128   const webrtc::AudioSendStream::Config& GetSendStreamConfig(uint32_t ssrc) {  in GetSendStreamConfig()  function in WebRtcVoiceEngineTestFake
205     EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());  in TestSetSendRtpHeaderExtensions()
211     EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());  in TestSetSendRtpHeaderExtensions()
216     EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());  in TestSetSendRtpHeaderExtensions()
222     EXPECT_EQ(1u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());  in TestSetSendRtpHeaderExtensions()
223     EXPECT_EQ(ext, GetSendStreamConfig(kSsrc1).rtp.extensions[0].name);  in TestSetSendRtpHeaderExtensions()
224     EXPECT_EQ(id, GetSendStreamConfig(kSsrc1).rtp.extensions[0].id);  in TestSetSendRtpHeaderExtensions()
231     EXPECT_EQ(1u, GetSendStreamConfig(kSsrc2).rtp.extensions.size());  in TestSetSendRtpHeaderExtensions()
232     EXPECT_EQ(ext, GetSendStreamConfig(kSsrc2).rtp.extensions[0].name);  in TestSetSendRtpHeaderExtensions()
233     EXPECT_EQ(id, GetSendStreamConfig(kSsrc2).rtp.extensions[0].id);  in TestSetSendRtpHeaderExtensions()
239     EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());  in TestSetSendRtpHeaderExtensions()
240     EXPECT_EQ(0u, GetSendStreamConfig(kSsrc2).rtp.extensions.size());  in TestSetSendRtpHeaderExtensions()
1970     EXPECT_TRUE(voe_.GetSend(GetSendStreamConfig(ssrc).voe_channel_id));  in TEST_F()
2003     int channel_num = GetSendStreamConfig(ssrc).voe_channel_id;  in TEST_F()
2014     int channel_num = GetSendStreamConfig(ssrc).voe_channel_id;  in TEST_F()
2037     int channel_num = GetSendStreamConfig(ssrc).voe_channel_id;  in TEST_F()
2045     int channel_num = GetSendStreamConfig(ssrc).voe_channel_id;  in TEST_F()