Index: webrtc/video/video_quality_test.cc |
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc |
index 6e8c9902c65df7b072ce256c52093205f8439f42..e38203d4d04427b1b2f7edf9b1ca57a6d9a65929 100644 |
--- a/webrtc/video/video_quality_test.cc |
+++ b/webrtc/video/video_quality_test.cc |
@@ -44,8 +44,7 @@ class VideoAnalyzer : public PacketReceiver, |
public Transport, |
public VideoRenderer, |
public VideoCaptureInput, |
- public EncodedFrameObserver, |
- public EncodingTimeObserver { |
+ public EncodedFrameObserver { |
public: |
VideoAnalyzer(test::LayerFilteringTransport* transport, |
const std::string& test_label, |
@@ -126,8 +125,7 @@ class VideoAnalyzer : public PacketReceiver, |
return receiver_->DeliverPacket(media_type, packet, length, packet_time); |
} |
- // EncodingTimeObserver. |
- void OnReportEncodedTime(int64_t ntp_time_ms, int encode_time_ms) override { |
+ void OnEncodeTiming(int64_t ntp_time_ms, int encode_time_ms) override { |
rtc::CritScope crit(&comparison_lock_); |
samples_encode_time_ms_[ntp_time_ms] = encode_time_ms; |
} |
@@ -203,7 +201,7 @@ class VideoAnalyzer : public PacketReceiver, |
assert(!reference_frame.IsZeroSize()); |
if (send_timestamp == reference_frame.timestamp() - 1) { |
// TODO(ivica): Make this work for > 2 streams. |
- // Look at rtp_sender.c:RTPSender::BuildRTPHeader. |
+ // Look at RTPSender::BuildRTPHeader. |
++send_timestamp; |
} |
EXPECT_EQ(reference_frame.timestamp(), send_timestamp); |
@@ -967,7 +965,6 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) { |
recv_transport.SetReceiver(sender_call_->Receiver()); |
SetupCommon(&analyzer, &recv_transport); |
- video_send_config_.encoding_time_observer = &analyzer; |
video_receive_configs_[params_.ss.selected_stream].renderer = &analyzer; |
for (auto& config : video_receive_configs_) |
config.pre_decode_callback = &analyzer; |