Searched refs:GetSendStreamConfig (Results 1 – 1 of 1) sorted by relevance
128 const webrtc::AudioSendStream::Config& GetSendStreamConfig(uint32_t ssrc) { in GetSendStreamConfig() function in WebRtcVoiceEngineTestFake205 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()[all …]