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
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
|
/*
* 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_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
#define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
#include <cstddef>
#include <cstdint>
#include <memory>
#include <utility>
#include "api/audio_codecs/audio_encoder.h"
#include "api/function_view.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "rtc_base/checks.h"
namespace webrtc {
// forward declarations
class AudioDecoder;
class AudioEncoder;
class AudioFrame;
struct RTPHeader;
// Callback class used for sending data ready to be packetized
class AudioPacketizationCallback {
public:
virtual ~AudioPacketizationCallback() {}
virtual int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
int64_t /* absolute_capture_timestamp_ms */) {
// TODO(bugs.webrtc.org/10739): Deprecate the old SendData and make this one
// pure virtual.
return SendData(frame_type, payload_type, timestamp, payload_data,
payload_len_bytes);
}
virtual int32_t SendData(AudioFrameType /* frame_type */,
uint8_t /* payload_type */,
uint32_t /* timestamp */,
const uint8_t* /* payload_data */,
size_t /* payload_len_bytes */) {
RTC_DCHECK_NOTREACHED() << "This method must be overridden, or not used.";
return -1;
}
};
class AudioCodingModule {
protected:
AudioCodingModule() {}
public:
static std::unique_ptr<AudioCodingModule> Create();
virtual ~AudioCodingModule() = default;
// `modifier` is called exactly once with one argument: a pointer to the
// unique_ptr that holds the current encoder (which is null if there is no
// current encoder). For the duration of the call, `modifier` has exclusive
// access to the unique_ptr; it may call the encoder, steal the encoder and
// replace it with another encoder or with nullptr, etc.
virtual void ModifyEncoder(
FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0;
// Utility method for simply replacing the existing encoder with a new one.
void SetEncoder(std::unique_ptr<AudioEncoder> new_encoder) {
ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
*encoder = std::move(new_encoder);
});
}
// Reset encoder and audio coding module. This throws away any audio passed
// and starts fresh.
virtual void Reset() = 0;
// int32_t RegisterTransportCallback()
// Register a transport callback which will be called to deliver
// the encoded buffers whenever Process() is called and a
// bit-stream is ready.
//
// Input:
// -transport : pointer to the callback class
// transport->SendData() is called whenever
// Process() is called and bit-stream is ready
// to deliver.
//
// Return value:
// -1 if the transport callback could not be registered
// 0 if registration is successful.
//
virtual int32_t RegisterTransportCallback(
AudioPacketizationCallback* transport) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t Add10MsData()
// Add 10MS of raw (PCM) audio data and encode it. If the sampling
// frequency of the audio does not match the sampling frequency of the
// current encoder ACM will resample the audio. If an encoded packet was
// produced, it will be delivered via the callback object registered using
// RegisterTransportCallback, and the return value from this function will
// be the number of bytes encoded.
//
// Input:
// -audio_frame : the input audio frame, containing raw audio
// sampling frequency etc.
//
// Return value:
// >= 0 number of bytes encoded.
// -1 some error occurred.
//
virtual int32_t Add10MsData(const AudioFrame& audio_frame) = 0;
///////////////////////////////////////////////////////////////////////////
// int SetPacketLossRate()
// Sets expected packet loss rate for encoding. Some encoders provide packet
// loss gnostic encoding to make stream less sensitive to packet losses,
// through e.g., FEC. No effects on codecs that do not provide such encoding.
//
// Input:
// -packet_loss_rate : expected packet loss rate (0 -- 100 inclusive).
//
// Return value
// -1 if failed to set packet loss rate,
// 0 if succeeded.
//
// This is only used in test code that rely on old ACM APIs.
// TODO(minyue): Remove it when possible.
virtual int SetPacketLossRate(int packet_loss_rate) = 0;
///////////////////////////////////////////////////////////////////////////
// statistics
//
virtual ANAStats GetANAStats() const = 0;
virtual int GetTargetBitrate() const = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
|
