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Unified Diff: webrtc/pc/rtpsenderreceiver_unittest.cc

Issue 2675173003: Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. (Closed)
Patch Set: Add memcheck suppression for end-to-end tests. Created 3 years, 10 months ago
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Index: webrtc/pc/rtpsenderreceiver_unittest.cc
diff --git a/webrtc/pc/rtpsenderreceiver_unittest.cc b/webrtc/pc/rtpsenderreceiver_unittest.cc
index 105d9d31d49b8a936782d73260b778ebe07b621d..5ddf6c4c81bc4ce5e0ae38026ab0c676c93b1511 100644
--- a/webrtc/pc/rtpsenderreceiver_unittest.cc
+++ b/webrtc/pc/rtpsenderreceiver_unittest.cc
@@ -59,7 +59,7 @@ class RtpSenderReceiverTest : public testing::Test,
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_),
@@ -67,7 +67,7 @@ class RtpSenderReceiverTest : public testing::Test,
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;
@@ -126,17 +126,17 @@ class RtpSenderReceiverTest : public testing::Test,
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);
@@ -149,8 +149,9 @@ class RtpSenderReceiverTest : public testing::Test,
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();
}
@@ -166,19 +167,15 @@ class RtpSenderReceiverTest : public testing::Test,
}
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();
}
@@ -263,7 +260,7 @@ class RtpSenderReceiverTest : public testing::Test,
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;
@@ -717,8 +714,9 @@ TEST_F(RtpSenderReceiverTest,
// 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.
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