Index: webrtc/call/call_perf_tests.cc |
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc |
index beb05a047d9f9a11f8268142754399538479771e..7501ada956325b032151409e94054cf042f56475 100644 |
--- a/webrtc/call/call_perf_tests.cc |
+++ b/webrtc/call/call_perf_tests.cc |
@@ -88,8 +88,7 @@ class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver, |
first_time_in_sync_(-1), |
receive_stream_(nullptr) {} |
- void RenderFrame(const VideoFrame& video_frame, |
- int time_to_render_ms) override { |
+ void OnFrame(const VideoFrame& video_frame) override { |
VideoReceiveStream::Stats stats; |
{ |
rtc::CritScope lock(&crit_); |
@@ -129,8 +128,6 @@ class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver, |
} |
} |
- bool IsTextureSupported() const override { return false; } |
- |
void set_receive_stream(VideoReceiveStream* receive_stream) { |
rtc::CritScope lock(&crit_); |
receive_stream_ = receive_stream; |
@@ -377,8 +374,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, |
nullptr, this, test::PacketTransport::kReceiver, net_config_); |
} |
- void RenderFrame(const VideoFrame& video_frame, |
- int time_to_render_ms) override { |
+ void OnFrame(const VideoFrame& video_frame) override { |
rtc::CritScope lock(&crit_); |
if (video_frame.ntp_time_ms() <= 0) { |
// Haven't got enough RTCP SR in order to calculate the capture ntp |
@@ -417,8 +413,6 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, |
EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_); |
} |
- bool IsTextureSupported() const override { return false; } |
- |
virtual Action OnSendRtp(const uint8_t* packet, size_t length) { |
rtc::CritScope lock(&crit_); |
RTPHeader header; |