Add muted_output parameter to ACM
The new parameter indicates if the output in the AudioFrame is muted. If
so, the output samples are not written, but should be interpreted as all
zero.
A version of AudioCodingModule::PlayoutData10Ms() without the new
parameter is maintained while waiting for downstream dependencies to
conform.
BUG=webrtc:5609
Review-Url: https://codereview.webrtc.org/1976913002
Cr-Commit-Position: refs/heads/master@{#12719}
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 57d7bd3..8c07631 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -132,16 +132,16 @@
return 0;
}
-int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
+int AcmReceiver::GetAudio(int desired_freq_hz,
+ AudioFrame* audio_frame,
+ bool* muted) {
// Accessing members, take the lock.
rtc::CritScope lock(&crit_sect_);
- bool muted;
- if (neteq_->GetAudio(audio_frame, &muted) != NetEq::kOK) {
+ if (neteq_->GetAudio(audio_frame, muted) != NetEq::kOK) {
LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
return -1;
}
- RTC_DCHECK(!muted);
const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();