| Index: webrtc/video/end_to_end_tests.cc
|
| diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
|
| index d93dd247d1da3c16e775bb66bed129ebc12a3d24..e406d0fab221a8105b1f3e09923c67e5f816b3cd 100644
|
| --- a/webrtc/video/end_to_end_tests.cc
|
| +++ b/webrtc/video/end_to_end_tests.cc
|
| @@ -122,11 +122,11 @@ class EndToEndTest : public test::CallTest {
|
| void TestRtpStatePreservation(bool use_rtx, bool provoke_rtcpsr_before_rtp);
|
| void VerifyHistogramStats(bool use_rtx, bool use_red, bool screenshare);
|
| void VerifyNewVideoSendStreamsRespectNetworkState(
|
| - MediaType network_to_bring_down,
|
| + MediaType network_to_bring_up,
|
| VideoEncoder* encoder,
|
| Transport* transport);
|
| void VerifyNewVideoReceiveStreamsRespectNetworkState(
|
| - MediaType network_to_bring_down,
|
| + MediaType network_to_bring_up,
|
| Transport* transport);
|
| };
|
|
|
| @@ -3671,11 +3671,11 @@ TEST_F(EndToEndTest, CallReportsRttForSender) {
|
| }
|
|
|
| void EndToEndTest::VerifyNewVideoSendStreamsRespectNetworkState(
|
| - MediaType network_to_bring_down,
|
| + MediaType network_to_bring_up,
|
| VideoEncoder* encoder,
|
| Transport* transport) {
|
| CreateSenderCall(Call::Config(&event_log_));
|
| - sender_call_->SignalChannelNetworkState(network_to_bring_down, kNetworkDown);
|
| + sender_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);
|
|
|
| CreateSendConfig(1, 0, 0, transport);
|
| video_send_config_.encoder_settings.encoder = encoder;
|
| @@ -3691,12 +3691,11 @@ void EndToEndTest::VerifyNewVideoSendStreamsRespectNetworkState(
|
| }
|
|
|
| void EndToEndTest::VerifyNewVideoReceiveStreamsRespectNetworkState(
|
| - MediaType network_to_bring_down,
|
| + MediaType network_to_bring_up,
|
| Transport* transport) {
|
| Call::Config config(&event_log_);
|
| CreateCalls(config, config);
|
| - receiver_call_->SignalChannelNetworkState(network_to_bring_down,
|
| - kNetworkDown);
|
| + receiver_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);
|
|
|
| test::DirectTransport sender_transport(sender_call_.get());
|
| sender_transport.SetReceiver(receiver_call_->Receiver());
|
| @@ -3738,7 +3737,7 @@ TEST_F(EndToEndTest, NewVideoSendStreamsRespectVideoNetworkDown) {
|
| UnusedEncoder unused_encoder;
|
| UnusedTransport unused_transport;
|
| VerifyNewVideoSendStreamsRespectNetworkState(
|
| - MediaType::VIDEO, &unused_encoder, &unused_transport);
|
| + MediaType::AUDIO, &unused_encoder, &unused_transport);
|
| }
|
|
|
| TEST_F(EndToEndTest, NewVideoSendStreamsIgnoreAudioNetworkDown) {
|
| @@ -3766,17 +3765,17 @@ TEST_F(EndToEndTest, NewVideoSendStreamsIgnoreAudioNetworkDown) {
|
| RequiredTransport required_transport(true /*rtp*/, false /*rtcp*/);
|
| RequiredEncoder required_encoder;
|
| VerifyNewVideoSendStreamsRespectNetworkState(
|
| - MediaType::AUDIO, &required_encoder, &required_transport);
|
| + MediaType::VIDEO, &required_encoder, &required_transport);
|
| }
|
|
|
| TEST_F(EndToEndTest, NewVideoReceiveStreamsRespectVideoNetworkDown) {
|
| UnusedTransport transport;
|
| - VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::VIDEO, &transport);
|
| + VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::AUDIO, &transport);
|
| }
|
|
|
| TEST_F(EndToEndTest, NewVideoReceiveStreamsIgnoreAudioNetworkDown) {
|
| RequiredTransport transport(false /*rtp*/, true /*rtcp*/);
|
| - VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::AUDIO, &transport);
|
| + VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::VIDEO, &transport);
|
| }
|
|
|
| void VerifyEmptyNackConfig(const NackConfig& config) {
|
|
|