Avoid clicks when muting/unmuting a 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 748 matching lines @@
       _moduleProcessThreadPtr(NULL),
       _audioDeviceModulePtr(NULL),
       _voiceEngineObserverPtr(NULL),
       _callbackCritSectPtr(NULL),
       _transportPtr(NULL),
       _rxVadObserverPtr(NULL),
       _oldVadDecision(-1),
       _sendFrameType(0),
       _externalMixing(false),
       _mixFileWithMicrophone(false),
-      _mute(false),
+      input_mute_(false),
+      previous_frame_muted_(false),
       _panLeft(1.0f),
       _panRight(1.0f),
       _outputGain(1.0f),
       _lastLocalTimeStamp(0),
       _lastPayloadType(0),
       _includeAudioLevelIndication(false),
       _outputSpeechType(AudioFrame::kNormalSpeech),
       _average_jitter_buffer_delay_us(0),
       _previousTimestamp(0),
       _recPacketDelayMs(20),
@@ skipping 1351 matching lines @@
   level = static_cast<int32_t>(currentLevel);
   return 0;
 }
 
 int Channel::GetSpeechOutputLevelFullRange(uint32_t& level) const {
   int16_t currentLevel = _outputAudioLevel.LevelFullRange();
   level = static_cast<int32_t>(currentLevel);
   return 0;
 }
 
-int Channel::SetMute(bool enable) {
+int Channel::SetInputMute(bool enable) {
   rtc::CritScope cs(&volume_settings_critsect_);
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetMute(enable=%d)", enable);
-  _mute = enable;
+  input_mute_ = enable;
   return 0;
 }
 
-bool Channel::Mute() const {
+bool Channel::InputMute() const {
   rtc::CritScope cs(&volume_settings_critsect_);
-  return _mute;
+  return input_mute_;
 }
 
 int Channel::SetOutputVolumePan(float left, float right) {
   rtc::CritScope cs(&volume_settings_critsect_);
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetOutputVolumePan()");
   _panLeft = left;
   _panRight = right;
   return 0;
 }
@@ skipping 784 matching lines @@
   if (_audioFrame.samples_per_channel_ == 0) {
     WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::PrepareEncodeAndSend() invalid audio frame");
     return 0xFFFFFFFF;
   }
 
   if (channel_state_.Get().input_file_playing) {
     MixOrReplaceAudioWithFile(mixingFrequency);
   }
 
-  bool is_muted = Mute();  // Cache locally as Mute() takes a lock.
-  if (is_muted) {
-    AudioFrameOperations::Mute(_audioFrame);
-  }
+  bool is_muted = InputMute();  // Cache locally as InputMute() takes a lock.
+  AudioFrameOperations::Mute(&_audioFrame, previous_frame_muted_, is_muted);
 
   if (channel_state_.Get().input_external_media) {
     rtc::CritScope cs(&_callbackCritSect);
     const bool isStereo = (_audioFrame.num_channels_ == 2);
     if (_inputExternalMediaCallbackPtr) {
       _inputExternalMediaCallbackPtr->Process(
           _channelId, kRecordingPerChannel, (int16_t*)_audioFrame.data_,
           _audioFrame.samples_per_channel_, _audioFrame.sample_rate_hz_,
           isStereo);
     }
   }
 
   if (_includeAudioLevelIndication) {
     size_t length =
         _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
-    if (is_muted) {
+    if (is_muted && previous_frame_muted_) {
       rms_level_.ProcessMuted(length);
     } else {
       rms_level_.Process(_audioFrame.data_, length);
     }
   }
+  previous_frame_muted_ = is_muted;
 
   return 0;
 }
 
 uint32_t Channel::EncodeAndSend() {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::EncodeAndSend()");
 
   assert(_audioFrame.num_channels_ <= 2);
   if (_audioFrame.samples_per_channel_ == 0) {
@@ skipping 538 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