Enable muted state by default in VoE

webrtc/voice_engine/channel.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 478 matching lines @@
   if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame,
                                      &muted) == -1) {
     WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::GetAudioFrame() PlayoutData10Ms() failed!");
     // In all likelihood, the audio in this frame is garbage. We return an
     // error so that the audio mixer module doesn't add it to the mix. As
     // a result, it won't be played out and the actions skipped here are
     // irrelevant.
     return MixerParticipant::AudioFrameInfo::kError;
   }
-  RTC_DCHECK(!muted);
+
+  if (muted) {
+    // TODO(henrik.lundin): We should be able to do better than this. But we
+    // will have to go through all the cases below where the audio samples may
+    // be used, and handle the muted case in some way.
+    audioFrame->Mute();
+  }
 
   if (_RxVadDetection) {
     UpdateRxVadDetection(*audioFrame);
   }
 
   // Convert module ID to internal VoE channel ID
   audioFrame->id_ = VoEChannelId(audioFrame->id_);
   // Store speech type for dead-or-alive detection
   _outputSpeechType = audioFrame->speech_type_;
 
@@ skipping 50 matching lines @@
     // action is needed.
 
     // Do the panning operation (the audio frame contains stereo at this
     // stage)
     AudioFrameOperations::Scale(left_pan, right_pan, *audioFrame);
   }
 
   // Mix decoded PCM output with file if file mixing is enabled
   if (state.output_file_playing) {
     MixAudioWithFile(*audioFrame, audioFrame->sample_rate_hz_);
+    muted = false;  // We may have added non-zero samples.
   }
 
   // External media
   if (_outputExternalMedia) {
     rtc::CritScope cs(&_callbackCritSect);
     const bool isStereo = (audioFrame->num_channels_ == 2);
     if (_outputExternalMediaCallbackPtr) {
       _outputExternalMediaCallbackPtr->Process(
           _channelId, kPlaybackPerChannel, (int16_t*)audioFrame->data_,
           audioFrame->samples_per_channel_, audioFrame->sample_rate_hz_,
           isStereo);
     }
   }
 
   // Record playout if enabled
   {
     rtc::CritScope cs(&_fileCritSect);
 
     if (_outputFileRecording && _outputFileRecorderPtr) {
       _outputFileRecorderPtr->RecordAudioToFile(*audioFrame);
     }
   }
 
   // Measure audio level (0-9)
+  // TODO(henrik.lundin) Use the |muted| information here too.
   _outputAudioLevel.ComputeLevel(*audioFrame);
 
   if (capture_start_rtp_time_stamp_ < 0 && audioFrame->timestamp_ != 0) {
     // The first frame with a valid rtp timestamp.
     capture_start_rtp_time_stamp_ = audioFrame->timestamp_;
   }
 
   if (capture_start_rtp_time_stamp_ >= 0) {
     // audioFrame.timestamp_ should be valid from now on.
 
@@ skipping 205 matching lines @@
   AudioCodingModule::Config acm_config;
   acm_config.id = VoEModuleId(instanceId, channelId);
   if (config.Get<NetEqCapacityConfig>().enabled) {
     // Clamping the buffer capacity at 20 packets. While going lower will
     // probably work, it makes little sense.
     acm_config.neteq_config.max_packets_in_buffer =
         std::max(20, config.Get<NetEqCapacityConfig>().capacity);
   }
   acm_config.neteq_config.enable_fast_accelerate =
       config.Get<NetEqFastAccelerate>().enabled;
-  acm_config.neteq_config.enable_muted_state = false;
+  acm_config.neteq_config.enable_muted_state = true;
   audio_coding_.reset(AudioCodingModule::Create(acm_config));
 
   _outputAudioLevel.Clear();
 
   RtpRtcp::Configuration configuration;
   configuration.audio = true;
   configuration.outgoing_transport = this;
   configuration.receive_statistics = rtp_receive_statistics_.get();
   configuration.bandwidth_callback = rtcp_observer_.get();
   if (pacing_enabled_) {
@@ skipping 2721 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc