Propagate muted parameter to VoE::Channel

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 465 matching lines @@
   return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
 }
 
 int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
   if (event_log_) {
     unsigned int ssrc;
     RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
     event_log_->LogAudioPlayout(ssrc);
   }
   // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
-  if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame) ==
-      -1) {
+  bool muted;
+  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 -1;
   }
+  RTC_DCHECK(!muted);
 
   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 298 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;
   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