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 607d654c929d896f3e8f3f02f6a0178b1b8bf020..f785199bcc246ac6d476ace3df6f9079e58b6c58 100644 |
--- a/webrtc/video/end_to_end_tests.cc |
+++ b/webrtc/video/end_to_end_tests.cc |
@@ -1432,8 +1432,8 @@ TEST_F(EndToEndTest, AssignsTransportSequenceNumbers) { |
VideoEncoderConfig* encoder_config, |
test::FrameGeneratorCapturer** frame_generator) override { |
send_config->rtp.extensions.clear(); |
- send_config->rtp.extensions.push_back( |
- RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); |
+ send_config->rtp.extensions.push_back(RtpExtension( |
+ RtpExtension::kTransportSequenceNumberUri, kExtensionId)); |
// Force some padding to be sent. |
const int kPaddingBitrateBps = 50000; |
@@ -1459,8 +1459,8 @@ TEST_F(EndToEndTest, AssignsTransportSequenceNumbers) { |
VideoReceiveStream::Config* receive_config) override { |
receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
receive_config->rtp.extensions.clear(); |
- receive_config->rtp.extensions.push_back( |
- RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); |
+ receive_config->rtp.extensions.push_back(RtpExtension( |
+ RtpExtension::kTransportSequenceNumberUri, kExtensionId)); |
} |
test::DirectTransport* CreateSendTransport(Call* sender_call) override { |
@@ -1539,7 +1539,7 @@ class TransportFeedbackTester : public test::EndToEndTest { |
VideoEncoderConfig* encoder_config) override { |
send_config->rtp.extensions.clear(); |
send_config->rtp.extensions.push_back( |
- RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); |
+ RtpExtension(RtpExtension::kTransportSequenceNumberUri, kExtensionId)); |
(*receive_configs)[0].rtp.extensions = send_config->rtp.extensions; |
(*receive_configs)[0].rtp.transport_cc = feedback_enabled_; |
} |
@@ -1549,7 +1549,7 @@ class TransportFeedbackTester : public test::EndToEndTest { |
std::vector<AudioReceiveStream::Config>* receive_configs) override { |
send_config->rtp.extensions.clear(); |
send_config->rtp.extensions.push_back( |
- RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); |
+ 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 = feedback_enabled_; |
@@ -1787,7 +1787,7 @@ TEST_F(EndToEndTest, RembWithSendSideBwe) { |
ASSERT_EQ(1u, send_config->rtp.ssrcs.size()); |
send_config->rtp.extensions.clear(); |
send_config->rtp.extensions.push_back( |
- RtpExtension(RtpExtension::kTransportSequenceNumber, |
+ RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
test::kTransportSequenceNumberExtensionId)); |
sender_ssrc_ = send_config->rtp.ssrcs[0]; |
@@ -3452,8 +3452,8 @@ TEST_F(EndToEndTest, TransportSeqNumOnAudioAndVideo) { |
std::vector<VideoReceiveStream::Config>* receive_configs, |
VideoEncoderConfig* encoder_config) override { |
send_config->rtp.extensions.clear(); |
- send_config->rtp.extensions.push_back( |
- RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); |
+ send_config->rtp.extensions.push_back(RtpExtension( |
+ RtpExtension::kTransportSequenceNumberUri, kExtensionId)); |
(*receive_configs)[0].rtp.extensions = send_config->rtp.extensions; |
} |
@@ -3461,8 +3461,8 @@ TEST_F(EndToEndTest, TransportSeqNumOnAudioAndVideo) { |
AudioSendStream::Config* send_config, |
std::vector<AudioReceiveStream::Config>* receive_configs) override { |
send_config->rtp.extensions.clear(); |
- send_config->rtp.extensions.push_back( |
- RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); |
+ 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; |
} |