Lock scheme #8: Introduced the new locking scheme

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 18 matching lines @@
 #include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
 #include "webrtc/modules/audio_processing/echo_control_mobile_impl.h"
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
 #include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
 #include "webrtc/modules/audio_processing/level_estimator_impl.h"
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 #include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
 #include "webrtc/modules/audio_processing/voice_detection_impl.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
 #else
 #include "webrtc/audio_processing/debug.pb.h"
@@ skipping 17 matching lines @@
     case AudioProcessing::kStereo:
       return false;
     case AudioProcessing::kMonoAndKeyboard:
     case AudioProcessing::kStereoAndKeyboard:
       return true;
   }
 
   assert(false);
   return false;
 }
+}  // namespace
 
-}  // namespace
+struct ApmExternallyAccessibleSubModules {
+  ApmExternallyAccessibleSubModules()
+      : echo_cancellation(NULL),
+        echo_control_mobile(NULL),
+        gain_control(NULL),
+        high_pass_filter(NULL),
+        level_estimator(NULL),
+        noise_suppression(NULL),
+        voice_detection(NULL) {}
+  // Accessed externally of APM without any lock acquired.
+  EchoCancellationImpl* echo_cancellation;
+  EchoControlMobileImpl* echo_control_mobile;
+  GainControlImpl* gain_control;
+  HighPassFilterImpl* high_pass_filter;
+  LevelEstimatorImpl* level_estimator;
+  NoiseSuppressionImpl* noise_suppression;
+  VoiceDetectionImpl* voice_detection;
+  rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc;
+
+  // Accessed internally from both render and capture.
+  rtc::scoped_ptr<TransientSuppressor> transient_suppressor;
+  rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer;
+};
+
+struct ApmInternalSubModules {
+  explicit ApmInternalSubModules(Beamformer<float>* beamformer)
+      : beamformer(beamformer) {}
+  // Accessed internally from capture or during initialization
+  std::list<ProcessingComponent*> component_list;
+  rtc::scoped_ptr<Beamformer<float>> beamformer;
+  rtc::scoped_ptr<AgcManagerDirect> agc_manager;
+};
 
 // Throughout webrtc, it's assumed that success is represented by zero.
 static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
 
 // This class has two main functionalities:
 //
 // 1) It is returned instead of the real GainControl after the new AGC has been
 //    enabled in order to prevent an outside user from overriding compression
 //    settings. It doesn't do anything in its implementation, except for
 //    delegating the const methods and Enable calls to the real GainControl, so
@@ skipping 86 matching lines @@
   }
 
   return apm;
 }
 
 AudioProcessingImpl::AudioProcessingImpl(const Config& config)
     : AudioProcessingImpl(config, nullptr) {}
 
 AudioProcessingImpl::AudioProcessingImpl(const Config& config,
                                          Beamformer<float>* beamformer)
-    : echo_cancellation_(NULL),
-      echo_control_mobile_(NULL),
-      gain_control_(NULL),
-      high_pass_filter_(NULL),
-      level_estimator_(NULL),
-      noise_suppression_(NULL),
-      voice_detection_(NULL),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()),
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-      debug_file_(FileWrapper::Create()),
-      event_msg_(new audioproc::Event()),
+    : public_submodules_(new ApmExternallyAccessibleSubModules()),
+      private_submodules_(new ApmInternalSubModules(beamformer)),
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+      constants_(config.Get<ExperimentalAgc>().startup_min_volume,
+                 config.Get<Beamforming>().array_geometry,
+                 false,
+                 config.Get<Intelligibility>().enabled),
+#else
+      constants_(config.Get<ExperimentalAgc>().startup_min_volume,
+                 config.Get<Beamforming>().array_geometry,
+                 config.Get<ExperimentalAgc>().enabled,
+                 config.Get<Intelligibility>().enabled),
 #endif
-      fwd_proc_format_(kSampleRate16kHz),
-      rev_proc_format_(kSampleRate16kHz, 1),
-      split_rate_(kSampleRate16kHz),
-      stream_delay_ms_(0),
-      delay_offset_ms_(0),
-      was_stream_delay_set_(false),
-      last_stream_delay_ms_(0),
-      last_aec_system_delay_ms_(0),
-      stream_delay_jumps_(-1),
-      aec_system_delay_jumps_(-1),
-      output_will_be_muted_(false),
-      key_pressed_(false),
+
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
-      use_new_agc_(false),
+      capture_(false, config.Get<Beamforming>().enabled)
 #else
-      use_new_agc_(config.Get<ExperimentalAgc>().enabled),
+      capture_(config.Get<ExperimentalNs>().enabled,
+               config.Get<Beamforming>().enabled)
 #endif
-      agc_startup_min_volume_(config.Get<ExperimentalAgc>().startup_min_volume),
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
-      transient_suppressor_enabled_(false),
-#else
-      transient_suppressor_enabled_(config.Get<ExperimentalNs>().enabled),
-#endif
-      beamformer_enabled_(config.Get<Beamforming>().enabled),
-      beamformer_(beamformer),
-      array_geometry_(config.Get<Beamforming>().array_geometry),
-      target_direction_(config.Get<Beamforming>().target_direction),
-      intelligibility_enabled_(config.Get<Intelligibility>().enabled) {
+{
   render_thread_checker_.DetachFromThread();
   capture_thread_checker_.DetachFromThread();
 
-  echo_cancellation_ =
-      new EchoCancellationImpl(this, crit_, &render_thread_checker_);
-  component_list_.push_back(echo_cancellation_);
+  {
+    rtc::CritScope cs_render(&crit_render_);

[hlundin-webrtc, 2015/11/05 16:11:21]

Can you avoid taking the locks in the ctor?

[peah-webrtc, 2015/11/06 09:54:31]

Seems to work. I thought that the threadannotation would complain, but it does
not :-)

Done.

[hlundin-webrtc, 2015/11/20 10:36:30]

I think the it intentionally turns a blind eye to ctors and dtors.

+    rtc::CritScope cs_capture(&crit_capture_);
 
-  echo_control_mobile_ =
-      new EchoControlMobileImpl(this, crit_, &render_thread_checker_);
-  component_list_.push_back(echo_control_mobile_);
+    public_submodules_->echo_cancellation = new EchoCancellationImpl(
+        this, &crit_render_, &crit_capture_, &render_thread_checker_);
+    public_submodules_->echo_control_mobile = new EchoControlMobileImpl(
+        this, &crit_render_, &crit_capture_, &render_thread_checker_);
+    public_submodules_->gain_control =
+        new GainControlImpl(this, &crit_capture_, &crit_capture_,
+                            &render_thread_checker_, &capture_thread_checker_);
+    public_submodules_->high_pass_filter =
+        new HighPassFilterImpl(this, &crit_capture_);
+    public_submodules_->level_estimator = new LevelEstimatorImpl(this);
+    public_submodules_->noise_suppression =
+        new NoiseSuppressionImpl(this, &crit_capture_);
+    public_submodules_->voice_detection =
+        new VoiceDetectionImpl(this, &crit_capture_);
+    public_submodules_->gain_control_for_new_agc.reset(
+        new GainControlForNewAgc(public_submodules_->gain_control));
 
-  gain_control_ = new GainControlImpl(this, crit_, &render_thread_checker_,
-                                      &capture_thread_checker_);
-  component_list_.push_back(gain_control_);
-
-  high_pass_filter_ = new HighPassFilterImpl(this, crit_);
-  component_list_.push_back(high_pass_filter_);
-
-  level_estimator_ = new LevelEstimatorImpl(this, crit_);
-  component_list_.push_back(level_estimator_);
-
-  noise_suppression_ = new NoiseSuppressionImpl(this, crit_);
-  component_list_.push_back(noise_suppression_);
-
-  voice_detection_ = new VoiceDetectionImpl(this, crit_);
-  component_list_.push_back(voice_detection_);
-
-  gain_control_for_new_agc_.reset(new GainControlForNewAgc(gain_control_));
+    private_submodules_->component_list.push_back(
+        public_submodules_->echo_cancellation);
+    private_submodules_->component_list.push_back(
+        public_submodules_->echo_control_mobile);
+    private_submodules_->component_list.push_back(
+        public_submodules_->gain_control);
+    private_submodules_->component_list.push_back(
+        public_submodules_->high_pass_filter);
+    private_submodules_->component_list.push_back(
+        public_submodules_->level_estimator);
+    private_submodules_->component_list.push_back(
+        public_submodules_->noise_suppression);
+    private_submodules_->component_list.push_back(
+        public_submodules_->voice_detection);
+  }
 
   SetExtraOptions(config);
 }
 
 AudioProcessingImpl::~AudioProcessingImpl() {
   {
-    CriticalSectionScoped crit_scoped(crit_);
-    // Depends on gain_control_ and gain_control_for_new_agc_.
-    agc_manager_.reset();
+    rtc::CritScope cs_render(&crit_render_);

[hlundin-webrtc, 2015/11/05 16:11:21]

There is no point in acquiring locks in the dtor. All other threads that could
possibly access this object must already be stopped (or at least been made aware
of that this object is no longer).

[peah-webrtc, 2015/11/06 09:54:31]

Done.

+    rtc::CritScope cs_capture(&crit_capture_);
+
+    // Set submodule pointers to NULL to ensure proper behavior if the accessor

[hlundin-webrtc, 2015/11/05 16:11:21]

This also makes no sense.

[peah-webrtc, 2015/11/06 09:54:31]

Done.

+    // methods
+    // are called during APM destruction. Note that the submodule objects are
+    // deleted
+    // below using the component vector.
+    public_submodules_->echo_cancellation = nullptr;
+    public_submodules_->echo_control_mobile = nullptr;
+    public_submodules_->gain_control = nullptr;
+    public_submodules_->high_pass_filter = nullptr;
+    public_submodules_->level_estimator = nullptr;
+    public_submodules_->noise_suppression = nullptr;
+    public_submodules_->voice_detection = nullptr;
+
+    // Depends on gain_control_ and
+    // public_submodules_->gain_control_for_new_agc.
+    private_submodules_->agc_manager.reset();
     // Depends on gain_control_.
-    gain_control_for_new_agc_.reset();
-    while (!component_list_.empty()) {
-      ProcessingComponent* component = component_list_.front();
+    public_submodules_->gain_control_for_new_agc.reset();
+    while (!private_submodules_->component_list.empty()) {
+      ProcessingComponent* component =
+          private_submodules_->component_list.front();
       component->Destroy();
       delete component;
-      component_list_.pop_front();
+      private_submodules_->component_list.pop_front();
     }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-    if (debug_file_->Open()) {
-      debug_file_->CloseFile();
+    if (debug_dump_.debug_file->Open()) {
+      debug_dump_.debug_file->CloseFile();
     }
 #endif
   }
-  delete crit_;
-  crit_ = NULL;
 }
 
 int AudioProcessingImpl::Initialize() {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner during initialization.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
   return InitializeLocked();
 }
 
 int AudioProcessingImpl::Initialize(int input_sample_rate_hz,
                                     int output_sample_rate_hz,
                                     int reverse_sample_rate_hz,
                                     ChannelLayout input_layout,
                                     ChannelLayout output_layout,
                                     ChannelLayout reverse_layout) {
+  // Run in a single-threaded manner during initialization.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+
   const ProcessingConfig processing_config = {
       {{input_sample_rate_hz,
         ChannelsFromLayout(input_layout),
         LayoutHasKeyboard(input_layout)},
        {output_sample_rate_hz,
         ChannelsFromLayout(output_layout),
         LayoutHasKeyboard(output_layout)},
        {reverse_sample_rate_hz,
         ChannelsFromLayout(reverse_layout),
         LayoutHasKeyboard(reverse_layout)},
        {reverse_sample_rate_hz,
         ChannelsFromLayout(reverse_layout),
         LayoutHasKeyboard(reverse_layout)}}};
 
   return Initialize(processing_config);
 }
 
 int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner during initialization.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
   return InitializeLocked(processing_config);
 }
 
 // Calls InitializeLocked() if any of the audio parameters have changed from
-// their current values.
-int AudioProcessingImpl::MaybeInitializeLocked(
+// their current values (needs to be called while holding the crit_render_
+// lock).
+int AudioProcessingImpl::MaybeInitialize(
     const ProcessingConfig& processing_config) {
-  if (processing_config == shared_state_.api_format_) {
+  if (processing_config == formats_.api_format) {
     return kNoError;
   }
+
+  rtc::CritScope cs_capture(&crit_capture_);
   return InitializeLocked(processing_config);
 }
 
 int AudioProcessingImpl::InitializeLocked() {
   const int fwd_audio_buffer_channels =
-      beamformer_enabled_
-          ? shared_state_.api_format_.input_stream().num_channels()
-          : shared_state_.api_format_.output_stream().num_channels();
+      capture_.beamformer_enabled
+          ? formats_.api_format.input_stream().num_channels()
+          : formats_.api_format.output_stream().num_channels();
   const int rev_audio_buffer_out_num_frames =
-      shared_state_.api_format_.reverse_output_stream().num_frames() == 0
-          ? rev_proc_format_.num_frames()
-          : shared_state_.api_format_.reverse_output_stream().num_frames();
-  if (shared_state_.api_format_.reverse_input_stream().num_channels() > 0) {
-    render_audio_.reset(new AudioBuffer(
-        shared_state_.api_format_.reverse_input_stream().num_frames(),
-        shared_state_.api_format_.reverse_input_stream().num_channels(),
-        rev_proc_format_.num_frames(), rev_proc_format_.num_channels(),
+      formats_.api_format.reverse_output_stream().num_frames() == 0
+          ? formats_.rev_proc_format.num_frames()
+          : formats_.api_format.reverse_output_stream().num_frames();
+  if (formats_.api_format.reverse_input_stream().num_channels() > 0) {
+    render_.render_audio.reset(new AudioBuffer(
+        formats_.api_format.reverse_input_stream().num_frames(),
+        formats_.api_format.reverse_input_stream().num_channels(),
+        formats_.rev_proc_format.num_frames(),
+        formats_.rev_proc_format.num_channels(),
         rev_audio_buffer_out_num_frames));
     if (rev_conversion_needed()) {
-      render_converter_ = AudioConverter::Create(
-          shared_state_.api_format_.reverse_input_stream().num_channels(),
-          shared_state_.api_format_.reverse_input_stream().num_frames(),
-          shared_state_.api_format_.reverse_output_stream().num_channels(),
-          shared_state_.api_format_.reverse_output_stream().num_frames());
+      render_.render_converter = AudioConverter::Create(
+          formats_.api_format.reverse_input_stream().num_channels(),
+          formats_.api_format.reverse_input_stream().num_frames(),
+          formats_.api_format.reverse_output_stream().num_channels(),
+          formats_.api_format.reverse_output_stream().num_frames());
     } else {
-      render_converter_.reset(nullptr);
+      render_.render_converter.reset(nullptr);
     }
   } else {
-    render_audio_.reset(nullptr);
-    render_converter_.reset(nullptr);
+    render_.render_audio.reset(nullptr);
+    render_.render_converter.reset(nullptr);
   }
-  capture_audio_.reset(
-      new AudioBuffer(shared_state_.api_format_.input_stream().num_frames(),
-                      shared_state_.api_format_.input_stream().num_channels(),
-                      fwd_proc_format_.num_frames(), fwd_audio_buffer_channels,
-                      shared_state_.api_format_.output_stream().num_frames()));
+  capture_.capture_audio.reset(
+      new AudioBuffer(formats_.api_format.input_stream().num_frames(),
+                      formats_.api_format.input_stream().num_channels(),
+                      capture_nonlocked_.fwd_proc_format.num_frames(),
+                      fwd_audio_buffer_channels,
+                      formats_.api_format.output_stream().num_frames()));
 
   // Initialize all components.
-  for (auto item : component_list_) {
+  for (auto item : private_submodules_->component_list) {
     int err = item->Initialize();
     if (err != kNoError) {
       return err;
     }
   }
 
   InitializeExperimentalAgc();
 
   InitializeTransient();
 
   InitializeBeamformer();
 
   InitializeIntelligibility();
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
+  if (debug_dump_.debug_file->Open()) {
     int err = WriteInitMessage();
     if (err != kNoError) {
       return err;
     }
   }
 #endif
 
   return kNoError;
 }
 
@@ skipping 10 matching lines @@
   const int num_in_channels = config.input_stream().num_channels();
   const int num_out_channels = config.output_stream().num_channels();
 
   // Need at least one input channel.
   // Need either one output channel or as many outputs as there are inputs.
   if (num_in_channels == 0 ||
       !(num_out_channels == 1 || num_out_channels == num_in_channels)) {
     return kBadNumberChannelsError;
   }
 
-  if (beamformer_enabled_ &&
-      (static_cast<size_t>(num_in_channels) != array_geometry_.size() ||
-       num_out_channels > 1)) {
+  if (capture_.beamformer_enabled && (static_cast<size_t>(num_in_channels) !=
+                                          constants_.array_geometry.size() ||
+                                      num_out_channels > 1)) {
     return kBadNumberChannelsError;
   }
 
-  shared_state_.api_format_ = config;
+  formats_.api_format = config;
 
   // We process at the closest native rate >= min(input rate, output rate)...
   const int min_proc_rate =
-      std::min(shared_state_.api_format_.input_stream().sample_rate_hz(),
-               shared_state_.api_format_.output_stream().sample_rate_hz());
+      std::min(formats_.api_format.input_stream().sample_rate_hz(),
+               formats_.api_format.output_stream().sample_rate_hz());
   int fwd_proc_rate;
   for (size_t i = 0; i < kNumNativeSampleRates; ++i) {
     fwd_proc_rate = kNativeSampleRatesHz[i];
     if (fwd_proc_rate >= min_proc_rate) {
       break;
     }
   }
   // ...with one exception.
-  if (echo_control_mobile_->is_enabled() &&
+  if (public_submodules_->echo_control_mobile->is_enabled() &&
       min_proc_rate > kMaxAECMSampleRateHz) {
     fwd_proc_rate = kMaxAECMSampleRateHz;
   }
 
-  fwd_proc_format_ = StreamConfig(fwd_proc_rate);
+  capture_nonlocked_.fwd_proc_format = StreamConfig(fwd_proc_rate);
 
   // We normally process the reverse stream at 16 kHz. Unless...
   int rev_proc_rate = kSampleRate16kHz;
-  if (fwd_proc_format_.sample_rate_hz() == kSampleRate8kHz) {
+  if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate8kHz) {
     // ...the forward stream is at 8 kHz.
     rev_proc_rate = kSampleRate8kHz;
   } else {
-    if (shared_state_.api_format_.reverse_input_stream().sample_rate_hz() ==
+    if (formats_.api_format.reverse_input_stream().sample_rate_hz() ==
         kSampleRate32kHz) {
       // ...or the input is at 32 kHz, in which case we use the splitting
       // filter rather than the resampler.
       rev_proc_rate = kSampleRate32kHz;
     }
   }
 
   // Always downmix the reverse stream to mono for analysis. This has been
   // demonstrated to work well for AEC in most practical scenarios.
-  rev_proc_format_ = StreamConfig(rev_proc_rate, 1);
+  formats_.rev_proc_format = StreamConfig(rev_proc_rate, 1);
 
-  if (fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
-      fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz) {
-    split_rate_ = kSampleRate16kHz;
+  if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate32kHz ||
+      capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate48kHz) {
+    capture_nonlocked_.split_rate = kSampleRate16kHz;
   } else {
-    split_rate_ = fwd_proc_format_.sample_rate_hz();
+    capture_nonlocked_.split_rate =
+        capture_nonlocked_.fwd_proc_format.sample_rate_hz();
   }
 
   return InitializeLocked();
 }
 
 void AudioProcessingImpl::SetExtraOptions(const Config& config) {
-  CriticalSectionScoped crit_scoped(crit_);
-  for (auto item : component_list_) {
+  // Run in a single-threaded manner when setting the extra options.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+  for (auto item : private_submodules_->component_list) {
     item->SetExtraOptions(config);
   }
 
-  if (transient_suppressor_enabled_ != config.Get<ExperimentalNs>().enabled) {
-    transient_suppressor_enabled_ = config.Get<ExperimentalNs>().enabled;
+  if (capture_.transient_suppressor_enabled !=
+      config.Get<ExperimentalNs>().enabled) {
+    capture_.transient_suppressor_enabled =
+        config.Get<ExperimentalNs>().enabled;
     InitializeTransient();
   }
 }
 
 
 int AudioProcessingImpl::proc_sample_rate_hz() const {
-  return fwd_proc_format_.sample_rate_hz();
+  // Only called from submodules beneath APM, hence locking is not needed.
+  return capture_nonlocked_.fwd_proc_format.sample_rate_hz();
 }
 
 int AudioProcessingImpl::proc_split_sample_rate_hz() const {
-  return split_rate_;
+  // Only called from submodules/tests beneath APM, hence locking is not needed.
+  return capture_nonlocked_.split_rate;
 }
 
 int AudioProcessingImpl::num_reverse_channels() const {
-  return rev_proc_format_.num_channels();
+  // Only called from submodules/tests beneath APM, hence locking is not needed.
+  return formats_.rev_proc_format.num_channels();
 }
 
 int AudioProcessingImpl::num_input_channels() const {
-  return shared_state_.api_format_.input_stream().num_channels();
+  // Only called from submodules/tests beneath APM, hence locking is not needed.
+  return formats_.api_format.input_stream().num_channels();
 }
 
 int AudioProcessingImpl::num_output_channels() const {
-  return shared_state_.api_format_.output_stream().num_channels();
+  // Only called from submodules/tests beneath APM, hence locking is not needed.
+  return formats_.api_format.output_stream().num_channels();
 }
 
 void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
-  CriticalSectionScoped lock(crit_);
+  rtc::CritScope cs(&crit_capture_);
   RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
-  output_will_be_muted_ = muted;
-  if (agc_manager_.get()) {
-    agc_manager_->SetCaptureMuted(output_will_be_muted_);
+  capture_.output_will_be_muted = muted;
+  if (private_submodules_->agc_manager.get()) {
+    private_submodules_->agc_manager->SetCaptureMuted(
+        capture_.output_will_be_muted);
   }
 }
 
 
 int AudioProcessingImpl::ProcessStream(const float* const* src,
                                        size_t samples_per_channel,
                                        int input_sample_rate_hz,
                                        ChannelLayout input_layout,
                                        int output_sample_rate_hz,
                                        ChannelLayout output_layout,
                                        float* const* dest) {
-  CriticalSectionScoped crit_scoped(crit_);
   RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
-  StreamConfig input_stream = shared_state_.api_format_.input_stream();
+  StreamConfig input_stream;
+  StreamConfig output_stream;
+  {
+    // Access the formats_.api_format.input_stream beneath the capture
+    // lock.

[hlundin-webrtc, 2015/11/05 16:11:21]

Widow.

[peah-webrtc, 2015/11/06 09:54:31]

Done.

+    // The lock must be released as it is later required in the call
+    // to ProcessStream(,,,);
+    rtc::CritScope cs(&crit_capture_);
+    input_stream = formats_.api_format.input_stream();
+    output_stream = formats_.api_format.output_stream();
+  }
+
   input_stream.set_sample_rate_hz(input_sample_rate_hz);
   input_stream.set_num_channels(ChannelsFromLayout(input_layout));
   input_stream.set_has_keyboard(LayoutHasKeyboard(input_layout));
-
-  StreamConfig output_stream = shared_state_.api_format_.output_stream();
   output_stream.set_sample_rate_hz(output_sample_rate_hz);
   output_stream.set_num_channels(ChannelsFromLayout(output_layout));
   output_stream.set_has_keyboard(LayoutHasKeyboard(output_layout));
 
   if (samples_per_channel != input_stream.num_frames()) {
     return kBadDataLengthError;
   }
   return ProcessStream(src, input_stream, output_stream, dest);
 }
 
 int AudioProcessingImpl::ProcessStream(const float* const* src,
                                        const StreamConfig& input_config,
                                        const StreamConfig& output_config,
                                        float* const* dest) {
-  CriticalSectionScoped crit_scoped(crit_);
   RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
+  {
+    // Acquire the capture lock in order to safely call the function
+    // that retrieves the render side data. This function accesses apm
+    // getters that need the capture lock held when being called.
+    rtc::CritScope cs_capture(&crit_capture_);
+    public_submodules_->echo_cancellation->ReadQueuedRenderData();
+    public_submodules_->echo_control_mobile->ReadQueuedRenderData();
+    public_submodules_->gain_control->ReadQueuedRenderData();
+  }
   if (!src || !dest) {
     return kNullPointerError;
   }
 
-  echo_cancellation_->ReadQueuedRenderData();
-  echo_control_mobile_->ReadQueuedRenderData();
-  gain_control_->ReadQueuedRenderData();
-
-  ProcessingConfig processing_config = shared_state_.api_format_;
+  ProcessingConfig processing_config = formats_.api_format;
   processing_config.input_stream() = input_config;
   processing_config.output_stream() = output_config;
 
-  RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
+  {
+    // Do conditional reinitialization.
+    rtc::CritScope cs_render(&crit_render_);
+    RETURN_ON_ERR(MaybeInitialize(processing_config));
+  }
+  rtc::CritScope cs_capture(&crit_capture_);
+
   assert(processing_config.input_stream().num_frames() ==
-         shared_state_.api_format_.input_stream().num_frames());
+         formats_.api_format.input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
+  if (debug_dump_.debug_file->Open()) {
     RETURN_ON_ERR(WriteConfigMessage(false));
 
-    event_msg_->set_type(audioproc::Event::STREAM);
-    audioproc::Stream* msg = event_msg_->mutable_stream();
+    debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM);
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t channel_size =
-        sizeof(float) * shared_state_.api_format_.input_stream().num_frames();
-    for (int i = 0; i < shared_state_.api_format_.input_stream().num_channels();
-         ++i)
+        sizeof(float) * formats_.api_format.input_stream().num_frames();
+    for (int i = 0; i < formats_.api_format.input_stream().num_channels(); ++i)
       msg->add_input_channel(src[i], channel_size);
   }
 #endif
 
-  capture_audio_->CopyFrom(src, shared_state_.api_format_.input_stream());
+  capture_.capture_audio->CopyFrom(src, formats_.api_format.input_stream());
   RETURN_ON_ERR(ProcessStreamLocked());
-  capture_audio_->CopyTo(shared_state_.api_format_.output_stream(), dest);
+  capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest);
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    audioproc::Stream* msg = event_msg_->mutable_stream();
+  if (debug_dump_.debug_file->Open()) {
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t channel_size =
-        sizeof(float) * shared_state_.api_format_.output_stream().num_frames();
-    for (int i = 0;
-         i < shared_state_.api_format_.output_stream().num_channels(); ++i)
+        sizeof(float) * formats_.api_format.output_stream().num_frames();
+    for (int i = 0; i < formats_.api_format.output_stream().num_channels(); ++i)
       msg->add_output_channel(dest[i], channel_size);
-    RETURN_ON_ERR(WriteMessageToDebugFile());
+    RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &crit_debug_, &debug_dump_.capture));
   }
 #endif
 
   return kNoError;
 }
 
 int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
-  CriticalSectionScoped crit_scoped(crit_);
-  echo_cancellation_->ReadQueuedRenderData();
-  echo_control_mobile_->ReadQueuedRenderData();
-  gain_control_->ReadQueuedRenderData();
+  // TODO(peah): Investigate why it is not possible to use a
+  // thread checker for this method.
+  {
+    // Acquire the capture lock in order to safely call the function
+    // that retrieves the render side data. This function accesses apm
+    // getters that need the capture lock held when being called.
+    // The lock needs to be released as
+    // public_submodules_->echo_control_mobile->is_enabled() aquires this lock
+    // as well.
+    rtc::CritScope cs_capture(&crit_capture_);
+    public_submodules_->echo_cancellation->ReadQueuedRenderData();
+    public_submodules_->echo_control_mobile->ReadQueuedRenderData();
+    public_submodules_->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;
   }
 
-  if (echo_control_mobile_->is_enabled() &&
+  if (public_submodules_->echo_control_mobile->is_enabled() &&
       frame->sample_rate_hz_ > kMaxAECMSampleRateHz) {
     LOG(LS_ERROR) << "AECM only supports 16 or 8 kHz sample rates";
     return kUnsupportedComponentError;
   }
 
-  // TODO(ajm): The input and output rates and channels are currently
-  // constrained to be identical in the int16 interface.
-  ProcessingConfig processing_config = shared_state_.api_format_;
+  ProcessingConfig processing_config;
+  {
+    // Aquire lock for the access of api_format.
+    // The lock is released immediately due to the conditional
+    // reinitialization.
+    rtc::CritScope cs_capture(&crit_capture_);
+    // TODO(ajm): The input and output rates and channels are currently
+    // constrained to be identical in the int16 interface.
+    processing_config = formats_.api_format;
+  }
   processing_config.input_stream().set_sample_rate_hz(frame->sample_rate_hz_);
   processing_config.input_stream().set_num_channels(frame->num_channels_);
   processing_config.output_stream().set_sample_rate_hz(frame->sample_rate_hz_);
   processing_config.output_stream().set_num_channels(frame->num_channels_);
 
-  RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
+  {
+    // Do conditional reinitialization.
+    rtc::CritScope cs_render(&crit_render_);
+    RETURN_ON_ERR(MaybeInitialize(processing_config));
+  }
+  rtc::CritScope cs_capture(&crit_capture_);
   if (frame->samples_per_channel_ !=
-      shared_state_.api_format_.input_stream().num_frames()) {
+      formats_.api_format.input_stream().num_frames()) {
     return kBadDataLengthError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    event_msg_->set_type(audioproc::Event::STREAM);
-    audioproc::Stream* msg = event_msg_->mutable_stream();
+  if (debug_dump_.debug_file->Open()) {
+    debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM);
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_input_data(frame->data_, data_size);
   }
 #endif
 
-  capture_audio_->DeinterleaveFrom(frame);
+  capture_.capture_audio->DeinterleaveFrom(frame);
   RETURN_ON_ERR(ProcessStreamLocked());
-  capture_audio_->InterleaveTo(frame, output_copy_needed(is_data_processed()));
+  capture_.capture_audio->InterleaveTo(frame,
+                                       output_copy_needed(is_data_processed()));
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    audioproc::Stream* msg = event_msg_->mutable_stream();
+  if (debug_dump_.debug_file->Open()) {
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_output_data(frame->data_, data_size);
-    RETURN_ON_ERR(WriteMessageToDebugFile());
+    RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &crit_debug_, &debug_dump_.capture));
   }
 #endif
 
   return kNoError;
 }
 
 int AudioProcessingImpl::ProcessStreamLocked() {
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    audioproc::Stream* msg = event_msg_->mutable_stream();
-    msg->set_delay(stream_delay_ms_);
-    msg->set_drift(echo_cancellation_->stream_drift_samples());
+  if (debug_dump_.debug_file->Open()) {
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
+    msg->set_delay(capture_nonlocked_.stream_delay_ms);
+    msg->set_drift(
+        public_submodules_->echo_cancellation->stream_drift_samples());
     msg->set_level(gain_control()->stream_analog_level());
-    msg->set_keypress(key_pressed_);
+    msg->set_keypress(capture_.key_pressed);
   }
 #endif
 
   MaybeUpdateHistograms();
 
-  AudioBuffer* ca = capture_audio_.get();  // For brevity.
+  AudioBuffer* ca = capture_.capture_audio.get();  // For brevity.
 
-  if (use_new_agc_ && gain_control_->is_enabled()) {
-    agc_manager_->AnalyzePreProcess(ca->channels()[0], ca->num_channels(),
-                                    fwd_proc_format_.num_frames());
+  if (constants_.use_new_agc &&
+      public_submodules_->gain_control->is_enabled()) {
+    private_submodules_->agc_manager->AnalyzePreProcess(
+        ca->channels()[0], ca->num_channels(),
+        capture_nonlocked_.fwd_proc_format.num_frames());
   }
 
   bool data_processed = is_data_processed();
   if (analysis_needed(data_processed)) {
     ca->SplitIntoFrequencyBands();
   }
 
-  if (intelligibility_enabled_) {
-    intelligibility_enhancer_->AnalyzeCaptureAudio(
-        ca->split_channels_f(kBand0To8kHz), split_rate_, ca->num_channels());
+  if (constants_.intelligibility_enabled) {
+    public_submodules_->intelligibility_enhancer->AnalyzeCaptureAudio(
+        ca->split_channels_f(kBand0To8kHz), capture_nonlocked_.split_rate,
+        ca->num_channels());
   }
 
-  if (beamformer_enabled_) {
-    beamformer_->ProcessChunk(*ca->split_data_f(), ca->split_data_f());
+  if (capture_.beamformer_enabled) {
+    private_submodules_->beamformer->ProcessChunk(*ca->split_data_f(),
+                                                  ca->split_data_f());
     ca->set_num_channels(1);
   }
 
-  RETURN_ON_ERR(high_pass_filter_->ProcessCaptureAudio(ca));
-  RETURN_ON_ERR(gain_control_->AnalyzeCaptureAudio(ca));
-  RETURN_ON_ERR(noise_suppression_->AnalyzeCaptureAudio(ca));
-  RETURN_ON_ERR(echo_cancellation_->ProcessCaptureAudio(ca));
+  RETURN_ON_ERR(public_submodules_->high_pass_filter->ProcessCaptureAudio(ca));
+  RETURN_ON_ERR(public_submodules_->gain_control->AnalyzeCaptureAudio(ca));
+  RETURN_ON_ERR(public_submodules_->noise_suppression->AnalyzeCaptureAudio(ca));
+  RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessCaptureAudio(ca));
 
-  if (echo_control_mobile_->is_enabled() && noise_suppression_->is_enabled()) {
+  if (public_submodules_->echo_control_mobile->is_enabled() &&
+      public_submodules_->noise_suppression->is_enabled()) {
     ca->CopyLowPassToReference();
   }
-  RETURN_ON_ERR(noise_suppression_->ProcessCaptureAudio(ca));
-  RETURN_ON_ERR(echo_control_mobile_->ProcessCaptureAudio(ca));
-  RETURN_ON_ERR(voice_detection_->ProcessCaptureAudio(ca));
+  RETURN_ON_ERR(public_submodules_->noise_suppression->ProcessCaptureAudio(ca));
+  RETURN_ON_ERR(
+      public_submodules_->echo_control_mobile->ProcessCaptureAudio(ca));
+  RETURN_ON_ERR(public_submodules_->voice_detection->ProcessCaptureAudio(ca));
 
-  if (use_new_agc_ && gain_control_->is_enabled() &&
-      (!beamformer_enabled_ || beamformer_->is_target_present())) {
-    agc_manager_->Process(ca->split_bands_const(0)[kBand0To8kHz],
-                          ca->num_frames_per_band(), split_rate_);
+  if (constants_.use_new_agc &&
+      public_submodules_->gain_control->is_enabled() &&
+      (!capture_.beamformer_enabled ||
+       private_submodules_->beamformer->is_target_present())) {
+    private_submodules_->agc_manager->Process(
+        ca->split_bands_const(0)[kBand0To8kHz], ca->num_frames_per_band(),
+        capture_nonlocked_.split_rate);
   }
-  RETURN_ON_ERR(gain_control_->ProcessCaptureAudio(ca));
+  RETURN_ON_ERR(public_submodules_->gain_control->ProcessCaptureAudio(ca));
 
   if (synthesis_needed(data_processed)) {
     ca->MergeFrequencyBands();
   }
 
   // TODO(aluebs): Investigate if the transient suppression placement should be
   // before or after the AGC.
-  if (transient_suppressor_enabled_) {
+  if (capture_.transient_suppressor_enabled) {
     float voice_probability =
-        agc_manager_.get() ? agc_manager_->voice_probability() : 1.f;
+        private_submodules_->agc_manager.get()
+            ? private_submodules_->agc_manager->voice_probability()
+            : 1.f;
 
-    transient_suppressor_->Suppress(
+    public_submodules_->transient_suppressor->Suppress(
         ca->channels_f()[0], ca->num_frames(), ca->num_channels(),
         ca->split_bands_const_f(0)[kBand0To8kHz], ca->num_frames_per_band(),
         ca->keyboard_data(), ca->num_keyboard_frames(), voice_probability,
-        key_pressed_);
+        capture_.key_pressed);
   }
 
   // The level estimator operates on the recombined data.
-  RETURN_ON_ERR(level_estimator_->ProcessStream(ca));
+  RETURN_ON_ERR(public_submodules_->level_estimator->ProcessStream(ca));
 
-  was_stream_delay_set_ = false;
+  capture_.was_stream_delay_set = false;
   return kNoError;
 }
 
 int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
                                               size_t samples_per_channel,
                                               int rev_sample_rate_hz,
                                               ChannelLayout layout) {
   RTC_DCHECK(render_thread_checker_.CalledOnValidThread());
+  rtc::CritScope cs(&crit_render_);
   const StreamConfig reverse_config = {
       rev_sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout),
   };
   if (samples_per_channel != reverse_config.num_frames()) {
     return kBadDataLengthError;
   }
-  return AnalyzeReverseStream(data, reverse_config, reverse_config);
+  return AnalyzeReverseStreamLocked(data, reverse_config, reverse_config);
 }
 
 int AudioProcessingImpl::ProcessReverseStream(
     const float* const* src,
     const StreamConfig& reverse_input_config,
     const StreamConfig& reverse_output_config,
     float* const* dest) {
   RTC_DCHECK(render_thread_checker_.CalledOnValidThread());
-  RETURN_ON_ERR(
-      AnalyzeReverseStream(src, reverse_input_config, reverse_output_config));
+  rtc::CritScope cs(&crit_render_);
+  RETURN_ON_ERR(AnalyzeReverseStreamLocked(src, reverse_input_config,
+                                           reverse_output_config));
   if (is_rev_processed()) {
-    render_audio_->CopyTo(shared_state_.api_format_.reverse_output_stream(),
-                          dest);
+    render_.render_audio->CopyTo(formats_.api_format.reverse_output_stream(),
+                                 dest);
   } else if (rev_conversion_needed()) {
-    render_converter_->Convert(src, reverse_input_config.num_samples(), dest,
-                               reverse_output_config.num_samples());
+    render_.render_converter->Convert(src, reverse_input_config.num_samples(),
+                                      dest,
+                                      reverse_output_config.num_samples());
   } else {
     CopyAudioIfNeeded(src, reverse_input_config.num_frames(),
                       reverse_input_config.num_channels(), dest);
   }
 
   return kNoError;
 }
 
-int AudioProcessingImpl::AnalyzeReverseStream(
+int AudioProcessingImpl::AnalyzeReverseStreamLocked(
     const float* const* src,
     const StreamConfig& reverse_input_config,
     const StreamConfig& reverse_output_config) {
-  CriticalSectionScoped crit_scoped(crit_);
   if (src == NULL) {
     return kNullPointerError;
   }
 
   if (reverse_input_config.num_channels() <= 0) {
     return kBadNumberChannelsError;
   }
 
-  ProcessingConfig processing_config = shared_state_.api_format_;
+  ProcessingConfig processing_config = formats_.api_format;
   processing_config.reverse_input_stream() = reverse_input_config;
   processing_config.reverse_output_stream() = reverse_output_config;
 
-  RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
+  RETURN_ON_ERR(MaybeInitialize(processing_config));
   assert(reverse_input_config.num_frames() ==
-         shared_state_.api_format_.reverse_input_stream().num_frames());
+         formats_.api_format.reverse_input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
-    audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
+  if (debug_dump_.debug_file->Open()) {
+    debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
+    audioproc::ReverseStream* msg =
+        debug_dump_.render.event_msg->mutable_reverse_stream();
     const size_t channel_size =
-        sizeof(float) *
-        shared_state_.api_format_.reverse_input_stream().num_frames();
+        sizeof(float) * formats_.api_format.reverse_input_stream().num_frames();
     for (int i = 0;
-         i < shared_state_.api_format_.reverse_input_stream().num_channels();
-         ++i)
+         i < formats_.api_format.reverse_input_stream().num_channels(); ++i)
       msg->add_channel(src[i], channel_size);
-    RETURN_ON_ERR(WriteMessageToDebugFile());
+    RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &crit_debug_, &debug_dump_.render));
   }
 #endif
 
-  render_audio_->CopyFrom(src,
-                          shared_state_.api_format_.reverse_input_stream());
+  render_.render_audio->CopyFrom(src,
+                                 formats_.api_format.reverse_input_stream());
   return ProcessReverseStreamLocked();
 }
 
 int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
   RTC_DCHECK(render_thread_checker_.CalledOnValidThread());
   RETURN_ON_ERR(AnalyzeReverseStream(frame));
+  rtc::CritScope cs(&crit_render_);
   if (is_rev_processed()) {
-    render_audio_->InterleaveTo(frame, true);
+    render_.render_audio->InterleaveTo(frame, true);
   }
 
   return kNoError;
 }
 
 int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
   RTC_DCHECK(render_thread_checker_.CalledOnValidThread());
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(&crit_render_);
   if (frame == NULL) {
     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;
   }
   // This interface does not tolerate different forward and reverse rates.
   if (frame->sample_rate_hz_ !=
-      shared_state_.api_format_.input_stream().sample_rate_hz()) {
+      formats_.api_format.input_stream().sample_rate_hz()) {
     return kBadSampleRateError;
   }
 
   if (frame->num_channels_ <= 0) {
     return kBadNumberChannelsError;
   }
 
-  ProcessingConfig processing_config = shared_state_.api_format_;
+  ProcessingConfig processing_config = formats_.api_format;
   processing_config.reverse_input_stream().set_sample_rate_hz(
       frame->sample_rate_hz_);
   processing_config.reverse_input_stream().set_num_channels(
       frame->num_channels_);
   processing_config.reverse_output_stream().set_sample_rate_hz(
       frame->sample_rate_hz_);
   processing_config.reverse_output_stream().set_num_channels(
       frame->num_channels_);
 
-  RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
+  RETURN_ON_ERR(MaybeInitialize(processing_config));
   if (frame->samples_per_channel_ !=
-      shared_state_.api_format_.reverse_input_stream().num_frames()) {
+      formats_.api_format.reverse_input_stream().num_frames()) {
     return kBadDataLengthError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
-    audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
+  if (debug_dump_.debug_file->Open()) {
+    debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
+    audioproc::ReverseStream* msg =
+        debug_dump_.render.event_msg->mutable_reverse_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_data(frame->data_, data_size);
-    RETURN_ON_ERR(WriteMessageToDebugFile());
+    RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &crit_debug_, &debug_dump_.render));
   }
 #endif
-  render_audio_->DeinterleaveFrom(frame);
+  render_.render_audio->DeinterleaveFrom(frame);
   return ProcessReverseStreamLocked();
 }
 
 int AudioProcessingImpl::ProcessReverseStreamLocked() {
-  AudioBuffer* ra = render_audio_.get();  // For brevity.
-  if (rev_proc_format_.sample_rate_hz() == kSampleRate32kHz) {
+  AudioBuffer* ra = render_.render_audio.get();  // For brevity.
+  if (formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz) {
     ra->SplitIntoFrequencyBands();
   }
 
-  if (intelligibility_enabled_) {
-    intelligibility_enhancer_->ProcessRenderAudio(
-        ra->split_channels_f(kBand0To8kHz), split_rate_, ra->num_channels());
+  if (constants_.intelligibility_enabled) {
+    // Currently run in single-threaded mode when the intelligibility
+    // enhancer is activated.
+    // TODO(peah): Fix to be properly multi-threaded.
+    rtc::CritScope cs(&crit_capture_);
+    public_submodules_->intelligibility_enhancer->ProcessRenderAudio(
+        ra->split_channels_f(kBand0To8kHz), capture_nonlocked_.split_rate,
+        ra->num_channels());
   }
 
-  RETURN_ON_ERR(echo_cancellation_->ProcessRenderAudio(ra));
-  RETURN_ON_ERR(echo_control_mobile_->ProcessRenderAudio(ra));
-  if (!use_new_agc_) {
-    RETURN_ON_ERR(gain_control_->ProcessRenderAudio(ra));
+  RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessRenderAudio(ra));
+  RETURN_ON_ERR(
+      public_submodules_->echo_control_mobile->ProcessRenderAudio(ra));
+  if (!constants_.use_new_agc) {
+    RETURN_ON_ERR(public_submodules_->gain_control->ProcessRenderAudio(ra));
   }
 
-  if (rev_proc_format_.sample_rate_hz() == kSampleRate32kHz &&
+  if (formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz &&
       is_rev_processed()) {
     ra->MergeFrequencyBands();
   }
 
   return kNoError;
 }
 
 int AudioProcessingImpl::set_stream_delay_ms(int delay) {
+  // TODO(peah): Refactor so that it is possible to add a threadchecker
+  // (currently not solely called from the capture thread in all places).
+  rtc::CritScope cs(&crit_capture_);
   Error retval = kNoError;
-  was_stream_delay_set_ = true;
-  delay += delay_offset_ms_;
+  capture_.was_stream_delay_set = true;
+  delay += capture_.delay_offset_ms;
 
   if (delay < 0) {
     delay = 0;
     retval = kBadStreamParameterWarning;
   }
 
   // TODO(ajm): the max is rather arbitrarily chosen; investigate.
   if (delay > 500) {
     delay = 500;
     retval = kBadStreamParameterWarning;
   }
 
-  stream_delay_ms_ = delay;
+  capture_nonlocked_.stream_delay_ms = delay;
   return retval;
 }
 
 int AudioProcessingImpl::stream_delay_ms() const {
-  return stream_delay_ms_;
+  return capture_nonlocked_.stream_delay_ms;
 }
 
 bool AudioProcessingImpl::was_stream_delay_set() const {
-  return was_stream_delay_set_;
+  // Only called from submodules/tests beneath APM, hence locking is not needed.
+  return capture_.was_stream_delay_set;
 }
 
 void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
-  key_pressed_ = key_pressed;
+  // Thread check not possible due to being called from different threads.
+  rtc::CritScope cs(&crit_capture_);
+  capture_.key_pressed = key_pressed;
 }
 
 void AudioProcessingImpl::set_delay_offset_ms(int offset) {
   RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
-  CriticalSectionScoped crit_scoped(crit_);
-  delay_offset_ms_ = offset;
+  rtc::CritScope cs(&crit_capture_);
+  capture_.delay_offset_ms = offset;
 }
 
 int AudioProcessingImpl::delay_offset_ms() const {
   RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
-  return delay_offset_ms_;
+  rtc::CritScope cs(&crit_capture_);
+  return capture_.delay_offset_ms;
 }
 
 int AudioProcessingImpl::StartDebugRecording(
     const char filename[AudioProcessing::kMaxFilenameSize]) {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+
   static_assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize, "");
 
   if (filename == NULL) {
     return kNullPointerError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // Stop any ongoing recording.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
+  if (debug_dump_.debug_file->Open()) {
+    if (debug_dump_.debug_file->CloseFile() == -1) {
       return kFileError;
     }
   }
 
-  if (debug_file_->OpenFile(filename, false) == -1) {
-    debug_file_->CloseFile();
+  if (debug_dump_.debug_file->OpenFile(filename, false) == -1) {
+    debug_dump_.debug_file->CloseFile();
     return kFileError;
   }
 
   RETURN_ON_ERR(WriteConfigMessage(true));
   RETURN_ON_ERR(WriteInitMessage());
   return kNoError;
 #else
   return kUnsupportedFunctionError;
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
 int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
 
   if (handle == NULL) {
     return kNullPointerError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // Stop any ongoing recording.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
+  if (debug_dump_.debug_file->Open()) {
+    if (debug_dump_.debug_file->CloseFile() == -1) {
       return kFileError;
     }
   }
 
-  if (debug_file_->OpenFromFileHandle(handle, true, false) == -1) {
+  if (debug_dump_.debug_file->OpenFromFileHandle(handle, true, false) == -1) {
     return kFileError;
   }
 
   RETURN_ON_ERR(WriteConfigMessage(true));
   RETURN_ON_ERR(WriteInitMessage());
   return kNoError;
 #else
   return kUnsupportedFunctionError;
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
 int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
     rtc::PlatformFile handle) {
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
   FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
   return StartDebugRecording(stream);
 }
 
 int AudioProcessingImpl::StopDebugRecording() {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // We just return if recording hasn't started.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
+  if (debug_dump_.debug_file->Open()) {
+    if (debug_dump_.debug_file->CloseFile() == -1) {
       return kFileError;
     }
   }
   return kNoError;
 #else
   return kUnsupportedFunctionError;
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
 EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
-  return echo_cancellation_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->echo_cancellation;
 }
 
 EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const {
-  return echo_control_mobile_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->echo_control_mobile;
 }
 
 GainControl* AudioProcessingImpl::gain_control() const {
-  if (use_new_agc_) {
-    return gain_control_for_new_agc_.get();
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  if (constants_.use_new_agc) {
+    return public_submodules_->gain_control_for_new_agc.get();
   }
-  return gain_control_;
+  return public_submodules_->gain_control;
 }
 
 HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
-  return high_pass_filter_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->high_pass_filter;
 }
 
 LevelEstimator* AudioProcessingImpl::level_estimator() const {
-  return level_estimator_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->level_estimator;
 }
 
 NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
-  return noise_suppression_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->noise_suppression;
 }
 
 VoiceDetection* AudioProcessingImpl::voice_detection() const {
-  return voice_detection_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->voice_detection;
 }
 
 bool AudioProcessingImpl::is_data_processed() const {
-  if (beamformer_enabled_) {
+  if (capture_.beamformer_enabled) {
     return true;
   }
 
   int enabled_count = 0;
-  for (auto item : component_list_) {
+  for (auto item : private_submodules_->component_list) {
     if (item->is_component_enabled()) {
       enabled_count++;
     }
   }
 
-  // Data is unchanged if no components are enabled, or if only level_estimator_
-  // or voice_detection_ is enabled.
+  // Data is unchanged if no components are enabled, or if only
+  // public_submodules_->level_estimator
+  // or public_submodules_->voice_detection is enabled.
   if (enabled_count == 0) {
     return false;
   } else if (enabled_count == 1) {
-    if (level_estimator_->is_enabled() || voice_detection_->is_enabled()) {
+    if (public_submodules_->level_estimator->is_enabled() ||
+        public_submodules_->voice_detection->is_enabled()) {
       return false;
     }
   } else if (enabled_count == 2) {
-    if (level_estimator_->is_enabled() && voice_detection_->is_enabled()) {
+    if (public_submodules_->level_estimator->is_enabled() &&
+        public_submodules_->voice_detection->is_enabled()) {
       return false;
     }
   }
   return true;
 }
 
 bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const {
   // Check if we've upmixed or downmixed the audio.
-  return ((shared_state_.api_format_.output_stream().num_channels() !=
-           shared_state_.api_format_.input_stream().num_channels()) ||
-          is_data_processed || transient_suppressor_enabled_);
+  return ((formats_.api_format.output_stream().num_channels() !=
+           formats_.api_format.input_stream().num_channels()) ||
+          is_data_processed || capture_.transient_suppressor_enabled);
 }
 
 bool AudioProcessingImpl::synthesis_needed(bool is_data_processed) const {
   return (is_data_processed &&
-          (fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
-           fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz));
+          (capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
+               kSampleRate32kHz ||
+           capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
+               kSampleRate48kHz));
 }
 
 bool AudioProcessingImpl::analysis_needed(bool is_data_processed) const {
-  if (!is_data_processed && !voice_detection_->is_enabled() &&
-      !transient_suppressor_enabled_) {
-    // Only level_estimator_ is enabled.
+  if (!is_data_processed &&
+      !public_submodules_->voice_detection->is_enabled() &&
+      !capture_.transient_suppressor_enabled) {
+    // Only public_submodules_->level_estimator is enabled.
     return false;
-  } else if (fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
-             fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz) {
-    // Something besides level_estimator_ is enabled, and we have super-wb.
+  } else if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
+                 kSampleRate32kHz ||
+             capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
+                 kSampleRate48kHz) {
+    // Something besides public_submodules_->level_estimator is enabled, and we
+    // have super-wb.
     return true;
   }
   return false;
 }
 
 bool AudioProcessingImpl::is_rev_processed() const {
-  return intelligibility_enabled_ && intelligibility_enhancer_->active();
+  return constants_.intelligibility_enabled &&
+         public_submodules_->intelligibility_enhancer->active();
 }
 
 bool AudioProcessingImpl::rev_conversion_needed() const {
-  return (shared_state_.api_format_.reverse_input_stream() !=
-          shared_state_.api_format_.reverse_output_stream());
+  return (formats_.api_format.reverse_input_stream() !=
+          formats_.api_format.reverse_output_stream());
 }
 
 void AudioProcessingImpl::InitializeExperimentalAgc() {
-  if (use_new_agc_) {
-    if (!agc_manager_.get()) {
-      agc_manager_.reset(new AgcManagerDirect(gain_control_,
-                                              gain_control_for_new_agc_.get(),
-                                              agc_startup_min_volume_));
+  if (constants_.use_new_agc) {
+    if (!private_submodules_->agc_manager.get()) {
+      private_submodules_->agc_manager.reset(new AgcManagerDirect(
+          public_submodules_->gain_control,
+          public_submodules_->gain_control_for_new_agc.get(),
+          constants_.agc_startup_min_volume));
     }
-    agc_manager_->Initialize();
-    agc_manager_->SetCaptureMuted(output_will_be_muted_);
+    private_submodules_->agc_manager->Initialize();
+    private_submodules_->agc_manager->SetCaptureMuted(
+        capture_.output_will_be_muted);
   }
 }
 
 void AudioProcessingImpl::InitializeTransient() {
-  if (transient_suppressor_enabled_) {
-    if (!transient_suppressor_.get()) {
-      transient_suppressor_.reset(new TransientSuppressor());
+  if (capture_.transient_suppressor_enabled) {
+    if (!public_submodules_->transient_suppressor.get()) {
+      public_submodules_->transient_suppressor.reset(new TransientSuppressor());
     }
-    transient_suppressor_->Initialize(
-        fwd_proc_format_.sample_rate_hz(), split_rate_,
-        shared_state_.api_format_.output_stream().num_channels());
+    public_submodules_->transient_suppressor->Initialize(
+        capture_nonlocked_.fwd_proc_format.sample_rate_hz(),
+        capture_nonlocked_.split_rate,
+        formats_.api_format.output_stream().num_channels());
   }
 }
 
 void AudioProcessingImpl::InitializeBeamformer() {
-  if (beamformer_enabled_) {
-    if (!beamformer_) {
-      beamformer_.reset(
-          new NonlinearBeamformer(array_geometry_, target_direction_));
+  if (capture_.beamformer_enabled) {
+    if (!private_submodules_->beamformer) {
+      private_submodules_->beamformer.reset(
+          new NonlinearBeamformer(constants_.array_geometry));
     }
-    beamformer_->Initialize(kChunkSizeMs, split_rate_);
+    private_submodules_->beamformer->Initialize(kChunkSizeMs,
+                                                capture_nonlocked_.split_rate);
   }
 }
 
 void AudioProcessingImpl::InitializeIntelligibility() {
-  if (intelligibility_enabled_) {
+  if (constants_.intelligibility_enabled) {
     IntelligibilityEnhancer::Config config;
-    config.sample_rate_hz = split_rate_;
-    config.num_capture_channels = capture_audio_->num_channels();
-    config.num_render_channels = render_audio_->num_channels();
-    intelligibility_enhancer_.reset(new IntelligibilityEnhancer(config));
+    config.sample_rate_hz = capture_nonlocked_.split_rate;
+    config.num_capture_channels = capture_.capture_audio->num_channels();
+    config.num_render_channels = render_.render_audio->num_channels();
+    public_submodules_->intelligibility_enhancer.reset(
+        new IntelligibilityEnhancer(config));
   }
 }
 
 void AudioProcessingImpl::MaybeUpdateHistograms() {
   static const int kMinDiffDelayMs = 60;
 
   if (echo_cancellation()->is_enabled()) {
     // Activate delay_jumps_ counters if we know echo_cancellation is runnning.
     // If a stream has echo we know that the echo_cancellation is in process.
-    if (stream_delay_jumps_ == -1 && echo_cancellation()->stream_has_echo()) {
-      stream_delay_jumps_ = 0;
+    if (capture_.stream_delay_jumps == -1 &&
+        echo_cancellation()->stream_has_echo()) {
+      capture_.stream_delay_jumps = 0;
     }
-    if (aec_system_delay_jumps_ == -1 &&
+    if (capture_.aec_system_delay_jumps == -1 &&
         echo_cancellation()->stream_has_echo()) {
-      aec_system_delay_jumps_ = 0;
+      capture_.aec_system_delay_jumps = 0;
     }
 
     // Detect a jump in platform reported system delay and log the difference.
-    const int diff_stream_delay_ms = stream_delay_ms_ - last_stream_delay_ms_;
-    if (diff_stream_delay_ms > kMinDiffDelayMs && last_stream_delay_ms_ != 0) {
+    const int diff_stream_delay_ms =
+        capture_nonlocked_.stream_delay_ms - capture_.last_stream_delay_ms;
+    if (diff_stream_delay_ms > kMinDiffDelayMs &&
+        capture_.last_stream_delay_ms != 0) {
       RTC_HISTOGRAM_COUNTS("WebRTC.Audio.PlatformReportedStreamDelayJump",
                            diff_stream_delay_ms, kMinDiffDelayMs, 1000, 100);
-      if (stream_delay_jumps_ == -1) {
-        stream_delay_jumps_ = 0;  // Activate counter if needed.
+      if (capture_.stream_delay_jumps == -1) {
+        capture_.stream_delay_jumps = 0;  // Activate counter if needed.
       }
-      stream_delay_jumps_++;
+      capture_.stream_delay_jumps++;
     }
-    last_stream_delay_ms_ = stream_delay_ms_;
+    capture_.last_stream_delay_ms = capture_nonlocked_.stream_delay_ms;
 
     // Detect a jump in AEC system delay and log the difference.
-    const int frames_per_ms = rtc::CheckedDivExact(split_rate_, 1000);
+    const int frames_per_ms =
+        rtc::CheckedDivExact(capture_nonlocked_.split_rate, 1000);
     const int aec_system_delay_ms =
         WebRtcAec_system_delay(echo_cancellation()->aec_core()) / frames_per_ms;
     const int diff_aec_system_delay_ms =
-        aec_system_delay_ms - last_aec_system_delay_ms_;
+        aec_system_delay_ms - capture_.last_aec_system_delay_ms;
     if (diff_aec_system_delay_ms > kMinDiffDelayMs &&
-        last_aec_system_delay_ms_ != 0) {
+        capture_.last_aec_system_delay_ms != 0) {
       RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecSystemDelayJump",
                            diff_aec_system_delay_ms, kMinDiffDelayMs, 1000,
                            100);
-      if (aec_system_delay_jumps_ == -1) {
-        aec_system_delay_jumps_ = 0;  // Activate counter if needed.
+      if (capture_.aec_system_delay_jumps == -1) {
+        capture_.aec_system_delay_jumps = 0;  // Activate counter if needed.
       }
-      aec_system_delay_jumps_++;
+      capture_.aec_system_delay_jumps++;
     }
-    last_aec_system_delay_ms_ = aec_system_delay_ms;
+    capture_.last_aec_system_delay_ms = aec_system_delay_ms;
   }
 }
 
 void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (stream_delay_jumps_ > -1) {
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+
+  if (capture_.stream_delay_jumps > -1) {
     RTC_HISTOGRAM_ENUMERATION(
         "WebRTC.Audio.NumOfPlatformReportedStreamDelayJumps",
-        stream_delay_jumps_, 51);
+        capture_.stream_delay_jumps, 51);
   }
-  stream_delay_jumps_ = -1;
-  last_stream_delay_ms_ = 0;
+  capture_.stream_delay_jumps = -1;
+  capture_.last_stream_delay_ms = 0;
 
-  if (aec_system_delay_jumps_ > -1) {
+  if (capture_.aec_system_delay_jumps > -1) {
     RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.NumOfAecSystemDelayJumps",
-                              aec_system_delay_jumps_, 51);
+                              capture_.aec_system_delay_jumps, 51);
   }
-  aec_system_delay_jumps_ = -1;
-  last_aec_system_delay_ms_ = 0;
+  capture_.aec_system_delay_jumps = -1;
+  capture_.last_aec_system_delay_ms = 0;
 }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-int AudioProcessingImpl::WriteMessageToDebugFile() {
-  int32_t size = event_msg_->ByteSize();
+int AudioProcessingImpl::WriteMessageToDebugFile(
+    FileWrapper* debug_file,
+    rtc::CriticalSection* crit_debug,
+    ApmDebugDumpThreadState* debug_state) {
+  int32_t size = debug_state->event_msg->ByteSize();
   if (size <= 0) {
     return kUnspecifiedError;
   }
 #if defined(WEBRTC_ARCH_BIG_ENDIAN)
 // TODO(ajm): Use little-endian "on the wire". For the moment, we can be
 //            pretty safe in assuming little-endian.
 #endif
 
-  if (!event_msg_->SerializeToString(&event_str_)) {
+  if (!debug_state->event_msg->SerializeToString(&debug_state->event_str)) {
     return kUnspecifiedError;
   }
 
-  // Write message preceded by its size.
-  if (!debug_file_->Write(&size, sizeof(int32_t))) {
-    return kFileError;
-  }
-  if (!debug_file_->Write(event_str_.data(), event_str_.length())) {
-    return kFileError;
+  {
+    // Ensure atomic writes of the message.
+    rtc::CritScope cs_capture(crit_debug);
+    // Write message preceded by its size.
+    if (!debug_file->Write(&size, sizeof(int32_t))) {
+      return kFileError;
+    }
+    if (!debug_file->Write(debug_state->event_str.data(),
+                           debug_state->event_str.length())) {
+      return kFileError;
+    }
   }
 
-  event_msg_->Clear();
+  debug_state->event_msg->Clear();
 
   return kNoError;
 }
 
 int AudioProcessingImpl::WriteInitMessage() {
-  event_msg_->set_type(audioproc::Event::INIT);
-  audioproc::Init* msg = event_msg_->mutable_init();
-  msg->set_sample_rate(
-      shared_state_.api_format_.input_stream().sample_rate_hz());
+  debug_dump_.capture.event_msg->set_type(audioproc::Event::INIT);
+  audioproc::Init* msg = debug_dump_.capture.event_msg->mutable_init();
+  msg->set_sample_rate(formats_.api_format.input_stream().sample_rate_hz());
   msg->set_num_input_channels(
-      shared_state_.api_format_.input_stream().num_channels());
+      formats_.api_format.input_stream().num_channels());
   msg->set_num_output_channels(
-      shared_state_.api_format_.output_stream().num_channels());
+      formats_.api_format.output_stream().num_channels());
   msg->set_num_reverse_channels(
-      shared_state_.api_format_.reverse_input_stream().num_channels());
+      formats_.api_format.reverse_input_stream().num_channels());
   msg->set_reverse_sample_rate(
-      shared_state_.api_format_.reverse_input_stream().sample_rate_hz());
+      formats_.api_format.reverse_input_stream().sample_rate_hz());
   msg->set_output_sample_rate(
-      shared_state_.api_format_.output_stream().sample_rate_hz());
-  // TODO(ekmeyerson): Add reverse output fields to event_msg_.
+      formats_.api_format.output_stream().sample_rate_hz());
+  // TODO(ekmeyerson): Add reverse output fields to
+  // debug_dump_.capture.event_msg.
 
-  RETURN_ON_ERR(WriteMessageToDebugFile());
+  RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                        &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 
 int AudioProcessingImpl::WriteConfigMessage(bool forced) {
   audioproc::Config config;
 
-  config.set_aec_enabled(echo_cancellation_->is_enabled());
+  config.set_aec_enabled(public_submodules_->echo_cancellation->is_enabled());
   config.set_aec_delay_agnostic_enabled(
-      echo_cancellation_->is_delay_agnostic_enabled());
+      public_submodules_->echo_cancellation->is_delay_agnostic_enabled());
   config.set_aec_drift_compensation_enabled(
-      echo_cancellation_->is_drift_compensation_enabled());
+      public_submodules_->echo_cancellation->is_drift_compensation_enabled());
   config.set_aec_extended_filter_enabled(
-      echo_cancellation_->is_extended_filter_enabled());
-  config.set_aec_suppression_level(
-      static_cast<int>(echo_cancellation_->suppression_level()));
+      public_submodules_->echo_cancellation->is_extended_filter_enabled());
+  config.set_aec_suppression_level(static_cast<int>(
+      public_submodules_->echo_cancellation->suppression_level()));
 
-  config.set_aecm_enabled(echo_control_mobile_->is_enabled());
+  config.set_aecm_enabled(
+      public_submodules_->echo_control_mobile->is_enabled());
   config.set_aecm_comfort_noise_enabled(
-      echo_control_mobile_->is_comfort_noise_enabled());
-  config.set_aecm_routing_mode(
-      static_cast<int>(echo_control_mobile_->routing_mode()));
+      public_submodules_->echo_control_mobile->is_comfort_noise_enabled());
+  config.set_aecm_routing_mode(static_cast<int>(
+      public_submodules_->echo_control_mobile->routing_mode()));
 
-  config.set_agc_enabled(gain_control_->is_enabled());
-  config.set_agc_mode(static_cast<int>(gain_control_->mode()));
-  config.set_agc_limiter_enabled(gain_control_->is_limiter_enabled());
-  config.set_noise_robust_agc_enabled(use_new_agc_);
+  config.set_agc_enabled(public_submodules_->gain_control->is_enabled());
+  config.set_agc_mode(
+      static_cast<int>(public_submodules_->gain_control->mode()));
+  config.set_agc_limiter_enabled(
+      public_submodules_->gain_control->is_limiter_enabled());
+  config.set_noise_robust_agc_enabled(constants_.use_new_agc);
 
-  config.set_hpf_enabled(high_pass_filter_->is_enabled());
+  config.set_hpf_enabled(public_submodules_->high_pass_filter->is_enabled());
 
-  config.set_ns_enabled(noise_suppression_->is_enabled());
-  config.set_ns_level(static_cast<int>(noise_suppression_->level()));
+  config.set_ns_enabled(public_submodules_->noise_suppression->is_enabled());
+  config.set_ns_level(
+      static_cast<int>(public_submodules_->noise_suppression->level()));
 
-  config.set_transient_suppression_enabled(transient_suppressor_enabled_);
+  config.set_transient_suppression_enabled(
+      capture_.transient_suppressor_enabled);
 
   std::string serialized_config = config.SerializeAsString();
-  if (!forced && last_serialized_config_ == serialized_config) {
+  if (!forced &&
+      debug_dump_.capture.last_serialized_config == serialized_config) {
     return kNoError;
   }
 
-  last_serialized_config_ = serialized_config;
+  debug_dump_.capture.last_serialized_config = serialized_config;
 
-  event_msg_->set_type(audioproc::Event::CONFIG);
-  event_msg_->mutable_config()->CopyFrom(config);
+  debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG);
+  debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
-  RETURN_ON_ERR(WriteMessageToDebugFile());
+  RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                        &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc