Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

webrtc/pc/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 @@
 
 }  // namespace
 
 namespace webrtc {
 
 class RtpSenderReceiverTest : public testing::Test,
                               public sigslot::has_slots<> {
  public:
   RtpSenderReceiverTest()
       :  // Create fake media engine/etc. so we can create channels to use to
-        // test RtpSenders/RtpReceivers.
+         // test RtpSenders/RtpReceivers.
         media_engine_(new cricket::FakeMediaEngine()),
         channel_manager_(
             std::unique_ptr<cricket::MediaEngineInterface>(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)) {
+        local_stream_(MediaStream::Create(kStreamLabel1)) {
     // Create channels to be used by the RtpSenders and RtpReceivers.
     channel_manager_.Init();
     bool srtp_required = true;
     cricket::DtlsTransportInternal* rtp_transport =
         fake_transport_controller_.CreateDtlsTransport(
             cricket::CN_AUDIO, cricket::ICE_CANDIDATE_COMPONENT_RTP);
     voice_channel_ = channel_manager_.CreateVoiceChannel(
         &fake_media_controller_, rtp_transport, nullptr, rtc::Thread::Current(),
         cricket::CN_AUDIO, srtp_required, cricket::AudioOptions());
     video_channel_ = channel_manager_.CreateVideoChannel(
@@ skipping 38 matching lines @@
     params.codecs.push_back(kTelephoneEventCodec);
     voice_media_channel_->SetSendParameters(params);
   }
 
   void AddVideoTrack() { AddVideoTrack(false); }
 
   void AddVideoTrack(bool is_screencast) {
     rtc::scoped_refptr<VideoTrackSourceInterface> source(
         FakeVideoTrackSource::Create(is_screencast));
     video_track_ = VideoTrack::Create(kVideoTrackId, source);
-    EXPECT_TRUE(stream_->AddTrack(video_track_));
+    EXPECT_TRUE(local_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_));
+    EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
     audio_rtp_sender_ =
-        new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(),
-                           voice_channel_, nullptr);
+        new AudioRtpSender(local_stream_->GetAudioTracks()[0],
+                           local_stream_->label(), voice_channel_, nullptr);
     audio_rtp_sender_->SetSsrc(kAudioSsrc);
     audio_rtp_sender_->GetOnDestroyedSignal()->connect(
         this, &RtpSenderReceiverTest::OnAudioSenderDestroyed);
     VerifyVoiceChannelInput();
   }
 
   void OnAudioSenderDestroyed() { audio_sender_destroyed_signal_fired_ = true; }
 
   void CreateVideoRtpSender() { CreateVideoRtpSender(false); }
 
   void CreateVideoRtpSender(bool is_screencast) {
     AddVideoTrack(is_screencast);
-    video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
-                                           stream_->label(), video_channel_);
+    video_rtp_sender_ =
+        new VideoRtpSender(local_stream_->GetVideoTracks()[0],
+                           local_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_rtp_receiver_ =
+        new AudioRtpReceiver(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_rtp_receiver_ = new VideoRtpReceiver(
+        kVideoTrackId, rtc::Thread::Current(), kVideoSsrc, video_channel_);
     video_track_ = video_rtp_receiver_->video_track();
     VerifyVideoChannelOutput();
   }
 
   void DestroyAudioRtpReceiver() {
     audio_rtp_receiver_ = nullptr;
     VerifyVoiceChannelNoOutput();
   }
 
   void DestroyVideoRtpReceiver() {
@@ skipping 64 matching lines @@
   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<MediaStreamInterface> local_stream_;
   rtc::scoped_refptr<VideoTrackInterface> video_track_;
   rtc::scoped_refptr<AudioTrackInterface> audio_track_;
   bool audio_sender_destroyed_signal_fired_ = false;
 };
 
 // Test that |voice_channel_| is updated when an audio track is associated
 // and disassociated with an AudioRtpSender.
 TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpSender) {
   CreateAudioRtpSender();
   DestroyAudioRtpSender();
@@ skipping 433 matching lines @@
 }
 
 // Test that makes sure any content hints that are set on a track before
 // VideoRtpSender is ready to send are still applied when it gets ready to send.
 TEST_F(RtpSenderReceiverTest,
        PropagatesVideoTrackContentHintSetBeforeEnabling) {
   AddVideoTrack();
   // Setting detailed overrides the default non-screencast mode. This should be
   // applied even if the track is set on construction.
   video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
-  video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
-                                         stream_->label(), video_channel_);
+  video_rtp_sender_ =
+      new VideoRtpSender(local_stream_->GetVideoTracks()[0],
+                         local_stream_->label(), video_channel_);
   video_track_->set_enabled(true);
 
   // Sender is not ready to send (no SSRC) so no option should have been set.
   EXPECT_EQ(rtc::Optional<bool>(),
             video_media_channel_->options().is_screencast);
 
   // Verify that the content hint is accounted for when video_rtp_sender_ does
   // get enabled.
   video_rtp_sender_->SetSsrc(kVideoSsrc);
   EXPECT_EQ(rtc::Optional<bool>(true),
@@ skipping 64 matching lines @@
 // Make sure the signal from "GetOnDestroyedSignal()" fires when the sender is
 // destroyed, which is needed for the DTMF sender.
 TEST_F(RtpSenderReceiverTest, TestOnDestroyedSignal) {
   CreateAudioRtpSender();
   EXPECT_FALSE(audio_sender_destroyed_signal_fired_);
   audio_rtp_sender_ = nullptr;
   EXPECT_TRUE(audio_sender_destroyed_signal_fired_);
 }
 
 }  // namespace webrtc