Let NetEq reset the AudioFrame during muted state

webrtc/modules/audio_coding/neteq/neteq_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 182 matching lines @@
 
 int NetEqImpl::GetAudio(AudioFrame* audio_frame, bool* muted) {
   TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
   rtc::CritScope lock(&crit_sect_);
   if (GetAudioInternal(audio_frame, muted) != 0) {
     return kFail;
   }
   RTC_DCHECK_EQ(
       audio_frame->sample_rate_hz_,
       rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
+  RTC_DCHECK_EQ(*muted, audio_frame->muted());
   SetAudioFrameActivityAndType(vad_->enabled(), LastOutputType(),
                                last_vad_activity_, audio_frame);
   last_vad_activity_ = audio_frame->vad_activity_;
   last_output_sample_rate_hz_ = audio_frame->sample_rate_hz_;
   RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||
              last_output_sample_rate_hz_ == 16000 ||
              last_output_sample_rate_hz_ == 32000 ||
              last_output_sample_rate_hz_ == 48000)
       << "Unexpected sample rate " << last_output_sample_rate_hz_;
   return kOK;
@@ skipping 610 matching lines @@
   Operations operation;
   bool play_dtmf;
   *muted = false;
   last_decoded_timestamps_.clear();
   tick_timer_->Increment();
   stats_.IncreaseCounter(output_size_samples_, fs_hz_);
 
   // Check for muted state.
   if (enable_muted_state_ && expand_->Muted() && packet_buffer_->Empty()) {
     RTC_DCHECK_EQ(last_mode_, kModeExpand);
+    audio_frame->Reset();
+    RTC_DCHECK(audio_frame->muted());  // Reset() should mute the frame.
     playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
     audio_frame->sample_rate_hz_ = fs_hz_;
     audio_frame->samples_per_channel_ = output_size_samples_;
     audio_frame->timestamp_ =
         first_packet_
             ? 0
             : timestamp_scaler_->ToExternal(playout_timestamp_) -
                   static_cast<uint32_t>(audio_frame->samples_per_channel_);
     audio_frame->num_channels_ = sync_buffer_->Channels();
     stats_.ExpandedNoiseSamples(output_size_samples_);
@@ skipping 1264 matching lines @@
   }
 }
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(
       fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(),
       *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(),
       tick_timer_.get()));
 }
 }  // namespace webrtc