New module for the adaptive level controlling functionality in the audio processing module

webrtc/modules/audio_processing/level_controller/signal_classifier.cc

Index: webrtc/modules/audio_processing/level_controller/signal_classifier.cc
diff --git a/webrtc/modules/audio_processing/level_controller/signal_classifier.cc b/webrtc/modules/audio_processing/level_controller/signal_classifier.cc
new file mode 100644
index 0000000000000000000000000000000000000000..d09220365807c4a07c563160ae2f24000823b45f
--- /dev/null
+++ b/webrtc/modules/audio_processing/level_controller/signal_classifier.cc
@@ -0,0 +1,157 @@
+/*
+ *  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/level_controller/signal_classifier.h"
+
+#include <algorithm>
+#include <numeric>
+#include <vector>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_processing/aec/aec_rdft.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/level_controller/down_sampler.h"
+#include "webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.h"
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+namespace {
+
+void RemoveDcLevel(rtc::ArrayView<float> x) {
+  RTC_DCHECK_LT(0u, x.size());
+  float mean = std::accumulate(x.data(), x.data() + x.size(), 0.f);
+  mean /= x.size();
+
+  for (float& v : x) {
+    v -= mean;
+  }
+}
+
+void PowerSpectrum(rtc::ArrayView<const float> x,
+                   rtc::ArrayView<float> spectrum) {
+  RTC_DCHECK_EQ(65u, spectrum.size());
+  RTC_DCHECK_EQ(128u, x.size());
+  float X[128];
+  std::copy(x.data(), x.data() + x.size(), X);
+  aec_rdft_forward_128(X);
+
+  float* X_p = X;
+  RTC_CHECK_EQ(X_p, &X[0]);

[hlundin-webrtc, 2016/06/27 11:21:17]

Use DCHECK here and in the places below.

[peah-webrtc, 2016/06/27 22:51:50]

Done.

+  spectrum[0] = (*X_p) * (*X_p);
+  ++X_p;
+  RTC_CHECK_EQ(X_p, &X[1]);
+  spectrum[64] = (*X_p) * (*X_p);
+  for (int k = 1; k < 64; ++k) {
+    ++X_p;
+    RTC_CHECK_EQ(X_p, &X[2 * k]);
+    spectrum[k] += (*X_p) * (*X_p);

[hlundin-webrtc, 2016/06/27 11:21:17]

You are adding to an undefined value here. Use = instead.

[peah-webrtc, 2016/06/27 22:51:50]

Great find!!!!! 

Done.

+    ++X_p;
+    RTC_CHECK_EQ(X_p, &X[2 * k + 1]);
+    spectrum[k] += (*X_p) * (*X_p);
+  }
+}
+
+void ClassifySignal(rtc::ArrayView<const float> signal_spectrum,
+                    rtc::ArrayView<const float> noise_spectrum,
+                    ApmDataDumper* data_dumper,
+                    SignalClassifier::SignalType* signal_type) {

[hlundin-webrtc, 2016/06/27 11:21:17]

Return the signal type instead of having it as an output parameter.

[peah-webrtc, 2016/06/27 22:51:50]

Done.

+  int num_stationary_bands = 0;
+  int num_highly_nonstationary_bands = 0;
+
+  // Detect stationary and highly nonstationary bands.
+  for (int k = 1; k < 40; k++) {

[hlundin-webrtc, 2016/06/27 11:21:17]

size_t

[peah-webrtc, 2016/06/27 22:51:50]

Done.

+    if (signal_spectrum[k] < 3 * noise_spectrum[k]) {
+      ++num_stationary_bands;
+    } else if (signal_spectrum[k] > 9 * noise_spectrum[k]) {
+      ++num_highly_nonstationary_bands;
+    }
+  }
+
+  // Use the detected number of bands to classify the overall signal
+  // stationarity.
+  if (num_stationary_bands > 20) {
+    *signal_type = SignalClassifier::SignalType::kStationary;
+  } else if (num_highly_nonstationary_bands > 15) {
+    *signal_type = SignalClassifier::SignalType::kHighlyNonStationary;
+  } else {
+    *signal_type = SignalClassifier::SignalType::kNonStationary;
+  }
+
+  data_dumper->DumpRaw("lc_num_stationary_bands", 1, &num_stationary_bands);
+  data_dumper->DumpRaw("lc_num_highly_nonstationary_bands", 1,
+                       &num_highly_nonstationary_bands);
+}
+
+}  // namespace
+
+class FrameExtender {

[hlundin-webrtc, 2016/06/27 11:21:17]

Why is this outside of the unnamed namespace?

[peah-webrtc, 2016/06/27 22:51:50]

I think it needs to be outside of that for the reason that it is a member of the
state.

[hlundin-webrtc, 2016/06/28 11:29:01]

Oh, I see. Then I think I would prefer to have it as a private subclass inside
the SignalClassifier.

[peah-webrtc, 2016/06/28 22:19:37]

Done.

+ public:
+  FrameExtender(size_t frame_size, size_t extended_frame_size)
+      : x_old_(extended_frame_size - frame_size, 0.f) {}
+
+  void ExtendFrame(rtc::ArrayView<const float> x,
+                   rtc::ArrayView<float> x_extended) {
+    RTC_DCHECK_EQ(x_old_.size() + x.size(), x_extended.size());
+    std::copy(x_old_.data(), x_old_.data() + x_old_.size(), x_extended.data());
+    std::copy(x.data(), x.data() + x.size(), x_extended.data() + x_old_.size());
+    std::copy(x_extended.data() + x_extended.size() - x_old_.size(),
+              x_extended.data() + x_extended.size(), x_old_.data());
+  }
+
+ private:
+  std::vector<float> x_old_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameExtender);
+};
+
+SignalClassifier::SignalClassifier(ApmDataDumper* data_dumper)
+    : data_dumper_(data_dumper) {
+  Initialize(AudioProcessing::kSampleRate48kHz);
+}
+SignalClassifier::~SignalClassifier() {}
+
+void SignalClassifier::Initialize(int sample_rate_hz) {
+  aec_rdft_init();
+  down_sampler_.reset(new DownSampler(data_dumper_, sample_rate_hz));
+  frame_extender_.reset(new FrameExtender(80, 128));
+  noise_spectrum_estimator_.reset(new NoiseSpectrumEstimator(data_dumper_));
+  sample_rate_hz_ = sample_rate_hz;
+  initialization_frames_left_ = 2;
+}
+
+void SignalClassifier::Analyze(const AudioBuffer& audio,
+                               SignalType* signal_type) {
+  // Compute the signal power spectrum.

[hlundin-webrtc, 2016/06/27 11:21:17]

You may want to DCHECK some properties of the AudioBuffer, such as sample rate
and/or number of samples.

[peah-webrtc, 2016/06/27 22:51:50]

That would make sense, but I think AudioBuffer does not provide any
functionality to access the sample rate. And apart from that there is nothing in
this method (apart from the downsampler) which relies on the frame length so I
cannot find any good candidates to DCHECK on. WDYT?

[hlundin-webrtc, 2016/06/28 11:29:01]

Can't you just check that the number of samples in the audio frame adds up as
expected with the assumed sample rate?

[peah-webrtc, 2016/06/28 22:19:37]

True!
Good point!
Done.

+  float downsampled_frame[80];
+  down_sampler_->DownSample(
+      rtc::ArrayView<const float>(audio.channels_const_f()[0],
+                                  audio.num_frames()),
+      downsampled_frame);
+  float extended_frame[128];
+  frame_extender_->ExtendFrame(downsampled_frame, extended_frame);
+  RemoveDcLevel(extended_frame);
+  float signal_spectrum[65];
+  PowerSpectrum(extended_frame, signal_spectrum);
+
+  // Classify the signal based on the estimate of the noise spectrum and the
+  // signal spectrum estimate.
+  ClassifySignal(signal_spectrum, noise_spectrum_estimator_->GetNoiseSpectrum(),
+                 data_dumper_, signal_type);
+
+  // Update the noise spectrum based on the signal spectrum.
+  noise_spectrum_estimator_->Update(signal_spectrum,
+                                    initialization_frames_left_ > 0);
+
+  // Update the number of frames until a reliable signal spectrum is achieved.
+  initialization_frames_left_ = std::max(0, initialization_frames_left_ - 1);
+}
+
+}  // namespace webrtc