Preparational work for an upcoming addition of a threadchecking scheme for APM

webrtc/modules/audio_processing/gain_control_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 25 matching lines @@
 
 // Maximum length that a frame of samples can have.
 static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
 // Maximum number of frames to buffer in the render queue.
 // TODO(peah): Decrease this once we properly handle hugely unbalanced
 // reverse and forward call numbers.
 static const size_t kMaxNumFramesToBuffer = 100;
 
 }  // namespace
 
-GainControlImpl::GainControlImpl(const AudioProcessing* apm,
-                                 CriticalSectionWrapper* crit)
+GainControlImpl::GainControlImpl(
+    const AudioProcessing* apm,
+    CriticalSectionWrapper* crit,
+    const rtc::ThreadChecker* render_thread_checker,
+    const rtc::ThreadChecker* capture_thread_checker)
     : ProcessingComponent(),
       apm_(apm),
       crit_(crit),
+      render_thread_checker_(render_thread_checker),
+      capture_thread_checker_(capture_thread_checker),
       mode_(kAdaptiveAnalog),
       minimum_capture_level_(0),
       maximum_capture_level_(255),
       limiter_enabled_(true),
       target_level_dbfs_(3),
       compression_gain_db_(9),
       analog_capture_level_(0),
       was_analog_level_set_(false),
       stream_is_saturated_(false),
       render_queue_element_max_size_(0) {}
 
 GainControlImpl::~GainControlImpl() {}
 
 int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
+  RTC_DCHECK(render_thread_checker_->CalledOnValidThread());
   if (!is_component_enabled()) {
     return apm_->kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
 
   render_queue_buffer_.resize(0);
   for (int i = 0; i < num_handles(); i++) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
     int err =
@@ skipping 138 matching lines @@
 
     analog_capture_level_ /= num_handles();
   }
 
   was_analog_level_set_ = false;
   return apm_->kNoError;
 }
 
 // TODO(ajm): ensure this is called under kAdaptiveAnalog.
 int GainControlImpl::set_stream_analog_level(int level) {
+  RTC_DCHECK(capture_thread_checker_->CalledOnValidThread());
   CriticalSectionScoped crit_scoped(crit_);
   was_analog_level_set_ = true;
   if (level < minimum_capture_level_ || level > maximum_capture_level_) {
     return apm_->kBadParameterError;
   }
   analog_capture_level_ = level;
 
   return apm_->kNoError;
 }
 
 int GainControlImpl::stream_analog_level() {
+  RTC_DCHECK(capture_thread_checker_->CalledOnValidThread());
   // TODO(ajm): enable this assertion?
   //assert(mode_ == kAdaptiveAnalog);
 
   return analog_capture_level_;
 }
 
 int GainControlImpl::Enable(bool enable) {
   CriticalSectionScoped crit_scoped(crit_);
   return EnableComponent(enable);
 }
@@ skipping 156 matching lines @@
   return apm_->num_output_channels();
 }
 
 int GainControlImpl::GetHandleError(void* handle) const {
   // The AGC has no get_error() function.
   // (Despite listing errors in its interface...)
   assert(handle != NULL);
   return apm_->kUnspecifiedError;
 }
 }  // namespace webrtc