| 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..3df49c1e0fe9525aa88ccde2de1fdeb4bbcc5289 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_output;
|
| + int error = neteq->GetAudio(&out_frame, &muted_output);
|
| + RTC_CHECK(!muted_output);
|
| if (error != NetEq::kOK) {
|
| std::cerr << "GetAudio returned error code " <<
|
| neteq->LastError() << std::endl;
|
|
|