| Index: webrtc/video/stream_synchronization.cc
|
| diff --git a/webrtc/video/stream_synchronization.cc b/webrtc/video/stream_synchronization.cc
|
| index cb37d80ef5ab3106af16c3aff53593663e72d471..3727f8fdb53e5934c6a884e507ac79f52161e8da 100644
|
| --- a/webrtc/video/stream_synchronization.cc
|
| +++ b/webrtc/video/stream_synchronization.cc
|
| @@ -60,10 +60,6 @@ bool StreamSynchronization::ComputeRelativeDelay(
|
| const Measurements& video_measurement,
|
| int* relative_delay_ms) {
|
| assert(relative_delay_ms);
|
| - if (audio_measurement.rtcp.size() < 2 || video_measurement.rtcp.size() < 2) {
|
| - // We need two RTCP SR reports per stream to do synchronization.
|
| - return false;
|
| - }
|
| int64_t audio_last_capture_time_ms;
|
| if (!RtpToNtpMs(audio_measurement.latest_timestamp,
|
| audio_measurement.rtcp,
|
|
|