Adding SmoothingFilter to audio network adaptor.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2339523002
Cr-Commit-Position: refs/heads/master@{#14313}
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index 36ae692..4a565c1 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -245,6 +245,7 @@
"audio_coding/audio_network_adaptor/dtx_controller_unittest.cc",
"audio_coding/audio_network_adaptor/mock/mock_controller.h",
"audio_coding/audio_network_adaptor/mock/mock_controller_manager.h",
+ "audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc",
]
deps = [
"audio_coding:audio_network_adaptor",
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 4fe49e5..2c90c7a 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -710,6 +710,8 @@
"audio_network_adaptor/dtx_controller.cc",
"audio_network_adaptor/dtx_controller.h",
"audio_network_adaptor/include/audio_network_adaptor.h",
+ "audio_network_adaptor/smoothing_filter.cc",
+ "audio_network_adaptor/smoothing_filter.h",
]
configs += [ "../..:common_config" ]
public_configs = [ "../..:common_inherited_config" ]
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi
index 8e87815..2a591c3 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi
@@ -24,7 +24,9 @@
'controller_manager.h',
'dtx_controller.h',
'dtx_controller.cc',
- 'include/audio_network_adaptor.h'
+ 'include/audio_network_adaptor.h',
+ 'smoothing_filter.h',
+ 'smoothing_filter.cc',
], # source
},
], # targets
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.cc b/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.cc
new file mode 100644
index 0000000..8a81069
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.cc
@@ -0,0 +1,62 @@
+/*
+ * 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 <cmath>
+
+#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
+
+namespace webrtc {
+
+SmoothingFilterImpl::SmoothingFilterImpl(int time_constant_ms,
+ const Clock* clock)
+ : time_constant_ms_(time_constant_ms),
+ clock_(clock),
+ first_sample_received_(false),
+ initialized_(false),
+ first_sample_time_ms_(0),
+ last_sample_time_ms_(0),
+ filter_(0.0) {}
+
+void SmoothingFilterImpl::AddSample(float sample) {
+ if (!first_sample_received_) {
+ last_sample_time_ms_ = first_sample_time_ms_ = clock_->TimeInMilliseconds();
+ first_sample_received_ = true;
+ RTC_DCHECK_EQ(rtc::ExpFilter::kValueUndefined, filter_.filtered());
+
+ // Since this is first sample, any value for argument 1 should work.
+ filter_.Apply(0.0f, sample);
+ return;
+ }
+
+ int64_t now_ms = clock_->TimeInMilliseconds();
+ if (!initialized_) {
+ float duration = now_ms - first_sample_time_ms_;
+ if (duration < static_cast<int64_t>(time_constant_ms_)) {
+ filter_.UpdateBase(exp(1.0f / duration));
+ } else {
+ initialized_ = true;
+ filter_.UpdateBase(exp(1.0f / time_constant_ms_));
+ }
+ }
+
+ // The filter will do the following:
+ // float alpha = pow(base, last_update_time_ms_ - now_ms);
+ // filtered_ = alpha * filtered_ + (1 - alpha) * sample;
+ filter_.Apply(static_cast<float>(last_sample_time_ms_ - now_ms), sample);
+ last_sample_time_ms_ = now_ms;
+}
+
+rtc::Optional<float> SmoothingFilterImpl::GetAverage() const {
+ float value = filter_.filtered();
+ return value == rtc::ExpFilter::kValueUndefined ? rtc::Optional<float>()
+ : rtc::Optional<float>(value);
+}
+
+} // namespace webrtc
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 0000000..c4de7b5
--- /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_
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc
new file mode 100644
index 0000000..388ae4f
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc
@@ -0,0 +1,108 @@
+/*
+ * 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 <memory>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kTimeConstantMs = 1000;
+constexpr float kMaxAbsError = 0.0001f;
+constexpr int64_t kClockInitialTime = 123456;
+
+struct SmoothingFilterStates {
+ std::unique_ptr<SimulatedClock> simulated_clock;
+ std::unique_ptr<SmoothingFilter> smoothing_filter;
+};
+
+SmoothingFilterStates CreateSmoothingFilter() {
+ SmoothingFilterStates states;
+ states.simulated_clock.reset(new SimulatedClock(kClockInitialTime));
+ states.smoothing_filter.reset(
+ new SmoothingFilterImpl(kTimeConstantMs, states.simulated_clock.get()));
+ return states;
+}
+
+void CheckOutput(SmoothingFilterStates* states,
+ int advance_time_ms,
+ float sample,
+ float expected_ouput) {
+ states->simulated_clock->AdvanceTimeMilliseconds(advance_time_ms);
+ states->smoothing_filter->AddSample(sample);
+ auto output = states->smoothing_filter->GetAverage();
+ EXPECT_TRUE(output);
+ EXPECT_NEAR(expected_ouput, *output, kMaxAbsError);
+}
+
+} // namespace
+
+TEST(SmoothingFilterTest, NoOutputWhenNoSampleAdded) {
+ auto states = CreateSmoothingFilter();
+ EXPECT_FALSE(states.smoothing_filter->GetAverage());
+}
+
+// Python script to calculate the reference values used in this test.
+// import math
+//
+// class ExpFilter:
+// alpha = 0.0
+// old_value = 0.0
+// def calc(self, new_value):
+// self.old_value = self.old_value * self.alpha
+// + (1.0 - self.alpha) * new_value
+// return self.old_value
+//
+// delta_t = 100.0
+// filter = ExpFilter()
+// total_t = 100.0
+// filter.alpha = math.exp(-delta_t/ total_t)
+// print filter.calc(1.0)
+// total_t = 200.0
+// filter.alpha = math.exp(-delta_t/ total_t)
+// print filter.calc(0.0)
+// total_t = 300.0
+// filter.alpha = math.exp(-delta_t/ total_t)
+// print filter.calc(1.0)
+TEST(SmoothingFilterTest, CheckBehaviorBeforeInitialized) {
+ // Adding three samples, all added before |kTimeConstantMs| is reached.
+ constexpr int kTimeIntervalMs = 100;
+ auto states = CreateSmoothingFilter();
+ states.smoothing_filter->AddSample(0.0);
+ CheckOutput(&states, kTimeIntervalMs, 1.0, 0.63212f);
+ CheckOutput(&states, kTimeIntervalMs, 0.0, 0.38340f);
+ CheckOutput(&states, kTimeIntervalMs, 1.0, 0.55818f);
+}
+
+// Python script to calculate the reference value used in this test.
+// (after defining ExpFilter as for CheckBehaviorBeforeInitialized)
+// time_constant_ms = 1000.0
+// filter = ExpFilter()
+// delta_t = 1100.0
+// filter.alpha = math.exp(-delta_t/ time_constant_ms)
+// print filter.calc(1.0)
+// delta_t = 100.0
+// filter.alpha = math.exp(-delta_t/ time_constant_ms)
+// print filter.calc(0.0)
+// print filter.calc(1.0)
+TEST(SmoothingFilterTest, CheckBehaviorAfterInitialized) {
+ constexpr int kTimeIntervalMs = 100;
+ auto states = CreateSmoothingFilter();
+ states.smoothing_filter->AddSample(0.0);
+ states.simulated_clock->AdvanceTimeMilliseconds(kTimeConstantMs);
+ CheckOutput(&states, kTimeIntervalMs, 1.0, 0.66713f);
+ CheckOutput(&states, kTimeIntervalMs, 0.0, 0.60364f);
+ CheckOutput(&states, kTimeIntervalMs, 1.0, 0.64136f);
+}
+
+} // namespace webrtc
