| 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);
|
| + bool muted;
|
| + int error = neteq->GetAudio(&out_frame, &muted);
|
| + RTC_CHECK(!muted);
|
| if (error != NetEq::kOK) {
|
| std::cerr << "GetAudio returned error code " <<
|
| neteq->LastError() << std::endl;
|
|
|