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Unified Diff: webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h

Issue 2339523002: Adding SmoothingFilter to audio network adaptor. (Closed)
Patch Set: fixing a gyp error Created 4 years, 3 months ago
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Index: webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h b/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h
new file mode 100644
index 0000000000000000000000000000000000000000..c4de7b5da5a5dc7f85b2f5e258505fb5025cb7b1
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h
@@ -0,0 +1,54 @@
+/*
+ * 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 WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_SMOOTHING_FILTER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_SMOOTHING_FILTER_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/exp_filter.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class SmoothingFilter {
+ public:
+ virtual ~SmoothingFilter() = default;
+ virtual void AddSample(float sample) = 0;
+ virtual rtc::Optional<float> GetAverage() const = 0;
+};
+
+// SmoothingFilterImpl applies an exponential filter
+// alpha = exp(-sample_interval / time_constant);
+// y[t] = alpha * y[t-1] + (1 - alpha) * sample;
+class SmoothingFilterImpl final : public SmoothingFilter {
+ public:
+ // |time_constant_ms| is the time constant for the exponential filter.
+ SmoothingFilterImpl(int time_constant_ms, const Clock* clock);
+
+ void AddSample(float sample) override;
+ rtc::Optional<float> GetAverage() const override;
+
+ private:
+ const int time_constant_ms_;
+ const Clock* const clock_;
+
+ bool first_sample_received_;
+ bool initialized_;
+ int64_t first_sample_time_ms_;
+ int64_t last_sample_time_ms_;
+ rtc::ExpFilter filter_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(SmoothingFilterImpl);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_SMOOTHING_FILTER_H_

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