Chromium Code Reviews

Side by Side Diff: webrtc/modules/audio_processing/utility/mean_calculator.h

Issue 1739993003: Adding fraction of filter divergence in AEC metrics. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: addressing Per's comments Created 4 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments.
Jump to:
View unified diff |
OLDNEW
(Empty)
1 /*
2 * Copyright 2016 The WebRTC Project Authors. All rights reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_MEAN_CALCULATOR_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_MEAN_CALCULATOR_H_
13
14 #include <stddef.h>
15 #include <vector>
16
17 #include "webrtc/base/criticalsection.h"
18 #include "webrtc/base/constructormagic.h"
19 #include "webrtc/base/optional.h"
20
21 namespace webrtc {
22
23 // MeanCalculator calculates the mean of a sample sequence of a certain length.
24 class MeanCalculator {
25 public:
26 explicit MeanCalculator(size_t window_length);
27
28 // Add one sample to the sequence.
29 void AddSample(float sample);
30
31 // Get the mean of the latest samples. Returns the mean if it is available,
32 // otherwise null, which happens when the added samples have not fully filled
33 // the window.
34 rtc::Optional<float> GetMean() const;
35
36 // Clear all samples added.
37 void Clear();
38
39 // Determines if the window is full. This is a quick way of checking if the
40 // mean is ready.
41 bool IsWindowFull() const;
42
43 private:
44 // Update |sum_| and |compensation_| using Kahan algorithm.
45 void KahanSum(float sample);
46
47 rtc::CriticalSection crit_;
48 size_t window_length_ GUARDED_BY(crit_);
49 std::vector<float> buffer_ GUARDED_BY(crit_);
50 size_t head_ GUARDED_BY(crit_);
51 bool full_ GUARDED_BY(crit_);
52 float sum_ GUARDED_BY(crit_);
53 float compensation_ GUARDED_BY(crit_);
54
55 RTC_DISALLOW_COPY_AND_ASSIGN(MeanCalculator);
56 };
57
58 } // namespace webrtc
59
60 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_MEAN_CALCULATOR_H
OLDNEW

Powered by Google App Engine