Index: webrtc/video/video_quality_test.cc |
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc |
index 96a488298e859229bbe2a80556c9b1f332e2f266..6e31307ecf297bf3e5159fed9c3946ea26a8c3c1 100644 |
--- a/webrtc/video/video_quality_test.cc |
+++ b/webrtc/video/video_quality_test.cc |
@@ -1287,6 +1287,8 @@ void VideoQualityTest::SetupVideo(Transport* send_transport, |
video_send_config_.rtp.extensions.push_back(RtpExtension( |
RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId)); |
} |
+ video_send_config_.rtp.extensions.push_back(RtpExtension( |
+ RtpExtension::kVideoContentTypeUri, test::kVideoContentTypeExtensionId)); |
video_encoder_config_.min_transmit_bitrate_bps = |
params_.video.min_transmit_bps; |
@@ -1314,6 +1316,8 @@ void VideoQualityTest::SetupVideo(Transport* send_transport, |
kSendRtxPayloadType; |
video_receive_configs_[i].rtp.transport_cc = params_.call.send_side_bwe; |
video_receive_configs_[i].rtp.remb = !params_.call.send_side_bwe; |
+ // Enable RTT calculation so NTP time estimator will work. |
+ video_receive_configs_[i].rtp.rtcp_xr.receiver_reference_time_report = true; |
// Force fake decoders on non-selected simulcast streams. |
if (i != params_.ss.selected_stream) { |
VideoReceiveStream::Decoder decoder; |