Removed the AudioProcessing dependency in EchoCancellerImpl

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.
  */
 
@@ skipping 39 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
 
+struct EchoCancellationImpl::StreamProperties {
+  StreamProperties() = delete;
+  StreamProperties(int sample_rate_hz,
+                   size_t num_reverse_channels,
+                   size_t num_output_channels,
+                   size_t num_proc_channels)
+      : sample_rate_hz(sample_rate_hz),
+        num_reverse_channels(num_reverse_channels),
+        num_output_channels(num_output_channels),
+        num_proc_channels(num_proc_channels) {}
+
+  const int sample_rate_hz;
+  const size_t num_reverse_channels;
+  const size_t num_output_channels;
+  const size_t num_proc_channels;
+};
+
 class EchoCancellationImpl::Canceller {
  public:
   Canceller() {
     state_ = WebRtcAec_Create();
     RTC_DCHECK(state_);
   }
 
   ~Canceller() {
     RTC_CHECK(state_);
     WebRtcAec_Free(state_);
   }
 
   void* state() { return state_; }
 
   void Initialize(int sample_rate_hz) {
     // 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.
     const int error = WebRtcAec_Init(state_, sample_rate_hz, 48000);
     RTC_DCHECK_EQ(0, error);
   }
 
  private:
   void* state_;
 };
 
-EchoCancellationImpl::EchoCancellationImpl(const AudioProcessing* apm,
-                                           rtc::CriticalSection* crit_render,
+EchoCancellationImpl::EchoCancellationImpl(rtc::CriticalSection* crit_render,
                                            rtc::CriticalSection* crit_capture)
-    : apm_(apm),
-      crit_render_(crit_render),
+    : 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),
       aec3_enabled_(false),
       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 (!enabled_) {
     return AudioProcessing::kNoError;
   }
 
+  RTC_DCHECK(stream_properties_);
   RTC_DCHECK_GE(160u, audio->num_frames_per_band());
-  RTC_DCHECK_EQ(audio->num_channels(), apm_->num_reverse_channels());
-  RTC_DCHECK_GE(cancellers_.size(),
-                apm_->num_output_channels() * audio->num_channels());
+  RTC_DCHECK_EQ(audio->num_channels(),
+                stream_properties_->num_reverse_channels);
+  RTC_DCHECK_GE(cancellers_.size(), stream_properties_->num_output_channels *
+                                        audio->num_channels());
 
   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 (size_t i = 0; i < apm_->num_output_channels(); i++) {
+  for (size_t i = 0; i < stream_properties_->num_output_channels; i++) {
     for (size_t j = 0; j < audio->num_channels(); j++) {
       // Retrieve any error code produced by the buffering of the farend
       // signal.
       err = WebRtcAec_GetBufferFarendError(
           cancellers_[handle_index++]->state(),
           audio->split_bands_const_f(j)[kBand0To8kHz],
           audio->num_frames_per_band());
 
       if (err != AudioProcessing::kNoError) {
         return MapError(err);  // TODO(ajm): warning possible?
@@ skipping 20 matching lines @@
 }
 
 // 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 (!enabled_) {
     return;
   }
 
+  RTC_DCHECK(stream_properties_);
   while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
     size_t handle_index = 0;
     size_t buffer_index = 0;
     const size_t num_frames_per_band =
         capture_queue_buffer_.size() /
-        (apm_->num_output_channels() * apm_->num_reverse_channels());
-    for (size_t i = 0; i < apm_->num_output_channels(); i++) {
-      for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
+        (stream_properties_->num_output_channels *
+         stream_properties_->num_reverse_channels);
+    for (size_t i = 0; i < stream_properties_->num_output_channels; i++) {
+      for (size_t j = 0; j < stream_properties_->num_reverse_channels; j++) {
         WebRtcAec_BufferFarend(cancellers_[handle_index++]->state(),
                                &capture_queue_buffer_[buffer_index],
                                num_frames_per_band);
 
         buffer_index += num_frames_per_band;
       }
     }
   }
 }
 
-int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio,
+                                              int stream_delay_ms) {
   rtc::CritScope cs_capture(crit_capture_);
   if (!enabled_) {
     return AudioProcessing::kNoError;
   }
 
-  if (!apm_->was_stream_delay_set()) {
-    return AudioProcessing::kStreamParameterNotSetError;
-  }
-
   if (drift_compensation_enabled_ && !was_stream_drift_set_) {
     return AudioProcessing::kStreamParameterNotSetError;
   }
 
+  RTC_DCHECK(stream_properties_);
   RTC_DCHECK_GE(160u, audio->num_frames_per_band());
-  RTC_DCHECK_EQ(audio->num_channels(), apm_->num_proc_channels());
+  RTC_DCHECK_EQ(audio->num_channels(), stream_properties_->num_proc_channels);
 
   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 (size_t i = 0; i < audio->num_channels(); i++) {
-    for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
-      err = WebRtcAec_Process(cancellers_[handle_index]->state(),
-                              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_);
+    for (size_t j = 0; j < stream_properties_->num_reverse_channels; j++) {
+      err = WebRtcAec_Process(
+          cancellers_[handle_index]->state(), audio->split_bands_const_f(i),
+          audio->num_bands(), audio->split_bands_f(i),
+          audio->num_frames_per_band(), stream_delay_ms, stream_drift_samples_);
 
       if (err != AudioProcessing::kNoError) {
         err = MapError(err);
         // TODO(ajm): Figure out how to return warnings properly.
         if (err != AudioProcessing::kBadStreamParameterWarning) {
           return err;
         }
       }
 
       int status = 0;
@@ skipping 12 matching lines @@
   }
 
   was_stream_drift_set_ = false;
   return AudioProcessing::kNoError;
 }
 
 int EchoCancellationImpl::Enable(bool enable) {
   // 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 AudioProcessing::kBadParameterError;
-  }
 
   if (enable && !enabled_) {
     enabled_ = enable;  // Must be set before Initialize() is called.
-    Initialize();
+
+    // TODO(peah): Simplify once the Enable function has been removed from
+    // the public APM API.
+    RTC_DCHECK(stream_properties_);
+    Initialize(stream_properties_->sample_rate_hz,
+               stream_properties_->num_reverse_channels,
+               stream_properties_->num_output_channels,
+               stream_properties_->num_proc_channels);
   } else {
     enabled_ = enable;
   }
   return AudioProcessing::kNoError;
 }
 
 bool EchoCancellationImpl::is_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return enabled_;
 }
@@ skipping 160 matching lines @@
 }
 
 struct AecCore* EchoCancellationImpl::aec_core() const {
   rtc::CritScope cs(crit_capture_);
   if (!enabled_) {
     return NULL;
   }
   return WebRtcAec_aec_core(cancellers_[0]->state());
 }
 
-void EchoCancellationImpl::Initialize() {
+void EchoCancellationImpl::Initialize(int sample_rate_hz,
+                                      size_t num_reverse_channels,
+                                      size_t num_output_channels,
+                                      size_t num_proc_channels) {
   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);
+
+  stream_properties_.reset(
+      new StreamProperties(sample_rate_hz, num_reverse_channels,
+                           num_output_channels, num_proc_channels));
+
   if (!enabled_) {
     return;
   }
 
-  if (num_handles_required() > cancellers_.size()) {
+  if (NumCancellersRequired() > cancellers_.size()) {
     const size_t cancellers_old_size = cancellers_.size();
-    cancellers_.resize(num_handles_required());
+    cancellers_.resize(NumCancellersRequired());
 
     for (size_t i = cancellers_old_size; i < cancellers_.size(); ++i) {
       cancellers_[i].reset(new Canceller());
     }
   }
 
-  const int sample_rate_hz = apm_->proc_sample_rate_hz();
   for (auto& canceller : cancellers_) {
     canceller->Initialize(sample_rate_hz);
   }
 
   Configure();
 
   AllocateRenderQueue();
 }
 
 int EchoCancellationImpl::GetSystemDelayInSamples() const {
   rtc::CritScope cs(crit_capture_);
   RTC_DCHECK(enabled_);
   // Report the delay for the first AEC component.
   return WebRtcAec_system_delay(
       WebRtcAec_aec_core(cancellers_[0]->state()));
 }
 
 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());
+      kMaxAllowedValuesOfSamplesPerFrame * NumCancellersRequired());
 
   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_);
@@ skipping 38 matching lines @@
     WebRtcAec_enable_aec3(WebRtcAec_aec_core(canceller->state()),
                           aec3_enabled_ ? 1 : 0);
     const int handle_error = WebRtcAec_set_config(canceller->state(), config);
     if (handle_error != AudioProcessing::kNoError) {
       error = AudioProcessing::kNoError;
     }
   }
   return error;
 }
 
-size_t 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();
+size_t EchoCancellationImpl::NumCancellersRequired() const {
+  RTC_DCHECK(stream_properties_);
+  return stream_properties_->num_output_channels *
+         stream_properties_->num_reverse_channels;
 }
 
 }  // namespace webrtc