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annotate DPF-Prymula-audioplugins/dpf/distrho/src/jackbridge/rtaudio/RtAudio.h @ 4:0962f0296c8a
ColorsMemory-0.210503-1.1
author | prymula <prymula76@outlook.com> |
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date | Tue, 30 Jan 2024 16:43:10 +0100 |
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1 /************************************************************************/ |
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2 /*! \class RtAudio |
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3 \brief Realtime audio i/o C++ classes. |
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4 |
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5 RtAudio provides a common API (Application Programming Interface) |
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6 for realtime audio input/output across Linux (native ALSA, Jack, |
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7 and OSS), Macintosh OS X (CoreAudio and Jack), and Windows |
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8 (DirectSound, ASIO and WASAPI) operating systems. |
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9 |
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10 RtAudio GitHub site: https://github.com/thestk/rtaudio |
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11 RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ |
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12 |
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13 RtAudio: realtime audio i/o C++ classes |
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14 Copyright (c) 2001-2019 Gary P. Scavone |
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15 |
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16 Permission is hereby granted, free of charge, to any person |
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17 obtaining a copy of this software and associated documentation files |
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18 (the "Software"), to deal in the Software without restriction, |
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19 including without limitation the rights to use, copy, modify, merge, |
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20 publish, distribute, sublicense, and/or sell copies of the Software, |
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21 and to permit persons to whom the Software is furnished to do so, |
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22 subject to the following conditions: |
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23 |
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24 The above copyright notice and this permission notice shall be |
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25 included in all copies or substantial portions of the Software. |
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26 |
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27 Any person wishing to distribute modifications to the Software is |
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28 asked to send the modifications to the original developer so that |
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29 they can be incorporated into the canonical version. This is, |
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30 however, not a binding provision of this license. |
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31 |
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32 THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, |
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33 EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF |
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34 MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. |
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35 IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR |
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36 ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF |
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37 CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION |
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38 WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. |
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39 */ |
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40 /************************************************************************/ |
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41 |
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42 /*! |
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43 \file RtAudio.h |
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44 */ |
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45 |
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46 #ifndef __RTAUDIO_H |
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47 #define __RTAUDIO_H |
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48 |
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49 #define RTAUDIO_VERSION "5.1.0" |
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50 |
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51 #if defined _WIN32 || defined __CYGWIN__ |
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52 #if defined(RTAUDIO_EXPORT) |
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53 #define RTAUDIO_DLL_PUBLIC __declspec(dllexport) |
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54 #else |
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55 #define RTAUDIO_DLL_PUBLIC |
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56 #endif |
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57 #else |
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58 #if __GNUC__ >= 4 |
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59 #define RTAUDIO_DLL_PUBLIC __attribute__( (visibility( "default" )) ) |
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60 #else |
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61 #define RTAUDIO_DLL_PUBLIC |
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62 #endif |
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63 #endif |
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64 |
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65 #include <string> |
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66 #include <vector> |
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67 #include <stdexcept> |
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68 #include <iostream> |
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69 |
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70 /*! \typedef typedef unsigned long RtAudioFormat; |
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71 \brief RtAudio data format type. |
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72 |
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73 Support for signed integers and floats. Audio data fed to/from an |
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74 RtAudio stream is assumed to ALWAYS be in host byte order. The |
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75 internal routines will automatically take care of any necessary |
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76 byte-swapping between the host format and the soundcard. Thus, |
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77 endian-ness is not a concern in the following format definitions. |
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78 |
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79 - \e RTAUDIO_SINT8: 8-bit signed integer. |
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80 - \e RTAUDIO_SINT16: 16-bit signed integer. |
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81 - \e RTAUDIO_SINT24: 24-bit signed integer. |
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82 - \e RTAUDIO_SINT32: 32-bit signed integer. |
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83 - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0. |
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84 - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0. |
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85 */ |
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86 typedef unsigned long RtAudioFormat; |
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87 static const RtAudioFormat RTAUDIO_SINT8 = 0x1; // 8-bit signed integer. |
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88 static const RtAudioFormat RTAUDIO_SINT16 = 0x2; // 16-bit signed integer. |
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89 static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // 24-bit signed integer. |
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90 static const RtAudioFormat RTAUDIO_SINT32 = 0x8; // 32-bit signed integer. |
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91 static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0. |
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92 static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0. |
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93 |
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94 /*! \typedef typedef unsigned long RtAudioStreamFlags; |
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95 \brief RtAudio stream option flags. |
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96 |
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97 The following flags can be OR'ed together to allow a client to |
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98 make changes to the default stream behavior: |
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99 |
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100 - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). |
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101 - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. |
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102 - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. |
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103 - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only). |
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104 - \e RTAUDIO_JACK_DONT_CONNECT: Do not automatically connect ports (JACK only). |
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105 |
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106 By default, RtAudio streams pass and receive audio data from the |
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107 client in an interleaved format. By passing the |
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108 RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio |
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109 data will instead be presented in non-interleaved buffers. In |
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110 this case, each buffer argument in the RtAudioCallback function |
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111 will point to a single array of data, with \c nFrames samples for |
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112 each channel concatenated back-to-back. For example, the first |
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113 sample of data for the second channel would be located at index \c |
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114 nFrames (assuming the \c buffer pointer was recast to the correct |
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115 data type for the stream). |
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116 |
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117 Certain audio APIs offer a number of parameters that influence the |
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118 I/O latency of a stream. By default, RtAudio will attempt to set |
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119 these parameters internally for robust (glitch-free) performance |
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120 (though some APIs, like Windows DirectSound, make this difficult). |
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121 By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() |
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122 function, internal stream settings will be influenced in an attempt |
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123 to minimize stream latency, though possibly at the expense of stream |
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124 performance. |
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125 |
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126 If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to |
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127 open the input and/or output stream device(s) for exclusive use. |
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128 Note that this is not possible with all supported audio APIs. |
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129 |
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130 If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt |
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131 to select realtime scheduling (round-robin) for the callback thread. |
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132 |
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133 If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to |
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134 open the "default" PCM device when using the ALSA API. Note that this |
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135 will override any specified input or output device id. |
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136 |
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137 If the RTAUDIO_JACK_DONT_CONNECT flag is set, RtAudio will not attempt |
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138 to automatically connect the ports of the client to the audio device. |
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139 */ |
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140 typedef unsigned int RtAudioStreamFlags; |
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141 static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved). |
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142 static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency. |
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143 static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others. |
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144 static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread. |
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145 static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only). |
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146 static const RtAudioStreamFlags RTAUDIO_JACK_DONT_CONNECT = 0x20; // Do not automatically connect ports (JACK only). |
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147 |
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148 /*! \typedef typedef unsigned long RtAudioStreamStatus; |
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149 \brief RtAudio stream status (over- or underflow) flags. |
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150 |
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151 Notification of a stream over- or underflow is indicated by a |
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152 non-zero stream \c status argument in the RtAudioCallback function. |
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153 The stream status can be one of the following two options, |
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154 depending on whether the stream is open for output and/or input: |
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155 |
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156 - \e RTAUDIO_INPUT_OVERFLOW: Input data was discarded because of an overflow condition at the driver. |
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157 - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound. |
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158 */ |
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159 typedef unsigned int RtAudioStreamStatus; |
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160 static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1; // Input data was discarded because of an overflow condition at the driver. |
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161 static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output buffer ran low, likely causing a gap in the output sound. |
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162 |
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163 //! RtAudio callback function prototype. |
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164 /*! |
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165 All RtAudio clients must create a function of type RtAudioCallback |
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166 to read and/or write data from/to the audio stream. When the |
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167 underlying audio system is ready for new input or output data, this |
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168 function will be invoked. |
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169 |
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170 \param outputBuffer For output (or duplex) streams, the client |
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171 should write \c nFrames of audio sample frames into this |
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172 buffer. This argument should be recast to the datatype |
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173 specified when the stream was opened. For input-only |
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174 streams, this argument will be NULL. |
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175 |
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176 \param inputBuffer For input (or duplex) streams, this buffer will |
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177 hold \c nFrames of input audio sample frames. This |
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178 argument should be recast to the datatype specified when the |
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179 stream was opened. For output-only streams, this argument |
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180 will be NULL. |
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181 |
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182 \param nFrames The number of sample frames of input or output |
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183 data in the buffers. The actual buffer size in bytes is |
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184 dependent on the data type and number of channels in use. |
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185 |
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186 \param streamTime The number of seconds that have elapsed since the |
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187 stream was started. |
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188 |
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189 \param status If non-zero, this argument indicates a data overflow |
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190 or underflow condition for the stream. The particular |
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191 condition can be determined by comparison with the |
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192 RtAudioStreamStatus flags. |
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193 |
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194 \param userData A pointer to optional data provided by the client |
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195 when opening the stream (default = NULL). |
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196 |
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197 \return |
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198 To continue normal stream operation, the RtAudioCallback function |
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199 should return a value of zero. To stop the stream and drain the |
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200 output buffer, the function should return a value of one. To abort |
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201 the stream immediately, the client should return a value of two. |
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202 */ |
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203 typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer, |
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204 unsigned int nFrames, |
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205 double streamTime, |
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206 RtAudioStreamStatus status, |
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207 void *userData ); |
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208 |
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209 /************************************************************************/ |
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210 /*! \class RtAudioError |
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211 \brief Exception handling class for RtAudio. |
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212 |
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213 The RtAudioError class is quite simple but it does allow errors to be |
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214 "caught" by RtAudioError::Type. See the RtAudio documentation to know |
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215 which methods can throw an RtAudioError. |
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216 */ |
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217 /************************************************************************/ |
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218 |
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219 class RTAUDIO_DLL_PUBLIC RtAudioError : public std::runtime_error |
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220 { |
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221 public: |
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222 //! Defined RtAudioError types. |
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223 enum Type { |
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224 WARNING, /*!< A non-critical error. */ |
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225 DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */ |
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226 UNSPECIFIED, /*!< The default, unspecified error type. */ |
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227 NO_DEVICES_FOUND, /*!< No devices found on system. */ |
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228 INVALID_DEVICE, /*!< An invalid device ID was specified. */ |
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229 MEMORY_ERROR, /*!< An error occured during memory allocation. */ |
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230 INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */ |
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231 INVALID_USE, /*!< The function was called incorrectly. */ |
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232 DRIVER_ERROR, /*!< A system driver error occured. */ |
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233 SYSTEM_ERROR, /*!< A system error occured. */ |
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234 THREAD_ERROR /*!< A thread error occured. */ |
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235 }; |
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236 |
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237 //! The constructor. |
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238 RtAudioError( const std::string& message, |
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239 Type type = RtAudioError::UNSPECIFIED ) |
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240 : std::runtime_error(message), type_(type) {} |
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241 |
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242 //! Prints thrown error message to stderr. |
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243 virtual void printMessage( void ) const |
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244 { std::cerr << '\n' << what() << "\n\n"; } |
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245 |
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246 //! Returns the thrown error message type. |
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247 virtual const Type& getType(void) const { return type_; } |
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248 |
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249 //! Returns the thrown error message string. |
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250 virtual const std::string getMessage(void) const |
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251 { return std::string(what()); } |
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252 |
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253 protected: |
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254 Type type_; |
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255 }; |
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256 |
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257 //! RtAudio error callback function prototype. |
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258 /*! |
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259 \param type Type of error. |
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260 \param errorText Error description. |
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261 */ |
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262 typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string &errorText ); |
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263 |
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264 // **************************************************************** // |
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265 // |
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266 // RtAudio class declaration. |
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267 // |
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268 // RtAudio is a "controller" used to select an available audio i/o |
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269 // interface. It presents a common API for the user to call but all |
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270 // functionality is implemented by the class RtApi and its |
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271 // subclasses. RtAudio creates an instance of an RtApi subclass |
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272 // based on the user's API choice. If no choice is made, RtAudio |
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273 // attempts to make a "logical" API selection. |
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274 // |
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275 // **************************************************************** // |
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276 |
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277 class RtApi; |
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278 |
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279 class RTAUDIO_DLL_PUBLIC RtAudio |
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280 { |
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281 public: |
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282 |
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283 //! Audio API specifier arguments. |
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284 enum Api { |
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285 UNSPECIFIED, /*!< Search for a working compiled API. */ |
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286 LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */ |
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287 LINUX_PULSE, /*!< The Linux PulseAudio API. */ |
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288 LINUX_OSS, /*!< The Linux Open Sound System API. */ |
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289 UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */ |
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290 MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */ |
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291 WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */ |
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292 WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */ |
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293 WINDOWS_DS, /*!< The Microsoft DirectSound API. */ |
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294 RTAUDIO_DUMMY, /*!< A compilable but non-functional API. */ |
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295 NUM_APIS /*!< Number of values in this enum. */ |
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296 }; |
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297 |
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298 //! The public device information structure for returning queried values. |
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299 struct DeviceInfo { |
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300 bool probed; /*!< true if the device capabilities were successfully probed. */ |
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301 std::string name; /*!< Character string device identifier. */ |
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302 unsigned int outputChannels{}; /*!< Maximum output channels supported by device. */ |
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303 unsigned int inputChannels{}; /*!< Maximum input channels supported by device. */ |
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304 unsigned int duplexChannels{}; /*!< Maximum simultaneous input/output channels supported by device. */ |
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305 bool isDefaultOutput{false}; /*!< true if this is the default output device. */ |
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306 bool isDefaultInput{false}; /*!< true if this is the default input device. */ |
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307 std::vector<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */ |
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308 unsigned int preferredSampleRate{}; /*!< Preferred sample rate, e.g. for WASAPI the system sample rate. */ |
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309 RtAudioFormat nativeFormats{}; /*!< Bit mask of supported data formats. */ |
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310 }; |
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311 |
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312 //! The structure for specifying input or ouput stream parameters. |
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313 struct StreamParameters { |
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314 unsigned int deviceId{}; /*!< Device index (0 to getDeviceCount() - 1). */ |
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315 unsigned int nChannels{}; /*!< Number of channels. */ |
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316 unsigned int firstChannel{}; /*!< First channel index on device (default = 0). */ |
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317 }; |
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318 |
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319 //! The structure for specifying stream options. |
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320 /*! |
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321 The following flags can be OR'ed together to allow a client to |
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322 make changes to the default stream behavior: |
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323 |
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324 - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). |
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325 - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. |
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326 - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. |
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327 - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread. |
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328 - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only). |
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329 |
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330 By default, RtAudio streams pass and receive audio data from the |
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331 client in an interleaved format. By passing the |
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332 RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio |
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333 data will instead be presented in non-interleaved buffers. In |
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334 this case, each buffer argument in the RtAudioCallback function |
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335 will point to a single array of data, with \c nFrames samples for |
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336 each channel concatenated back-to-back. For example, the first |
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337 sample of data for the second channel would be located at index \c |
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338 nFrames (assuming the \c buffer pointer was recast to the correct |
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339 data type for the stream). |
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340 |
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341 Certain audio APIs offer a number of parameters that influence the |
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342 I/O latency of a stream. By default, RtAudio will attempt to set |
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343 these parameters internally for robust (glitch-free) performance |
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344 (though some APIs, like Windows DirectSound, make this difficult). |
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345 By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() |
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346 function, internal stream settings will be influenced in an attempt |
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347 to minimize stream latency, though possibly at the expense of stream |
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348 performance. |
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349 |
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350 If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to |
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351 open the input and/or output stream device(s) for exclusive use. |
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352 Note that this is not possible with all supported audio APIs. |
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353 |
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354 If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt |
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355 to select realtime scheduling (round-robin) for the callback thread. |
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356 The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME |
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357 flag is set. It defines the thread's realtime priority. |
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358 |
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359 If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to |
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360 open the "default" PCM device when using the ALSA API. Note that this |
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361 will override any specified input or output device id. |
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362 |
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363 The \c numberOfBuffers parameter can be used to control stream |
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364 latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs |
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365 only. A value of two is usually the smallest allowed. Larger |
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366 numbers can potentially result in more robust stream performance, |
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367 though likely at the cost of stream latency. The value set by the |
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368 user is replaced during execution of the RtAudio::openStream() |
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369 function by the value actually used by the system. |
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370 |
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371 The \c streamName parameter can be used to set the client name |
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372 when using the Jack API. By default, the client name is set to |
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373 RtApiJack. However, if you wish to create multiple instances of |
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374 RtAudio with Jack, each instance must have a unique client name. |
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375 */ |
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376 struct StreamOptions { |
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377 RtAudioStreamFlags flags{}; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */ |
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378 unsigned int numberOfBuffers{}; /*!< Number of stream buffers. */ |
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379 std::string streamName; /*!< A stream name (currently used only in Jack). */ |
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380 int priority{}; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */ |
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381 }; |
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382 |
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383 //! A static function to determine the current RtAudio version. |
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384 static std::string getVersion( void ); |
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385 |
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386 //! A static function to determine the available compiled audio APIs. |
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387 /*! |
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388 The values returned in the std::vector can be compared against |
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389 the enumerated list values. Note that there can be more than one |
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390 API compiled for certain operating systems. |
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391 */ |
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392 static void getCompiledApi( std::vector<RtAudio::Api> &apis ); |
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393 |
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394 //! Return the name of a specified compiled audio API. |
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395 /*! |
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396 This obtains a short lower-case name used for identification purposes. |
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397 This value is guaranteed to remain identical across library versions. |
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398 If the API is unknown, this function will return the empty string. |
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399 */ |
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400 static std::string getApiName( RtAudio::Api api ); |
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401 |
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402 //! Return the display name of a specified compiled audio API. |
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403 /*! |
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404 This obtains a long name used for display purposes. |
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405 If the API is unknown, this function will return the empty string. |
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406 */ |
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407 static std::string getApiDisplayName( RtAudio::Api api ); |
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408 |
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409 //! Return the compiled audio API having the given name. |
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410 /*! |
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411 A case insensitive comparison will check the specified name |
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412 against the list of compiled APIs, and return the one which |
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413 matches. On failure, the function returns UNSPECIFIED. |
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414 */ |
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415 static RtAudio::Api getCompiledApiByName( const std::string &name ); |
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416 |
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417 //! The class constructor. |
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418 /*! |
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419 The constructor performs minor initialization tasks. An exception |
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420 can be thrown if no API support is compiled. |
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421 |
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422 If no API argument is specified and multiple API support has been |
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423 compiled, the default order of use is JACK, ALSA, OSS (Linux |
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424 systems) and ASIO, DS (Windows systems). |
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425 */ |
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426 RtAudio( RtAudio::Api api=UNSPECIFIED ); |
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427 |
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428 //! The destructor. |
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429 /*! |
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430 If a stream is running or open, it will be stopped and closed |
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431 automatically. |
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432 */ |
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433 ~RtAudio(); |
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434 |
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435 //! Returns the audio API specifier for the current instance of RtAudio. |
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436 RtAudio::Api getCurrentApi( void ); |
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437 |
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438 //! A public function that queries for the number of audio devices available. |
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439 /*! |
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440 This function performs a system query of available devices each time it |
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441 is called, thus supporting devices connected \e after instantiation. If |
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442 a system error occurs during processing, a warning will be issued. |
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443 */ |
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444 unsigned int getDeviceCount( void ); |
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445 |
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446 //! Return an RtAudio::DeviceInfo structure for a specified device number. |
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447 /*! |
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448 |
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449 Any device integer between 0 and getDeviceCount() - 1 is valid. |
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450 If an invalid argument is provided, an RtAudioError (type = INVALID_USE) |
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451 will be thrown. If a device is busy or otherwise unavailable, the |
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452 structure member "probed" will have a value of "false" and all |
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453 other members are undefined. If the specified device is the |
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454 current default input or output device, the corresponding |
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455 "isDefault" member will have a value of "true". |
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456 */ |
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457 RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); |
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458 |
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459 //! A function that returns the index of the default output device. |
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460 /*! |
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461 If the underlying audio API does not provide a "default |
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462 device", or if no devices are available, the return value will be |
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463 0. Note that this is a valid device identifier and it is the |
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464 client's responsibility to verify that a device is available |
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465 before attempting to open a stream. |
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466 */ |
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467 unsigned int getDefaultOutputDevice( void ); |
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468 |
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469 //! A function that returns the index of the default input device. |
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470 /*! |
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471 If the underlying audio API does not provide a "default |
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472 device", or if no devices are available, the return value will be |
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473 0. Note that this is a valid device identifier and it is the |
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474 client's responsibility to verify that a device is available |
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475 before attempting to open a stream. |
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476 */ |
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477 unsigned int getDefaultInputDevice( void ); |
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478 |
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479 //! A public function for opening a stream with the specified parameters. |
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480 /*! |
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481 An RtAudioError (type = SYSTEM_ERROR) is thrown if a stream cannot be |
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482 opened with the specified parameters or an error occurs during |
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483 processing. An RtAudioError (type = INVALID_USE) is thrown if any |
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484 invalid device ID or channel number parameters are specified. |
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485 |
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486 \param outputParameters Specifies output stream parameters to use |
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487 when opening a stream, including a device ID, number of channels, |
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488 and starting channel number. For input-only streams, this |
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489 argument should be NULL. The device ID is an index value between |
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490 0 and getDeviceCount() - 1. |
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491 \param inputParameters Specifies input stream parameters to use |
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492 when opening a stream, including a device ID, number of channels, |
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493 and starting channel number. For output-only streams, this |
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494 argument should be NULL. The device ID is an index value between |
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495 0 and getDeviceCount() - 1. |
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496 \param format An RtAudioFormat specifying the desired sample data format. |
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497 \param sampleRate The desired sample rate (sample frames per second). |
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498 \param *bufferFrames A pointer to a value indicating the desired |
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499 internal buffer size in sample frames. The actual value |
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500 used by the device is returned via the same pointer. A |
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501 value of zero can be specified, in which case the lowest |
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502 allowable value is determined. |
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503 \param callback A client-defined function that will be invoked |
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504 when input data is available and/or output data is needed. |
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505 \param userData An optional pointer to data that can be accessed |
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506 from within the callback function. |
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507 \param options An optional pointer to a structure containing various |
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508 global stream options, including a list of OR'ed RtAudioStreamFlags |
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509 and a suggested number of stream buffers that can be used to |
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510 control stream latency. More buffers typically result in more |
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511 robust performance, though at a cost of greater latency. If a |
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512 value of zero is specified, a system-specific median value is |
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513 chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the |
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514 lowest allowable value is used. The actual value used is |
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515 returned via the structure argument. The parameter is API dependent. |
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516 \param errorCallback A client-defined function that will be invoked |
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517 when an error has occured. |
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518 */ |
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519 void openStream( RtAudio::StreamParameters *outputParameters, |
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520 RtAudio::StreamParameters *inputParameters, |
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521 RtAudioFormat format, unsigned int sampleRate, |
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522 unsigned int *bufferFrames, RtAudioCallback callback, |
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523 void *userData = NULL, RtAudio::StreamOptions *options = NULL, RtAudioErrorCallback errorCallback = NULL ); |
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524 |
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525 //! A function that closes a stream and frees any associated stream memory. |
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526 /*! |
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527 If a stream is not open, this function issues a warning and |
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528 returns (no exception is thrown). |
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529 */ |
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530 void closeStream( void ); |
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531 |
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532 //! A function that starts a stream. |
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533 /*! |
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534 An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs |
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535 during processing. An RtAudioError (type = INVALID_USE) is thrown if a |
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536 stream is not open. A warning is issued if the stream is already |
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537 running. |
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538 */ |
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539 void startStream( void ); |
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540 |
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541 //! Stop a stream, allowing any samples remaining in the output queue to be played. |
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542 /*! |
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543 An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs |
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544 during processing. An RtAudioError (type = INVALID_USE) is thrown if a |
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545 stream is not open. A warning is issued if the stream is already |
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546 stopped. |
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547 */ |
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548 void stopStream( void ); |
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549 |
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550 //! Stop a stream, discarding any samples remaining in the input/output queue. |
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551 /*! |
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552 An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs |
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553 during processing. An RtAudioError (type = INVALID_USE) is thrown if a |
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554 stream is not open. A warning is issued if the stream is already |
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555 stopped. |
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556 */ |
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557 void abortStream( void ); |
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558 |
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559 //! Returns true if a stream is open and false if not. |
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560 bool isStreamOpen( void ) const; |
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561 |
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562 //! Returns true if the stream is running and false if it is stopped or not open. |
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563 bool isStreamRunning( void ) const; |
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564 |
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565 //! Returns the number of elapsed seconds since the stream was started. |
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566 /*! |
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567 If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. |
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568 */ |
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569 double getStreamTime( void ); |
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570 |
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571 //! Set the stream time to a time in seconds greater than or equal to 0.0. |
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572 /*! |
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573 If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. |
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574 */ |
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575 void setStreamTime( double time ); |
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576 |
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577 //! Returns the internal stream latency in sample frames. |
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578 /*! |
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579 The stream latency refers to delay in audio input and/or output |
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580 caused by internal buffering by the audio system and/or hardware. |
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581 For duplex streams, the returned value will represent the sum of |
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582 the input and output latencies. If a stream is not open, an |
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583 RtAudioError (type = INVALID_USE) will be thrown. If the API does not |
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584 report latency, the return value will be zero. |
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585 */ |
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586 long getStreamLatency( void ); |
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587 |
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588 //! Returns actual sample rate in use by the stream. |
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589 /*! |
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590 On some systems, the sample rate used may be slightly different |
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591 than that specified in the stream parameters. If a stream is not |
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592 open, an RtAudioError (type = INVALID_USE) will be thrown. |
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593 */ |
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594 unsigned int getStreamSampleRate( void ); |
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595 |
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596 //! Specify whether warning messages should be printed to stderr. |
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597 void showWarnings( bool value = true ); |
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598 |
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599 protected: |
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600 |
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601 void openRtApi( RtAudio::Api api ); |
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602 RtApi *rtapi_; |
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603 }; |
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604 |
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605 // Operating system dependent thread functionality. |
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606 #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) |
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607 |
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608 #ifndef NOMINMAX |
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609 #define NOMINMAX |
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610 #endif |
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611 #include <windows.h> |
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612 #include <process.h> |
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613 #include <stdint.h> |
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614 |
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615 typedef uintptr_t ThreadHandle; |
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616 typedef CRITICAL_SECTION StreamMutex; |
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617 |
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618 #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) |
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619 // Using pthread library for various flavors of unix. |
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620 #include <pthread.h> |
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621 |
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622 typedef pthread_t ThreadHandle; |
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623 typedef pthread_mutex_t StreamMutex; |
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624 |
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625 #else // Setup for "dummy" behavior |
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626 |
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627 #define __RTAUDIO_DUMMY__ |
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628 typedef int ThreadHandle; |
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629 typedef int StreamMutex; |
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630 |
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631 #endif |
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632 |
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633 // This global structure type is used to pass callback information |
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634 // between the private RtAudio stream structure and global callback |
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635 // handling functions. |
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636 struct CallbackInfo { |
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637 void *object{}; // Used as a "this" pointer. |
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638 ThreadHandle thread{}; |
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639 void *callback{}; |
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640 void *userData{}; |
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641 void *errorCallback{}; |
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642 void *apiInfo{}; // void pointer for API specific callback information |
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643 bool isRunning{false}; |
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644 bool doRealtime{false}; |
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645 int priority{}; |
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646 }; |
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647 |
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648 // **************************************************************** // |
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649 // |
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650 // RtApi class declaration. |
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651 // |
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652 // Subclasses of RtApi contain all API- and OS-specific code necessary |
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653 // to fully implement the RtAudio API. |
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654 // |
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655 // Note that RtApi is an abstract base class and cannot be |
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656 // explicitly instantiated. The class RtAudio will create an |
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657 // instance of an RtApi subclass (RtApiOss, RtApiAlsa, |
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658 // RtApiJack, RtApiCore, RtApiDs, or RtApiAsio). |
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659 // |
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660 // **************************************************************** // |
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661 |
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662 #pragma pack(push, 1) |
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663 class S24 { |
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664 |
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665 protected: |
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666 unsigned char c3[3]; |
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667 |
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668 public: |
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669 S24() {} |
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670 |
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671 S24& operator = ( const int& i ) { |
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672 c3[0] = (i & 0x000000ff); |
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673 c3[1] = (i & 0x0000ff00) >> 8; |
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674 c3[2] = (i & 0x00ff0000) >> 16; |
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675 return *this; |
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676 } |
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677 |
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678 S24( const double& d ) { *this = (int) d; } |
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679 S24( const float& f ) { *this = (int) f; } |
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680 S24( const signed short& s ) { *this = (int) s; } |
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681 S24( const char& c ) { *this = (int) c; } |
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682 |
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683 int asInt() { |
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684 int i = c3[0] | (c3[1] << 8) | (c3[2] << 16); |
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685 if (i & 0x800000) i |= ~0xffffff; |
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686 return i; |
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687 } |
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688 }; |
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689 #pragma pack(pop) |
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690 |
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691 #if defined( HAVE_GETTIMEOFDAY ) |
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692 #include <sys/time.h> |
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693 #endif |
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694 |
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695 #include <sstream> |
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696 |
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697 class RTAUDIO_DLL_PUBLIC RtApi |
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698 { |
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699 public: |
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700 |
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701 RtApi(); |
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702 virtual ~RtApi(); |
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703 virtual RtAudio::Api getCurrentApi( void ) = 0; |
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704 virtual unsigned int getDeviceCount( void ) = 0; |
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705 virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0; |
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706 virtual unsigned int getDefaultInputDevice( void ); |
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707 virtual unsigned int getDefaultOutputDevice( void ); |
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708 void openStream( RtAudio::StreamParameters *outputParameters, |
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709 RtAudio::StreamParameters *inputParameters, |
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710 RtAudioFormat format, unsigned int sampleRate, |
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711 unsigned int *bufferFrames, RtAudioCallback callback, |
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712 void *userData, RtAudio::StreamOptions *options, |
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713 RtAudioErrorCallback errorCallback ); |
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714 virtual void closeStream( void ); |
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715 virtual void startStream( void ) = 0; |
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716 virtual void stopStream( void ) = 0; |
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717 virtual void abortStream( void ) = 0; |
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718 long getStreamLatency( void ); |
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719 unsigned int getStreamSampleRate( void ); |
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720 virtual double getStreamTime( void ); |
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721 virtual void setStreamTime( double time ); |
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722 bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; } |
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723 bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; } |
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724 void showWarnings( bool value ) { showWarnings_ = value; } |
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725 |
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726 |
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727 protected: |
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728 |
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729 static const unsigned int MAX_SAMPLE_RATES; |
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730 static const unsigned int SAMPLE_RATES[]; |
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731 |
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732 enum { FAILURE, SUCCESS }; |
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733 |
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734 enum StreamState { |
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735 STREAM_STOPPED, |
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736 STREAM_STOPPING, |
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737 STREAM_RUNNING, |
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738 STREAM_CLOSED = -50 |
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739 }; |
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740 |
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741 enum StreamMode { |
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742 OUTPUT, |
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743 INPUT, |
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744 DUPLEX, |
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745 UNINITIALIZED = -75 |
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746 }; |
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747 |
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748 // A protected structure used for buffer conversion. |
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749 struct ConvertInfo { |
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750 int channels; |
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751 int inJump, outJump; |
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752 RtAudioFormat inFormat, outFormat; |
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753 std::vector<int> inOffset; |
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754 std::vector<int> outOffset; |
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755 }; |
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756 |
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757 // A protected structure for audio streams. |
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758 struct RtApiStream { |
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759 unsigned int device[2]; // Playback and record, respectively. |
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760 void *apiHandle; // void pointer for API specific stream handle information |
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761 StreamMode mode; // OUTPUT, INPUT, or DUPLEX. |
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762 StreamState state; // STOPPED, RUNNING, or CLOSED |
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763 char *userBuffer[2]; // Playback and record, respectively. |
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764 char *deviceBuffer; |
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765 bool doConvertBuffer[2]; // Playback and record, respectively. |
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766 bool userInterleaved; |
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767 bool deviceInterleaved[2]; // Playback and record, respectively. |
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768 bool doByteSwap[2]; // Playback and record, respectively. |
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769 unsigned int sampleRate; |
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770 unsigned int bufferSize; |
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771 unsigned int nBuffers; |
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772 unsigned int nUserChannels[2]; // Playback and record, respectively. |
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773 unsigned int nDeviceChannels[2]; // Playback and record channels, respectively. |
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774 unsigned int channelOffset[2]; // Playback and record, respectively. |
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775 unsigned long latency[2]; // Playback and record, respectively. |
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776 RtAudioFormat userFormat; |
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777 RtAudioFormat deviceFormat[2]; // Playback and record, respectively. |
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778 StreamMutex mutex; |
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779 CallbackInfo callbackInfo; |
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780 ConvertInfo convertInfo[2]; |
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781 double streamTime; // Number of elapsed seconds since the stream started. |
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782 |
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783 #if defined(HAVE_GETTIMEOFDAY) |
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784 struct timeval lastTickTimestamp; |
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785 #endif |
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786 |
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787 RtApiStream() |
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788 :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; } |
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789 }; |
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790 |
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791 typedef S24 Int24; |
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792 typedef signed short Int16; |
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793 typedef signed int Int32; |
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794 typedef float Float32; |
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795 typedef double Float64; |
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796 |
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797 std::ostringstream errorStream_; |
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798 std::string errorText_; |
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799 bool showWarnings_; |
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800 RtApiStream stream_; |
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801 bool firstErrorOccurred_; |
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802 |
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803 /*! |
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804 Protected, api-specific method that attempts to open a device |
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805 with the given parameters. This function MUST be implemented by |
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806 all subclasses. If an error is encountered during the probe, a |
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807 "warning" message is reported and FAILURE is returned. A |
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808 successful probe is indicated by a return value of SUCCESS. |
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809 */ |
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810 virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, |
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811 unsigned int firstChannel, unsigned int sampleRate, |
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812 RtAudioFormat format, unsigned int *bufferSize, |
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813 RtAudio::StreamOptions *options ); |
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814 |
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815 //! A protected function used to increment the stream time. |
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816 void tickStreamTime( void ); |
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817 |
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818 //! Protected common method to clear an RtApiStream structure. |
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819 void clearStreamInfo(); |
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820 |
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821 /*! |
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822 Protected common method that throws an RtAudioError (type = |
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823 INVALID_USE) if a stream is not open. |
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824 */ |
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825 void verifyStream( void ); |
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826 |
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827 //! Protected common error method to allow global control over error handling. |
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828 void error( RtAudioError::Type type ); |
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829 |
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830 /*! |
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831 Protected method used to perform format, channel number, and/or interleaving |
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832 conversions between the user and device buffers. |
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833 */ |
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834 void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info ); |
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835 |
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836 //! Protected common method used to perform byte-swapping on buffers. |
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837 void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ); |
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838 |
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839 //! Protected common method that returns the number of bytes for a given format. |
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840 unsigned int formatBytes( RtAudioFormat format ); |
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841 |
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842 //! Protected common method that sets up the parameters for buffer conversion. |
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843 void setConvertInfo( StreamMode mode, unsigned int firstChannel ); |
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844 }; |
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845 |
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846 // **************************************************************** // |
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847 // |
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848 // Inline RtAudio definitions. |
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849 // |
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850 // **************************************************************** // |
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851 |
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852 inline RtAudio::Api RtAudio :: getCurrentApi( void ) { return rtapi_->getCurrentApi(); } |
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853 inline unsigned int RtAudio :: getDeviceCount( void ) { return rtapi_->getDeviceCount(); } |
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854 inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); } |
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855 inline unsigned int RtAudio :: getDefaultInputDevice( void ) { return rtapi_->getDefaultInputDevice(); } |
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856 inline unsigned int RtAudio :: getDefaultOutputDevice( void ) { return rtapi_->getDefaultOutputDevice(); } |
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857 inline void RtAudio :: closeStream( void ) { return rtapi_->closeStream(); } |
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858 inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); } |
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859 inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); } |
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860 inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); } |
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861 inline bool RtAudio :: isStreamOpen( void ) const { return rtapi_->isStreamOpen(); } |
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862 inline bool RtAudio :: isStreamRunning( void ) const { return rtapi_->isStreamRunning(); } |
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863 inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); } |
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864 inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); } |
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865 inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); } |
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866 inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); } |
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867 inline void RtAudio :: showWarnings( bool value ) { rtapi_->showWarnings( value ); } |
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868 |
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869 // RtApi Subclass prototypes. |
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870 |
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871 #if defined(__MACOSX_CORE__) |
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872 |
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873 #include <CoreAudio/AudioHardware.h> |
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874 |
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875 class RtApiCore: public RtApi |
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876 { |
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877 public: |
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878 |
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879 RtApiCore(); |
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880 ~RtApiCore(); |
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881 RtAudio::Api getCurrentApi( void ) override { return RtAudio::MACOSX_CORE; } |
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882 unsigned int getDeviceCount( void ) override; |
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883 RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; |
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884 unsigned int getDefaultOutputDevice( void ) override; |
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885 unsigned int getDefaultInputDevice( void ) override; |
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886 void closeStream( void ) override; |
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887 void startStream( void ) override; |
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888 void stopStream( void ) override; |
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889 void abortStream( void ) override; |
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890 |
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891 // This function is intended for internal use only. It must be |
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892 // public because it is called by the internal callback handler, |
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893 // which is not a member of RtAudio. External use of this function |
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894 // will most likely produce highly undesireable results! |
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895 bool callbackEvent( AudioDeviceID deviceId, |
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896 const AudioBufferList *inBufferList, |
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897 const AudioBufferList *outBufferList ); |
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898 |
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899 private: |
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900 |
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901 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, |
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902 unsigned int firstChannel, unsigned int sampleRate, |
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903 RtAudioFormat format, unsigned int *bufferSize, |
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904 RtAudio::StreamOptions *options ) override; |
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905 static const char* getErrorCode( OSStatus code ); |
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906 }; |
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907 |
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908 #endif |
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909 |
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910 #if defined(__UNIX_JACK__) |
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911 |
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912 class RtApiJack: public RtApi |
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913 { |
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914 public: |
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915 |
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916 RtApiJack(); |
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917 ~RtApiJack(); |
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918 RtAudio::Api getCurrentApi( void ) override { return RtAudio::UNIX_JACK; } |
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919 unsigned int getDeviceCount( void ) override; |
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920 RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; |
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921 void closeStream( void ) override; |
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922 void startStream( void ) override; |
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923 void stopStream( void ) override; |
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924 void abortStream( void ) override; |
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925 |
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926 // This function is intended for internal use only. It must be |
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927 // public because it is called by the internal callback handler, |
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928 // which is not a member of RtAudio. External use of this function |
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929 // will most likely produce highly undesireable results! |
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930 bool callbackEvent( unsigned long nframes ); |
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931 |
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932 private: |
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933 |
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934 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, |
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935 unsigned int firstChannel, unsigned int sampleRate, |
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936 RtAudioFormat format, unsigned int *bufferSize, |
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937 RtAudio::StreamOptions *options ) override; |
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938 |
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939 bool shouldAutoconnect_; |
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940 }; |
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941 |
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942 #endif |
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943 |
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944 #if defined(__WINDOWS_ASIO__) |
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945 |
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946 class RtApiAsio: public RtApi |
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947 { |
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948 public: |
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949 |
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950 RtApiAsio(); |
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951 ~RtApiAsio(); |
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952 RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_ASIO; } |
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953 unsigned int getDeviceCount( void ) override; |
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954 RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; |
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955 void closeStream( void ) override; |
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956 void startStream( void ) override; |
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957 void stopStream( void ) override; |
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958 void abortStream( void ) override; |
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959 |
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960 // This function is intended for internal use only. It must be |
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961 // public because it is called by the internal callback handler, |
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962 // which is not a member of RtAudio. External use of this function |
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963 // will most likely produce highly undesireable results! |
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964 bool callbackEvent( long bufferIndex ); |
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965 |
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966 private: |
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967 |
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968 std::vector<RtAudio::DeviceInfo> devices_; |
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969 void saveDeviceInfo( void ); |
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970 bool coInitialized_; |
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971 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, |
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972 unsigned int firstChannel, unsigned int sampleRate, |
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973 RtAudioFormat format, unsigned int *bufferSize, |
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974 RtAudio::StreamOptions *options ) override; |
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975 }; |
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976 |
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977 #endif |
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978 |
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979 #if defined(__WINDOWS_DS__) |
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980 |
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981 class RtApiDs: public RtApi |
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982 { |
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983 public: |
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984 |
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985 RtApiDs(); |
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986 ~RtApiDs(); |
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987 RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_DS; } |
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988 unsigned int getDeviceCount( void ) override; |
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989 unsigned int getDefaultOutputDevice( void ) override; |
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990 unsigned int getDefaultInputDevice( void ) override; |
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991 RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; |
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992 void closeStream( void ) override; |
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993 void startStream( void ) override; |
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994 void stopStream( void ) override; |
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995 void abortStream( void ) override; |
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996 |
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997 // This function is intended for internal use only. It must be |
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998 // public because it is called by the internal callback handler, |
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999 // which is not a member of RtAudio. External use of this function |
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1000 // will most likely produce highly undesireable results! |
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1001 void callbackEvent( void ); |
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1002 |
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1003 private: |
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1004 |
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1005 bool coInitialized_; |
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1006 bool buffersRolling; |
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1007 long duplexPrerollBytes; |
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1008 std::vector<struct DsDevice> dsDevices; |
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1009 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, |
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1010 unsigned int firstChannel, unsigned int sampleRate, |
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1011 RtAudioFormat format, unsigned int *bufferSize, |
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1012 RtAudio::StreamOptions *options ) override; |
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1013 }; |
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1014 |
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1015 #endif |
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1016 |
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1017 #if defined(__WINDOWS_WASAPI__) |
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1018 |
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1019 struct IMMDeviceEnumerator; |
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1020 |
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1021 class RtApiWasapi : public RtApi |
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1022 { |
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1023 public: |
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1024 RtApiWasapi(); |
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1025 virtual ~RtApiWasapi(); |
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1026 |
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1027 RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_WASAPI; } |
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1028 unsigned int getDeviceCount( void ) override; |
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1029 RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; |
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1030 unsigned int getDefaultOutputDevice( void ) override; |
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1031 unsigned int getDefaultInputDevice( void ) override; |
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1032 void closeStream( void ) override; |
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1033 void startStream( void ) override; |
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1034 void stopStream( void ) override; |
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1035 void abortStream( void ) override; |
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1036 |
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1037 private: |
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1038 bool coInitialized_; |
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1039 IMMDeviceEnumerator* deviceEnumerator_; |
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1040 |
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1041 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, |
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1042 unsigned int firstChannel, unsigned int sampleRate, |
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1043 RtAudioFormat format, unsigned int* bufferSize, |
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1044 RtAudio::StreamOptions* options ) override; |
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1045 |
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1046 static DWORD WINAPI runWasapiThread( void* wasapiPtr ); |
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1047 static DWORD WINAPI stopWasapiThread( void* wasapiPtr ); |
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1048 static DWORD WINAPI abortWasapiThread( void* wasapiPtr ); |
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1049 void wasapiThread(); |
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1050 }; |
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1051 |
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1052 #endif |
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1053 |
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1054 #if defined(__LINUX_ALSA__) |
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1055 |
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1056 class RtApiAlsa: public RtApi |
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1057 { |
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1058 public: |
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1059 |
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1060 RtApiAlsa(); |
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1061 ~RtApiAlsa(); |
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1062 RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_ALSA; } |
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1063 unsigned int getDeviceCount( void ) override; |
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1064 RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; |
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1065 void closeStream( void ) override; |
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1066 void startStream( void ) override; |
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1067 void stopStream( void ) override; |
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1068 void abortStream( void ) override; |
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1069 |
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1070 // This function is intended for internal use only. It must be |
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1071 // public because it is called by the internal callback handler, |
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1072 // which is not a member of RtAudio. External use of this function |
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1073 // will most likely produce highly undesireable results! |
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1074 void callbackEvent( void ); |
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1075 |
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1076 private: |
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1077 |
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1078 std::vector<RtAudio::DeviceInfo> devices_; |
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1079 void saveDeviceInfo( void ); |
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1080 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, |
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1081 unsigned int firstChannel, unsigned int sampleRate, |
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1082 RtAudioFormat format, unsigned int *bufferSize, |
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1083 RtAudio::StreamOptions *options ) override; |
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1084 }; |
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1085 |
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1086 #endif |
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1087 |
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1088 #if defined(__LINUX_PULSE__) |
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1089 |
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1090 class RtApiPulse: public RtApi |
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1091 { |
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1092 public: |
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1093 ~RtApiPulse(); |
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1094 RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_PULSE; } |
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1095 unsigned int getDeviceCount( void ) override; |
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1096 RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; |
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1097 void closeStream( void ) override; |
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1098 void startStream( void ) override; |
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1099 void stopStream( void ) override; |
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1100 void abortStream( void ) override; |
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1101 |
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1102 // This function is intended for internal use only. It must be |
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1103 // public because it is called by the internal callback handler, |
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1104 // which is not a member of RtAudio. External use of this function |
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1105 // will most likely produce highly undesireable results! |
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1106 void callbackEvent( void ); |
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1107 |
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1108 private: |
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1109 |
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1110 void collectDeviceInfo( void ); |
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1111 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, |
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1112 unsigned int firstChannel, unsigned int sampleRate, |
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1113 RtAudioFormat format, unsigned int *bufferSize, |
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1114 RtAudio::StreamOptions *options ) override; |
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1115 }; |
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1116 |
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1117 #endif |
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1118 |
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1119 #if defined(__LINUX_OSS__) |
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1120 |
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1121 class RtApiOss: public RtApi |
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1122 { |
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1123 public: |
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1124 |
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1125 RtApiOss(); |
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1126 ~RtApiOss(); |
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1127 RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_OSS; } |
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1128 unsigned int getDeviceCount( void ) override; |
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1129 RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; |
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1130 void closeStream( void ) override; |
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1131 void startStream( void ) override; |
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1132 void stopStream( void ) override; |
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1133 void abortStream( void ) override; |
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1134 |
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1135 // This function is intended for internal use only. It must be |
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1136 // public because it is called by the internal callback handler, |
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1137 // which is not a member of RtAudio. External use of this function |
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1138 // will most likely produce highly undesireable results! |
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1139 void callbackEvent( void ); |
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1140 |
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1141 private: |
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1142 |
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1143 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, |
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1144 unsigned int firstChannel, unsigned int sampleRate, |
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1145 RtAudioFormat format, unsigned int *bufferSize, |
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1146 RtAudio::StreamOptions *options ) override; |
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1147 }; |
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1148 |
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1149 #endif |
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1150 |
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1151 #if defined(__RTAUDIO_DUMMY__) |
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1152 |
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1153 class RtApiDummy: public RtApi |
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1154 { |
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1155 public: |
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1156 |
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1157 RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtAudioError::WARNING ); } |
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1158 RtAudio::Api getCurrentApi( void ) override { return RtAudio::RTAUDIO_DUMMY; } |
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1159 unsigned int getDeviceCount( void ) override { return 0; } |
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1160 RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) override { RtAudio::DeviceInfo info; return info; } |
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1161 void closeStream( void ) override {} |
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1162 void startStream( void ) override {} |
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1163 void stopStream( void ) override {} |
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1164 void abortStream( void ) override {} |
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1165 |
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1166 private: |
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1167 |
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1168 bool probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, |
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1169 unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, |
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1170 RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, |
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1171 RtAudio::StreamOptions * /*options*/ ) override { return false; } |
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1172 }; |
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1173 |
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1174 #endif |
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1175 |
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1176 #endif |
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1177 |
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1178 // Indentation settings for Vim and Emacs |
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1179 // |
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1180 // Local Variables: |
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1181 // c-basic-offset: 2 |
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1182 // indent-tabs-mode: nil |
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1183 // End: |
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1184 // |
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1185 // vim: et sts=2 sw=2 |