blob: 87d697274ad1957a93880716920aab147128e0c8 (
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
|
/*
* Copyright (c) 2011 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_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
#include "module_common_types.h"
#include "scoped_ptr.h"
#include "typedefs.h"
namespace webrtc {
struct AudioChannel;
struct SplitAudioChannel;
class AudioBuffer {
public:
AudioBuffer(int max_num_channels, int samples_per_channel);
virtual ~AudioBuffer();
int num_channels() const;
int samples_per_channel() const;
int samples_per_split_channel() const;
int16_t* data(int channel) const;
int16_t* low_pass_split_data(int channel) const;
int16_t* high_pass_split_data(int channel) const;
int16_t* mixed_data(int channel) const;
int16_t* mixed_low_pass_data(int channel) const;
int16_t* low_pass_reference(int channel) const;
int32_t* analysis_filter_state1(int channel) const;
int32_t* analysis_filter_state2(int channel) const;
int32_t* synthesis_filter_state1(int channel) const;
int32_t* synthesis_filter_state2(int channel) const;
void set_activity(AudioFrame::VADActivity activity);
AudioFrame::VADActivity activity() const;
bool is_muted() const;
void DeinterleaveFrom(AudioFrame* audioFrame);
void InterleaveTo(AudioFrame* audioFrame) const;
// If |data_changed| is false, only the non-audio data members will be copied
// to |frame|.
void InterleaveTo(AudioFrame* frame, bool data_changed) const;
void Mix(int num_mixed_channels);
void CopyAndMix(int num_mixed_channels);
void CopyAndMixLowPass(int num_mixed_channels);
void CopyLowPassToReference();
private:
const int max_num_channels_;
int num_channels_;
int num_mixed_channels_;
int num_mixed_low_pass_channels_;
// Whether the original data was replaced with mixed data.
bool data_was_mixed_;
const int samples_per_channel_;
int samples_per_split_channel_;
bool reference_copied_;
AudioFrame::VADActivity activity_;
bool is_muted_;
int16_t* data_;
scoped_array<AudioChannel> channels_;
scoped_array<SplitAudioChannel> split_channels_;
scoped_array<AudioChannel> mixed_channels_;
// TODO(andrew): improve this, we don't need the full 32 kHz space here.
scoped_array<AudioChannel> mixed_low_pass_channels_;
scoped_array<AudioChannel> low_pass_reference_channels_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
|
