Removed the AudioProcessing dependency in EchoCancellerImpl

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 150 matching lines @@
 #endif
                config.Get<Beamforming>().array_geometry,
                config.Get<Beamforming>().target_direction),
       capture_nonlocked_(config.Get<Beamforming>().enabled)
 {
   {
     rtc::CritScope cs_render(&crit_render_);
     rtc::CritScope cs_capture(&crit_capture_);
 
     public_submodules_->echo_cancellation.reset(
-        new EchoCancellationImpl(this, &crit_render_, &crit_capture_));
+        new EchoCancellationImpl(&crit_render_, &crit_capture_));
     public_submodules_->echo_control_mobile.reset(
         new EchoControlMobileImpl(&crit_render_, &crit_capture_));
     public_submodules_->gain_control.reset(
         new GainControlImpl(&crit_capture_, &crit_capture_));
     public_submodules_->high_pass_filter.reset(
         new HighPassFilterImpl(&crit_capture_));
     public_submodules_->level_estimator.reset(
         new LevelEstimatorImpl(&crit_capture_));
     public_submodules_->noise_suppression.reset(
         new NoiseSuppressionImpl(&crit_capture_));
@@ skipping 469 matching lines @@
     RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                           &debug_dump_.num_bytes_left_for_log_,
                                           &crit_debug_, &debug_dump_.capture));
   }
 #endif
 
   return kNoError;
 }
 
 int AudioProcessingImpl::ProcessStreamLocked() {
+  // Ensure that not both the AEC and AECM are active at the same time.
+  // TODO(peah): Simplify once the public API Enable functions for these
+  // are moved to APM.
+  RTC_DCHECK(!(public_submodules_->echo_cancellation->is_enabled() &&
+               public_submodules_->echo_control_mobile->is_enabled()));
+
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   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(capture_.key_pressed);
   }
 #endif
@@ skipping 16 matching lines @@
 
   if (capture_nonlocked_.beamformer_enabled) {
     private_submodules_->beamformer->ProcessChunk(*ca->split_data_f(),
                                                   ca->split_data_f());
     ca->set_num_channels(1);
   }
 
   public_submodules_->high_pass_filter->ProcessCaptureAudio(ca);
   RETURN_ON_ERR(public_submodules_->gain_control->AnalyzeCaptureAudio(ca));
   public_submodules_->noise_suppression->AnalyzeCaptureAudio(ca);
-  RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessCaptureAudio(ca));
+
+  // Ensure that the stream delay was set before the call to the
+  // AEC ProcessCaptureAudio function.
+  if (public_submodules_->echo_cancellation->is_enabled() &&
+      !was_stream_delay_set()) {
+    return AudioProcessing::kStreamParameterNotSetError;
+  }
+
+  RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessCaptureAudio(
+      ca, stream_delay_ms()));
 
   if (public_submodules_->echo_control_mobile->is_enabled() &&
       public_submodules_->noise_suppression->is_enabled()) {
     ca->CopyLowPassToReference();
   }
   public_submodules_->noise_suppression->ProcessCaptureAudio(ca);
   if (constants_.intelligibility_enabled) {
     RTC_DCHECK(public_submodules_->noise_suppression->is_enabled());
     public_submodules_->intelligibility_enhancer->SetCaptureNoiseEstimate(
         public_submodules_->noise_suppression->NoiseEstimate());
@@ skipping 512 matching lines @@
   public_submodules_->high_pass_filter->Initialize(num_proc_channels(),
                                                    proc_sample_rate_hz());
 }
 
 void AudioProcessingImpl::InitializeNoiseSuppression() {
   public_submodules_->noise_suppression->Initialize(num_proc_channels(),
                                                     proc_sample_rate_hz());
 }
 
 void AudioProcessingImpl::InitializeEchoCanceller() {
-  public_submodules_->echo_cancellation->Initialize();
+  public_submodules_->echo_cancellation->Initialize(
+      proc_sample_rate_hz(), num_reverse_channels(), num_output_channels(),
+      num_proc_channels());
 }
 
 void AudioProcessingImpl::InitializeGainController() {
   public_submodules_->gain_control->Initialize(num_proc_channels(),
                                                proc_sample_rate_hz());
 }
 
 void AudioProcessingImpl::InitializeEchoControlMobile() {
   public_submodules_->echo_control_mobile->Initialize(
       proc_sample_rate_hz(), num_reverse_channels(), num_output_channels());
@@ skipping 201 matching lines @@
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &debug_dump_.num_bytes_left_for_log_,
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc