| Index: webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
 | 
| diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
 | 
| index fdb66714cfbe20c465ae4f6f949237eb94ad50f7..1d462b3c9f24b0498d39941e2a73b6c343ef63b5 100644
 | 
| --- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
 | 
| +++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
 | 
| @@ -605,7 +605,9 @@ int main(int argc, char* argv[]) {
 | 
|      // Check if it is time to get output audio.
 | 
|      while (time_now_ms >= next_output_time_ms && output_event_available) {
 | 
|        webrtc::AudioFrame out_frame;
 | 
| -      int error = neteq->GetAudio(&out_frame);
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| +      bool muted;
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| +      int error = neteq->GetAudio(&out_frame, &muted);
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| +      RTC_CHECK(!muted);
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|        if (error != NetEq::kOK) {
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|          std::cerr << "GetAudio returned error code " <<
 | 
|              neteq->LastError() << std::endl;
 | 
| 
 |