Index: webrtc/api/rtpsenderreceiver_unittest.cc |
diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc |
index f8b968f54991841dd183bf88c7ff69580a1e4f87..3db9d5e00ee1d57af9ba98fd106b30c6aec21518 100644 |
--- a/webrtc/api/rtpsenderreceiver_unittest.cc |
+++ b/webrtc/api/rtpsenderreceiver_unittest.cc |
@@ -59,8 +59,12 @@ class MockAudioProvider : public AudioProviderInterface { |
const cricket::AudioOptions& options, |
cricket::AudioSource* source)); |
MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume)); |
- MOCK_CONST_METHOD1(GetAudioRtpParameters, RtpParameters(uint32_t ssrc)); |
- MOCK_METHOD2(SetAudioRtpParameters, |
+ 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( |
@@ -88,8 +92,12 @@ class MockVideoProvider : public VideoProviderInterface { |
bool enable, |
const cricket::VideoOptions* options)); |
- MOCK_CONST_METHOD1(GetVideoRtpParameters, RtpParameters(uint32_t ssrc)); |
- MOCK_METHOD2(SetVideoRtpParameters, |
+ 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&)); |
}; |
@@ -504,9 +512,9 @@ TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) { |
TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) { |
CreateAudioRtpSender(); |
- EXPECT_CALL(audio_provider_, GetAudioRtpParameters(kAudioSsrc)) |
+ EXPECT_CALL(audio_provider_, GetAudioRtpSendParameters(kAudioSsrc)) |
.WillOnce(Return(RtpParameters())); |
- EXPECT_CALL(audio_provider_, SetAudioRtpParameters(kAudioSsrc, _)) |
+ EXPECT_CALL(audio_provider_, SetAudioRtpSendParameters(kAudioSsrc, _)) |
.WillOnce(Return(true)); |
RtpParameters params = audio_rtp_sender_->GetParameters(); |
EXPECT_TRUE(audio_rtp_sender_->SetParameters(params)); |
@@ -517,9 +525,9 @@ TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) { |
TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) { |
CreateVideoRtpSender(); |
- EXPECT_CALL(video_provider_, GetVideoRtpParameters(kVideoSsrc)) |
+ EXPECT_CALL(video_provider_, GetVideoRtpSendParameters(kVideoSsrc)) |
.WillOnce(Return(RtpParameters())); |
- EXPECT_CALL(video_provider_, SetVideoRtpParameters(kVideoSsrc, _)) |
+ EXPECT_CALL(video_provider_, SetVideoRtpSendParameters(kVideoSsrc, _)) |
.WillOnce(Return(true)); |
RtpParameters params = video_rtp_sender_->GetParameters(); |
EXPECT_TRUE(video_rtp_sender_->SetParameters(params)); |
@@ -527,4 +535,30 @@ TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) { |
DestroyVideoRtpSender(); |
} |
+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_TRUE(audio_rtp_receiver_->SetParameters(params)); |
+ |
+ DestroyAudioRtpReceiver(); |
+} |
+ |
+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_TRUE(video_rtp_receiver_->SetParameters(params)); |
+ |
+ DestroyVideoRtpReceiver(); |
+} |
+ |
} // namespace webrtc |