Index: webrtc/video/call_perf_tests.cc |
diff --git a/webrtc/video/call_perf_tests.cc b/webrtc/video/call_perf_tests.cc |
index 481abb6e2d86cafe19d1fa761880bf54bcb9d4e4..5fa520b2e6445f7db9a968340d1e979326f7f43d 100644 |
--- a/webrtc/video/call_perf_tests.cc |
+++ b/webrtc/video/call_perf_tests.cc |
@@ -197,8 +197,10 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) { |
: channel_(channel), |
voe_network_(voe_network), |
parser_(RtpHeaderParser::Create()) {} |
- DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet, |
- size_t length) override { |
+ DeliveryStatus DeliverPacket(MediaType media_type, |
+ const uint8_t* packet, |
+ size_t length, |
+ const PacketTime& packet_time) override { |
EXPECT_TRUE(media_type == MediaType::ANY || |
media_type == MediaType::AUDIO); |
int ret; |
@@ -545,8 +547,10 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) { |
test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver); |
} |
- DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet, |
- size_t length) override { |
+ DeliveryStatus DeliverPacket(MediaType media_type, |
+ const uint8_t* packet, |
+ size_t length, |
+ const PacketTime& packet_time) override { |
VideoSendStream::Stats stats = send_stream_->GetStats(); |
if (stats.substreams.size() > 0) { |
DCHECK_EQ(1u, stats.substreams.size()); |
@@ -580,8 +584,8 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) { |
observation_complete_->Set(); |
} |
} |
- return send_transport_receiver_->DeliverPacket(media_type, packet, |
- length); |
+ return send_transport_receiver_->DeliverPacket(media_type, packet, length, |
+ packet_time); |
} |
void OnStreamsCreated( |