| 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(
|
|
|