Index: webrtc/api/rtpsenderreceiver_unittest.cc |
diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc |
index 8c91e148bef38daff9050d1d266baf803ca22469..a08e1b4c634a34083e01b451fd8080560504d30b 100644 |
--- a/webrtc/api/rtpsenderreceiver_unittest.cc |
+++ b/webrtc/api/rtpsenderreceiver_unittest.cc |
@@ -15,8 +15,6 @@ |
#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" |
@@ -27,10 +25,6 @@ |
#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; |
@@ -47,56 +41,69 @@ |
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: |
- 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(); } |
+ virtual void SetUp() { |
+ stream_ = MediaStream::Create(kStreamLabel1); |
+ } |
void AddVideoTrack() { |
rtc::scoped_refptr<VideoTrackSourceInterface> source( |
@@ -105,128 +112,68 @@ |
EXPECT_TRUE(stream_->AddTrack(video_track_)); |
} |
- void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); } |
- |
- void CreateAudioRtpSender(rtc::scoped_refptr<LocalAudioSource> source) { |
- audio_track_ = AudioTrack::Create(kAudioTrackId, source); |
+ void CreateAudioRtpSender() { |
+ 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], stream_->label(), |
- voice_channel_, nullptr); |
+ &audio_provider_, 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_channel_); |
+ stream_->label(), &video_provider_); |
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, voice_channel_); |
+ kAudioSsrc, &audio_provider_); |
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_channel_); |
+ kVideoSsrc, &video_provider_); |
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: |
- 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_; |
+ MockAudioProvider audio_provider_; |
+ MockVideoProvider video_provider_; |
rtc::scoped_refptr<AudioRtpSender> audio_rtp_sender_; |
rtc::scoped_refptr<VideoRtpSender> video_rtp_sender_; |
rtc::scoped_refptr<AudioRtpReceiver> audio_rtp_receiver_; |
@@ -236,96 +183,72 @@ |
rtc::scoped_refptr<AudioTrackInterface> audio_track_; |
}; |
-// Test that |voice_channel_| is updated when an audio track is associated |
+// Test that |audio_provider_| is notified 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 |
+// Test that |video_provider_| is notified 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 |
+// Test that |audio_provider_| is notified when a remote audio and 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 that |video_provider_| is notified 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(); |
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)); |
audio_track_->set_enabled(false); |
- EXPECT_TRUE(voice_media_channel_->IsStreamMuted(kAudioSsrc)); |
- |
+ |
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _)); |
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); |
- |
+ EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, false)); |
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(); |
@@ -340,223 +263,257 @@ |
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); |
+ double volume = 0.5; |
+ EXPECT_CALL(audio_provider_, SetAudioPlayoutVolume(kAudioSsrc, volume)); |
+ audio_track_->GetSource()->SetVolume(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(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_->GetSource()->SetVolume(1.0); |
+ |
+ EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true)); |
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); |
+ |
+ double new_volume = 0.8; |
+ EXPECT_CALL(audio_provider_, SetAudioPlayoutVolume(kAudioSsrc, new_volume)); |
+ audio_track_->GetSource()->SetVolume(new_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 that provider methods aren't called without both a track and an SSRC. |
TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) { |
- audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr); |
+ rtc::scoped_refptr<AudioRtpSender> sender = |
+ new AudioRtpSender(&audio_provider_, 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. |
+ 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. |
+} |
+ |
+// Test that provider methods aren't called without both a track and an 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. |
+ 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. |
+} |
+ |
+// 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) { |
- audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr); |
+ rtc::scoped_refptr<AudioRtpSender> sender = |
+ new AudioRtpSender(&audio_provider_, 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. |
+ 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); |
+} |
+ |
+// 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) { |
- audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr); |
+ rtc::scoped_refptr<AudioRtpSender> sender = |
+ new AudioRtpSender(&audio_provider_, 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. |
+ 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) { |
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. |
+ 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_); |
+ |
+ // Calls expected from destructor. |
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr)) |
+ .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) { |
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. |
+ 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); |
+ |
+ // Calls expected from destructor. |
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr)) |
+ .Times(1); |
+} |
+ |
+// Test that the sender is disconnected from the provider when its 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. |
+ 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) { |
- CreateAudioRtpSender(); |
- |
- audio_rtp_sender_->SetSsrc(0); |
- VerifyVideoChannelNoInput(); |
-} |
- |
-// Test that the media channel stops sending when the audio sender's track is |
-// set to null. |
+ 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); |
+ |
+ 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); |
+} |
+ |
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. |
+ 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 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); |
+} |
+ |
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. |
+ 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); |
+ |
+ // 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); |
+} |
+ |
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. |
+ 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); |
+ |
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1); |
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc2, true, _, _)).Times(1); |
+ sender->SetSsrc(kAudioSsrc2); |
+ |
+ // Calls expected from destructor. |
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc2, false, _, _)).Times(1); |
+} |
+ |
TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) { |
- CreateVideoRtpSender(); |
- |
- video_rtp_sender_->SetSsrc(kVideoSsrc2); |
- VerifyVideoChannelNoInput(kVideoSsrc); |
- VerifyVideoChannelInput(kVideoSsrc2); |
- |
- video_rtp_sender_ = nullptr; |
- VerifyVideoChannelNoInput(kVideoSsrc2); |
+ 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); |
+ |
+ 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); |
+ |
+ // Calls expected from destructor. |
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _, nullptr)) |
+ .Times(1); |
} |
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(); |
} |
@@ -564,35 +521,12 @@ |
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(); |
} |
@@ -600,8 +534,11 @@ |
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(); |
@@ -610,8 +547,11 @@ |
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(); |