Chromium Code Reviews| Index: webrtc/media/engine/webrtcvoiceengine_unittest.cc |
| diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc |
| index f88b99ba103960591bdd6b47406e635abf9fcc65..b79c47e92b241b28c9d30b0107eadb6fd322a76c 100644 |
| --- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc |
| +++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc |
| @@ -910,6 +910,44 @@ TEST_F(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) { |
| EXPECT_EQ(64000, GetCodecBitrate(kSsrcs4[2])); |
| } |
| +// Test that GetRtpParameters returns the currently configured codecs. |
| +TEST_F(WebRtcVoiceEngineTestFake, GetRtpParametersCodecs) { |
| + EXPECT_TRUE(SetupSendStream()); |
| + cricket::AudioSendParameters parameters; |
| + parameters.codecs.push_back(kIsacCodec); |
| + parameters.codecs.push_back(kPcmuCodec); |
| + EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| + |
| + webrtc::RtpParameters rtp_parameters = channel_->GetRtpParameters(kSsrc1); |
| + ASSERT_EQ(2u, rtp_parameters.codecs.size()); |
| + EXPECT_EQ(kIsacCodec.id, rtp_parameters.codecs[0].payload_type); |
| + EXPECT_EQ(kIsacCodec.name, rtp_parameters.codecs[0].mime_type); |
| + EXPECT_EQ(kIsacCodec.clockrate, rtp_parameters.codecs[0].clock_rate); |
| + EXPECT_EQ(kIsacCodec.channels, rtp_parameters.codecs[0].channels); |
| + EXPECT_EQ(kPcmuCodec.id, rtp_parameters.codecs[1].payload_type); |
| + EXPECT_EQ(kPcmuCodec.name, rtp_parameters.codecs[1].mime_type); |
| + EXPECT_EQ(kPcmuCodec.clockrate, rtp_parameters.codecs[1].clock_rate); |
| + EXPECT_EQ(kPcmuCodec.channels, rtp_parameters.codecs[1].channels); |
|
pthatcher1
2016/04/14 15:58:12
Since you implemented the == operator, you could j
Taylor Brandstetter
2016/04/14 16:30:54
I didn't implement the == operator for webrtc::Rtp
|
| +} |
| + |
| +// Test that if we set/get parameters multiple times, we get the same results. |
| +TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpParameters) { |
| + EXPECT_TRUE(SetupSendStream()); |
| + cricket::AudioSendParameters parameters; |
| + parameters.codecs.push_back(kIsacCodec); |
| + parameters.codecs.push_back(kPcmuCodec); |
| + EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| + |
| + webrtc::RtpParameters initial_params = channel_->GetRtpParameters(kSsrc1); |
| + |
| + // We should be able to set the params we just got. |
| + EXPECT_TRUE(channel_->SetRtpParameters(kSsrc1, initial_params)); |
| + |
| + // ... And this shouldn't change the params returned by GetRtpParameters. |
| + webrtc::RtpParameters new_params = channel_->GetRtpParameters(kSsrc1); |
| + EXPECT_EQ(initial_params, channel_->GetRtpParameters(kSsrc1)); |
| +} |
| + |
| // Test that we apply codecs properly. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecs) { |
| EXPECT_TRUE(SetupSendStream()); |