Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(418)

Unified Diff: talk/app/webrtc/rtpsenderreceiver_unittest.cc

Issue 1413713003: Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Adding some unit tests for new methods on the sender. Created 5 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: talk/app/webrtc/rtpsenderreceiver_unittest.cc
diff --git a/talk/app/webrtc/rtpsenderreceiver_unittest.cc b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
index c9d7e008c300475dd8c06aa09d4e4cd141381987..557d452f1601ac1ed040989efb0f8c8f83da2931 100644
--- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc
+++ b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
@@ -120,8 +120,10 @@ class RtpSenderReceiverTest : public testing::Test {
audio_track_ = AudioTrack::Create(kAudioTrackId, NULL);
EXPECT_TRUE(stream_->AddTrack(audio_track_));
EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
- audio_rtp_sender_ = new AudioRtpSender(stream_->GetAudioTracks()[0],
- kAudioSsrc, &audio_provider_);
+ audio_rtp_sender_ =
+ new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(),
+ &audio_provider_, nullptr);
+ audio_rtp_sender_->SetSsrc(kAudioSsrc);
}
void CreateVideoRtpSender() {
@@ -130,7 +132,8 @@ class RtpSenderReceiverTest : public testing::Test {
kVideoSsrc, video_track_->GetSource()->GetVideoCapturer()));
EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
- kVideoSsrc, &video_provider_);
+ stream_->label(), &video_provider_);
+ video_rtp_sender_->SetSsrc(kVideoSsrc);
}
void DestroyAudioRtpSender() {
@@ -280,4 +283,121 @@ TEST_F(RtpSenderReceiverTest, RemoteAudioTrackSetVolume) {
DestroyAudioRtpReceiver();
}
+TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrack) {
+ rtc::scoped_refptr<AudioRtpSender> sender =
+ new AudioRtpSender(&audio_provider_, nullptr);
+ // Just let it get destroyed and make sure it doesn't call any methods on the
+ // provider interface.
+}
+
+TEST_F(RtpSenderReceiverTest, VideoSenderWithoutTrack) {
+ rtc::scoped_refptr<VideoRtpSender> sender =
+ new VideoRtpSender(&video_provider_);
+ // Just let it get destroyed and make sure it doesn't call any methods on the
+ // provider interface.
+}
+
+// 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_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupSsrcThenTrack) {
+ rtc::scoped_refptr<AudioRtpSender> sender =
+ new AudioRtpSender(&audio_provider_, nullptr);
+ rtc::scoped_refptr<AudioTrackInterface> track =
+ AudioTrack::Create(kAudioTrackId, nullptr);
+ sender->SetSsrc(kAudioSsrc);
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
+ sender->SetTrack(track);
+
+ // Calls expected from destructor.
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
pthatcher1 2015/10/20 17:42:49 Should we test that this gets called even if SetTr
Taylor Brandstetter 2015/10/21 00:22:09 It shouldn't get called if SetTrack is never calle
+}
+
+// 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_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupTrackThenSsrc) {
+ rtc::scoped_refptr<AudioRtpSender> sender =
+ new AudioRtpSender(&audio_provider_, nullptr);
+ rtc::scoped_refptr<AudioTrackInterface> track =
+ AudioTrack::Create(kAudioTrackId, nullptr);
+ sender->SetTrack(track);
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
+ sender->SetSsrc(kAudioSsrc);
+
+ // Calls expected from destructor.
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
+}
+
+// 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_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) {
+ rtc::scoped_refptr<VideoRtpSender> sender =
+ new VideoRtpSender(&video_provider_);
+ sender->SetSsrc(kVideoSsrc);
+ EXPECT_CALL(video_provider_,
+ SetCaptureDevice(kVideoSsrc,
+ video_track_->GetSource()->GetVideoCapturer()));
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
+ sender->SetTrack(video_track_);
+
+ // Calls expected from destructor.
+ EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
+}
+
+// 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_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) {
+ rtc::scoped_refptr<VideoRtpSender> sender =
+ new VideoRtpSender(&video_provider_);
+ sender->SetTrack(video_track_);
+ EXPECT_CALL(video_provider_,
+ SetCaptureDevice(kVideoSsrc,
+ video_track_->GetSource()->GetVideoCapturer()));
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
+ sender->SetSsrc(kVideoSsrc);
+
+ // Calls expected from destructor.
+ EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
+}
pthatcher1 2015/10/20 17:42:49 It seems you don't have any tests for setting the
Taylor Brandstetter 2015/10/21 00:22:09 No, I just wasn't thorough enough. I added some mo
+
+// Test that the sender is disconnected from the provider when its 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);
+
+ 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);
+}
+
+// Test that the sender is disconnected from the provider when its SSRC is
+// set to 0.
+TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) {
+ EXPECT_CALL(video_provider_,
+ SetCaptureDevice(kVideoSsrc,
+ video_track_->GetSource()->GetVideoCapturer()));
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
+ rtc::scoped_refptr<VideoRtpSender> sender =
+ new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
+ sender->SetSsrc(kVideoSsrc);
+
+ EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
+ sender->SetSsrc(0);
+
+ // Make sure it's SetSsrc that called methods on the provider, and not the
+ // destructor.
+ EXPECT_CALL(video_provider_, SetCaptureDevice(_, _)).Times(0);
+ EXPECT_CALL(video_provider_, SetVideoSend(_, _, _)).Times(0);
+}
+
} // namespace webrtc

Powered by Google App Engine
This is Rietveld 408576698