| Index: webrtc/media/engine/webrtcvoiceengine_unittest.cc
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
|
| index e5af99fca7fbbcea92563a1c163cbeaa71d66adc..5d8dd90d02beff0c6370ada650a3618d5e92e422 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
|
| +++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
|
| @@ -494,6 +494,35 @@ TEST_F(WebRtcVoiceEngineTestFake, CreateChannel) {
|
| EXPECT_TRUE(SetupChannel());
|
| }
|
|
|
| +// Test that we can add a send stream and that it has the correct defaults.
|
| +TEST_F(WebRtcVoiceEngineTestFake, CreateSendStream) {
|
| + EXPECT_TRUE(SetupChannel());
|
| + EXPECT_TRUE(
|
| + channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc1)));
|
| + const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrc1);
|
| + EXPECT_EQ(kSsrc1, config.rtp.ssrc);
|
| + EXPECT_EQ("", config.rtp.c_name);
|
| + EXPECT_EQ(0u, config.rtp.extensions.size());
|
| + EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_),
|
| + config.send_transport);
|
| +}
|
| +
|
| +// Test that we can add a receive stream and that it has the correct defaults.
|
| +TEST_F(WebRtcVoiceEngineTestFake, CreateRecvStream) {
|
| + EXPECT_TRUE(SetupChannel());
|
| + EXPECT_TRUE(
|
| + channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc1)));
|
| + const webrtc::AudioReceiveStream::Config& config =
|
| + GetRecvStreamConfig(kSsrc1);
|
| + EXPECT_EQ(kSsrc1, config.rtp.remote_ssrc);
|
| + EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc);
|
| + EXPECT_FALSE(config.rtp.transport_cc);
|
| + EXPECT_EQ(0u, config.rtp.extensions.size());
|
| + EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_),
|
| + config.rtcp_send_transport);
|
| + EXPECT_EQ("", config.sync_group);
|
| +}
|
| +
|
| // Tests that the list of supported codecs is created properly and ordered
|
| // correctly (such that opus appears first).
|
| TEST_F(WebRtcVoiceEngineTestFake, CodecOrder) {
|
|
|