blob: aa84a2e0f12e81ea80572cc57540462d8191c2ba (
plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
|
/*
* Copyright (c) 2018 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 MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
#include <array>
#include <vector>
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
class ApmDataDumper;
// Produces a smooth signal level estimate from an input audio
// stream. The estimate smoothing is done through exponential
// filtering.
class FixedDigitalLevelEstimator {
public:
// Sample rates are allowed if the number of samples in a frame
// (sample_rate_hz * kFrameDurationMs / 1000) is divisible by
// kSubFramesInSample. For kFrameDurationMs=10 and
// kSubFramesInSample=20, this means that sample_rate_hz has to be
// divisible by 2000.
FixedDigitalLevelEstimator(size_t sample_rate_hz,
ApmDataDumper* apm_data_dumper);
// The input is assumed to be in FloatS16 format. Scaled input will
// produce similarly scaled output. A frame of with kFrameDurationMs
// ms of audio produces a level estimates in the same scale. The
// level estimate contains kSubFramesInFrame values.
std::array<float, kSubFramesInFrame> ComputeLevel(
const AudioFrameView<const float>& float_frame);
// Rate may be changed at any time (but not concurrently) from the
// value passed to the constructor. The class is not thread safe.
void SetSampleRate(size_t sample_rate_hz);
// Resets the level estimator internal state.
void Reset();
float LastAudioLevel() const { return filter_state_level_; }
private:
void CheckParameterCombination();
ApmDataDumper* const apm_data_dumper_ = nullptr;
float filter_state_level_;
size_t samples_in_frame_;
size_t samples_in_sub_frame_;
RTC_DISALLOW_COPY_AND_ASSIGN(FixedDigitalLevelEstimator);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
|
