Index: webrtc/video/video_send_stream_tests.cc |
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc |
index ba15b17c85b229604bd7ba082e836b00ac54fc4f..04b954ac12ef0d47e15c1972ccf2ce09e700b7e2 100644 |
--- a/webrtc/video/video_send_stream_tests.cc |
+++ b/webrtc/video/video_send_stream_tests.cc |
@@ -1136,19 +1136,43 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) { |
} |
TEST_F(VideoSendStreamTest, ChangingNetworkRoute) { |
+ static const int kStartBitrateBps = 300000; |
+ static const int kNewMaxBitrateBps = 1234567; |
+ static const uint8_t kExtensionId = 13; |
class ChangingNetworkRouteTest : public test::EndToEndTest { |
public: |
- const int kStartBitrateBps = 300000; |
- const int kNewMaxBitrateBps = 1234567; |
- |
ChangingNetworkRouteTest() |
- : EndToEndTest(test::CallTest::kDefaultTimeoutMs), |
- call_(nullptr) {} |
+ : EndToEndTest(test::CallTest::kDefaultTimeoutMs), call_(nullptr) { |
+ EXPECT_TRUE(parser_->RegisterRtpHeaderExtension( |
+ kRtpExtensionTransportSequenceNumber, kExtensionId)); |
+ } |
void OnCallsCreated(Call* sender_call, Call* receiver_call) override { |
call_ = sender_call; |
} |
+ void ModifyVideoConfigs( |
+ VideoSendStream::Config* send_config, |
+ std::vector<VideoReceiveStream::Config>* receive_configs, |
+ VideoEncoderConfig* encoder_config) override { |
+ send_config->rtp.extensions.clear(); |
+ send_config->rtp.extensions.push_back(RtpExtension( |
+ RtpExtension::kTransportSequenceNumberUri, kExtensionId)); |
+ (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions; |
+ (*receive_configs)[0].rtp.transport_cc = true; |
+ } |
+ |
+ void ModifyAudioConfigs( |
+ AudioSendStream::Config* send_config, |
+ std::vector<AudioReceiveStream::Config>* receive_configs) override { |
+ send_config->rtp.extensions.clear(); |
+ send_config->rtp.extensions.push_back(RtpExtension( |
+ RtpExtension::kTransportSequenceNumberUri, kExtensionId)); |
+ (*receive_configs)[0].rtp.extensions.clear(); |
+ (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions; |
+ (*receive_configs)[0].rtp.transport_cc = true; |
+ } |
stefan-webrtc
2016/09/27 10:58:55
Changes to enable send-side BWE for this test. Thi
|
+ |
Action OnSendRtp(const uint8_t* packet, size_t length) override { |
if (call_->GetStats().send_bandwidth_bps > kStartBitrateBps) { |
observation_complete_.Set(); |