blob: 1095971040d37e41afc846f079c28977bcf39cd3 (
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
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
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
|
/*
* 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 WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class CriticalSectionWrapper;
const uint32_t kPulsePeriodMs = 1000;
const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
class AudioDeviceObserver;
class AudioDeviceBuffer
{
public:
AudioDeviceBuffer();
virtual ~AudioDeviceBuffer();
void SetId(uint32_t id);
int32_t RegisterAudioCallback(AudioTransport* audioCallback);
int32_t InitPlayout();
int32_t InitRecording();
virtual int32_t SetRecordingSampleRate(uint32_t fsHz);
virtual int32_t SetPlayoutSampleRate(uint32_t fsHz);
int32_t RecordingSampleRate() const;
int32_t PlayoutSampleRate() const;
virtual int32_t SetRecordingChannels(size_t channels);
virtual int32_t SetPlayoutChannels(size_t channels);
size_t RecordingChannels() const;
size_t PlayoutChannels() const;
int32_t SetRecordingChannel(
const AudioDeviceModule::ChannelType channel);
int32_t RecordingChannel(
AudioDeviceModule::ChannelType& channel) const;
virtual int32_t SetRecordedBuffer(const void* audioBuffer,
size_t nSamples);
int32_t SetCurrentMicLevel(uint32_t level);
virtual void SetVQEData(int playDelayMS,
int recDelayMS,
int clockDrift);
virtual int32_t DeliverRecordedData();
uint32_t NewMicLevel() const;
virtual int32_t RequestPlayoutData(size_t nSamples);
virtual int32_t GetPlayoutData(void* audioBuffer);
int32_t StartInputFileRecording(
const char fileName[kAdmMaxFileNameSize]);
int32_t StopInputFileRecording();
int32_t StartOutputFileRecording(
const char fileName[kAdmMaxFileNameSize]);
int32_t StopOutputFileRecording();
int32_t SetTypingStatus(bool typingStatus);
private:
int32_t _id;
CriticalSectionWrapper& _critSect;
CriticalSectionWrapper& _critSectCb;
AudioTransport* _ptrCbAudioTransport;
uint32_t _recSampleRate;
uint32_t _playSampleRate;
size_t _recChannels;
size_t _playChannels;
// selected recording channel (left/right/both)
AudioDeviceModule::ChannelType _recChannel;
// 2 or 4 depending on mono or stereo
size_t _recBytesPerSample;
size_t _playBytesPerSample;
// 10ms in stereo @ 96kHz
int8_t _recBuffer[kMaxBufferSizeBytes];
// one sample <=> 2 or 4 bytes
size_t _recSamples;
size_t _recSize; // in bytes
// 10ms in stereo @ 96kHz
int8_t _playBuffer[kMaxBufferSizeBytes];
// one sample <=> 2 or 4 bytes
size_t _playSamples;
size_t _playSize; // in bytes
FileWrapper& _recFile;
FileWrapper& _playFile;
uint32_t _currentMicLevel;
uint32_t _newMicLevel;
bool _typingStatus;
int _playDelayMS;
int _recDelayMS;
int _clockDrift;
int high_delay_counter_;
};
} // namespace webrtc
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
|
