Pull out the PostFilter to its own NonlinearBeamformer API

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 109 matching lines @@
   std::unique_ptr<VoiceDetectionImpl> voice_detection;
   std::unique_ptr<GainControlForExperimentalAgc>
       gain_control_for_experimental_agc;
 
   // Accessed internally from both render and capture.
   std::unique_ptr<TransientSuppressor> transient_suppressor;
   std::unique_ptr<IntelligibilityEnhancer> intelligibility_enhancer;
 };
 
 struct AudioProcessingImpl::ApmPrivateSubmodules {
-  explicit ApmPrivateSubmodules(Beamformer<float>* beamformer)
+  explicit ApmPrivateSubmodules(NonlinearBeamformer* beamformer)
       : beamformer(beamformer) {}
   // Accessed internally from capture or during initialization
-  std::unique_ptr<Beamformer<float>> beamformer;
+  std::unique_ptr<NonlinearBeamformer> beamformer;
   std::unique_ptr<AgcManagerDirect> agc_manager;
 };
 
 AudioProcessing* AudioProcessing::Create() {
   Config config;
   return Create(config, nullptr);
 }
 
 AudioProcessing* AudioProcessing::Create(const Config& config) {
   return Create(config, nullptr);
 }
 
 AudioProcessing* AudioProcessing::Create(const Config& config,
-                                         Beamformer<float>* beamformer) {
+                                         NonlinearBeamformer* beamformer) {
   AudioProcessingImpl* apm = new AudioProcessingImpl(config, beamformer);
   if (apm->Initialize() != kNoError) {
     delete apm;
     apm = nullptr;
   }
 
   return apm;
 }
 
 AudioProcessingImpl::AudioProcessingImpl(const Config& config)
     : AudioProcessingImpl(config, nullptr) {}
 
 AudioProcessingImpl::AudioProcessingImpl(const Config& config,
-                                         Beamformer<float>* beamformer)
+                                         NonlinearBeamformer* beamformer)
     : public_submodules_(new ApmPublicSubmodules()),
       private_submodules_(new ApmPrivateSubmodules(beamformer)),
       constants_(config.Get<ExperimentalAgc>().startup_min_volume,
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
                  false,
 #else
                  config.Get<ExperimentalAgc>().enabled,
 #endif
                  config.Get<Intelligibility>().enabled),
 
@@ skipping 501 matching lines @@
     private_submodules_->agc_manager->AnalyzePreProcess(
         ca->channels()[0], ca->num_channels(),
         capture_nonlocked_.fwd_proc_format.num_frames());
   }
 
   if (fwd_analysis_needed()) {
     ca->SplitIntoFrequencyBands();
   }
 
   if (capture_nonlocked_.beamformer_enabled) {
-    private_submodules_->beamformer->ProcessChunk(*ca->split_data_f(),
-                                                  ca->split_data_f());
+    private_submodules_->beamformer->AnalyzeChunk(*ca->split_data_f());

[peah-webrtc, 2016/06/08 12:04:55]

What has changed here, is that now all the downmixing is done by the
set_num_channels call, in contrast to that before the downmixing was done inside
the beamformer, but the set_num_channels call was needed to correct for the new
number of channels? Is that correct?

[aluebs-webrtc, 2016/06/09 02:11:46]

What do you mean with downmixing? Before there was a frequency domain fixed
delay and sum beamformer and then the number of channels was corrected in the
AudioBuffer. Now all channels except the first one are discarded.

[peah-webrtc, 2016/06/09 09:36:30]

By downmixing I mean going from 2 to 1 channel, so discarding the leftmost
channel definitely counts as that.

[aluebs-webrtc, 2016/06/09 19:21:42]

Personally I find "downmixing" to have a blending implication. But acknowledged.

+    // Discards all channels by the leftmost one.
     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);
 
   // Ensure that the stream delay was set before the call to the
   // AEC ProcessCaptureAudio function.
   if (public_submodules_->echo_cancellation->is_enabled() &&
@@ skipping 20 matching lines @@
   // Ensure that the stream delay was set before the call to the
   // AECM ProcessCaptureAudio function.
   if (public_submodules_->echo_control_mobile->is_enabled() &&
       !was_stream_delay_set()) {
     return AudioProcessing::kStreamParameterNotSetError;
   }
 
   RETURN_ON_ERR(public_submodules_->echo_control_mobile->ProcessCaptureAudio(
       ca, stream_delay_ms()));
 
+  if (capture_nonlocked_.beamformer_enabled) {
+    private_submodules_->beamformer->PostFilter(ca->split_data_f());
+  }
+
   public_submodules_->voice_detection->ProcessCaptureAudio(ca);
 
   if (constants_.use_experimental_agc &&
       public_submodules_->gain_control->is_enabled() &&
       (!capture_nonlocked_.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);
   }
@@ skipping 726 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