| Index: webrtc/video/video_quality_test.cc
|
| diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
|
| index e43ddb16ce4055c765076db7c2196d8db9efd35b..c8829925e43fb720508bd950a9bd5d27d176cce5 100644
|
| --- a/webrtc/video/video_quality_test.cc
|
| +++ b/webrtc/video/video_quality_test.cc
|
| @@ -161,13 +161,13 @@ class VideoAnalyzer : public PacketReceiver,
|
| bool result = transport_->SendRtp(packet, length, options);
|
| {
|
| rtc::CritScope lock(&crit_);
|
| - int64_t timestamp = wrap_handler_.Unwrap(header.timestamp);
|
|
|
| if (rtp_timestamp_delta_ == 0) {
|
| - rtp_timestamp_delta_ = timestamp - first_send_frame_.timestamp();
|
| + rtp_timestamp_delta_ = header.timestamp - first_send_frame_.timestamp();
|
| first_send_frame_.Reset();
|
| }
|
| - timestamp -= rtp_timestamp_delta_;
|
| + int64_t timestamp =
|
| + wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
|
| send_times_[timestamp] = current_time;
|
| if (!transport_->DiscardedLastPacket() &&
|
| header.ssrc == ssrc_to_analyze_) {
|
|
|