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

webrtc/modules/audio_processing/level_controller/gain_applier.cc

Index: webrtc/modules/audio_processing/level_controller/gain_applier.cc
diff --git a/webrtc/modules/audio_processing/level_controller/gain_applier.cc b/webrtc/modules/audio_processing/level_controller/gain_applier.cc
new file mode 100644
index 0000000000000000000000000000000000000000..11b60af228d92715c86cd6beaebc0505c17ae1ce
--- /dev/null
+++ b/webrtc/modules/audio_processing/level_controller/gain_applier.cc
@@ -0,0 +1,143 @@
+/*
+ *  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/gain_applier.h"
+
+#include <algorithm>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/checks.h"
+
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+namespace {
+
+const float kMaxSampleValue = 32767.f;
+const float kMinSampleValue = -32767.f;
+
+int CountSaturations(rtc::ArrayView<const float> in) {
+  return std::count_if(in.begin(), in.end(), [](const float& v) {
+    return v >= kMaxSampleValue || v <= kMinSampleValue;
+  });
+}
+
+int CountSaturations(const AudioBuffer& audio) {
+  int num_saturations = 0;
+  for (size_t k = 0; k < audio.num_channels(); ++k) {
+    num_saturations += CountSaturations(rtc::ArrayView<const float>(
+        audio.channels_const_f()[k], audio.num_frames()));
+  }
+  return num_saturations;
+}
+
+void LimitToAllowedRange(rtc::ArrayView<float> x) {
+  for (auto& v : x) {
+    v = std::max(kMinSampleValue, v);
+    v = std::min(kMaxSampleValue, v);
+  }
+}
+
+void LimitToAllowedRange(AudioBuffer* audio) {
+  for (size_t k = 0; k < audio->num_channels(); ++k) {
+    LimitToAllowedRange(
+        rtc::ArrayView<float>(audio->channels_f()[k], audio->num_frames()));
+  }
+}
+
+float ApplyIncreasingGain(float new_gain,
+                          float old_gain,
+                          float step_size,
+                          rtc::ArrayView<float> x) {
+  RTC_DCHECK_LT(0.f, step_size);
+  float gain = old_gain;
+  for (auto& v : x) {
+    gain = std::min(new_gain, gain + step_size);
+    v *= gain;
+  }
+  return gain;
+}
+
+float ApplyDecreasingGain(float new_gain,
+                          float old_gain,
+                          float step_size,
+                          rtc::ArrayView<float> x) {
+  RTC_DCHECK_LT(0.f, step_size);
+  float gain = old_gain;
+  for (auto& v : x) {
+    gain = std::max(new_gain, gain - step_size);
+    v *= gain;
+  }
+  return gain;
+}
+
+float ApplyConstantGain(float gain, rtc::ArrayView<float> x) {
+  for (auto& v : x) {
+    v *= gain;
+  }
+
+  return gain;
+}
+
+float ApplyGain(float new_gain,
+                float old_gain,
+                float step_size,
+                rtc::ArrayView<float> x) {
+  if (new_gain == old_gain) {
+    return ApplyConstantGain(new_gain, x);
+  } else if (new_gain > old_gain) {
+    return ApplyIncreasingGain(new_gain, old_gain, step_size, x);
+  } else {
+    return ApplyDecreasingGain(new_gain, old_gain, step_size, x);
+  }
+}
+
+}  // namespace
+
+GainApplier::GainApplier(ApmDataDumper* data_dumper)
+    : data_dumper_(data_dumper) {}
+
+void GainApplier::Initialize(int sample_rate_hz) {
+  RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
+             sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
+             sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
+             sample_rate_hz == AudioProcessing::kSampleRate48kHz);
+  const float kStepSize48kHz = 0.001f;
+  old_gain_ = 1.f;
+  gain_change_step_size_ =
+      kStepSize48kHz *
+      (static_cast<float>(AudioProcessing::kSampleRate48kHz) / sample_rate_hz);
+}
+
+int GainApplier::Process(float new_gain, AudioBuffer* audio) {
+  RTC_CHECK_NE(0.f, gain_change_step_size_);
+  int num_saturations = 0;
+  if (new_gain != 1.f) {
+    float last_applied_gain = 1.f;
+    for (size_t k = 0; k < audio->num_channels(); ++k) {
+      // TODO(peah): Consider using a faster update rate downwards than upwards.
+      last_applied_gain = ApplyGain(
+          new_gain, old_gain_, gain_change_step_size_,
+          rtc::ArrayView<float>(audio->channels_f()[k], audio->num_frames()));
+    }
+    // TODO(peah): Consider the need for faster gain reduction in case of
+    // excessive saturation.
+    num_saturations = CountSaturations(*audio);
+    LimitToAllowedRange(audio);
+    old_gain_ = last_applied_gain;
+  }
+
+  data_dumper_->DumpRaw("lc_last_applied_gain", 1, &old_gain_);
+
+  return num_saturations;
+}
+
+}  // namespace webrtc