Create //webrtc/api:libjingle_peerconnection_api + refactorings.

webrtc/api/rtpsenderreceiver_unittest.cc

Index: webrtc/api/rtpsenderreceiver_unittest.cc
diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc
deleted file mode 100644
index f4f550e9068dd872f608a0e442df37584a5cd6aa..0000000000000000000000000000000000000000
--- a/webrtc/api/rtpsenderreceiver_unittest.cc
+++ /dev/null
@@ -1,624 +0,0 @@
-/*
- *  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 <memory>
-#include <string>
-#include <utility>
-
-#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"
-#include "webrtc/api/rtpsender.h"
-#include "webrtc/api/streamcollection.h"
-#include "webrtc/api/test/fakevideotracksource.h"
-#include "webrtc/api/videotrack.h"
-#include "webrtc/api/videotracksource.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/media/base/fakemediaengine.h"
-#include "webrtc/media/base/mediachannel.h"
-#include "webrtc/media/engine/fakewebrtccall.h"
-#include "webrtc/p2p/base/faketransportcontroller.h"
-#include "webrtc/pc/channelmanager.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
-
-using ::testing::_;
-using ::testing::Exactly;
-using ::testing::InvokeWithoutArgs;
-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 {
-
-class RtpSenderReceiverTest : public testing::Test {
- public:
-  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_(Call::Config(&event_log_)),
-        fake_media_controller_(&channel_manager_, &fake_call_),
-        stream_(MediaStream::Create(kStreamLabel1)) {
-    // Create channels to be used by the RtpSenders and RtpReceivers.
-    channel_manager_.Init();
-    bool rtcp_enabled = false;
-    bool srtp_required = true;
-    voice_channel_ = channel_manager_.CreateVoiceChannel(
-        &fake_media_controller_, &fake_transport_controller_, cricket::CN_AUDIO,
-        nullptr, rtcp_enabled, srtp_required, cricket::AudioOptions());
-    video_channel_ = channel_manager_.CreateVideoChannel(
-        &fake_media_controller_, &fake_transport_controller_, cricket::CN_VIDEO,
-        nullptr, rtcp_enabled, srtp_required, 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());
-    video_track_ = VideoTrack::Create(kVideoTrackId, source);
-    EXPECT_TRUE(stream_->AddTrack(video_track_));
-  }
-
-  void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); }
-
-  void CreateAudioRtpSender(rtc::scoped_refptr<LocalAudioSource> source) {
-    audio_track_ = AudioTrack::Create(kAudioTrackId, source);
-    EXPECT_TRUE(stream_->AddTrack(audio_track_));
-    audio_rtp_sender_ =
-        new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(),
-                           voice_channel_, nullptr);
-    audio_rtp_sender_->SetSsrc(kAudioSsrc);
-    VerifyVoiceChannelInput();
-  }
-
-  void CreateVideoRtpSender() {
-    AddVideoTrack();
-    video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
-                                           stream_->label(), video_channel_);
-    video_rtp_sender_->SetSsrc(kVideoSsrc);
-    VerifyVideoChannelInput();
-  }
-
-  void DestroyAudioRtpSender() {
-    audio_rtp_sender_ = nullptr;
-    VerifyVoiceChannelNoInput();
-  }
-
-  void DestroyVideoRtpSender() {
-    video_rtp_sender_ = nullptr;
-    VerifyVideoChannelNoInput();
-  }
-
-  void CreateAudioRtpReceiver() {
-    audio_track_ = AudioTrack::Create(
-        kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL));
-    EXPECT_TRUE(stream_->AddTrack(audio_track_));
-    audio_rtp_receiver_ = new AudioRtpReceiver(stream_, kAudioTrackId,
-                                               kAudioSsrc, voice_channel_);
-    audio_track_ = audio_rtp_receiver_->audio_track();
-    VerifyVoiceChannelOutput();
-  }
-
-  void CreateVideoRtpReceiver() {
-    video_rtp_receiver_ =
-        new VideoRtpReceiver(stream_, kVideoTrackId, rtc::Thread::Current(),
-                             kVideoSsrc, video_channel_);
-    video_track_ = video_rtp_receiver_->video_track();
-    VerifyVideoChannelOutput();
-  }
-
-  void DestroyAudioRtpReceiver() {
-    audio_rtp_receiver_ = nullptr;
-    VerifyVoiceChannelNoOutput();
-  }
-
-  void DestroyVideoRtpReceiver() {
-    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:
-  webrtc::RtcEventLogNullImpl event_log_;
-  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_;
-  rtc::scoped_refptr<VideoRtpReceiver> video_rtp_receiver_;
-  rtc::scoped_refptr<MediaStreamInterface> stream_;
-  rtc::scoped_refptr<VideoTrackInterface> video_track_;
-  rtc::scoped_refptr<AudioTrackInterface> audio_track_;
-};
-
-// 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_channel_| is updated when a video track is associated and
-// disassociated with a VideoRtpSender.
-TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpSender) {
-  CreateVideoRtpSender();
-  DestroyVideoRtpSender();
-}
-
-// 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_channel_| is updated when a remote video track is
-// associated and disassociated with a VideoRtpReceiver.
-TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiver) {
-  CreateVideoRtpReceiver();
-  DestroyVideoRtpReceiver();
-}
-
-// 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_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();
-
-  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);
-
-  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();
-
-  video_track_->set_enabled(false);
-  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();
-
-  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track_->state());
-  EXPECT_EQ(webrtc::MediaSourceInterface::kLive,
-            video_track_->GetSource()->state());
-
-  DestroyVideoRtpReceiver();
-
-  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state());
-  EXPECT_EQ(webrtc::MediaSourceInterface::kEnded,
-            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;
-  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.
-  audio_track_->set_enabled(false);
-  audio_track_->GetSource()->SetVolume(0.8);
-  EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
-  EXPECT_EQ(0, volume);
-
-  // 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);
-
-  // 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 the media channel isn't enabled for sending if the audio sender
-// doesn't have both a track and SSRC.
-TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) {
-  audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr);
-  rtc::scoped_refptr<AudioTrackInterface> track =
-      AudioTrack::Create(kAudioTrackId, nullptr);
-
-  // 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 the media channel isn't enabled for sending if the video sender
-// doesn't have both a track and SSRC.
-TEST_F(RtpSenderReceiverTest, VideoSenderWithoutTrackAndSsrc) {
-  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 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) {
-  audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr);
-  rtc::scoped_refptr<AudioTrackInterface> track =
-      AudioTrack::Create(kAudioTrackId, nullptr);
-  audio_rtp_sender_->SetSsrc(kAudioSsrc);
-  audio_rtp_sender_->SetTrack(track);
-  VerifyVoiceChannelInput();
-
-  DestroyAudioRtpSender();
-}
-
-// 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) {
-  audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr);
-  rtc::scoped_refptr<AudioTrackInterface> track =
-      AudioTrack::Create(kAudioTrackId, nullptr);
-  audio_rtp_sender_->SetTrack(track);
-  audio_rtp_sender_->SetSsrc(kAudioSsrc);
-  VerifyVoiceChannelInput();
-
-  DestroyAudioRtpSender();
-}
-
-// 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();
-  video_rtp_sender_ = new VideoRtpSender(video_channel_);
-  video_rtp_sender_->SetSsrc(kVideoSsrc);
-  video_rtp_sender_->SetTrack(video_track_);
-  VerifyVideoChannelInput();
-
-  DestroyVideoRtpSender();
-}
-
-// 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();
-  video_rtp_sender_ = new VideoRtpSender(video_channel_);
-  video_rtp_sender_->SetTrack(video_track_);
-  video_rtp_sender_->SetSsrc(kVideoSsrc);
-  VerifyVideoChannelInput();
-
-  DestroyVideoRtpSender();
-}
-
-// Test that the media channel stops sending when the audio sender's SSRC is set
-// to 0.
-TEST_F(RtpSenderReceiverTest, AudioSenderSsrcSetToZero) {
-  CreateAudioRtpSender();
-
-  audio_rtp_sender_->SetSsrc(0);
-  VerifyVoiceChannelNoInput();
-}
-
-// Test that the media channel stops sending when the video sender's SSRC is set
-// to 0.
-TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) {
-  CreateAudioRtpSender();
-
-  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) {
-  CreateAudioRtpSender();
-
-  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) {
-  CreateVideoRtpSender();
-
-  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) {
-  CreateAudioRtpSender();
-
-  audio_rtp_sender_->SetSsrc(kAudioSsrc2);
-  VerifyVoiceChannelNoInput(kAudioSsrc);
-  VerifyVoiceChannelInput(kAudioSsrc2);
-
-  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) {
-  CreateVideoRtpSender();
-
-  video_rtp_sender_->SetSsrc(kVideoSsrc2);
-  VerifyVideoChannelNoInput(kVideoSsrc);
-  VerifyVideoChannelInput(kVideoSsrc2);
-
-  video_rtp_sender_ = nullptr;
-  VerifyVideoChannelNoInput(kVideoSsrc2);
-}
-
-TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) {
-  CreateAudioRtpSender();
-
-  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();
-
-  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();
-
-  RtpParameters params = audio_rtp_receiver_->GetParameters();
-  EXPECT_EQ(1u, params.encodings.size());
-  EXPECT_TRUE(audio_rtp_receiver_->SetParameters(params));
-
-  DestroyAudioRtpReceiver();
-}
-
-TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetParameters) {
-  CreateVideoRtpReceiver();
-
-  RtpParameters params = video_rtp_receiver_->GetParameters();
-  EXPECT_EQ(1u, params.encodings.size());
-  EXPECT_TRUE(video_rtp_receiver_->SetParameters(params));
-
-  DestroyVideoRtpReceiver();
-}
-
-}  // namespace webrtc