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
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
|
/*
* 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_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
#include "webrtc/common_types.h"
#include "webrtc/modules/interface/module.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/media_file/interface/media_file_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class MediaFile : public Module
{
public:
// Factory method. Constructor disabled. id is the identifier for the
// MediaFile instance.
static MediaFile* CreateMediaFile(const int32_t id);
static void DestroyMediaFile(MediaFile* module);
// Put 10-60ms of audio data from file into the audioBuffer depending on
// codec frame size. dataLengthInBytes is both an input and output
// parameter. As input parameter it indicates the size of audioBuffer.
// As output parameter it indicates the number of bytes written to
// audioBuffer.
// Note: This API only play mono audio but can be used on file containing
// audio with more channels (in which case the audio will be converted to
// mono).
virtual int32_t PlayoutAudioData(
int8_t* audioBuffer,
size_t& dataLengthInBytes) = 0;
// Put 10-60ms, depending on codec frame size, of audio data from file into
// audioBufferLeft and audioBufferRight. The buffers contain the left and
// right channel of played out stereo audio.
// dataLengthInBytes is both an input and output parameter. As input
// parameter it indicates the size of both audioBufferLeft and
// audioBufferRight. As output parameter it indicates the number of bytes
// written to both audio buffers.
// Note: This API can only be successfully called for WAV files with stereo
// audio.
virtual int32_t PlayoutStereoData(
int8_t* audioBufferLeft,
int8_t* audioBufferRight,
size_t& dataLengthInBytes) = 0;
// Open the file specified by fileName (relative path is allowed) for
// reading. FileCallback::PlayNotification(..) will be called after
// notificationTimeMs of the file has been played if notificationTimeMs is
// greater than zero. If loop is true the file will be played until
// StopPlaying() is called. When end of file is reached the file is read
// from the start. format specifies the type of file fileName refers to.
// codecInst specifies the encoding of the audio data. Note that
// file formats that contain this information (like WAV files) don't need to
// provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
// specify what part of the file should be read. From startPointMs ms to
// stopPointMs ms.
// Note: codecInst.channels should be set to 2 for stereo (and 1 for
// mono). Stereo audio is only supported for WAV files.
virtual int32_t StartPlayingAudioFile(
const char* fileName,
const uint32_t notificationTimeMs = 0,
const bool loop = false,
const FileFormats format = kFileFormatPcm16kHzFile,
const CodecInst* codecInst = NULL,
const uint32_t startPointMs = 0,
const uint32_t stopPointMs = 0) = 0;
// Prepare for playing audio from stream.
// FileCallback::PlayNotification(..) will be called after
// notificationTimeMs of the file has been played if notificationTimeMs is
// greater than zero. format specifies the type of file fileName refers to.
// codecInst specifies the encoding of the audio data. Note that
// file formats that contain this information (like WAV files) don't need to
// provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
// specify what part of the file should be read. From startPointMs ms to
// stopPointMs ms.
// Note: codecInst.channels should be set to 2 for stereo (and 1 for
// mono). Stereo audio is only supported for WAV files.
virtual int32_t StartPlayingAudioStream(
InStream& stream,
const uint32_t notificationTimeMs = 0,
const FileFormats format = kFileFormatPcm16kHzFile,
const CodecInst* codecInst = NULL,
const uint32_t startPointMs = 0,
const uint32_t stopPointMs = 0) = 0;
// Stop playing from file or stream.
virtual int32_t StopPlaying() = 0;
// Return true if playing.
virtual bool IsPlaying() = 0;
// Set durationMs to the number of ms that has been played from file.
virtual int32_t PlayoutPositionMs(
uint32_t& durationMs) const = 0;
// Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
// to file. The audio frame size is determined by the codecInst.pacsize
// parameter of the last sucessfull StartRecordingAudioFile(..) call.
// Note: bufferLength must be exactly one frame.
virtual int32_t IncomingAudioData(
const int8_t* audioBuffer,
const size_t bufferLength) = 0;
// Open/creates file specified by fileName for writing (relative path is
// allowed). FileCallback::RecordNotification(..) will be called after
// notificationTimeMs of audio data has been recorded if
// notificationTimeMs is greater than zero.
// format specifies the type of file that should be created/opened.
// codecInst specifies the encoding of the audio data. maxSizeBytes
// specifies the number of bytes allowed to be written to file if it is
// greater than zero.
// Note: codecInst.channels should be set to 2 for stereo (and 1 for
// mono). Stereo is only supported for WAV files.
virtual int32_t StartRecordingAudioFile(
const char* fileName,
const FileFormats format,
const CodecInst& codecInst,
const uint32_t notificationTimeMs = 0,
const uint32_t maxSizeBytes = 0) = 0;
// Prepare for recording audio to stream.
// FileCallback::RecordNotification(..) will be called after
// notificationTimeMs of audio data has been recorded if
// notificationTimeMs is greater than zero.
// format specifies the type of file that stream should correspond to.
// codecInst specifies the encoding of the audio data.
// Note: codecInst.channels should be set to 2 for stereo (and 1 for
// mono). Stereo is only supported for WAV files.
virtual int32_t StartRecordingAudioStream(
OutStream& stream,
const FileFormats format,
const CodecInst& codecInst,
const uint32_t notificationTimeMs = 0) = 0;
// Stop recording to file or stream.
virtual int32_t StopRecording() = 0;
// Return true if recording.
virtual bool IsRecording() = 0;
// Set durationMs to the number of ms that has been recorded to file.
virtual int32_t RecordDurationMs(uint32_t& durationMs) = 0;
// Return true if recording or playing is stereo.
virtual bool IsStereo() = 0;
// Register callback to receive media file related notifications. Disables
// callbacks if callback is NULL.
virtual int32_t SetModuleFileCallback(FileCallback* callback) = 0;
// Set durationMs to the size of the file (in ms) specified by fileName.
// format specifies the type of file fileName refers to. freqInHz specifies
// the sampling frequency of the file.
virtual int32_t FileDurationMs(
const char* fileName,
uint32_t& durationMs,
const FileFormats format,
const uint32_t freqInHz = 16000) = 0;
// Update codecInst according to the current audio codec being used for
// reading or writing.
virtual int32_t codec_info(CodecInst& codecInst) const = 0;
protected:
MediaFile() {}
virtual ~MediaFile() {}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
|
