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_) { |