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, |