1 files changed, 75 insertions, 0 deletions
diff --git a/webrtc/common_audio/smoothing_filter.h b/webrtc/common_audio/smoothing_filter.h
new file mode 100644
index 0000000..e96d52a
--- /dev/null
+++ b/webrtc/common_audio/smoothing_filter.h
@@ -0,0 +1,75 @@
+/*
+ *  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.
+ */
+
+#ifndef COMMON_AUDIO_SMOOTHING_FILTER_H_
+#define COMMON_AUDIO_SMOOTHING_FILTER_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+class SmoothingFilter {
+ public:
+  virtual ~SmoothingFilter() = default;
+  virtual void AddSample(float sample) = 0;
+  virtual absl::optional<float> GetAverage() = 0;
+  virtual bool SetTimeConstantMs(int time_constant_ms) = 0;
+};
+
+// SmoothingFilterImpl applies an exponential filter
+//   alpha = exp(-1.0 / time_constant_ms);
+//   y[t] = alpha * y[t-1] + (1 - alpha) * sample;
+// This implies a sample rate of 1000 Hz, i.e., 1 sample / ms.
+// But SmoothingFilterImpl allows sparse samples. All missing samples will be
+// assumed to equal the last received sample.
+class SmoothingFilterImpl final : public SmoothingFilter {
+ public:
+  // |init_time_ms| is initialization time. It defines a period starting from
+  // the arriving time of the first sample. During this period, the exponential
+  // filter uses a varying time constant so that a smaller time constant will be
+  // applied to the earlier samples. This is to allow the the filter to adapt to
+  // earlier samples quickly. After the initialization period, the time constant
+  // will be set to |init_time_ms| first and can be changed through
+  // |SetTimeConstantMs|.
+  explicit SmoothingFilterImpl(int init_time_ms);
+
+  SmoothingFilterImpl() = delete;
+  SmoothingFilterImpl(const SmoothingFilterImpl&) = delete;
+  SmoothingFilterImpl& operator=(const SmoothingFilterImpl&) = delete;
+
+  ~SmoothingFilterImpl() override;
+
+  void AddSample(float sample) override;
+  absl::optional<float> GetAverage() override;
+  bool SetTimeConstantMs(int time_constant_ms) override;
+
+  // Methods used for unittests.
+  float alpha() const { return alpha_; }
+
+ private:
+  void UpdateAlpha(int time_constant_ms);
+  void ExtrapolateLastSample(int64_t time_ms);
+
+  const int init_time_ms_;
+  const float init_factor_;
+  const float init_const_;
+
+  absl::optional<int64_t> init_end_time_ms_;
+  float last_sample_;
+  float alpha_;
+  float state_;
+  int64_t last_state_time_ms_;
+};
+
+}  // namespace webrtc
+
+#endif  // COMMON_AUDIO_SMOOTHING_FILTER_H_