blob: 4ab5cd268ccb16176ce9a4e6db6932643633933c (
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
|
/*
* Copyright (c) 2013 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_DEVICE_FINE_AUDIO_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioDeviceBuffer;
// FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with audio data
// corresponding to 10ms of data. It then allows for this data to be pulled in
// a finer or coarser granularity. I.e. interacting with this class instead of
// directly with the AudioDeviceBuffer one can ask for any number of audio data
// samples. This class also ensures that audio data can be delivered to the ADB
// in 10ms chunks when the size of the provided audio buffers differs from 10ms.
// As an example: calling DeliverRecordedData() with 5ms buffers will deliver
// accumulated 10ms worth of data to the ADB every second call.
class FineAudioBuffer {
public:
// |device_buffer| is a buffer that provides 10ms of audio data.
// |desired_frame_size_bytes| is the number of bytes of audio data
// GetPlayoutData() should return on success. It is also the required size of
// each recorded buffer used in DeliverRecordedData() calls.
// |sample_rate| is the sample rate of the audio data. This is needed because
// |device_buffer| delivers 10ms of data. Given the sample rate the number
// of samples can be calculated.
FineAudioBuffer(AudioDeviceBuffer* device_buffer,
size_t desired_frame_size_bytes,
int sample_rate);
~FineAudioBuffer();
// Returns the required size of |buffer| when calling GetPlayoutData(). If
// the buffer is smaller memory trampling will happen.
size_t RequiredPlayoutBufferSizeBytes();
// Clears buffers and counters dealing with playour and/or recording.
void ResetPlayout();
void ResetRecord();
// |buffer| must be of equal or greater size than what is returned by
// RequiredBufferSize(). This is to avoid unnecessary memcpy.
void GetPlayoutData(int8_t* buffer);
// Consumes the audio data in |buffer| and sends it to the WebRTC layer in
// chunks of 10ms. The provided delay estimates in |playout_delay_ms| and
// |record_delay_ms| are given to the AEC in the audio processing module.
// They can be fixed values on most platforms and they are ignored if an
// external (hardware/built-in) AEC is used.
// The size of |buffer| is given by |size_in_bytes| and must be equal to
// |desired_frame_size_bytes_|. A RTC_CHECK will be hit if this is not the
// case.
// Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores
// 5ms of data and sends a total of 10ms to WebRTC and clears the intenal
// cache. Call #3 restarts the scheme above.
void DeliverRecordedData(const int8_t* buffer,
size_t size_in_bytes,
int playout_delay_ms,
int record_delay_ms);
private:
// Device buffer that works with 10ms chunks of data both for playout and
// for recording. I.e., the WebRTC side will always be asked for audio to be
// played out in 10ms chunks and recorded audio will be sent to WebRTC in
// 10ms chunks as well. This pointer is owned by the constructor of this
// class and the owner must ensure that the pointer is valid during the life-
// time of this object.
AudioDeviceBuffer* const device_buffer_;
// Number of bytes delivered by GetPlayoutData() call and provided to
// DeliverRecordedData().
const size_t desired_frame_size_bytes_;
// Sample rate in Hertz.
const int sample_rate_;
// Number of audio samples per 10ms.
const size_t samples_per_10_ms_;
// Number of audio bytes per 10ms.
const size_t bytes_per_10_ms_;
// Storage for output samples that are not yet asked for.
rtc::scoped_ptr<int8_t[]> playout_cache_buffer_;
// Location of first unread output sample.
size_t playout_cached_buffer_start_;
// Number of bytes stored in output (contain samples to be played out) cache.
size_t playout_cached_bytes_;
// Storage for input samples that are about to be delivered to the WebRTC
// ADB or remains from the last successful delivery of a 10ms audio buffer.
rtc::scoped_ptr<int8_t[]> record_cache_buffer_;
// Required (max) size in bytes of the |record_cache_buffer_|.
const size_t required_record_buffer_size_bytes_;
// Number of bytes in input (contains recorded samples) cache.
size_t record_cached_bytes_;
// Read and write pointers used in the buffering scheme on the recording side.
size_t record_read_pos_;
size_t record_write_pos_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
|
