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