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Unified Diff: webrtc/media/engine/webrtcvoiceengine_unittest.cc

Issue 1951833002: Set rtcp_send_transport for AudioReceiveStreams. This was forgotten in https://codereview.webrtc.or… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Android gunit idiosyncracies Created 4 years, 7 months ago
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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) {
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