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; |