Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(177)

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

Issue 2339523002: Adding SmoothingFilter to audio network adaptor. (Closed)
Patch Set: fixing a gyp error Created 4 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.cc
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 0000000000000000000000000000000000000000..8a8106918a9366310df7a434c4e28a383fcf8c23
--- /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

Powered by Google App Engine
This is Rietveld 408576698