times(1);
+  audio_rtp_sender_ = nullptr;
+  VerifyVoiceChannelNoInput(kAudioSsrc2);
 }
 
+// Test that when the audio sender's SSRC is changed, the media channel stops
+// sending with the old SSRC and starts sending with the new one.
 TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) {
-  AddVideoTrack();
-  EXPECT_CALL(video_provider_,
-              SetVideoSend(kVideoSsrc, true, _, video_track_.get()));
-  rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
-  sender->SetSsrc(kVideoSsrc);
+  CreateVideoRtpSender();
 
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr))
-      .Times(1);
-  EXPECT_CALL(video_provider_,
-              SetVideoSend(kVideoSsrc2, true, _, video_track_.get()))
-      .Times(1);
-  sender->SetSsrc(kVideoSsrc2);
+  video_rtp_sender_->SetSsrc(kVideoSsrc2);
+  VerifyVideoChannelNoInput(kVideoSsrc);
+  VerifyVideoChannelInput(kVideoSsrc2);
 
-  // Calls expected from destructor.
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _, nullptr))
-      .Times(1);
+  video_rtp_sender_ = nullptr;
+  VerifyVideoChannelNoInput(kVideoSsrc2);
 }
 
 TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) {
   CreateAudioRtpSender();
 
-  EXPECT_CALL(audio_provider_, GetAudioRtpSendParameters(kAudioSsrc))
-      .WillOnce(Return(RtpParameters()));
-  EXPECT_CALL(audio_provider_, SetAudioRtpSendParameters(kAudioSsrc, _))
-      .WillOnce(Return(true));
   RtpParameters params = audio_rtp_sender_->GetParameters();
+  EXPECT_EQ(1u, params.encodings.size());
   EXPECT_TRUE(audio_rtp_sender_->SetParameters(params));
 
   DestroyAudioRtpSender();
 }
 
+TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) {
+  CreateAudioRtpSender();
+
+  EXPECT_EQ(-1, voice_media_channel_->max_bps());
+  webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
+  params.encodings[0].max_bitrate_bps = 1000;
+  EXPECT_TRUE(audio_rtp_sender_->SetParameters(params));
+
+  // Read back the parameters and verify they have been changed.
+  params = audio_rtp_sender_->GetParameters();
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the audio channel received the new parameters.
+  params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the global bitrate limit has not been changed.
+  EXPECT_EQ(-1, voice_media_channel_->max_bps());
+
+  DestroyAudioRtpSender();
+}
+
 TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
   CreateVideoRtpSender();
 
-  EXPECT_CALL(video_provider_, GetVideoRtpSendParameters(kVideoSsrc))
-      .WillOnce(Return(RtpParameters()));
-  EXPECT_CALL(video_provider_, SetVideoRtpSendParameters(kVideoSsrc, _))
-      .WillOnce(Return(true));
   RtpParameters params = video_rtp_sender_->GetParameters();
+  EXPECT_EQ(1u, params.encodings.size());
   EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
 
   DestroyVideoRtpSender();
 }
 
+TEST_F(RtpSenderReceiverTest, SetVideoMaxSendBitrate) {
+  CreateVideoRtpSender();
+
+  EXPECT_EQ(-1, video_media_channel_->max_bps());
+  webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
+  params.encodings[0].max_bitrate_bps = 1000;
+  EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
+
+  // Read back the parameters and verify they have been changed.
+  params = video_rtp_sender_->GetParameters();
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the video channel received the new parameters.
+  params = video_media_channel_->GetRtpSendParameters(kVideoSsrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the global bitrate limit has not been changed.
+  EXPECT_EQ(-1, video_media_channel_->max_bps());
+
+  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_EQ(1u, params.encodings.size());
   EXPECT_TRUE(audio_rtp_receiver_->SetParameters(params));
 
   DestroyAudioRtpReceiver();
@@ -547,11 +610,8 @@ TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) {
 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_EQ(1u, params.encodings.size());
   EXPECT_TRUE(video_rtp_receiver_->SetParameters(params));
 
   DestroyVideoRtpReceiver();