Moving the residual echo detector outside of band-scheme in APM

webrtc/modules/audio_processing/residual_echo_detector.cc

 /*
  *  Copyright (c) 2016 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/residual_echo_detector.h"
 
 #include <algorithm>
 #include <numeric>
 
+#include "webrtc/base/atomicops.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 namespace {
 
 float Power(rtc::ArrayView<const float> input) {
-  return std::inner_product(input.begin(), input.end(), input.begin(), 0.f);
+  if (input.size() == 0) {
+    return 0.f;
+  }
+  return std::inner_product(input.begin(), input.end(), input.begin(), 0.f) /
+         input.size();
 }
 
 constexpr size_t kLookbackFrames = 650;
 // TODO(ivoc): Verify the size of this buffer.
 constexpr size_t kRenderBufferSize = 30;
 constexpr float kAlpha = 0.001f;
 // 10 seconds of data, updated every 10 ms.
 constexpr size_t kAggregationBufferSize = 10 * 100;
 
 }  // namespace
 
 namespace webrtc {
 
+int ResidualEchoDetector::instance_count_ = 0;
+
 ResidualEchoDetector::ResidualEchoDetector()
-    : render_buffer_(kRenderBufferSize),
+    : data_dumper_(
+          new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
+      render_buffer_(kRenderBufferSize),
       render_power_(kLookbackFrames),
       render_power_mean_(kLookbackFrames),
       render_power_std_dev_(kLookbackFrames),
       covariances_(kLookbackFrames),
       recent_likelihood_max_(kAggregationBufferSize) {}
 
 ResidualEchoDetector::~ResidualEchoDetector() = default;
 
 void ResidualEchoDetector::AnalyzeRenderAudio(
     rtc::ArrayView<const float> render_audio) {
+  // Dump debug data assuming 48 kHz sample rate (if this assumption is not
+  // valid the dumped audio will need to be converted offline accordingly).
+  data_dumper_->DumpWav("ed_render", render_audio.size(), render_audio.data(),
+                        48000, 1);
+
   if (render_buffer_.Size() == 0) {
     frames_since_zero_buffer_size_ = 0;
   } else if (frames_since_zero_buffer_size_ >= kRenderBufferSize) {
     // This can happen in a few cases: at the start of a call, due to a glitch
     // or due to clock drift. The excess capture value will be ignored.
     // TODO(ivoc): Include how often this happens in APM stats.
     render_buffer_.Pop();
     frames_since_zero_buffer_size_ = 0;
   }
   ++frames_since_zero_buffer_size_;
   float power = Power(render_audio);
   render_buffer_.Push(power);
 }
 
 void ResidualEchoDetector::AnalyzeCaptureAudio(
     rtc::ArrayView<const float> capture_audio) {
+  // Dump debug data assuming 48 kHz sample rate (if this assumption is not
+  // valid the dumped audio will need to be converted offline accordingly).
+  data_dumper_->DumpWav("ed_capture", capture_audio.size(),
+                        capture_audio.data(), 48000, 1);
+
   if (first_process_call_) {
     // On the first process call (so the start of a call), we must flush the
     // render buffer, otherwise the render data will be delayed.
     render_buffer_.Clear();
     first_process_call_ = false;
   }
 
   // Get the next render value.
   const rtc::Optional<float> buffered_render_power = render_buffer_.Pop();
   if (!buffered_render_power) {
@@ skipping 59 matching lines @@
     cov.Clear();
   }
   echo_likelihood_ = 0.f;
   next_insertion_index_ = 0;
   reliability_ = 0.f;
 }
 
 void ResidualEchoDetector::PackRenderAudioBuffer(
     AudioBuffer* audio,
     std::vector<float>* packed_buffer) {
-  RTC_DCHECK_GE(160, audio->num_frames_per_band());
-
   packed_buffer->clear();
-  packed_buffer->insert(packed_buffer->end(),
-                        audio->split_bands_const_f(0)[kBand0To8kHz],
-                        (audio->split_bands_const_f(0)[kBand0To8kHz] +
-                         audio->num_frames_per_band()));
+  packed_buffer->insert(packed_buffer->end(), audio->channels_f()[0],
+                        audio->channels_f()[0] + audio->num_frames());
 }
 
 }  // namespace webrtc