Lock scheme #5: Applied the render queueing to the agc

webrtc/modules/audio_processing/audio_processing_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 513 matching lines @@
                                        const StreamConfig& input_config,
                                        const StreamConfig& output_config,
                                        float* const* dest) {
   CriticalSectionScoped crit_scoped(crit_);
   if (!src || !dest) {
     return kNullPointerError;
   }
 
   echo_cancellation_->ReadQueuedRenderData();
   echo_control_mobile_->ReadQueuedRenderData();
+  gain_control_->ReadQueuedRenderData();
 
   ProcessingConfig processing_config = api_format_;
   processing_config.input_stream() = input_config;
   processing_config.output_stream() = output_config;
 
   RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
   assert(processing_config.input_stream().num_frames() ==
          api_format_.input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
@@ skipping 24 matching lines @@
   }
 #endif
 
   return kNoError;
 }
 
 int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
   CriticalSectionScoped crit_scoped(crit_);
   echo_cancellation_->ReadQueuedRenderData();
   echo_control_mobile_->ReadQueuedRenderData();
+  gain_control_->ReadQueuedRenderData();
 
   if (!frame) {
     return kNullPointerError;
   }
   // Must be a native rate.
   if (frame->sample_rate_hz_ != kSampleRate8kHz &&
       frame->sample_rate_hz_ != kSampleRate16kHz &&
       frame->sample_rate_hz_ != kSampleRate32kHz &&
       frame->sample_rate_hz_ != kSampleRate48kHz) {
     return kBadSampleRateError;
@@ skipping 682 matching lines @@
 
   event_msg_->set_type(audioproc::Event::CONFIG);
   event_msg_->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile());
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc