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) { |