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
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
|
/*
* Copyright (c) 2012 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_GAIN_CONTROL_IMPL_H_
#define MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
#include <memory>
#include <vector>
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/render_queue_item_verifier.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/swap_queue.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class ApmDataDumper;
class AudioBuffer;
class GainControlImpl : public GainControl {
public:
GainControlImpl(rtc::CriticalSection* crit_render,
rtc::CriticalSection* crit_capture);
~GainControlImpl() override;
void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
int AnalyzeCaptureAudio(AudioBuffer* audio);
int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
void Initialize(size_t num_proc_channels, int sample_rate_hz);
static void PackRenderAudioBuffer(AudioBuffer* audio,
std::vector<int16_t>* packed_buffer);
// GainControl implementation.
bool is_enabled() const override;
int stream_analog_level() override;
bool is_limiter_enabled() const override;
Mode mode() const override;
int compression_gain_db() const override;
private:
class GainController;
// GainControl implementation.
int Enable(bool enable) override;
int set_stream_analog_level(int level) override;
int set_mode(Mode mode) override;
int set_target_level_dbfs(int level) override;
int target_level_dbfs() const override;
int set_compression_gain_db(int gain) override;
int enable_limiter(bool enable) override;
int set_analog_level_limits(int minimum, int maximum) override;
int analog_level_minimum() const override;
int analog_level_maximum() const override;
bool stream_is_saturated() const override;
int Configure();
rtc::CriticalSection* const crit_render_ RTC_ACQUIRED_BEFORE(crit_capture_);
rtc::CriticalSection* const crit_capture_;
std::unique_ptr<ApmDataDumper> data_dumper_;
bool enabled_ = false;
Mode mode_ RTC_GUARDED_BY(crit_capture_);
int minimum_capture_level_ RTC_GUARDED_BY(crit_capture_);
int maximum_capture_level_ RTC_GUARDED_BY(crit_capture_);
bool limiter_enabled_ RTC_GUARDED_BY(crit_capture_);
int target_level_dbfs_ RTC_GUARDED_BY(crit_capture_);
int compression_gain_db_ RTC_GUARDED_BY(crit_capture_);
int analog_capture_level_ RTC_GUARDED_BY(crit_capture_);
bool was_analog_level_set_ RTC_GUARDED_BY(crit_capture_);
bool stream_is_saturated_ RTC_GUARDED_BY(crit_capture_);
std::vector<std::unique_ptr<GainController>> gain_controllers_;
rtc::Optional<size_t> num_proc_channels_ RTC_GUARDED_BY(crit_capture_);
rtc::Optional<int> sample_rate_hz_ RTC_GUARDED_BY(crit_capture_);
static int instance_counter_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
|
