Lock scheme #8: Introduced the new locking scheme

webrtc/modules/audio_processing/echo_cancellation_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.
  */
 
 #include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
 
 #include <assert.h>
 #include <string.h>
 
 extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 }
 #include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
 typedef void Handle;
 
 namespace {
 int16_t MapSetting(EchoCancellation::SuppressionLevel level) {
   switch (level) {
     case EchoCancellation::kLowSuppression:
       return kAecNlpConservative;
@@ skipping 24 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
 
 EchoCancellationImpl::EchoCancellationImpl(const AudioProcessing* apm,
-                                           CriticalSectionWrapper* crit)
+                                           rtc::CriticalSection* crit_render,
+                                           rtc::CriticalSection* crit_capture)
     : ProcessingComponent(),
       apm_(apm),
-      crit_(crit),
+      crit_render_(crit_render),
+      crit_capture_(crit_capture),
       drift_compensation_enabled_(false),
       metrics_enabled_(false),
       suppression_level_(kModerateSuppression),
       stream_drift_samples_(0),
       was_stream_drift_set_(false),
       stream_has_echo_(false),
       delay_logging_enabled_(false),
       extended_filter_enabled_(false),
       delay_agnostic_enabled_(false),
-      render_queue_element_max_size_(0) {}
+      render_queue_element_max_size_(0) {
+  RTC_DCHECK(apm);
+  RTC_DCHECK(crit_render);
+  RTC_DCHECK(crit_capture);
+}
 
 EchoCancellationImpl::~EchoCancellationImpl() {}
 
 int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) {
+  rtc::CritScope cs_render(crit_render_);
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == apm_->num_reverse_channels());
 
-  int err = apm_->kNoError;
+  int err = AudioProcessing::kNoError;
 
   // The ordering convention must be followed to pass to the correct AEC.
   size_t handle_index = 0;
   render_queue_buffer_.clear();
   for (int i = 0; i < apm_->num_output_channels(); i++) {
     for (int j = 0; j < audio->num_channels(); j++) {
       Handle* my_handle = static_cast<Handle*>(handle(handle_index));
       // Retrieve any error code produced by the buffering of the farend
       // signal
       err = WebRtcAec_GetBufferFarendError(
           my_handle, audio->split_bands_const_f(j)[kBand0To8kHz],
           audio->num_frames_per_band());
 
-      if (err != apm_->kNoError) {
+      if (err != AudioProcessing::kNoError) {
         return MapError(err);  // TODO(ajm): warning possible?
       }
 
       // Buffer the samples in the render queue.
       render_queue_buffer_.insert(render_queue_buffer_.end(),
                                   audio->split_bands_const_f(j)[kBand0To8kHz],
                                   (audio->split_bands_const_f(j)[kBand0To8kHz] +
                                    audio->num_frames_per_band()));
     }
   }
 
   // Insert the samples into the queue.
   if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
+    // The data queue is full and needs to be emptied.
     ReadQueuedRenderData();
 
     // Retry the insert (should always work).
     RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
   }
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 // Read chunks of data that were received and queued on the render side from
 // a queue. All the data chunks are buffered into the farend signal of the AEC.
 void EchoCancellationImpl::ReadQueuedRenderData() {
+  rtc::CritScope cs_capture(crit_capture_);
   if (!is_component_enabled()) {
     return;
   }
 
   while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
     size_t handle_index = 0;
     int buffer_index = 0;
     const int num_frames_per_band =
         capture_queue_buffer_.size() /
         (apm_->num_output_channels() * apm_->num_reverse_channels());
     for (int i = 0; i < apm_->num_output_channels(); i++) {
       for (int j = 0; j < apm_->num_reverse_channels(); j++) {
         Handle* my_handle = static_cast<Handle*>(handle(handle_index));
         WebRtcAec_BufferFarend(my_handle, &capture_queue_buffer_[buffer_index],
                                num_frames_per_band);
 
         buffer_index += num_frames_per_band;
         handle_index++;
       }
     }
   }
 }
 
 int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  rtc::CritScope cs_capture(crit_capture_);
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   if (!apm_->was_stream_delay_set()) {
-    return apm_->kStreamParameterNotSetError;
+    return AudioProcessing::kStreamParameterNotSetError;
   }
 
   if (drift_compensation_enabled_ && !was_stream_drift_set_) {
-    return apm_->kStreamParameterNotSetError;
+    return AudioProcessing::kStreamParameterNotSetError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == apm_->num_output_channels());
 
-  int err = apm_->kNoError;
+  int err = AudioProcessing::kNoError;
 
   // The ordering convention must be followed to pass to the correct AEC.
   size_t handle_index = 0;
   stream_has_echo_ = false;
   for (int i = 0; i < audio->num_channels(); i++) {
     for (int j = 0; j < apm_->num_reverse_channels(); j++) {
       Handle* my_handle = handle(handle_index);
-      err = WebRtcAec_Process(
-          my_handle,
-          audio->split_bands_const_f(i),
-          audio->num_bands(),
-          audio->split_bands_f(i),
-          audio->num_frames_per_band(),
-          apm_->stream_delay_ms(),
-          stream_drift_samples_);
+      err = WebRtcAec_Process(my_handle, audio->split_bands_const_f(i),
+                              audio->num_bands(), audio->split_bands_f(i),
+                              audio->num_frames_per_band(),
+                              apm_->stream_delay_ms(), stream_drift_samples_);
 
-      if (err != apm_->kNoError) {
+      if (err != AudioProcessing::kNoError) {
         err = MapError(err);
         // TODO(ajm): Figure out how to return warnings properly.
-        if (err != apm_->kBadStreamParameterWarning) {
+        if (err != AudioProcessing::kBadStreamParameterWarning) {
           return err;
         }
       }
 
       int status = 0;
       err = WebRtcAec_get_echo_status(my_handle, &status);
-      if (err != apm_->kNoError) {
+      if (err != AudioProcessing::kNoError) {
         return MapError(err);
       }
 
       if (status == 1) {
         stream_has_echo_ = true;
       }
 
       handle_index++;
     }
   }
 
   was_stream_drift_set_ = false;
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int EchoCancellationImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   // Ensure AEC and AECM are not both enabled.
+  // The is_enabled call is safe from a deadlock perspective
+  // as both locks are already held in the correct order.
   if (enable && apm_->echo_control_mobile()->is_enabled()) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   return EnableComponent(enable);
 }
 
 bool EchoCancellationImpl::is_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return is_component_enabled();
 }
 
 int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (MapSetting(level) == -1) {
-    return apm_->kBadParameterError;
+  {
+    if (MapSetting(level) == -1) {
+      return AudioProcessing::kBadParameterError;
+    }
+    rtc::CritScope cs(crit_capture_);
+    suppression_level_ = level;
   }
-
-  suppression_level_ = level;
   return Configure();
 }
 
 EchoCancellation::SuppressionLevel EchoCancellationImpl::suppression_level()
     const {
+  rtc::CritScope cs(crit_capture_);
   return suppression_level_;
 }
 
 int EchoCancellationImpl::enable_drift_compensation(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  drift_compensation_enabled_ = enable;
+  {
+    rtc::CritScope cs(crit_capture_);
+    drift_compensation_enabled_ = enable;
+  }
   return Configure();
 }
 
 bool EchoCancellationImpl::is_drift_compensation_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return drift_compensation_enabled_;
 }
 
 void EchoCancellationImpl::set_stream_drift_samples(int drift) {
+  rtc::CritScope cs(crit_capture_);
   was_stream_drift_set_ = true;
   stream_drift_samples_ = drift;
 }
 
 int EchoCancellationImpl::stream_drift_samples() const {
+  rtc::CritScope cs(crit_capture_);
   return stream_drift_samples_;
 }
 
 int EchoCancellationImpl::enable_metrics(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  metrics_enabled_ = enable;
+  {
+    rtc::CritScope cs(crit_capture_);
+    metrics_enabled_ = enable;
+  }
   return Configure();
 }
 
 bool EchoCancellationImpl::are_metrics_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return metrics_enabled_;
 }
 
 // TODO(ajm): we currently just use the metrics from the first AEC. Think more
 //            aboue the best way to extend this to multi-channel.
 int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (metrics == NULL) {
-    return apm_->kNullPointerError;
+    return AudioProcessing::kNullPointerError;
   }
 
   if (!is_component_enabled() || !metrics_enabled_) {
-    return apm_->kNotEnabledError;
+    return AudioProcessing::kNotEnabledError;
   }
 
   AecMetrics my_metrics;
   memset(&my_metrics, 0, sizeof(my_metrics));
   memset(metrics, 0, sizeof(Metrics));
 
   Handle* my_handle = static_cast<Handle*>(handle(0));
   int err = WebRtcAec_GetMetrics(my_handle, &my_metrics);
-  if (err != apm_->kNoError) {
+  if (err != AudioProcessing::kNoError) {
     return MapError(err);
   }
 
   metrics->residual_echo_return_loss.instant = my_metrics.rerl.instant;
   metrics->residual_echo_return_loss.average = my_metrics.rerl.average;
   metrics->residual_echo_return_loss.maximum = my_metrics.rerl.max;
   metrics->residual_echo_return_loss.minimum = my_metrics.rerl.min;
 
   metrics->echo_return_loss.instant = my_metrics.erl.instant;
   metrics->echo_return_loss.average = my_metrics.erl.average;
   metrics->echo_return_loss.maximum = my_metrics.erl.max;
   metrics->echo_return_loss.minimum = my_metrics.erl.min;
 
   metrics->echo_return_loss_enhancement.instant = my_metrics.erle.instant;
   metrics->echo_return_loss_enhancement.average = my_metrics.erle.average;
   metrics->echo_return_loss_enhancement.maximum = my_metrics.erle.max;
   metrics->echo_return_loss_enhancement.minimum = my_metrics.erle.min;
 
   metrics->a_nlp.instant = my_metrics.aNlp.instant;
   metrics->a_nlp.average = my_metrics.aNlp.average;
   metrics->a_nlp.maximum = my_metrics.aNlp.max;
   metrics->a_nlp.minimum = my_metrics.aNlp.min;
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 bool EchoCancellationImpl::stream_has_echo() const {
+  rtc::CritScope cs(crit_capture_);
   return stream_has_echo_;
 }
 
 int EchoCancellationImpl::enable_delay_logging(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  delay_logging_enabled_ = enable;
+  {
+    rtc::CritScope cs(crit_capture_);
+    delay_logging_enabled_ = enable;
+  }
   return Configure();
 }
 
 bool EchoCancellationImpl::is_delay_logging_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return delay_logging_enabled_;
 }
 
 bool EchoCancellationImpl::is_delay_agnostic_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return delay_agnostic_enabled_;
 }
 
 bool EchoCancellationImpl::is_extended_filter_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return extended_filter_enabled_;
 }
 
 // TODO(bjornv): How should we handle the multi-channel case?
 int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
+  rtc::CritScope cs(crit_capture_);
   float fraction_poor_delays = 0;
   return GetDelayMetrics(median, std, &fraction_poor_delays);
 }
 
 int EchoCancellationImpl::GetDelayMetrics(int* median, int* std,
                                           float* fraction_poor_delays) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (median == NULL) {
-    return apm_->kNullPointerError;
+    return AudioProcessing::kNullPointerError;
   }
   if (std == NULL) {
-    return apm_->kNullPointerError;
+    return AudioProcessing::kNullPointerError;
   }
 
   if (!is_component_enabled() || !delay_logging_enabled_) {
-    return apm_->kNotEnabledError;
+    return AudioProcessing::kNotEnabledError;
   }
 
   Handle* my_handle = static_cast<Handle*>(handle(0));
   const int err =
       WebRtcAec_GetDelayMetrics(my_handle, median, std, fraction_poor_delays);
-  if (err != apm_->kNoError) {
+  if (err != AudioProcessing::kNoError) {
     return MapError(err);
   }
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 struct AecCore* EchoCancellationImpl::aec_core() const {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (!is_component_enabled()) {
     return NULL;
   }
   Handle* my_handle = static_cast<Handle*>(handle(0));
   return WebRtcAec_aec_core(my_handle);
 }
 
 int EchoCancellationImpl::Initialize() {
   int err = ProcessingComponent::Initialize();
-  if (err != apm_->kNoError || !is_component_enabled()) {
-    return err;
+  {
+    rtc::CritScope cs(crit_capture_);
+    if (err != AudioProcessing::kNoError || !is_component_enabled()) {
+      return err;
+    }
   }
 
   AllocateRenderQueue();
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 void EchoCancellationImpl::AllocateRenderQueue() {
   const size_t new_render_queue_element_max_size = std::max<size_t>(
       static_cast<size_t>(1),
       kMaxAllowedValuesOfSamplesPerFrame * num_handles_required());
 
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
+
   // Reallocate the queue if the queue item size is too small to fit the
   // data to put in the queue.
   if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
     render_queue_element_max_size_ = new_render_queue_element_max_size;
 
     std::vector<float> template_queue_element(render_queue_element_max_size_);
 
     render_signal_queue_.reset(
         new SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>(
             kMaxNumFramesToBuffer, template_queue_element,
             RenderQueueItemVerifier<float>(render_queue_element_max_size_)));
 
     render_queue_buffer_.resize(render_queue_element_max_size_);
     capture_queue_buffer_.resize(render_queue_element_max_size_);
   } else {
     render_signal_queue_->Clear();
   }
 }
 
 void EchoCancellationImpl::SetExtraOptions(const Config& config) {
-  extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
-  delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
+  {
+    rtc::CritScope cs(crit_capture_);
+    extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
+    delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
+  }
   Configure();
 }
 
 void* EchoCancellationImpl::CreateHandle() const {
   return WebRtcAec_Create();
 }
 
 void EchoCancellationImpl::DestroyHandle(void* handle) const {
   assert(handle != NULL);
   WebRtcAec_Free(static_cast<Handle*>(handle));
 }
 
 int EchoCancellationImpl::InitializeHandle(void* handle) const {
+  // Not locked as it only relies on APM public API which is threadsafe.
+
   assert(handle != NULL);
   // TODO(ajm): Drift compensation is disabled in practice. If restored, it
   // should be managed internally and not depend on the hardware sample rate.
   // For now, just hardcode a 48 kHz value.
   return WebRtcAec_Init(static_cast<Handle*>(handle),
-                       apm_->proc_sample_rate_hz(),
-                       48000);
+                        apm_->proc_sample_rate_hz(), 48000);
 }
 
 int EchoCancellationImpl::ConfigureHandle(void* handle) const {
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   assert(handle != NULL);
   AecConfig config;
   config.metricsMode = metrics_enabled_;
   config.nlpMode = MapSetting(suppression_level_);
   config.skewMode = drift_compensation_enabled_;
   config.delay_logging = delay_logging_enabled_;
-
   WebRtcAec_enable_extended_filter(
       WebRtcAec_aec_core(static_cast<Handle*>(handle)),
       extended_filter_enabled_ ? 1 : 0);
   WebRtcAec_enable_delay_agnostic(
       WebRtcAec_aec_core(static_cast<Handle*>(handle)),
       delay_agnostic_enabled_ ? 1 : 0);
   return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
 }
 
 int EchoCancellationImpl::num_handles_required() const {
+  // Not locked as it only relies on APM public API which is threadsafe.
   return apm_->num_output_channels() *
          apm_->num_reverse_channels();
 }
 
 int EchoCancellationImpl::GetHandleError(void* handle) const {
+  // Not locked as it does not rely on anything in the state.
   assert(handle != NULL);
   return AudioProcessing::kUnspecifiedError;
 }
 }  // namespace webrtc