1 /************************************************************************/
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3 \brief Realtime audio i/o C++ classes.
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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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10 RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
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12 RtAudio: realtime audio i/o C++ classes
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13 Copyright (c) 2001-2016 Gary P. Scavone
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15 Permission is hereby granted, free of charge, to any person
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16 obtaining a copy of this software and associated documentation files
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17 (the "Software"), to deal in the Software without restriction,
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18 including without limitation the rights to use, copy, modify, merge,
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19 publish, distribute, sublicense, and/or sell copies of the Software,
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20 and to permit persons to whom the Software is furnished to do so,
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21 subject to the following conditions:
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23 The above copyright notice and this permission notice shall be
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24 included in all copies or substantial portions of the Software.
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26 Any person wishing to distribute modifications to the Software is
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27 asked to send the modifications to the original developer so that
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28 they can be incorporated into the canonical version. This is,
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29 however, not a binding provision of this license.
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31 THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
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32 EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
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33 MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
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34 IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
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35 ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
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36 CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
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37 WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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39 /************************************************************************/
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41 // RtAudio: Version 4.1.2
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43 #include "RtAudio.h"
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48 #include <algorithm>
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50 // Static variable definitions.
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51 const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
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52 const unsigned int RtApi::SAMPLE_RATES[] = {
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53 4000, 5512, 8000, 9600, 11025, 16000, 22050,
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54 32000, 44100, 48000, 88200, 96000, 176400, 192000
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57 #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
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58 #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
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59 #define MUTEX_DESTROY(A) DeleteCriticalSection(A)
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60 #define MUTEX_LOCK(A) EnterCriticalSection(A)
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61 #define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
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65 static std::string convertCharPointerToStdString(const char *text)
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67 return std::string(text);
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70 static std::string convertCharPointerToStdString(const wchar_t *text)
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72 int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);
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73 std::string s( length-1, '\0' );
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74 WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL);
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78 #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
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80 #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
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81 #define MUTEX_DESTROY(A) pthread_mutex_destroy(A)
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82 #define MUTEX_LOCK(A) pthread_mutex_lock(A)
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83 #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
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85 #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
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86 #define MUTEX_DESTROY(A) abs(*A) // dummy definitions
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89 // *************************************************** //
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91 // RtAudio definitions.
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93 // *************************************************** //
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95 std::string RtAudio :: getVersion( void ) throw()
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97 return RTAUDIO_VERSION;
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100 void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
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104 // The order here will control the order of RtAudio's API search in
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105 // the constructor.
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106 #if defined(__UNIX_JACK__)
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107 apis.push_back( UNIX_JACK );
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109 #if defined(__LINUX_ALSA__)
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110 apis.push_back( LINUX_ALSA );
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112 #if defined(__LINUX_PULSE__)
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113 apis.push_back( LINUX_PULSE );
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115 #if defined(__LINUX_OSS__)
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116 apis.push_back( LINUX_OSS );
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118 #if defined(__WINDOWS_ASIO__)
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119 apis.push_back( WINDOWS_ASIO );
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121 #if defined(__WINDOWS_WASAPI__)
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122 apis.push_back( WINDOWS_WASAPI );
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124 #if defined(__WINDOWS_DS__)
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125 apis.push_back( WINDOWS_DS );
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127 #if defined(__MACOSX_CORE__)
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128 apis.push_back( MACOSX_CORE );
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130 #if defined(__RTAUDIO_DUMMY__)
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131 apis.push_back( RTAUDIO_DUMMY );
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135 void RtAudio :: openRtApi( RtAudio::Api api )
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141 #if defined(__UNIX_JACK__)
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142 if ( api == UNIX_JACK )
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143 rtapi_ = new RtApiJack();
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145 #if defined(__LINUX_ALSA__)
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146 if ( api == LINUX_ALSA )
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147 rtapi_ = new RtApiAlsa();
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149 #if defined(__LINUX_PULSE__)
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150 if ( api == LINUX_PULSE )
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151 rtapi_ = new RtApiPulse();
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153 #if defined(__LINUX_OSS__)
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154 if ( api == LINUX_OSS )
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155 rtapi_ = new RtApiOss();
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157 #if defined(__WINDOWS_ASIO__)
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158 if ( api == WINDOWS_ASIO )
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159 rtapi_ = new RtApiAsio();
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161 #if defined(__WINDOWS_WASAPI__)
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162 if ( api == WINDOWS_WASAPI )
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163 rtapi_ = new RtApiWasapi();
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165 #if defined(__WINDOWS_DS__)
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166 if ( api == WINDOWS_DS )
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167 rtapi_ = new RtApiDs();
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169 #if defined(__MACOSX_CORE__)
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170 if ( api == MACOSX_CORE )
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171 rtapi_ = new RtApiCore();
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173 #if defined(__RTAUDIO_DUMMY__)
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174 if ( api == RTAUDIO_DUMMY )
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175 rtapi_ = new RtApiDummy();
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179 RtAudio :: RtAudio( RtAudio::Api api )
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183 if ( api != UNSPECIFIED ) {
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184 // Attempt to open the specified API.
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186 if ( rtapi_ ) return;
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188 // No compiled support for specified API value. Issue a debug
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189 // warning and continue as if no API was specified.
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190 std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
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193 // Iterate through the compiled APIs and return as soon as we find
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194 // one with at least one device or we reach the end of the list.
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195 std::vector< RtAudio::Api > apis;
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196 getCompiledApi( apis );
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197 for ( unsigned int i=0; i<apis.size(); i++ ) {
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198 openRtApi( apis[i] );
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199 if ( rtapi_ && rtapi_->getDeviceCount() ) break;
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202 if ( rtapi_ ) return;
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204 // It should not be possible to get here because the preprocessor
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205 // definition __RTAUDIO_DUMMY__ is automatically defined if no
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206 // API-specific definitions are passed to the compiler. But just in
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207 // case something weird happens, we'll thow an error.
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208 std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
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209 throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
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212 RtAudio :: ~RtAudio() throw()
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218 void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
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219 RtAudio::StreamParameters *inputParameters,
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220 RtAudioFormat format, unsigned int sampleRate,
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221 unsigned int *bufferFrames,
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222 RtAudioCallback callback, void *userData,
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223 RtAudio::StreamOptions *options,
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224 RtAudioErrorCallback errorCallback )
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226 return rtapi_->openStream( outputParameters, inputParameters, format,
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227 sampleRate, bufferFrames, callback,
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228 userData, options, errorCallback );
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231 // *************************************************** //
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233 // Public RtApi definitions (see end of file for
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234 // private or protected utility functions).
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236 // *************************************************** //
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240 stream_.state = STREAM_CLOSED;
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241 stream_.mode = UNINITIALIZED;
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242 stream_.apiHandle = 0;
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243 stream_.userBuffer[0] = 0;
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244 stream_.userBuffer[1] = 0;
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245 MUTEX_INITIALIZE( &stream_.mutex );
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246 showWarnings_ = true;
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247 firstErrorOccurred_ = false;
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252 MUTEX_DESTROY( &stream_.mutex );
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255 void RtApi :: openStream( RtAudio::StreamParameters *oParams,
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256 RtAudio::StreamParameters *iParams,
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257 RtAudioFormat format, unsigned int sampleRate,
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258 unsigned int *bufferFrames,
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259 RtAudioCallback callback, void *userData,
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260 RtAudio::StreamOptions *options,
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261 RtAudioErrorCallback errorCallback )
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263 if ( stream_.state != STREAM_CLOSED ) {
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264 errorText_ = "RtApi::openStream: a stream is already open!";
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265 error( RtAudioError::INVALID_USE );
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269 // Clear stream information potentially left from a previously open stream.
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272 if ( oParams && oParams->nChannels < 1 ) {
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273 errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
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274 error( RtAudioError::INVALID_USE );
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278 if ( iParams && iParams->nChannels < 1 ) {
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279 errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
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280 error( RtAudioError::INVALID_USE );
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284 if ( oParams == NULL && iParams == NULL ) {
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285 errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
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286 error( RtAudioError::INVALID_USE );
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290 if ( formatBytes(format) == 0 ) {
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291 errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
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292 error( RtAudioError::INVALID_USE );
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296 unsigned int nDevices = getDeviceCount();
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297 unsigned int oChannels = 0;
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299 oChannels = oParams->nChannels;
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300 if ( oParams->deviceId >= nDevices ) {
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301 errorText_ = "RtApi::openStream: output device parameter value is invalid.";
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302 error( RtAudioError::INVALID_USE );
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307 unsigned int iChannels = 0;
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309 iChannels = iParams->nChannels;
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310 if ( iParams->deviceId >= nDevices ) {
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311 errorText_ = "RtApi::openStream: input device parameter value is invalid.";
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312 error( RtAudioError::INVALID_USE );
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319 if ( oChannels > 0 ) {
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321 result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
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322 sampleRate, format, bufferFrames, options );
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323 if ( result == false ) {
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324 error( RtAudioError::SYSTEM_ERROR );
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329 if ( iChannels > 0 ) {
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331 result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
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332 sampleRate, format, bufferFrames, options );
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333 if ( result == false ) {
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334 if ( oChannels > 0 ) closeStream();
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335 error( RtAudioError::SYSTEM_ERROR );
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340 stream_.callbackInfo.callback = (void *) callback;
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341 stream_.callbackInfo.userData = userData;
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342 stream_.callbackInfo.errorCallback = (void *) errorCallback;
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344 if ( options ) options->numberOfBuffers = stream_.nBuffers;
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345 stream_.state = STREAM_STOPPED;
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348 unsigned int RtApi :: getDefaultInputDevice( void )
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350 // Should be implemented in subclasses if possible.
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354 unsigned int RtApi :: getDefaultOutputDevice( void )
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356 // Should be implemented in subclasses if possible.
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360 void RtApi :: closeStream( void )
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362 // MUST be implemented in subclasses!
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366 bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
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367 unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
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368 RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
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369 RtAudio::StreamOptions * /*options*/ )
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371 // MUST be implemented in subclasses!
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375 void RtApi :: tickStreamTime( void )
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377 // Subclasses that do not provide their own implementation of
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378 // getStreamTime should call this function once per buffer I/O to
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379 // provide basic stream time support.
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381 stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
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383 #if defined( HAVE_GETTIMEOFDAY )
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384 gettimeofday( &stream_.lastTickTimestamp, NULL );
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388 long RtApi :: getStreamLatency( void )
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392 long totalLatency = 0;
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393 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
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394 totalLatency = stream_.latency[0];
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395 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
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396 totalLatency += stream_.latency[1];
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398 return totalLatency;
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401 double RtApi :: getStreamTime( void )
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405 #if defined( HAVE_GETTIMEOFDAY )
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406 // Return a very accurate estimate of the stream time by
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407 // adding in the elapsed time since the last tick.
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408 struct timeval then;
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409 struct timeval now;
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411 if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
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412 return stream_.streamTime;
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414 gettimeofday( &now, NULL );
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415 then = stream_.lastTickTimestamp;
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416 return stream_.streamTime +
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417 ((now.tv_sec + 0.000001 * now.tv_usec) -
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418 (then.tv_sec + 0.000001 * then.tv_usec));
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420 return stream_.streamTime;
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424 void RtApi :: setStreamTime( double time )
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429 stream_.streamTime = time;
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432 unsigned int RtApi :: getStreamSampleRate( void )
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436 return stream_.sampleRate;
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440 // *************************************************** //
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442 // OS/API-specific methods.
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444 // *************************************************** //
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446 #if defined(__MACOSX_CORE__)
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448 // The OS X CoreAudio API is designed to use a separate callback
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449 // procedure for each of its audio devices. A single RtAudio duplex
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450 // stream using two different devices is supported here, though it
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451 // cannot be guaranteed to always behave correctly because we cannot
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452 // synchronize these two callbacks.
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454 // A property listener is installed for over/underrun information.
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455 // However, no functionality is currently provided to allow property
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456 // listeners to trigger user handlers because it is unclear what could
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457 // be done if a critical stream parameter (buffer size, sample rate,
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458 // device disconnect) notification arrived. The listeners entail
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459 // quite a bit of extra code and most likely, a user program wouldn't
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460 // be prepared for the result anyway. However, we do provide a flag
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461 // to the client callback function to inform of an over/underrun.
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463 // A structure to hold various information related to the CoreAudio API
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465 struct CoreHandle {
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466 AudioDeviceID id[2]; // device ids
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467 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
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468 AudioDeviceIOProcID procId[2];
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470 UInt32 iStream[2]; // device stream index (or first if using multiple)
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471 UInt32 nStreams[2]; // number of streams to use
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473 char *deviceBuffer;
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474 pthread_cond_t condition;
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475 int drainCounter; // Tracks callback counts when draining
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476 bool internalDrain; // Indicates if stop is initiated from callback or not.
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479 :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
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482 RtApiCore:: RtApiCore()
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484 #if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
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485 // This is a largely undocumented but absolutely necessary
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486 // requirement starting with OS-X 10.6. If not called, queries and
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487 // updates to various audio device properties are not handled
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489 CFRunLoopRef theRunLoop = NULL;
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490 AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
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491 kAudioObjectPropertyScopeGlobal,
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492 kAudioObjectPropertyElementMaster };
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493 OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
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494 if ( result != noErr ) {
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495 errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
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496 error( RtAudioError::WARNING );
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501 RtApiCore :: ~RtApiCore()
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503 // The subclass destructor gets called before the base class
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504 // destructor, so close an existing stream before deallocating
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505 // apiDeviceId memory.
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506 if ( stream_.state != STREAM_CLOSED ) closeStream();
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509 unsigned int RtApiCore :: getDeviceCount( void )
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511 // Find out how many audio devices there are, if any.
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513 AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
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514 OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
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515 if ( result != noErr ) {
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516 errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
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517 error( RtAudioError::WARNING );
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521 return dataSize / sizeof( AudioDeviceID );
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524 unsigned int RtApiCore :: getDefaultInputDevice( void )
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526 unsigned int nDevices = getDeviceCount();
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527 if ( nDevices <= 1 ) return 0;
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530 UInt32 dataSize = sizeof( AudioDeviceID );
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531 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
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532 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
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533 if ( result != noErr ) {
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534 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
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535 error( RtAudioError::WARNING );
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539 dataSize *= nDevices;
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540 AudioDeviceID deviceList[ nDevices ];
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541 property.mSelector = kAudioHardwarePropertyDevices;
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542 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
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543 if ( result != noErr ) {
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544 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
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545 error( RtAudioError::WARNING );
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549 for ( unsigned int i=0; i<nDevices; i++ )
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550 if ( id == deviceList[i] ) return i;
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552 errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
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553 error( RtAudioError::WARNING );
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557 unsigned int RtApiCore :: getDefaultOutputDevice( void )
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559 unsigned int nDevices = getDeviceCount();
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560 if ( nDevices <= 1 ) return 0;
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563 UInt32 dataSize = sizeof( AudioDeviceID );
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564 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
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565 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
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566 if ( result != noErr ) {
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567 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
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568 error( RtAudioError::WARNING );
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572 dataSize = sizeof( AudioDeviceID ) * nDevices;
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573 AudioDeviceID deviceList[ nDevices ];
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574 property.mSelector = kAudioHardwarePropertyDevices;
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575 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
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576 if ( result != noErr ) {
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577 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
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578 error( RtAudioError::WARNING );
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582 for ( unsigned int i=0; i<nDevices; i++ )
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583 if ( id == deviceList[i] ) return i;
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585 errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
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586 error( RtAudioError::WARNING );
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590 RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
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592 RtAudio::DeviceInfo info;
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593 info.probed = false;
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596 unsigned int nDevices = getDeviceCount();
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597 if ( nDevices == 0 ) {
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598 errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
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599 error( RtAudioError::INVALID_USE );
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603 if ( device >= nDevices ) {
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604 errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
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605 error( RtAudioError::INVALID_USE );
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609 AudioDeviceID deviceList[ nDevices ];
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610 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
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611 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
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612 kAudioObjectPropertyScopeGlobal,
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613 kAudioObjectPropertyElementMaster };
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614 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
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615 0, NULL, &dataSize, (void *) &deviceList );
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616 if ( result != noErr ) {
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617 errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
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618 error( RtAudioError::WARNING );
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622 AudioDeviceID id = deviceList[ device ];
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624 // Get the device name.
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626 CFStringRef cfname;
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627 dataSize = sizeof( CFStringRef );
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628 property.mSelector = kAudioObjectPropertyManufacturer;
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629 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
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630 if ( result != noErr ) {
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631 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
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632 errorText_ = errorStream_.str();
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633 error( RtAudioError::WARNING );
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637 //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
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638 int length = CFStringGetLength(cfname);
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639 char *mname = (char *)malloc(length * 3 + 1);
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640 #if defined( UNICODE ) || defined( _UNICODE )
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641 CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
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643 CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
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645 info.name.append( (const char *)mname, strlen(mname) );
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646 info.name.append( ": " );
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647 CFRelease( cfname );
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650 property.mSelector = kAudioObjectPropertyName;
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651 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
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652 if ( result != noErr ) {
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653 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
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654 errorText_ = errorStream_.str();
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655 error( RtAudioError::WARNING );
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659 //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
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660 length = CFStringGetLength(cfname);
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661 char *name = (char *)malloc(length * 3 + 1);
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662 #if defined( UNICODE ) || defined( _UNICODE )
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663 CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
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665 CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
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667 info.name.append( (const char *)name, strlen(name) );
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668 CFRelease( cfname );
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671 // Get the output stream "configuration".
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672 AudioBufferList *bufferList = nil;
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673 property.mSelector = kAudioDevicePropertyStreamConfiguration;
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674 property.mScope = kAudioDevicePropertyScopeOutput;
\r
675 // property.mElement = kAudioObjectPropertyElementWildcard;
\r
677 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
\r
678 if ( result != noErr || dataSize == 0 ) {
\r
679 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
\r
680 errorText_ = errorStream_.str();
\r
681 error( RtAudioError::WARNING );
\r
685 // Allocate the AudioBufferList.
\r
686 bufferList = (AudioBufferList *) malloc( dataSize );
\r
687 if ( bufferList == NULL ) {
\r
688 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
\r
689 error( RtAudioError::WARNING );
\r
693 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
\r
694 if ( result != noErr || dataSize == 0 ) {
\r
695 free( bufferList );
\r
696 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
\r
697 errorText_ = errorStream_.str();
\r
698 error( RtAudioError::WARNING );
\r
702 // Get output channel information.
\r
703 unsigned int i, nStreams = bufferList->mNumberBuffers;
\r
704 for ( i=0; i<nStreams; i++ )
\r
705 info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
\r
706 free( bufferList );
\r
708 // Get the input stream "configuration".
\r
709 property.mScope = kAudioDevicePropertyScopeInput;
\r
710 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
\r
711 if ( result != noErr || dataSize == 0 ) {
\r
712 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
\r
713 errorText_ = errorStream_.str();
\r
714 error( RtAudioError::WARNING );
\r
718 // Allocate the AudioBufferList.
\r
719 bufferList = (AudioBufferList *) malloc( dataSize );
\r
720 if ( bufferList == NULL ) {
\r
721 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
\r
722 error( RtAudioError::WARNING );
\r
726 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
\r
727 if (result != noErr || dataSize == 0) {
\r
728 free( bufferList );
\r
729 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
\r
730 errorText_ = errorStream_.str();
\r
731 error( RtAudioError::WARNING );
\r
735 // Get input channel information.
\r
736 nStreams = bufferList->mNumberBuffers;
\r
737 for ( i=0; i<nStreams; i++ )
\r
738 info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
\r
739 free( bufferList );
\r
741 // If device opens for both playback and capture, we determine the channels.
\r
742 if ( info.outputChannels > 0 && info.inputChannels > 0 )
\r
743 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
\r
745 // Probe the device sample rates.
\r
746 bool isInput = false;
\r
747 if ( info.outputChannels == 0 ) isInput = true;
\r
749 // Determine the supported sample rates.
\r
750 property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
\r
751 if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
\r
752 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
\r
753 if ( result != kAudioHardwareNoError || dataSize == 0 ) {
\r
754 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
\r
755 errorText_ = errorStream_.str();
\r
756 error( RtAudioError::WARNING );
\r
760 UInt32 nRanges = dataSize / sizeof( AudioValueRange );
\r
761 AudioValueRange rangeList[ nRanges ];
\r
762 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
\r
763 if ( result != kAudioHardwareNoError ) {
\r
764 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
\r
765 errorText_ = errorStream_.str();
\r
766 error( RtAudioError::WARNING );
\r
770 // The sample rate reporting mechanism is a bit of a mystery. It
\r
771 // seems that it can either return individual rates or a range of
\r
772 // rates. I assume that if the min / max range values are the same,
\r
773 // then that represents a single supported rate and if the min / max
\r
774 // range values are different, the device supports an arbitrary
\r
775 // range of values (though there might be multiple ranges, so we'll
\r
776 // use the most conservative range).
\r
777 Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
\r
778 bool haveValueRange = false;
\r
779 info.sampleRates.clear();
\r
780 for ( UInt32 i=0; i<nRanges; i++ ) {
\r
781 if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
\r
782 unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
\r
783 info.sampleRates.push_back( tmpSr );
\r
785 if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
\r
786 info.preferredSampleRate = tmpSr;
\r
789 haveValueRange = true;
\r
790 if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
\r
791 if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
\r
795 if ( haveValueRange ) {
\r
796 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
\r
797 if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) {
\r
798 info.sampleRates.push_back( SAMPLE_RATES[k] );
\r
800 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
\r
801 info.preferredSampleRate = SAMPLE_RATES[k];
\r
806 // Sort and remove any redundant values
\r
807 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
\r
808 info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
\r
810 if ( info.sampleRates.size() == 0 ) {
\r
811 errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
\r
812 errorText_ = errorStream_.str();
\r
813 error( RtAudioError::WARNING );
\r
817 // CoreAudio always uses 32-bit floating point data for PCM streams.
\r
818 // Thus, any other "physical" formats supported by the device are of
\r
819 // no interest to the client.
\r
820 info.nativeFormats = RTAUDIO_FLOAT32;
\r
822 if ( info.outputChannels > 0 )
\r
823 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
\r
824 if ( info.inputChannels > 0 )
\r
825 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
\r
827 info.probed = true;
\r
831 static OSStatus callbackHandler( AudioDeviceID inDevice,
\r
832 const AudioTimeStamp* /*inNow*/,
\r
833 const AudioBufferList* inInputData,
\r
834 const AudioTimeStamp* /*inInputTime*/,
\r
835 AudioBufferList* outOutputData,
\r
836 const AudioTimeStamp* /*inOutputTime*/,
\r
837 void* infoPointer )
\r
839 CallbackInfo *info = (CallbackInfo *) infoPointer;
\r
841 RtApiCore *object = (RtApiCore *) info->object;
\r
842 if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
\r
843 return kAudioHardwareUnspecifiedError;
\r
845 return kAudioHardwareNoError;
\r
848 static OSStatus xrunListener( AudioObjectID /*inDevice*/,
\r
850 const AudioObjectPropertyAddress properties[],
\r
851 void* handlePointer )
\r
853 CoreHandle *handle = (CoreHandle *) handlePointer;
\r
854 for ( UInt32 i=0; i<nAddresses; i++ ) {
\r
855 if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
\r
856 if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
\r
857 handle->xrun[1] = true;
\r
859 handle->xrun[0] = true;
\r
863 return kAudioHardwareNoError;
\r
866 static OSStatus rateListener( AudioObjectID inDevice,
\r
867 UInt32 /*nAddresses*/,
\r
868 const AudioObjectPropertyAddress /*properties*/[],
\r
869 void* ratePointer )
\r
871 Float64 *rate = (Float64 *) ratePointer;
\r
872 UInt32 dataSize = sizeof( Float64 );
\r
873 AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
\r
874 kAudioObjectPropertyScopeGlobal,
\r
875 kAudioObjectPropertyElementMaster };
\r
876 AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
\r
877 return kAudioHardwareNoError;
\r
880 bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
881 unsigned int firstChannel, unsigned int sampleRate,
\r
882 RtAudioFormat format, unsigned int *bufferSize,
\r
883 RtAudio::StreamOptions *options )
\r
886 unsigned int nDevices = getDeviceCount();
\r
887 if ( nDevices == 0 ) {
\r
888 // This should not happen because a check is made before this function is called.
\r
889 errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
\r
893 if ( device >= nDevices ) {
\r
894 // This should not happen because a check is made before this function is called.
\r
895 errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
\r
899 AudioDeviceID deviceList[ nDevices ];
\r
900 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
\r
901 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
\r
902 kAudioObjectPropertyScopeGlobal,
\r
903 kAudioObjectPropertyElementMaster };
\r
904 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
\r
905 0, NULL, &dataSize, (void *) &deviceList );
\r
906 if ( result != noErr ) {
\r
907 errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
\r
911 AudioDeviceID id = deviceList[ device ];
\r
913 // Setup for stream mode.
\r
914 bool isInput = false;
\r
915 if ( mode == INPUT ) {
\r
917 property.mScope = kAudioDevicePropertyScopeInput;
\r
920 property.mScope = kAudioDevicePropertyScopeOutput;
\r
922 // Get the stream "configuration".
\r
923 AudioBufferList *bufferList = nil;
\r
925 property.mSelector = kAudioDevicePropertyStreamConfiguration;
\r
926 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
\r
927 if ( result != noErr || dataSize == 0 ) {
\r
928 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
\r
929 errorText_ = errorStream_.str();
\r
933 // Allocate the AudioBufferList.
\r
934 bufferList = (AudioBufferList *) malloc( dataSize );
\r
935 if ( bufferList == NULL ) {
\r
936 errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
\r
940 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
\r
941 if (result != noErr || dataSize == 0) {
\r
942 free( bufferList );
\r
943 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
\r
944 errorText_ = errorStream_.str();
\r
948 // Search for one or more streams that contain the desired number of
\r
949 // channels. CoreAudio devices can have an arbitrary number of
\r
950 // streams and each stream can have an arbitrary number of channels.
\r
951 // For each stream, a single buffer of interleaved samples is
\r
952 // provided. RtAudio prefers the use of one stream of interleaved
\r
953 // data or multiple consecutive single-channel streams. However, we
\r
954 // now support multiple consecutive multi-channel streams of
\r
955 // interleaved data as well.
\r
956 UInt32 iStream, offsetCounter = firstChannel;
\r
957 UInt32 nStreams = bufferList->mNumberBuffers;
\r
958 bool monoMode = false;
\r
959 bool foundStream = false;
\r
961 // First check that the device supports the requested number of
\r
963 UInt32 deviceChannels = 0;
\r
964 for ( iStream=0; iStream<nStreams; iStream++ )
\r
965 deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
\r
967 if ( deviceChannels < ( channels + firstChannel ) ) {
\r
968 free( bufferList );
\r
969 errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
\r
970 errorText_ = errorStream_.str();
\r
974 // Look for a single stream meeting our needs.
\r
975 UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
\r
976 for ( iStream=0; iStream<nStreams; iStream++ ) {
\r
977 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
\r
978 if ( streamChannels >= channels + offsetCounter ) {
\r
979 firstStream = iStream;
\r
980 channelOffset = offsetCounter;
\r
981 foundStream = true;
\r
984 if ( streamChannels > offsetCounter ) break;
\r
985 offsetCounter -= streamChannels;
\r
988 // If we didn't find a single stream above, then we should be able
\r
989 // to meet the channel specification with multiple streams.
\r
990 if ( foundStream == false ) {
\r
992 offsetCounter = firstChannel;
\r
993 for ( iStream=0; iStream<nStreams; iStream++ ) {
\r
994 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
\r
995 if ( streamChannels > offsetCounter ) break;
\r
996 offsetCounter -= streamChannels;
\r
999 firstStream = iStream;
\r
1000 channelOffset = offsetCounter;
\r
1001 Int32 channelCounter = channels + offsetCounter - streamChannels;
\r
1003 if ( streamChannels > 1 ) monoMode = false;
\r
1004 while ( channelCounter > 0 ) {
\r
1005 streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
\r
1006 if ( streamChannels > 1 ) monoMode = false;
\r
1007 channelCounter -= streamChannels;
\r
1012 free( bufferList );
\r
1014 // Determine the buffer size.
\r
1015 AudioValueRange bufferRange;
\r
1016 dataSize = sizeof( AudioValueRange );
\r
1017 property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
\r
1018 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
\r
1020 if ( result != noErr ) {
\r
1021 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
\r
1022 errorText_ = errorStream_.str();
\r
1026 if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
\r
1027 else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
\r
1028 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
\r
1030 // Set the buffer size. For multiple streams, I'm assuming we only
\r
1031 // need to make this setting for the master channel.
\r
1032 UInt32 theSize = (UInt32) *bufferSize;
\r
1033 dataSize = sizeof( UInt32 );
\r
1034 property.mSelector = kAudioDevicePropertyBufferFrameSize;
\r
1035 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
\r
1037 if ( result != noErr ) {
\r
1038 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
\r
1039 errorText_ = errorStream_.str();
\r
1043 // If attempting to setup a duplex stream, the bufferSize parameter
\r
1044 // MUST be the same in both directions!
\r
1045 *bufferSize = theSize;
\r
1046 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
\r
1047 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
\r
1048 errorText_ = errorStream_.str();
\r
1052 stream_.bufferSize = *bufferSize;
\r
1053 stream_.nBuffers = 1;
\r
1055 // Try to set "hog" mode ... it's not clear to me this is working.
\r
1056 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
\r
1058 dataSize = sizeof( hog_pid );
\r
1059 property.mSelector = kAudioDevicePropertyHogMode;
\r
1060 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
\r
1061 if ( result != noErr ) {
\r
1062 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
\r
1063 errorText_ = errorStream_.str();
\r
1067 if ( hog_pid != getpid() ) {
\r
1068 hog_pid = getpid();
\r
1069 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
\r
1070 if ( result != noErr ) {
\r
1071 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
\r
1072 errorText_ = errorStream_.str();
\r
1078 // Check and if necessary, change the sample rate for the device.
\r
1079 Float64 nominalRate;
\r
1080 dataSize = sizeof( Float64 );
\r
1081 property.mSelector = kAudioDevicePropertyNominalSampleRate;
\r
1082 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
\r
1083 if ( result != noErr ) {
\r
1084 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
\r
1085 errorText_ = errorStream_.str();
\r
1089 // Only change the sample rate if off by more than 1 Hz.
\r
1090 if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
\r
1092 // Set a property listener for the sample rate change
\r
1093 Float64 reportedRate = 0.0;
\r
1094 AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
\r
1095 result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
\r
1096 if ( result != noErr ) {
\r
1097 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
\r
1098 errorText_ = errorStream_.str();
\r
1102 nominalRate = (Float64) sampleRate;
\r
1103 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
\r
1104 if ( result != noErr ) {
\r
1105 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
\r
1106 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
\r
1107 errorText_ = errorStream_.str();
\r
1111 // Now wait until the reported nominal rate is what we just set.
\r
1112 UInt32 microCounter = 0;
\r
1113 while ( reportedRate != nominalRate ) {
\r
1114 microCounter += 5000;
\r
1115 if ( microCounter > 5000000 ) break;
\r
1119 // Remove the property listener.
\r
1120 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
\r
1122 if ( microCounter > 5000000 ) {
\r
1123 errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
\r
1124 errorText_ = errorStream_.str();
\r
1129 // Now set the stream format for all streams. Also, check the
\r
1130 // physical format of the device and change that if necessary.
\r
1131 AudioStreamBasicDescription description;
\r
1132 dataSize = sizeof( AudioStreamBasicDescription );
\r
1133 property.mSelector = kAudioStreamPropertyVirtualFormat;
\r
1134 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
\r
1135 if ( result != noErr ) {
\r
1136 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
\r
1137 errorText_ = errorStream_.str();
\r
1141 // Set the sample rate and data format id. However, only make the
\r
1142 // change if the sample rate is not within 1.0 of the desired
\r
1143 // rate and the format is not linear pcm.
\r
1144 bool updateFormat = false;
\r
1145 if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
\r
1146 description.mSampleRate = (Float64) sampleRate;
\r
1147 updateFormat = true;
\r
1150 if ( description.mFormatID != kAudioFormatLinearPCM ) {
\r
1151 description.mFormatID = kAudioFormatLinearPCM;
\r
1152 updateFormat = true;
\r
1155 if ( updateFormat ) {
\r
1156 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
\r
1157 if ( result != noErr ) {
\r
1158 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
\r
1159 errorText_ = errorStream_.str();
\r
1164 // Now check the physical format.
\r
1165 property.mSelector = kAudioStreamPropertyPhysicalFormat;
\r
1166 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
\r
1167 if ( result != noErr ) {
\r
1168 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
\r
1169 errorText_ = errorStream_.str();
\r
1173 //std::cout << "Current physical stream format:" << std::endl;
\r
1174 //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
\r
1175 //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
\r
1176 //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
\r
1177 //std::cout << " sample rate = " << description.mSampleRate << std::endl;
\r
1179 if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
\r
1180 description.mFormatID = kAudioFormatLinearPCM;
\r
1181 //description.mSampleRate = (Float64) sampleRate;
\r
1182 AudioStreamBasicDescription testDescription = description;
\r
1183 UInt32 formatFlags;
\r
1185 // We'll try higher bit rates first and then work our way down.
\r
1186 std::vector< std::pair<UInt32, UInt32> > physicalFormats;
\r
1187 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
\r
1188 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
\r
1189 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
\r
1190 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
\r
1191 physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed
\r
1192 formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
\r
1193 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
\r
1194 formatFlags |= kAudioFormatFlagIsAlignedHigh;
\r
1195 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
\r
1196 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
\r
1197 physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
\r
1198 physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
\r
1200 bool setPhysicalFormat = false;
\r
1201 for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
\r
1202 testDescription = description;
\r
1203 testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
\r
1204 testDescription.mFormatFlags = physicalFormats[i].second;
\r
1205 if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
\r
1206 testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
\r
1208 testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
\r
1209 testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
\r
1210 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
\r
1211 if ( result == noErr ) {
\r
1212 setPhysicalFormat = true;
\r
1213 //std::cout << "Updated physical stream format:" << std::endl;
\r
1214 //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
\r
1215 //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
\r
1216 //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
\r
1217 //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
\r
1222 if ( !setPhysicalFormat ) {
\r
1223 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
\r
1224 errorText_ = errorStream_.str();
\r
1227 } // done setting virtual/physical formats.
\r
1229 // Get the stream / device latency.
\r
1231 dataSize = sizeof( UInt32 );
\r
1232 property.mSelector = kAudioDevicePropertyLatency;
\r
1233 if ( AudioObjectHasProperty( id, &property ) == true ) {
\r
1234 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
\r
1235 if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
\r
1237 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
\r
1238 errorText_ = errorStream_.str();
\r
1239 error( RtAudioError::WARNING );
\r
1243 // Byte-swapping: According to AudioHardware.h, the stream data will
\r
1244 // always be presented in native-endian format, so we should never
\r
1245 // need to byte swap.
\r
1246 stream_.doByteSwap[mode] = false;
\r
1248 // From the CoreAudio documentation, PCM data must be supplied as
\r
1250 stream_.userFormat = format;
\r
1251 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
\r
1253 if ( streamCount == 1 )
\r
1254 stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
\r
1255 else // multiple streams
\r
1256 stream_.nDeviceChannels[mode] = channels;
\r
1257 stream_.nUserChannels[mode] = channels;
\r
1258 stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
\r
1259 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
\r
1260 else stream_.userInterleaved = true;
\r
1261 stream_.deviceInterleaved[mode] = true;
\r
1262 if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
\r
1264 // Set flags for buffer conversion.
\r
1265 stream_.doConvertBuffer[mode] = false;
\r
1266 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
1267 stream_.doConvertBuffer[mode] = true;
\r
1268 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
\r
1269 stream_.doConvertBuffer[mode] = true;
\r
1270 if ( streamCount == 1 ) {
\r
1271 if ( stream_.nUserChannels[mode] > 1 &&
\r
1272 stream_.userInterleaved != stream_.deviceInterleaved[mode] )
\r
1273 stream_.doConvertBuffer[mode] = true;
\r
1275 else if ( monoMode && stream_.userInterleaved )
\r
1276 stream_.doConvertBuffer[mode] = true;
\r
1278 // Allocate our CoreHandle structure for the stream.
\r
1279 CoreHandle *handle = 0;
\r
1280 if ( stream_.apiHandle == 0 ) {
\r
1282 handle = new CoreHandle;
\r
1284 catch ( std::bad_alloc& ) {
\r
1285 errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
\r
1289 if ( pthread_cond_init( &handle->condition, NULL ) ) {
\r
1290 errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
\r
1293 stream_.apiHandle = (void *) handle;
\r
1296 handle = (CoreHandle *) stream_.apiHandle;
\r
1297 handle->iStream[mode] = firstStream;
\r
1298 handle->nStreams[mode] = streamCount;
\r
1299 handle->id[mode] = id;
\r
1301 // Allocate necessary internal buffers.
\r
1302 unsigned long bufferBytes;
\r
1303 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
1304 // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
1305 stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
\r
1306 memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
\r
1307 if ( stream_.userBuffer[mode] == NULL ) {
\r
1308 errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
\r
1312 // If possible, we will make use of the CoreAudio stream buffers as
\r
1313 // "device buffers". However, we can't do this if using multiple
\r
1315 if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
\r
1317 bool makeBuffer = true;
\r
1318 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
\r
1319 if ( mode == INPUT ) {
\r
1320 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
1321 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
1322 if ( bufferBytes <= bytesOut ) makeBuffer = false;
\r
1326 if ( makeBuffer ) {
\r
1327 bufferBytes *= *bufferSize;
\r
1328 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
1329 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
1330 if ( stream_.deviceBuffer == NULL ) {
\r
1331 errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
\r
1337 stream_.sampleRate = sampleRate;
\r
1338 stream_.device[mode] = device;
\r
1339 stream_.state = STREAM_STOPPED;
\r
1340 stream_.callbackInfo.object = (void *) this;
\r
1342 // Setup the buffer conversion information structure.
\r
1343 if ( stream_.doConvertBuffer[mode] ) {
\r
1344 if ( streamCount > 1 ) setConvertInfo( mode, 0 );
\r
1345 else setConvertInfo( mode, channelOffset );
\r
1348 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
\r
1349 // Only one callback procedure per device.
\r
1350 stream_.mode = DUPLEX;
\r
1352 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
\r
1353 result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
\r
1355 // deprecated in favor of AudioDeviceCreateIOProcID()
\r
1356 result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
\r
1358 if ( result != noErr ) {
\r
1359 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
\r
1360 errorText_ = errorStream_.str();
\r
1363 if ( stream_.mode == OUTPUT && mode == INPUT )
\r
1364 stream_.mode = DUPLEX;
\r
1366 stream_.mode = mode;
\r
1369 // Setup the device property listener for over/underload.
\r
1370 property.mSelector = kAudioDeviceProcessorOverload;
\r
1371 property.mScope = kAudioObjectPropertyScopeGlobal;
\r
1372 result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
\r
1378 pthread_cond_destroy( &handle->condition );
\r
1380 stream_.apiHandle = 0;
\r
1383 for ( int i=0; i<2; i++ ) {
\r
1384 if ( stream_.userBuffer[i] ) {
\r
1385 free( stream_.userBuffer[i] );
\r
1386 stream_.userBuffer[i] = 0;
\r
1390 if ( stream_.deviceBuffer ) {
\r
1391 free( stream_.deviceBuffer );
\r
1392 stream_.deviceBuffer = 0;
\r
1395 stream_.state = STREAM_CLOSED;
\r
1399 void RtApiCore :: closeStream( void )
\r
1401 if ( stream_.state == STREAM_CLOSED ) {
\r
1402 errorText_ = "RtApiCore::closeStream(): no open stream to close!";
\r
1403 error( RtAudioError::WARNING );
\r
1407 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
\r
1408 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
1410 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
\r
1411 kAudioObjectPropertyScopeGlobal,
\r
1412 kAudioObjectPropertyElementMaster };
\r
1414 property.mSelector = kAudioDeviceProcessorOverload;
\r
1415 property.mScope = kAudioObjectPropertyScopeGlobal;
\r
1416 if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
\r
1417 errorText_ = "RtApiCore::closeStream(): error removing property listener!";
\r
1418 error( RtAudioError::WARNING );
\r
1421 if ( stream_.state == STREAM_RUNNING )
\r
1422 AudioDeviceStop( handle->id[0], callbackHandler );
\r
1423 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
\r
1424 AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
\r
1426 // deprecated in favor of AudioDeviceDestroyIOProcID()
\r
1427 AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
\r
1431 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
\r
1433 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
\r
1434 kAudioObjectPropertyScopeGlobal,
\r
1435 kAudioObjectPropertyElementMaster };
\r
1437 property.mSelector = kAudioDeviceProcessorOverload;
\r
1438 property.mScope = kAudioObjectPropertyScopeGlobal;
\r
1439 if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
\r
1440 errorText_ = "RtApiCore::closeStream(): error removing property listener!";
\r
1441 error( RtAudioError::WARNING );
\r
1444 if ( stream_.state == STREAM_RUNNING )
\r
1445 AudioDeviceStop( handle->id[1], callbackHandler );
\r
1446 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
\r
1447 AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
\r
1449 // deprecated in favor of AudioDeviceDestroyIOProcID()
\r
1450 AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
\r
1454 for ( int i=0; i<2; i++ ) {
\r
1455 if ( stream_.userBuffer[i] ) {
\r
1456 free( stream_.userBuffer[i] );
\r
1457 stream_.userBuffer[i] = 0;
\r
1461 if ( stream_.deviceBuffer ) {
\r
1462 free( stream_.deviceBuffer );
\r
1463 stream_.deviceBuffer = 0;
\r
1466 // Destroy pthread condition variable.
\r
1467 pthread_cond_destroy( &handle->condition );
\r
1469 stream_.apiHandle = 0;
\r
1471 stream_.mode = UNINITIALIZED;
\r
1472 stream_.state = STREAM_CLOSED;
\r
1475 void RtApiCore :: startStream( void )
\r
1478 if ( stream_.state == STREAM_RUNNING ) {
\r
1479 errorText_ = "RtApiCore::startStream(): the stream is already running!";
\r
1480 error( RtAudioError::WARNING );
\r
1484 OSStatus result = noErr;
\r
1485 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
\r
1486 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
1488 result = AudioDeviceStart( handle->id[0], callbackHandler );
\r
1489 if ( result != noErr ) {
\r
1490 errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
\r
1491 errorText_ = errorStream_.str();
\r
1496 if ( stream_.mode == INPUT ||
\r
1497 ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
\r
1499 result = AudioDeviceStart( handle->id[1], callbackHandler );
\r
1500 if ( result != noErr ) {
\r
1501 errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
\r
1502 errorText_ = errorStream_.str();
\r
1507 handle->drainCounter = 0;
\r
1508 handle->internalDrain = false;
\r
1509 stream_.state = STREAM_RUNNING;
\r
1512 if ( result == noErr ) return;
\r
1513 error( RtAudioError::SYSTEM_ERROR );
\r
1516 void RtApiCore :: stopStream( void )
\r
1519 if ( stream_.state == STREAM_STOPPED ) {
\r
1520 errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
\r
1521 error( RtAudioError::WARNING );
\r
1525 OSStatus result = noErr;
\r
1526 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
\r
1527 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
1529 if ( handle->drainCounter == 0 ) {
\r
1530 handle->drainCounter = 2;
\r
1531 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
\r
1534 result = AudioDeviceStop( handle->id[0], callbackHandler );
\r
1535 if ( result != noErr ) {
\r
1536 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
\r
1537 errorText_ = errorStream_.str();
\r
1542 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
\r
1544 result = AudioDeviceStop( handle->id[1], callbackHandler );
\r
1545 if ( result != noErr ) {
\r
1546 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
\r
1547 errorText_ = errorStream_.str();
\r
1552 stream_.state = STREAM_STOPPED;
\r
1555 if ( result == noErr ) return;
\r
1556 error( RtAudioError::SYSTEM_ERROR );
\r
1559 void RtApiCore :: abortStream( void )
\r
1562 if ( stream_.state == STREAM_STOPPED ) {
\r
1563 errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
\r
1564 error( RtAudioError::WARNING );
\r
1568 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
\r
1569 handle->drainCounter = 2;
\r
1574 // This function will be called by a spawned thread when the user
\r
1575 // callback function signals that the stream should be stopped or
\r
1576 // aborted. It is better to handle it this way because the
\r
1577 // callbackEvent() function probably should return before the AudioDeviceStop()
\r
1578 // function is called.
\r
1579 static void *coreStopStream( void *ptr )
\r
1581 CallbackInfo *info = (CallbackInfo *) ptr;
\r
1582 RtApiCore *object = (RtApiCore *) info->object;
\r
1584 object->stopStream();
\r
1585 pthread_exit( NULL );
\r
1588 bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
\r
1589 const AudioBufferList *inBufferList,
\r
1590 const AudioBufferList *outBufferList )
\r
1592 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
\r
1593 if ( stream_.state == STREAM_CLOSED ) {
\r
1594 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
\r
1595 error( RtAudioError::WARNING );
\r
1599 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
\r
1600 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
\r
1602 // Check if we were draining the stream and signal is finished.
\r
1603 if ( handle->drainCounter > 3 ) {
\r
1604 ThreadHandle threadId;
\r
1606 stream_.state = STREAM_STOPPING;
\r
1607 if ( handle->internalDrain == true )
\r
1608 pthread_create( &threadId, NULL, coreStopStream, info );
\r
1609 else // external call to stopStream()
\r
1610 pthread_cond_signal( &handle->condition );
\r
1614 AudioDeviceID outputDevice = handle->id[0];
\r
1616 // Invoke user callback to get fresh output data UNLESS we are
\r
1617 // draining stream or duplex mode AND the input/output devices are
\r
1618 // different AND this function is called for the input device.
\r
1619 if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
\r
1620 RtAudioCallback callback = (RtAudioCallback) info->callback;
\r
1621 double streamTime = getStreamTime();
\r
1622 RtAudioStreamStatus status = 0;
\r
1623 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
\r
1624 status |= RTAUDIO_OUTPUT_UNDERFLOW;
\r
1625 handle->xrun[0] = false;
\r
1627 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
\r
1628 status |= RTAUDIO_INPUT_OVERFLOW;
\r
1629 handle->xrun[1] = false;
\r
1632 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
\r
1633 stream_.bufferSize, streamTime, status, info->userData );
\r
1634 if ( cbReturnValue == 2 ) {
\r
1635 stream_.state = STREAM_STOPPING;
\r
1636 handle->drainCounter = 2;
\r
1640 else if ( cbReturnValue == 1 ) {
\r
1641 handle->drainCounter = 1;
\r
1642 handle->internalDrain = true;
\r
1646 if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
\r
1648 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
\r
1650 if ( handle->nStreams[0] == 1 ) {
\r
1651 memset( outBufferList->mBuffers[handle->iStream[0]].mData,
\r
1653 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
\r
1655 else { // fill multiple streams with zeros
\r
1656 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
\r
1657 memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
\r
1659 outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
\r
1663 else if ( handle->nStreams[0] == 1 ) {
\r
1664 if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
\r
1665 convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
\r
1666 stream_.userBuffer[0], stream_.convertInfo[0] );
\r
1668 else { // copy from user buffer
\r
1669 memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
\r
1670 stream_.userBuffer[0],
\r
1671 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
\r
1674 else { // fill multiple streams
\r
1675 Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
\r
1676 if ( stream_.doConvertBuffer[0] ) {
\r
1677 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
\r
1678 inBuffer = (Float32 *) stream_.deviceBuffer;
\r
1681 if ( stream_.deviceInterleaved[0] == false ) { // mono mode
\r
1682 UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
\r
1683 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
\r
1684 memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
\r
1685 (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
\r
1688 else { // fill multiple multi-channel streams with interleaved data
\r
1689 UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
\r
1690 Float32 *out, *in;
\r
1692 bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
\r
1693 UInt32 inChannels = stream_.nUserChannels[0];
\r
1694 if ( stream_.doConvertBuffer[0] ) {
\r
1695 inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
\r
1696 inChannels = stream_.nDeviceChannels[0];
\r
1699 if ( inInterleaved ) inOffset = 1;
\r
1700 else inOffset = stream_.bufferSize;
\r
1702 channelsLeft = inChannels;
\r
1703 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
\r
1705 out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
\r
1706 streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
\r
1709 // Account for possible channel offset in first stream
\r
1710 if ( i == 0 && stream_.channelOffset[0] > 0 ) {
\r
1711 streamChannels -= stream_.channelOffset[0];
\r
1712 outJump = stream_.channelOffset[0];
\r
1716 // Account for possible unfilled channels at end of the last stream
\r
1717 if ( streamChannels > channelsLeft ) {
\r
1718 outJump = streamChannels - channelsLeft;
\r
1719 streamChannels = channelsLeft;
\r
1722 // Determine input buffer offsets and skips
\r
1723 if ( inInterleaved ) {
\r
1724 inJump = inChannels;
\r
1725 in += inChannels - channelsLeft;
\r
1729 in += (inChannels - channelsLeft) * inOffset;
\r
1732 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
\r
1733 for ( unsigned int j=0; j<streamChannels; j++ ) {
\r
1734 *out++ = in[j*inOffset];
\r
1739 channelsLeft -= streamChannels;
\r
1745 // Don't bother draining input
\r
1746 if ( handle->drainCounter ) {
\r
1747 handle->drainCounter++;
\r
1751 AudioDeviceID inputDevice;
\r
1752 inputDevice = handle->id[1];
\r
1753 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
\r
1755 if ( handle->nStreams[1] == 1 ) {
\r
1756 if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
\r
1757 convertBuffer( stream_.userBuffer[1],
\r
1758 (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
\r
1759 stream_.convertInfo[1] );
\r
1761 else { // copy to user buffer
\r
1762 memcpy( stream_.userBuffer[1],
\r
1763 inBufferList->mBuffers[handle->iStream[1]].mData,
\r
1764 inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
\r
1767 else { // read from multiple streams
\r
1768 Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
\r
1769 if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
\r
1771 if ( stream_.deviceInterleaved[1] == false ) { // mono mode
\r
1772 UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
\r
1773 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
\r
1774 memcpy( (void *)&outBuffer[i*stream_.bufferSize],
\r
1775 inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
\r
1778 else { // read from multiple multi-channel streams
\r
1779 UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
\r
1780 Float32 *out, *in;
\r
1782 bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
\r
1783 UInt32 outChannels = stream_.nUserChannels[1];
\r
1784 if ( stream_.doConvertBuffer[1] ) {
\r
1785 outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
\r
1786 outChannels = stream_.nDeviceChannels[1];
\r
1789 if ( outInterleaved ) outOffset = 1;
\r
1790 else outOffset = stream_.bufferSize;
\r
1792 channelsLeft = outChannels;
\r
1793 for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
\r
1795 in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
\r
1796 streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
\r
1799 // Account for possible channel offset in first stream
\r
1800 if ( i == 0 && stream_.channelOffset[1] > 0 ) {
\r
1801 streamChannels -= stream_.channelOffset[1];
\r
1802 inJump = stream_.channelOffset[1];
\r
1806 // Account for possible unread channels at end of the last stream
\r
1807 if ( streamChannels > channelsLeft ) {
\r
1808 inJump = streamChannels - channelsLeft;
\r
1809 streamChannels = channelsLeft;
\r
1812 // Determine output buffer offsets and skips
\r
1813 if ( outInterleaved ) {
\r
1814 outJump = outChannels;
\r
1815 out += outChannels - channelsLeft;
\r
1819 out += (outChannels - channelsLeft) * outOffset;
\r
1822 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
\r
1823 for ( unsigned int j=0; j<streamChannels; j++ ) {
\r
1824 out[j*outOffset] = *in++;
\r
1829 channelsLeft -= streamChannels;
\r
1833 if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
\r
1834 convertBuffer( stream_.userBuffer[1],
\r
1835 stream_.deviceBuffer,
\r
1836 stream_.convertInfo[1] );
\r
1842 //MUTEX_UNLOCK( &stream_.mutex );
\r
1844 RtApi::tickStreamTime();
\r
1848 const char* RtApiCore :: getErrorCode( OSStatus code )
\r
1852 case kAudioHardwareNotRunningError:
\r
1853 return "kAudioHardwareNotRunningError";
\r
1855 case kAudioHardwareUnspecifiedError:
\r
1856 return "kAudioHardwareUnspecifiedError";
\r
1858 case kAudioHardwareUnknownPropertyError:
\r
1859 return "kAudioHardwareUnknownPropertyError";
\r
1861 case kAudioHardwareBadPropertySizeError:
\r
1862 return "kAudioHardwareBadPropertySizeError";
\r
1864 case kAudioHardwareIllegalOperationError:
\r
1865 return "kAudioHardwareIllegalOperationError";
\r
1867 case kAudioHardwareBadObjectError:
\r
1868 return "kAudioHardwareBadObjectError";
\r
1870 case kAudioHardwareBadDeviceError:
\r
1871 return "kAudioHardwareBadDeviceError";
\r
1873 case kAudioHardwareBadStreamError:
\r
1874 return "kAudioHardwareBadStreamError";
\r
1876 case kAudioHardwareUnsupportedOperationError:
\r
1877 return "kAudioHardwareUnsupportedOperationError";
\r
1879 case kAudioDeviceUnsupportedFormatError:
\r
1880 return "kAudioDeviceUnsupportedFormatError";
\r
1882 case kAudioDevicePermissionsError:
\r
1883 return "kAudioDevicePermissionsError";
\r
1886 return "CoreAudio unknown error";
\r
1890 //******************** End of __MACOSX_CORE__ *********************//
\r
1893 #if defined(__UNIX_JACK__)
\r
1895 // JACK is a low-latency audio server, originally written for the
\r
1896 // GNU/Linux operating system and now also ported to OS-X. It can
\r
1897 // connect a number of different applications to an audio device, as
\r
1898 // well as allowing them to share audio between themselves.
\r
1900 // When using JACK with RtAudio, "devices" refer to JACK clients that
\r
1901 // have ports connected to the server. The JACK server is typically
\r
1902 // started in a terminal as follows:
\r
1904 // .jackd -d alsa -d hw:0
\r
1906 // or through an interface program such as qjackctl. Many of the
\r
1907 // parameters normally set for a stream are fixed by the JACK server
\r
1908 // and can be specified when the JACK server is started. In
\r
1911 // .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
\r
1913 // specifies a sample rate of 44100 Hz, a buffer size of 512 sample
\r
1914 // frames, and number of buffers = 4. Once the server is running, it
\r
1915 // is not possible to override these values. If the values are not
\r
1916 // specified in the command-line, the JACK server uses default values.
\r
1918 // The JACK server does not have to be running when an instance of
\r
1919 // RtApiJack is created, though the function getDeviceCount() will
\r
1920 // report 0 devices found until JACK has been started. When no
\r
1921 // devices are available (i.e., the JACK server is not running), a
\r
1922 // stream cannot be opened.
\r
1924 #include <jack/jack.h>
\r
1925 #include <unistd.h>
\r
1928 // A structure to hold various information related to the Jack API
\r
1929 // implementation.
\r
1930 struct JackHandle {
\r
1931 jack_client_t *client;
\r
1932 jack_port_t **ports[2];
\r
1933 std::string deviceName[2];
\r
1935 pthread_cond_t condition;
\r
1936 int drainCounter; // Tracks callback counts when draining
\r
1937 bool internalDrain; // Indicates if stop is initiated from callback or not.
\r
1940 :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
\r
1943 static void jackSilentError( const char * ) {};
\r
1945 RtApiJack :: RtApiJack()
\r
1947 // Nothing to do here.
\r
1948 #if !defined(__RTAUDIO_DEBUG__)
\r
1949 // Turn off Jack's internal error reporting.
\r
1950 jack_set_error_function( &jackSilentError );
\r
1954 RtApiJack :: ~RtApiJack()
\r
1956 if ( stream_.state != STREAM_CLOSED ) closeStream();
\r
1959 unsigned int RtApiJack :: getDeviceCount( void )
\r
1961 // See if we can become a jack client.
\r
1962 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
\r
1963 jack_status_t *status = NULL;
\r
1964 jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
\r
1965 if ( client == 0 ) return 0;
\r
1967 const char **ports;
\r
1968 std::string port, previousPort;
\r
1969 unsigned int nChannels = 0, nDevices = 0;
\r
1970 ports = jack_get_ports( client, NULL, NULL, 0 );
\r
1972 // Parse the port names up to the first colon (:).
\r
1973 size_t iColon = 0;
\r
1975 port = (char *) ports[ nChannels ];
\r
1976 iColon = port.find(":");
\r
1977 if ( iColon != std::string::npos ) {
\r
1978 port = port.substr( 0, iColon + 1 );
\r
1979 if ( port != previousPort ) {
\r
1981 previousPort = port;
\r
1984 } while ( ports[++nChannels] );
\r
1988 jack_client_close( client );
\r
1992 RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
\r
1994 RtAudio::DeviceInfo info;
\r
1995 info.probed = false;
\r
1997 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
\r
1998 jack_status_t *status = NULL;
\r
1999 jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
\r
2000 if ( client == 0 ) {
\r
2001 errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
\r
2002 error( RtAudioError::WARNING );
\r
2006 const char **ports;
\r
2007 std::string port, previousPort;
\r
2008 unsigned int nPorts = 0, nDevices = 0;
\r
2009 ports = jack_get_ports( client, NULL, NULL, 0 );
\r
2011 // Parse the port names up to the first colon (:).
\r
2012 size_t iColon = 0;
\r
2014 port = (char *) ports[ nPorts ];
\r
2015 iColon = port.find(":");
\r
2016 if ( iColon != std::string::npos ) {
\r
2017 port = port.substr( 0, iColon );
\r
2018 if ( port != previousPort ) {
\r
2019 if ( nDevices == device ) info.name = port;
\r
2021 previousPort = port;
\r
2024 } while ( ports[++nPorts] );
\r
2028 if ( device >= nDevices ) {
\r
2029 jack_client_close( client );
\r
2030 errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
\r
2031 error( RtAudioError::INVALID_USE );
\r
2035 // Get the current jack server sample rate.
\r
2036 info.sampleRates.clear();
\r
2038 info.preferredSampleRate = jack_get_sample_rate( client );
\r
2039 info.sampleRates.push_back( info.preferredSampleRate );
\r
2041 // Count the available ports containing the client name as device
\r
2042 // channels. Jack "input ports" equal RtAudio output channels.
\r
2043 unsigned int nChannels = 0;
\r
2044 ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
\r
2046 while ( ports[ nChannels ] ) nChannels++;
\r
2048 info.outputChannels = nChannels;
\r
2051 // Jack "output ports" equal RtAudio input channels.
\r
2053 ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
\r
2055 while ( ports[ nChannels ] ) nChannels++;
\r
2057 info.inputChannels = nChannels;
\r
2060 if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
\r
2061 jack_client_close(client);
\r
2062 errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
\r
2063 error( RtAudioError::WARNING );
\r
2067 // If device opens for both playback and capture, we determine the channels.
\r
2068 if ( info.outputChannels > 0 && info.inputChannels > 0 )
\r
2069 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
\r
2071 // Jack always uses 32-bit floats.
\r
2072 info.nativeFormats = RTAUDIO_FLOAT32;
\r
2074 // Jack doesn't provide default devices so we'll use the first available one.
\r
2075 if ( device == 0 && info.outputChannels > 0 )
\r
2076 info.isDefaultOutput = true;
\r
2077 if ( device == 0 && info.inputChannels > 0 )
\r
2078 info.isDefaultInput = true;
\r
2080 jack_client_close(client);
\r
2081 info.probed = true;
\r
2085 static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
\r
2087 CallbackInfo *info = (CallbackInfo *) infoPointer;
\r
2089 RtApiJack *object = (RtApiJack *) info->object;
\r
2090 if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
\r
2095 // This function will be called by a spawned thread when the Jack
\r
2096 // server signals that it is shutting down. It is necessary to handle
\r
2097 // it this way because the jackShutdown() function must return before
\r
2098 // the jack_deactivate() function (in closeStream()) will return.
\r
2099 static void *jackCloseStream( void *ptr )
\r
2101 CallbackInfo *info = (CallbackInfo *) ptr;
\r
2102 RtApiJack *object = (RtApiJack *) info->object;
\r
2104 object->closeStream();
\r
2106 pthread_exit( NULL );
\r
2108 static void jackShutdown( void *infoPointer )
\r
2110 CallbackInfo *info = (CallbackInfo *) infoPointer;
\r
2111 RtApiJack *object = (RtApiJack *) info->object;
\r
2113 // Check current stream state. If stopped, then we'll assume this
\r
2114 // was called as a result of a call to RtApiJack::stopStream (the
\r
2115 // deactivation of a client handle causes this function to be called).
\r
2116 // If not, we'll assume the Jack server is shutting down or some
\r
2117 // other problem occurred and we should close the stream.
\r
2118 if ( object->isStreamRunning() == false ) return;
\r
2120 ThreadHandle threadId;
\r
2121 pthread_create( &threadId, NULL, jackCloseStream, info );
\r
2122 std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
\r
2125 static int jackXrun( void *infoPointer )
\r
2127 JackHandle *handle = (JackHandle *) infoPointer;
\r
2129 if ( handle->ports[0] ) handle->xrun[0] = true;
\r
2130 if ( handle->ports[1] ) handle->xrun[1] = true;
\r
2135 bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
2136 unsigned int firstChannel, unsigned int sampleRate,
\r
2137 RtAudioFormat format, unsigned int *bufferSize,
\r
2138 RtAudio::StreamOptions *options )
\r
2140 JackHandle *handle = (JackHandle *) stream_.apiHandle;
\r
2142 // Look for jack server and try to become a client (only do once per stream).
\r
2143 jack_client_t *client = 0;
\r
2144 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
\r
2145 jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
\r
2146 jack_status_t *status = NULL;
\r
2147 if ( options && !options->streamName.empty() )
\r
2148 client = jack_client_open( options->streamName.c_str(), jackoptions, status );
\r
2150 client = jack_client_open( "RtApiJack", jackoptions, status );
\r
2151 if ( client == 0 ) {
\r
2152 errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
\r
2153 error( RtAudioError::WARNING );
\r
2158 // The handle must have been created on an earlier pass.
\r
2159 client = handle->client;
\r
2162 const char **ports;
\r
2163 std::string port, previousPort, deviceName;
\r
2164 unsigned int nPorts = 0, nDevices = 0;
\r
2165 ports = jack_get_ports( client, NULL, NULL, 0 );
\r
2167 // Parse the port names up to the first colon (:).
\r
2168 size_t iColon = 0;
\r
2170 port = (char *) ports[ nPorts ];
\r
2171 iColon = port.find(":");
\r
2172 if ( iColon != std::string::npos ) {
\r
2173 port = port.substr( 0, iColon );
\r
2174 if ( port != previousPort ) {
\r
2175 if ( nDevices == device ) deviceName = port;
\r
2177 previousPort = port;
\r
2180 } while ( ports[++nPorts] );
\r
2184 if ( device >= nDevices ) {
\r
2185 errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
\r
2189 // Count the available ports containing the client name as device
\r
2190 // channels. Jack "input ports" equal RtAudio output channels.
\r
2191 unsigned int nChannels = 0;
\r
2192 unsigned long flag = JackPortIsInput;
\r
2193 if ( mode == INPUT ) flag = JackPortIsOutput;
\r
2194 ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
\r
2196 while ( ports[ nChannels ] ) nChannels++;
\r
2200 // Compare the jack ports for specified client to the requested number of channels.
\r
2201 if ( nChannels < (channels + firstChannel) ) {
\r
2202 errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
\r
2203 errorText_ = errorStream_.str();
\r
2207 // Check the jack server sample rate.
\r
2208 unsigned int jackRate = jack_get_sample_rate( client );
\r
2209 if ( sampleRate != jackRate ) {
\r
2210 jack_client_close( client );
\r
2211 errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
\r
2212 errorText_ = errorStream_.str();
\r
2215 stream_.sampleRate = jackRate;
\r
2217 // Get the latency of the JACK port.
\r
2218 ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
\r
2219 if ( ports[ firstChannel ] ) {
\r
2220 // Added by Ge Wang
\r
2221 jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
\r
2222 // the range (usually the min and max are equal)
\r
2223 jack_latency_range_t latrange; latrange.min = latrange.max = 0;
\r
2224 // get the latency range
\r
2225 jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
\r
2226 // be optimistic, use the min!
\r
2227 stream_.latency[mode] = latrange.min;
\r
2228 //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
\r
2232 // The jack server always uses 32-bit floating-point data.
\r
2233 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
\r
2234 stream_.userFormat = format;
\r
2236 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
\r
2237 else stream_.userInterleaved = true;
\r
2239 // Jack always uses non-interleaved buffers.
\r
2240 stream_.deviceInterleaved[mode] = false;
\r
2242 // Jack always provides host byte-ordered data.
\r
2243 stream_.doByteSwap[mode] = false;
\r
2245 // Get the buffer size. The buffer size and number of buffers
\r
2246 // (periods) is set when the jack server is started.
\r
2247 stream_.bufferSize = (int) jack_get_buffer_size( client );
\r
2248 *bufferSize = stream_.bufferSize;
\r
2250 stream_.nDeviceChannels[mode] = channels;
\r
2251 stream_.nUserChannels[mode] = channels;
\r
2253 // Set flags for buffer conversion.
\r
2254 stream_.doConvertBuffer[mode] = false;
\r
2255 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
2256 stream_.doConvertBuffer[mode] = true;
\r
2257 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
\r
2258 stream_.nUserChannels[mode] > 1 )
\r
2259 stream_.doConvertBuffer[mode] = true;
\r
2261 // Allocate our JackHandle structure for the stream.
\r
2262 if ( handle == 0 ) {
\r
2264 handle = new JackHandle;
\r
2266 catch ( std::bad_alloc& ) {
\r
2267 errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
\r
2271 if ( pthread_cond_init(&handle->condition, NULL) ) {
\r
2272 errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
\r
2275 stream_.apiHandle = (void *) handle;
\r
2276 handle->client = client;
\r
2278 handle->deviceName[mode] = deviceName;
\r
2280 // Allocate necessary internal buffers.
\r
2281 unsigned long bufferBytes;
\r
2282 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
2283 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
2284 if ( stream_.userBuffer[mode] == NULL ) {
\r
2285 errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
\r
2289 if ( stream_.doConvertBuffer[mode] ) {
\r
2291 bool makeBuffer = true;
\r
2292 if ( mode == OUTPUT )
\r
2293 bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
2294 else { // mode == INPUT
\r
2295 bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
\r
2296 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
2297 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
\r
2298 if ( bufferBytes < bytesOut ) makeBuffer = false;
\r
2302 if ( makeBuffer ) {
\r
2303 bufferBytes *= *bufferSize;
\r
2304 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
2305 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
2306 if ( stream_.deviceBuffer == NULL ) {
\r
2307 errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
\r
2313 // Allocate memory for the Jack ports (channels) identifiers.
\r
2314 handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
\r
2315 if ( handle->ports[mode] == NULL ) {
\r
2316 errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
\r
2320 stream_.device[mode] = device;
\r
2321 stream_.channelOffset[mode] = firstChannel;
\r
2322 stream_.state = STREAM_STOPPED;
\r
2323 stream_.callbackInfo.object = (void *) this;
\r
2325 if ( stream_.mode == OUTPUT && mode == INPUT )
\r
2326 // We had already set up the stream for output.
\r
2327 stream_.mode = DUPLEX;
\r
2329 stream_.mode = mode;
\r
2330 jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
\r
2331 jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
\r
2332 jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
\r
2335 // Register our ports.
\r
2337 if ( mode == OUTPUT ) {
\r
2338 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
\r
2339 snprintf( label, 64, "outport %d", i );
\r
2340 handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
\r
2341 JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
\r
2345 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
\r
2346 snprintf( label, 64, "inport %d", i );
\r
2347 handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
\r
2348 JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
\r
2352 // Setup the buffer conversion information structure. We don't use
\r
2353 // buffers to do channel offsets, so we override that parameter
\r
2355 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
\r
2361 pthread_cond_destroy( &handle->condition );
\r
2362 jack_client_close( handle->client );
\r
2364 if ( handle->ports[0] ) free( handle->ports[0] );
\r
2365 if ( handle->ports[1] ) free( handle->ports[1] );
\r
2368 stream_.apiHandle = 0;
\r
2371 for ( int i=0; i<2; i++ ) {
\r
2372 if ( stream_.userBuffer[i] ) {
\r
2373 free( stream_.userBuffer[i] );
\r
2374 stream_.userBuffer[i] = 0;
\r
2378 if ( stream_.deviceBuffer ) {
\r
2379 free( stream_.deviceBuffer );
\r
2380 stream_.deviceBuffer = 0;
\r
2386 void RtApiJack :: closeStream( void )
\r
2388 if ( stream_.state == STREAM_CLOSED ) {
\r
2389 errorText_ = "RtApiJack::closeStream(): no open stream to close!";
\r
2390 error( RtAudioError::WARNING );
\r
2394 JackHandle *handle = (JackHandle *) stream_.apiHandle;
\r
2397 if ( stream_.state == STREAM_RUNNING )
\r
2398 jack_deactivate( handle->client );
\r
2400 jack_client_close( handle->client );
\r
2404 if ( handle->ports[0] ) free( handle->ports[0] );
\r
2405 if ( handle->ports[1] ) free( handle->ports[1] );
\r
2406 pthread_cond_destroy( &handle->condition );
\r
2408 stream_.apiHandle = 0;
\r
2411 for ( int i=0; i<2; i++ ) {
\r
2412 if ( stream_.userBuffer[i] ) {
\r
2413 free( stream_.userBuffer[i] );
\r
2414 stream_.userBuffer[i] = 0;
\r
2418 if ( stream_.deviceBuffer ) {
\r
2419 free( stream_.deviceBuffer );
\r
2420 stream_.deviceBuffer = 0;
\r
2423 stream_.mode = UNINITIALIZED;
\r
2424 stream_.state = STREAM_CLOSED;
\r
2427 void RtApiJack :: startStream( void )
\r
2430 if ( stream_.state == STREAM_RUNNING ) {
\r
2431 errorText_ = "RtApiJack::startStream(): the stream is already running!";
\r
2432 error( RtAudioError::WARNING );
\r
2436 JackHandle *handle = (JackHandle *) stream_.apiHandle;
\r
2437 int result = jack_activate( handle->client );
\r
2439 errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
\r
2443 const char **ports;
\r
2445 // Get the list of available ports.
\r
2446 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
2448 ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
\r
2449 if ( ports == NULL) {
\r
2450 errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
\r
2454 // Now make the port connections. Since RtAudio wasn't designed to
\r
2455 // allow the user to select particular channels of a device, we'll
\r
2456 // just open the first "nChannels" ports with offset.
\r
2457 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
\r
2459 if ( ports[ stream_.channelOffset[0] + i ] )
\r
2460 result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
\r
2463 errorText_ = "RtApiJack::startStream(): error connecting output ports!";
\r
2470 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
2472 ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
\r
2473 if ( ports == NULL) {
\r
2474 errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
\r
2478 // Now make the port connections. See note above.
\r
2479 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
\r
2481 if ( ports[ stream_.channelOffset[1] + i ] )
\r
2482 result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
\r
2485 errorText_ = "RtApiJack::startStream(): error connecting input ports!";
\r
2492 handle->drainCounter = 0;
\r
2493 handle->internalDrain = false;
\r
2494 stream_.state = STREAM_RUNNING;
\r
2497 if ( result == 0 ) return;
\r
2498 error( RtAudioError::SYSTEM_ERROR );
\r
2501 void RtApiJack :: stopStream( void )
\r
2504 if ( stream_.state == STREAM_STOPPED ) {
\r
2505 errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
\r
2506 error( RtAudioError::WARNING );
\r
2510 JackHandle *handle = (JackHandle *) stream_.apiHandle;
\r
2511 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
2513 if ( handle->drainCounter == 0 ) {
\r
2514 handle->drainCounter = 2;
\r
2515 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
\r
2519 jack_deactivate( handle->client );
\r
2520 stream_.state = STREAM_STOPPED;
\r
2523 void RtApiJack :: abortStream( void )
\r
2526 if ( stream_.state == STREAM_STOPPED ) {
\r
2527 errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
\r
2528 error( RtAudioError::WARNING );
\r
2532 JackHandle *handle = (JackHandle *) stream_.apiHandle;
\r
2533 handle->drainCounter = 2;
\r
2538 // This function will be called by a spawned thread when the user
\r
2539 // callback function signals that the stream should be stopped or
\r
2540 // aborted. It is necessary to handle it this way because the
\r
2541 // callbackEvent() function must return before the jack_deactivate()
\r
2542 // function will return.
\r
2543 static void *jackStopStream( void *ptr )
\r
2545 CallbackInfo *info = (CallbackInfo *) ptr;
\r
2546 RtApiJack *object = (RtApiJack *) info->object;
\r
2548 object->stopStream();
\r
2549 pthread_exit( NULL );
\r
2552 bool RtApiJack :: callbackEvent( unsigned long nframes )
\r
2554 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
\r
2555 if ( stream_.state == STREAM_CLOSED ) {
\r
2556 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
\r
2557 error( RtAudioError::WARNING );
\r
2560 if ( stream_.bufferSize != nframes ) {
\r
2561 errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
\r
2562 error( RtAudioError::WARNING );
\r
2566 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
\r
2567 JackHandle *handle = (JackHandle *) stream_.apiHandle;
\r
2569 // Check if we were draining the stream and signal is finished.
\r
2570 if ( handle->drainCounter > 3 ) {
\r
2571 ThreadHandle threadId;
\r
2573 stream_.state = STREAM_STOPPING;
\r
2574 if ( handle->internalDrain == true )
\r
2575 pthread_create( &threadId, NULL, jackStopStream, info );
\r
2577 pthread_cond_signal( &handle->condition );
\r
2581 // Invoke user callback first, to get fresh output data.
\r
2582 if ( handle->drainCounter == 0 ) {
\r
2583 RtAudioCallback callback = (RtAudioCallback) info->callback;
\r
2584 double streamTime = getStreamTime();
\r
2585 RtAudioStreamStatus status = 0;
\r
2586 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
\r
2587 status |= RTAUDIO_OUTPUT_UNDERFLOW;
\r
2588 handle->xrun[0] = false;
\r
2590 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
\r
2591 status |= RTAUDIO_INPUT_OVERFLOW;
\r
2592 handle->xrun[1] = false;
\r
2594 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
\r
2595 stream_.bufferSize, streamTime, status, info->userData );
\r
2596 if ( cbReturnValue == 2 ) {
\r
2597 stream_.state = STREAM_STOPPING;
\r
2598 handle->drainCounter = 2;
\r
2600 pthread_create( &id, NULL, jackStopStream, info );
\r
2603 else if ( cbReturnValue == 1 ) {
\r
2604 handle->drainCounter = 1;
\r
2605 handle->internalDrain = true;
\r
2609 jack_default_audio_sample_t *jackbuffer;
\r
2610 unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
\r
2611 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
2613 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
\r
2615 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
\r
2616 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
\r
2617 memset( jackbuffer, 0, bufferBytes );
\r
2621 else if ( stream_.doConvertBuffer[0] ) {
\r
2623 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
\r
2625 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
\r
2626 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
\r
2627 memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
\r
2630 else { // no buffer conversion
\r
2631 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
\r
2632 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
\r
2633 memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
\r
2638 // Don't bother draining input
\r
2639 if ( handle->drainCounter ) {
\r
2640 handle->drainCounter++;
\r
2644 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
2646 if ( stream_.doConvertBuffer[1] ) {
\r
2647 for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
\r
2648 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
\r
2649 memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
\r
2651 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
\r
2653 else { // no buffer conversion
\r
2654 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
\r
2655 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
\r
2656 memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
\r
2662 RtApi::tickStreamTime();
\r
2665 //******************** End of __UNIX_JACK__ *********************//
\r
2668 #if defined(__WINDOWS_ASIO__) // ASIO API on Windows
\r
2670 // The ASIO API is designed around a callback scheme, so this
\r
2671 // implementation is similar to that used for OS-X CoreAudio and Linux
\r
2672 // Jack. The primary constraint with ASIO is that it only allows
\r
2673 // access to a single driver at a time. Thus, it is not possible to
\r
2674 // have more than one simultaneous RtAudio stream.
\r
2676 // This implementation also requires a number of external ASIO files
\r
2677 // and a few global variables. The ASIO callback scheme does not
\r
2678 // allow for the passing of user data, so we must create a global
\r
2679 // pointer to our callbackInfo structure.
\r
2681 // On unix systems, we make use of a pthread condition variable.
\r
2682 // Since there is no equivalent in Windows, I hacked something based
\r
2683 // on information found in
\r
2684 // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
\r
2686 #include "asiosys.h"
\r
2688 #include "iasiothiscallresolver.h"
\r
2689 #include "asiodrivers.h"
\r
2692 static AsioDrivers drivers;
\r
2693 static ASIOCallbacks asioCallbacks;
\r
2694 static ASIODriverInfo driverInfo;
\r
2695 static CallbackInfo *asioCallbackInfo;
\r
2696 static bool asioXRun;
\r
2698 struct AsioHandle {
\r
2699 int drainCounter; // Tracks callback counts when draining
\r
2700 bool internalDrain; // Indicates if stop is initiated from callback or not.
\r
2701 ASIOBufferInfo *bufferInfos;
\r
2705 :drainCounter(0), internalDrain(false), bufferInfos(0) {}
\r
2708 // Function declarations (definitions at end of section)
\r
2709 static const char* getAsioErrorString( ASIOError result );
\r
2710 static void sampleRateChanged( ASIOSampleRate sRate );
\r
2711 static long asioMessages( long selector, long value, void* message, double* opt );
\r
2713 RtApiAsio :: RtApiAsio()
\r
2715 // ASIO cannot run on a multi-threaded appartment. You can call
\r
2716 // CoInitialize beforehand, but it must be for appartment threading
\r
2717 // (in which case, CoInitilialize will return S_FALSE here).
\r
2718 coInitialized_ = false;
\r
2719 HRESULT hr = CoInitialize( NULL );
\r
2720 if ( FAILED(hr) ) {
\r
2721 errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
\r
2722 error( RtAudioError::WARNING );
\r
2724 coInitialized_ = true;
\r
2726 drivers.removeCurrentDriver();
\r
2727 driverInfo.asioVersion = 2;
\r
2729 // See note in DirectSound implementation about GetDesktopWindow().
\r
2730 driverInfo.sysRef = GetForegroundWindow();
\r
2733 RtApiAsio :: ~RtApiAsio()
\r
2735 if ( stream_.state != STREAM_CLOSED ) closeStream();
\r
2736 if ( coInitialized_ ) CoUninitialize();
\r
2739 unsigned int RtApiAsio :: getDeviceCount( void )
\r
2741 return (unsigned int) drivers.asioGetNumDev();
\r
2744 RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
\r
2746 RtAudio::DeviceInfo info;
\r
2747 info.probed = false;
\r
2750 unsigned int nDevices = getDeviceCount();
\r
2751 if ( nDevices == 0 ) {
\r
2752 errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
\r
2753 error( RtAudioError::INVALID_USE );
\r
2757 if ( device >= nDevices ) {
\r
2758 errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
\r
2759 error( RtAudioError::INVALID_USE );
\r
2763 // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
\r
2764 if ( stream_.state != STREAM_CLOSED ) {
\r
2765 if ( device >= devices_.size() ) {
\r
2766 errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
\r
2767 error( RtAudioError::WARNING );
\r
2770 return devices_[ device ];
\r
2773 char driverName[32];
\r
2774 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
\r
2775 if ( result != ASE_OK ) {
\r
2776 errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
\r
2777 errorText_ = errorStream_.str();
\r
2778 error( RtAudioError::WARNING );
\r
2782 info.name = driverName;
\r
2784 if ( !drivers.loadDriver( driverName ) ) {
\r
2785 errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
\r
2786 errorText_ = errorStream_.str();
\r
2787 error( RtAudioError::WARNING );
\r
2791 result = ASIOInit( &driverInfo );
\r
2792 if ( result != ASE_OK ) {
\r
2793 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
\r
2794 errorText_ = errorStream_.str();
\r
2795 error( RtAudioError::WARNING );
\r
2799 // Determine the device channel information.
\r
2800 long inputChannels, outputChannels;
\r
2801 result = ASIOGetChannels( &inputChannels, &outputChannels );
\r
2802 if ( result != ASE_OK ) {
\r
2803 drivers.removeCurrentDriver();
\r
2804 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
\r
2805 errorText_ = errorStream_.str();
\r
2806 error( RtAudioError::WARNING );
\r
2810 info.outputChannels = outputChannels;
\r
2811 info.inputChannels = inputChannels;
\r
2812 if ( info.outputChannels > 0 && info.inputChannels > 0 )
\r
2813 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
\r
2815 // Determine the supported sample rates.
\r
2816 info.sampleRates.clear();
\r
2817 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
\r
2818 result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
\r
2819 if ( result == ASE_OK ) {
\r
2820 info.sampleRates.push_back( SAMPLE_RATES[i] );
\r
2822 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
\r
2823 info.preferredSampleRate = SAMPLE_RATES[i];
\r
2827 // Determine supported data types ... just check first channel and assume rest are the same.
\r
2828 ASIOChannelInfo channelInfo;
\r
2829 channelInfo.channel = 0;
\r
2830 channelInfo.isInput = true;
\r
2831 if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
\r
2832 result = ASIOGetChannelInfo( &channelInfo );
\r
2833 if ( result != ASE_OK ) {
\r
2834 drivers.removeCurrentDriver();
\r
2835 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
\r
2836 errorText_ = errorStream_.str();
\r
2837 error( RtAudioError::WARNING );
\r
2841 info.nativeFormats = 0;
\r
2842 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
\r
2843 info.nativeFormats |= RTAUDIO_SINT16;
\r
2844 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
\r
2845 info.nativeFormats |= RTAUDIO_SINT32;
\r
2846 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
\r
2847 info.nativeFormats |= RTAUDIO_FLOAT32;
\r
2848 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
\r
2849 info.nativeFormats |= RTAUDIO_FLOAT64;
\r
2850 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
\r
2851 info.nativeFormats |= RTAUDIO_SINT24;
\r
2853 if ( info.outputChannels > 0 )
\r
2854 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
\r
2855 if ( info.inputChannels > 0 )
\r
2856 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
\r
2858 info.probed = true;
\r
2859 drivers.removeCurrentDriver();
\r
2863 static void bufferSwitch( long index, ASIOBool /*processNow*/ )
\r
2865 RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
\r
2866 object->callbackEvent( index );
\r
2869 void RtApiAsio :: saveDeviceInfo( void )
\r
2873 unsigned int nDevices = getDeviceCount();
\r
2874 devices_.resize( nDevices );
\r
2875 for ( unsigned int i=0; i<nDevices; i++ )
\r
2876 devices_[i] = getDeviceInfo( i );
\r
2879 bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
2880 unsigned int firstChannel, unsigned int sampleRate,
\r
2881 RtAudioFormat format, unsigned int *bufferSize,
\r
2882 RtAudio::StreamOptions *options )
\r
2883 {////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
\r
2885 bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT;
\r
2887 // For ASIO, a duplex stream MUST use the same driver.
\r
2888 if ( isDuplexInput && stream_.device[0] != device ) {
\r
2889 errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
\r
2893 char driverName[32];
\r
2894 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
\r
2895 if ( result != ASE_OK ) {
\r
2896 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
\r
2897 errorText_ = errorStream_.str();
\r
2901 // Only load the driver once for duplex stream.
\r
2902 if ( !isDuplexInput ) {
\r
2903 // The getDeviceInfo() function will not work when a stream is open
\r
2904 // because ASIO does not allow multiple devices to run at the same
\r
2905 // time. Thus, we'll probe the system before opening a stream and
\r
2906 // save the results for use by getDeviceInfo().
\r
2907 this->saveDeviceInfo();
\r
2909 if ( !drivers.loadDriver( driverName ) ) {
\r
2910 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
\r
2911 errorText_ = errorStream_.str();
\r
2915 result = ASIOInit( &driverInfo );
\r
2916 if ( result != ASE_OK ) {
\r
2917 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
\r
2918 errorText_ = errorStream_.str();
\r
2923 // keep them before any "goto error", they are used for error cleanup + goto device boundary checks
\r
2924 bool buffersAllocated = false;
\r
2925 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
\r
2926 unsigned int nChannels;
\r
2929 // Check the device channel count.
\r
2930 long inputChannels, outputChannels;
\r
2931 result = ASIOGetChannels( &inputChannels, &outputChannels );
\r
2932 if ( result != ASE_OK ) {
\r
2933 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
\r
2934 errorText_ = errorStream_.str();
\r
2938 if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
\r
2939 ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
\r
2940 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
\r
2941 errorText_ = errorStream_.str();
\r
2944 stream_.nDeviceChannels[mode] = channels;
\r
2945 stream_.nUserChannels[mode] = channels;
\r
2946 stream_.channelOffset[mode] = firstChannel;
\r
2948 // Verify the sample rate is supported.
\r
2949 result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
\r
2950 if ( result != ASE_OK ) {
\r
2951 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
\r
2952 errorText_ = errorStream_.str();
\r
2956 // Get the current sample rate
\r
2957 ASIOSampleRate currentRate;
\r
2958 result = ASIOGetSampleRate( ¤tRate );
\r
2959 if ( result != ASE_OK ) {
\r
2960 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
\r
2961 errorText_ = errorStream_.str();
\r
2965 // Set the sample rate only if necessary
\r
2966 if ( currentRate != sampleRate ) {
\r
2967 result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
\r
2968 if ( result != ASE_OK ) {
\r
2969 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
\r
2970 errorText_ = errorStream_.str();
\r
2975 // Determine the driver data type.
\r
2976 ASIOChannelInfo channelInfo;
\r
2977 channelInfo.channel = 0;
\r
2978 if ( mode == OUTPUT ) channelInfo.isInput = false;
\r
2979 else channelInfo.isInput = true;
\r
2980 result = ASIOGetChannelInfo( &channelInfo );
\r
2981 if ( result != ASE_OK ) {
\r
2982 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
\r
2983 errorText_ = errorStream_.str();
\r
2987 // Assuming WINDOWS host is always little-endian.
\r
2988 stream_.doByteSwap[mode] = false;
\r
2989 stream_.userFormat = format;
\r
2990 stream_.deviceFormat[mode] = 0;
\r
2991 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
\r
2992 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
2993 if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
\r
2995 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
\r
2996 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
\r
2997 if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
\r
2999 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
\r
3000 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
\r
3001 if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
\r
3003 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
\r
3004 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
\r
3005 if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
\r
3007 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
\r
3008 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
\r
3009 if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
\r
3012 if ( stream_.deviceFormat[mode] == 0 ) {
\r
3013 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
\r
3014 errorText_ = errorStream_.str();
\r
3018 // Set the buffer size. For a duplex stream, this will end up
\r
3019 // setting the buffer size based on the input constraints, which
\r
3021 long minSize, maxSize, preferSize, granularity;
\r
3022 result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
\r
3023 if ( result != ASE_OK ) {
\r
3024 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
\r
3025 errorText_ = errorStream_.str();
\r
3029 if ( isDuplexInput ) {
\r
3030 // When this is the duplex input (output was opened before), then we have to use the same
\r
3031 // buffersize as the output, because it might use the preferred buffer size, which most
\r
3032 // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
\r
3033 // So instead of throwing an error, make them equal. The caller uses the reference
\r
3034 // to the "bufferSize" param as usual to set up processing buffers.
\r
3036 *bufferSize = stream_.bufferSize;
\r
3039 if ( *bufferSize == 0 ) *bufferSize = preferSize;
\r
3040 else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
\r
3041 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
\r
3042 else if ( granularity == -1 ) {
\r
3043 // Make sure bufferSize is a power of two.
\r
3044 int log2_of_min_size = 0;
\r
3045 int log2_of_max_size = 0;
\r
3047 for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
\r
3048 if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
\r
3049 if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
\r
3052 long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
\r
3053 int min_delta_num = log2_of_min_size;
\r
3055 for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
\r
3056 long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
\r
3057 if (current_delta < min_delta) {
\r
3058 min_delta = current_delta;
\r
3059 min_delta_num = i;
\r
3063 *bufferSize = ( (unsigned int)1 << min_delta_num );
\r
3064 if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
\r
3065 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
\r
3067 else if ( granularity != 0 ) {
\r
3068 // Set to an even multiple of granularity, rounding up.
\r
3069 *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
\r
3074 // we don't use it anymore, see above!
\r
3075 // Just left it here for the case...
\r
3076 if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
\r
3077 errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
\r
3082 stream_.bufferSize = *bufferSize;
\r
3083 stream_.nBuffers = 2;
\r
3085 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
\r
3086 else stream_.userInterleaved = true;
\r
3088 // ASIO always uses non-interleaved buffers.
\r
3089 stream_.deviceInterleaved[mode] = false;
\r
3091 // Allocate, if necessary, our AsioHandle structure for the stream.
\r
3092 if ( handle == 0 ) {
\r
3094 handle = new AsioHandle;
\r
3096 catch ( std::bad_alloc& ) {
\r
3097 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
\r
3100 handle->bufferInfos = 0;
\r
3102 // Create a manual-reset event.
\r
3103 handle->condition = CreateEvent( NULL, // no security
\r
3104 TRUE, // manual-reset
\r
3105 FALSE, // non-signaled initially
\r
3106 NULL ); // unnamed
\r
3107 stream_.apiHandle = (void *) handle;
\r
3110 // Create the ASIO internal buffers. Since RtAudio sets up input
\r
3111 // and output separately, we'll have to dispose of previously
\r
3112 // created output buffers for a duplex stream.
\r
3113 if ( mode == INPUT && stream_.mode == OUTPUT ) {
\r
3114 ASIODisposeBuffers();
\r
3115 if ( handle->bufferInfos ) free( handle->bufferInfos );
\r
3118 // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
\r
3120 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
\r
3121 handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
\r
3122 if ( handle->bufferInfos == NULL ) {
\r
3123 errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
\r
3124 errorText_ = errorStream_.str();
\r
3128 ASIOBufferInfo *infos;
\r
3129 infos = handle->bufferInfos;
\r
3130 for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
\r
3131 infos->isInput = ASIOFalse;
\r
3132 infos->channelNum = i + stream_.channelOffset[0];
\r
3133 infos->buffers[0] = infos->buffers[1] = 0;
\r
3135 for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
\r
3136 infos->isInput = ASIOTrue;
\r
3137 infos->channelNum = i + stream_.channelOffset[1];
\r
3138 infos->buffers[0] = infos->buffers[1] = 0;
\r
3141 // prepare for callbacks
\r
3142 stream_.sampleRate = sampleRate;
\r
3143 stream_.device[mode] = device;
\r
3144 stream_.mode = isDuplexInput ? DUPLEX : mode;
\r
3146 // store this class instance before registering callbacks, that are going to use it
\r
3147 asioCallbackInfo = &stream_.callbackInfo;
\r
3148 stream_.callbackInfo.object = (void *) this;
\r
3150 // Set up the ASIO callback structure and create the ASIO data buffers.
\r
3151 asioCallbacks.bufferSwitch = &bufferSwitch;
\r
3152 asioCallbacks.sampleRateDidChange = &sampleRateChanged;
\r
3153 asioCallbacks.asioMessage = &asioMessages;
\r
3154 asioCallbacks.bufferSwitchTimeInfo = NULL;
\r
3155 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
\r
3156 if ( result != ASE_OK ) {
\r
3157 // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
\r
3158 // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver
\r
3159 // in that case, let's be naïve and try that instead
\r
3160 *bufferSize = preferSize;
\r
3161 stream_.bufferSize = *bufferSize;
\r
3162 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
\r
3165 if ( result != ASE_OK ) {
\r
3166 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
\r
3167 errorText_ = errorStream_.str();
\r
3170 buffersAllocated = true;
\r
3171 stream_.state = STREAM_STOPPED;
\r
3173 // Set flags for buffer conversion.
\r
3174 stream_.doConvertBuffer[mode] = false;
\r
3175 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
3176 stream_.doConvertBuffer[mode] = true;
\r
3177 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
\r
3178 stream_.nUserChannels[mode] > 1 )
\r
3179 stream_.doConvertBuffer[mode] = true;
\r
3181 // Allocate necessary internal buffers
\r
3182 unsigned long bufferBytes;
\r
3183 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
3184 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
3185 if ( stream_.userBuffer[mode] == NULL ) {
\r
3186 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
\r
3190 if ( stream_.doConvertBuffer[mode] ) {
\r
3192 bool makeBuffer = true;
\r
3193 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
\r
3194 if ( isDuplexInput && stream_.deviceBuffer ) {
\r
3195 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
3196 if ( bufferBytes <= bytesOut ) makeBuffer = false;
\r
3199 if ( makeBuffer ) {
\r
3200 bufferBytes *= *bufferSize;
\r
3201 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
3202 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
3203 if ( stream_.deviceBuffer == NULL ) {
\r
3204 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
\r
3210 // Determine device latencies
\r
3211 long inputLatency, outputLatency;
\r
3212 result = ASIOGetLatencies( &inputLatency, &outputLatency );
\r
3213 if ( result != ASE_OK ) {
\r
3214 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
\r
3215 errorText_ = errorStream_.str();
\r
3216 error( RtAudioError::WARNING); // warn but don't fail
\r
3219 stream_.latency[0] = outputLatency;
\r
3220 stream_.latency[1] = inputLatency;
\r
3223 // Setup the buffer conversion information structure. We don't use
\r
3224 // buffers to do channel offsets, so we override that parameter
\r
3226 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
\r
3231 if ( !isDuplexInput ) {
\r
3232 // the cleanup for error in the duplex input, is done by RtApi::openStream
\r
3233 // So we clean up for single channel only
\r
3235 if ( buffersAllocated )
\r
3236 ASIODisposeBuffers();
\r
3238 drivers.removeCurrentDriver();
\r
3241 CloseHandle( handle->condition );
\r
3242 if ( handle->bufferInfos )
\r
3243 free( handle->bufferInfos );
\r
3246 stream_.apiHandle = 0;
\r
3250 if ( stream_.userBuffer[mode] ) {
\r
3251 free( stream_.userBuffer[mode] );
\r
3252 stream_.userBuffer[mode] = 0;
\r
3255 if ( stream_.deviceBuffer ) {
\r
3256 free( stream_.deviceBuffer );
\r
3257 stream_.deviceBuffer = 0;
\r
3262 }////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
\r
3264 void RtApiAsio :: closeStream()
\r
3266 if ( stream_.state == STREAM_CLOSED ) {
\r
3267 errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
\r
3268 error( RtAudioError::WARNING );
\r
3272 if ( stream_.state == STREAM_RUNNING ) {
\r
3273 stream_.state = STREAM_STOPPED;
\r
3276 ASIODisposeBuffers();
\r
3277 drivers.removeCurrentDriver();
\r
3279 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
\r
3281 CloseHandle( handle->condition );
\r
3282 if ( handle->bufferInfos )
\r
3283 free( handle->bufferInfos );
\r
3285 stream_.apiHandle = 0;
\r
3288 for ( int i=0; i<2; i++ ) {
\r
3289 if ( stream_.userBuffer[i] ) {
\r
3290 free( stream_.userBuffer[i] );
\r
3291 stream_.userBuffer[i] = 0;
\r
3295 if ( stream_.deviceBuffer ) {
\r
3296 free( stream_.deviceBuffer );
\r
3297 stream_.deviceBuffer = 0;
\r
3300 stream_.mode = UNINITIALIZED;
\r
3301 stream_.state = STREAM_CLOSED;
\r
3304 bool stopThreadCalled = false;
\r
3306 void RtApiAsio :: startStream()
\r
3309 if ( stream_.state == STREAM_RUNNING ) {
\r
3310 errorText_ = "RtApiAsio::startStream(): the stream is already running!";
\r
3311 error( RtAudioError::WARNING );
\r
3315 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
\r
3316 ASIOError result = ASIOStart();
\r
3317 if ( result != ASE_OK ) {
\r
3318 errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
\r
3319 errorText_ = errorStream_.str();
\r
3323 handle->drainCounter = 0;
\r
3324 handle->internalDrain = false;
\r
3325 ResetEvent( handle->condition );
\r
3326 stream_.state = STREAM_RUNNING;
\r
3330 stopThreadCalled = false;
\r
3332 if ( result == ASE_OK ) return;
\r
3333 error( RtAudioError::SYSTEM_ERROR );
\r
3336 void RtApiAsio :: stopStream()
\r
3339 if ( stream_.state == STREAM_STOPPED ) {
\r
3340 errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
\r
3341 error( RtAudioError::WARNING );
\r
3345 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
\r
3346 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
3347 if ( handle->drainCounter == 0 ) {
\r
3348 handle->drainCounter = 2;
\r
3349 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
\r
3353 stream_.state = STREAM_STOPPED;
\r
3355 ASIOError result = ASIOStop();
\r
3356 if ( result != ASE_OK ) {
\r
3357 errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
\r
3358 errorText_ = errorStream_.str();
\r
3361 if ( result == ASE_OK ) return;
\r
3362 error( RtAudioError::SYSTEM_ERROR );
\r
3365 void RtApiAsio :: abortStream()
\r
3368 if ( stream_.state == STREAM_STOPPED ) {
\r
3369 errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
\r
3370 error( RtAudioError::WARNING );
\r
3374 // The following lines were commented-out because some behavior was
\r
3375 // noted where the device buffers need to be zeroed to avoid
\r
3376 // continuing sound, even when the device buffers are completely
\r
3377 // disposed. So now, calling abort is the same as calling stop.
\r
3378 // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
\r
3379 // handle->drainCounter = 2;
\r
3383 // This function will be called by a spawned thread when the user
\r
3384 // callback function signals that the stream should be stopped or
\r
3385 // aborted. It is necessary to handle it this way because the
\r
3386 // callbackEvent() function must return before the ASIOStop()
\r
3387 // function will return.
\r
3388 static unsigned __stdcall asioStopStream( void *ptr )
\r
3390 CallbackInfo *info = (CallbackInfo *) ptr;
\r
3391 RtApiAsio *object = (RtApiAsio *) info->object;
\r
3393 object->stopStream();
\r
3394 _endthreadex( 0 );
\r
3398 bool RtApiAsio :: callbackEvent( long bufferIndex )
\r
3400 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
\r
3401 if ( stream_.state == STREAM_CLOSED ) {
\r
3402 errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
\r
3403 error( RtAudioError::WARNING );
\r
3407 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
\r
3408 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
\r
3410 // Check if we were draining the stream and signal if finished.
\r
3411 if ( handle->drainCounter > 3 ) {
\r
3413 stream_.state = STREAM_STOPPING;
\r
3414 if ( handle->internalDrain == false )
\r
3415 SetEvent( handle->condition );
\r
3416 else { // spawn a thread to stop the stream
\r
3417 unsigned threadId;
\r
3418 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
\r
3419 &stream_.callbackInfo, 0, &threadId );
\r
3424 // Invoke user callback to get fresh output data UNLESS we are
\r
3425 // draining stream.
\r
3426 if ( handle->drainCounter == 0 ) {
\r
3427 RtAudioCallback callback = (RtAudioCallback) info->callback;
\r
3428 double streamTime = getStreamTime();
\r
3429 RtAudioStreamStatus status = 0;
\r
3430 if ( stream_.mode != INPUT && asioXRun == true ) {
\r
3431 status |= RTAUDIO_OUTPUT_UNDERFLOW;
\r
3434 if ( stream_.mode != OUTPUT && asioXRun == true ) {
\r
3435 status |= RTAUDIO_INPUT_OVERFLOW;
\r
3438 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
\r
3439 stream_.bufferSize, streamTime, status, info->userData );
\r
3440 if ( cbReturnValue == 2 ) {
\r
3441 stream_.state = STREAM_STOPPING;
\r
3442 handle->drainCounter = 2;
\r
3443 unsigned threadId;
\r
3444 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
\r
3445 &stream_.callbackInfo, 0, &threadId );
\r
3448 else if ( cbReturnValue == 1 ) {
\r
3449 handle->drainCounter = 1;
\r
3450 handle->internalDrain = true;
\r
3454 unsigned int nChannels, bufferBytes, i, j;
\r
3455 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
\r
3456 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
3458 bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
\r
3460 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
\r
3462 for ( i=0, j=0; i<nChannels; i++ ) {
\r
3463 if ( handle->bufferInfos[i].isInput != ASIOTrue )
\r
3464 memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
\r
3468 else if ( stream_.doConvertBuffer[0] ) {
\r
3470 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
\r
3471 if ( stream_.doByteSwap[0] )
\r
3472 byteSwapBuffer( stream_.deviceBuffer,
\r
3473 stream_.bufferSize * stream_.nDeviceChannels[0],
\r
3474 stream_.deviceFormat[0] );
\r
3476 for ( i=0, j=0; i<nChannels; i++ ) {
\r
3477 if ( handle->bufferInfos[i].isInput != ASIOTrue )
\r
3478 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
\r
3479 &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
\r
3485 if ( stream_.doByteSwap[0] )
\r
3486 byteSwapBuffer( stream_.userBuffer[0],
\r
3487 stream_.bufferSize * stream_.nUserChannels[0],
\r
3488 stream_.userFormat );
\r
3490 for ( i=0, j=0; i<nChannels; i++ ) {
\r
3491 if ( handle->bufferInfos[i].isInput != ASIOTrue )
\r
3492 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
\r
3493 &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
\r
3499 // Don't bother draining input
\r
3500 if ( handle->drainCounter ) {
\r
3501 handle->drainCounter++;
\r
3505 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
3507 bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
\r
3509 if (stream_.doConvertBuffer[1]) {
\r
3511 // Always interleave ASIO input data.
\r
3512 for ( i=0, j=0; i<nChannels; i++ ) {
\r
3513 if ( handle->bufferInfos[i].isInput == ASIOTrue )
\r
3514 memcpy( &stream_.deviceBuffer[j++*bufferBytes],
\r
3515 handle->bufferInfos[i].buffers[bufferIndex],
\r
3519 if ( stream_.doByteSwap[1] )
\r
3520 byteSwapBuffer( stream_.deviceBuffer,
\r
3521 stream_.bufferSize * stream_.nDeviceChannels[1],
\r
3522 stream_.deviceFormat[1] );
\r
3523 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
\r
3527 for ( i=0, j=0; i<nChannels; i++ ) {
\r
3528 if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
\r
3529 memcpy( &stream_.userBuffer[1][bufferBytes*j++],
\r
3530 handle->bufferInfos[i].buffers[bufferIndex],
\r
3535 if ( stream_.doByteSwap[1] )
\r
3536 byteSwapBuffer( stream_.userBuffer[1],
\r
3537 stream_.bufferSize * stream_.nUserChannels[1],
\r
3538 stream_.userFormat );
\r
3543 // The following call was suggested by Malte Clasen. While the API
\r
3544 // documentation indicates it should not be required, some device
\r
3545 // drivers apparently do not function correctly without it.
\r
3546 ASIOOutputReady();
\r
3548 RtApi::tickStreamTime();
\r
3552 static void sampleRateChanged( ASIOSampleRate sRate )
\r
3554 // The ASIO documentation says that this usually only happens during
\r
3555 // external sync. Audio processing is not stopped by the driver,
\r
3556 // actual sample rate might not have even changed, maybe only the
\r
3557 // sample rate status of an AES/EBU or S/PDIF digital input at the
\r
3560 RtApi *object = (RtApi *) asioCallbackInfo->object;
\r
3562 object->stopStream();
\r
3564 catch ( RtAudioError &exception ) {
\r
3565 std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
\r
3569 std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
\r
3572 static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
\r
3576 switch( selector ) {
\r
3577 case kAsioSelectorSupported:
\r
3578 if ( value == kAsioResetRequest
\r
3579 || value == kAsioEngineVersion
\r
3580 || value == kAsioResyncRequest
\r
3581 || value == kAsioLatenciesChanged
\r
3582 // The following three were added for ASIO 2.0, you don't
\r
3583 // necessarily have to support them.
\r
3584 || value == kAsioSupportsTimeInfo
\r
3585 || value == kAsioSupportsTimeCode
\r
3586 || value == kAsioSupportsInputMonitor)
\r
3589 case kAsioResetRequest:
\r
3590 // Defer the task and perform the reset of the driver during the
\r
3591 // next "safe" situation. You cannot reset the driver right now,
\r
3592 // as this code is called from the driver. Reset the driver is
\r
3593 // done by completely destruct is. I.e. ASIOStop(),
\r
3594 // ASIODisposeBuffers(), Destruction Afterwards you initialize the
\r
3596 std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
\r
3599 case kAsioResyncRequest:
\r
3600 // This informs the application that the driver encountered some
\r
3601 // non-fatal data loss. It is used for synchronization purposes
\r
3602 // of different media. Added mainly to work around the Win16Mutex
\r
3603 // problems in Windows 95/98 with the Windows Multimedia system,
\r
3604 // which could lose data because the Mutex was held too long by
\r
3605 // another thread. However a driver can issue it in other
\r
3606 // situations, too.
\r
3607 // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
\r
3611 case kAsioLatenciesChanged:
\r
3612 // This will inform the host application that the drivers were
\r
3613 // latencies changed. Beware, it this does not mean that the
\r
3614 // buffer sizes have changed! You might need to update internal
\r
3616 std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
\r
3619 case kAsioEngineVersion:
\r
3620 // Return the supported ASIO version of the host application. If
\r
3621 // a host application does not implement this selector, ASIO 1.0
\r
3622 // is assumed by the driver.
\r
3625 case kAsioSupportsTimeInfo:
\r
3626 // Informs the driver whether the
\r
3627 // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
\r
3628 // For compatibility with ASIO 1.0 drivers the host application
\r
3629 // should always support the "old" bufferSwitch method, too.
\r
3632 case kAsioSupportsTimeCode:
\r
3633 // Informs the driver whether application is interested in time
\r
3634 // code info. If an application does not need to know about time
\r
3635 // code, the driver has less work to do.
\r
3642 static const char* getAsioErrorString( ASIOError result )
\r
3647 const char*message;
\r
3650 static const Messages m[] =
\r
3652 { ASE_NotPresent, "Hardware input or output is not present or available." },
\r
3653 { ASE_HWMalfunction, "Hardware is malfunctioning." },
\r
3654 { ASE_InvalidParameter, "Invalid input parameter." },
\r
3655 { ASE_InvalidMode, "Invalid mode." },
\r
3656 { ASE_SPNotAdvancing, "Sample position not advancing." },
\r
3657 { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
\r
3658 { ASE_NoMemory, "Not enough memory to complete the request." }
\r
3661 for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
\r
3662 if ( m[i].value == result ) return m[i].message;
\r
3664 return "Unknown error.";
\r
3667 //******************** End of __WINDOWS_ASIO__ *********************//
\r
3671 #if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
\r
3673 // Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
\r
3674 // - Introduces support for the Windows WASAPI API
\r
3675 // - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
\r
3676 // - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
\r
3677 // - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
\r
3682 #include <audioclient.h>
\r
3684 #include <mmdeviceapi.h>
\r
3685 #include <functiondiscoverykeys_devpkey.h>
\r
3687 //=============================================================================
\r
3689 #define SAFE_RELEASE( objectPtr )\
\r
3692 objectPtr->Release();\
\r
3693 objectPtr = NULL;\
\r
3696 typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
\r
3698 //-----------------------------------------------------------------------------
\r
3700 // WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
\r
3701 // Therefore we must perform all necessary conversions to user buffers in order to satisfy these
\r
3702 // requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
\r
3703 // provide intermediate storage for read / write synchronization.
\r
3704 class WasapiBuffer
\r
3708 : buffer_( NULL ),
\r
3717 // sets the length of the internal ring buffer
\r
3718 void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
\r
3721 buffer_ = ( char* ) calloc( bufferSize, formatBytes );
\r
3723 bufferSize_ = bufferSize;
\r
3728 // attempt to push a buffer into the ring buffer at the current "in" index
\r
3729 bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
\r
3731 if ( !buffer || // incoming buffer is NULL
\r
3732 bufferSize == 0 || // incoming buffer has no data
\r
3733 bufferSize > bufferSize_ ) // incoming buffer too large
\r
3738 unsigned int relOutIndex = outIndex_;
\r
3739 unsigned int inIndexEnd = inIndex_ + bufferSize;
\r
3740 if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
\r
3741 relOutIndex += bufferSize_;
\r
3744 // "in" index can end on the "out" index but cannot begin at it
\r
3745 if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
\r
3746 return false; // not enough space between "in" index and "out" index
\r
3749 // copy buffer from external to internal
\r
3750 int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
\r
3751 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
\r
3752 int fromInSize = bufferSize - fromZeroSize;
\r
3756 case RTAUDIO_SINT8:
\r
3757 memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
\r
3758 memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
\r
3760 case RTAUDIO_SINT16:
\r
3761 memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
\r
3762 memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
\r
3764 case RTAUDIO_SINT24:
\r
3765 memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
\r
3766 memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
\r
3768 case RTAUDIO_SINT32:
\r
3769 memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
\r
3770 memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
\r
3772 case RTAUDIO_FLOAT32:
\r
3773 memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
\r
3774 memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
\r
3776 case RTAUDIO_FLOAT64:
\r
3777 memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
\r
3778 memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
\r
3782 // update "in" index
\r
3783 inIndex_ += bufferSize;
\r
3784 inIndex_ %= bufferSize_;
\r
3789 // attempt to pull a buffer from the ring buffer from the current "out" index
\r
3790 bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
\r
3792 if ( !buffer || // incoming buffer is NULL
\r
3793 bufferSize == 0 || // incoming buffer has no data
\r
3794 bufferSize > bufferSize_ ) // incoming buffer too large
\r
3799 unsigned int relInIndex = inIndex_;
\r
3800 unsigned int outIndexEnd = outIndex_ + bufferSize;
\r
3801 if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
\r
3802 relInIndex += bufferSize_;
\r
3805 // "out" index can begin at and end on the "in" index
\r
3806 if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
\r
3807 return false; // not enough space between "out" index and "in" index
\r
3810 // copy buffer from internal to external
\r
3811 int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
\r
3812 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
\r
3813 int fromOutSize = bufferSize - fromZeroSize;
\r
3817 case RTAUDIO_SINT8:
\r
3818 memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
\r
3819 memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
\r
3821 case RTAUDIO_SINT16:
\r
3822 memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
\r
3823 memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
\r
3825 case RTAUDIO_SINT24:
\r
3826 memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
\r
3827 memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
\r
3829 case RTAUDIO_SINT32:
\r
3830 memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
\r
3831 memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
\r
3833 case RTAUDIO_FLOAT32:
\r
3834 memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
\r
3835 memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
\r
3837 case RTAUDIO_FLOAT64:
\r
3838 memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
\r
3839 memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
\r
3843 // update "out" index
\r
3844 outIndex_ += bufferSize;
\r
3845 outIndex_ %= bufferSize_;
\r
3852 unsigned int bufferSize_;
\r
3853 unsigned int inIndex_;
\r
3854 unsigned int outIndex_;
\r
3857 //-----------------------------------------------------------------------------
\r
3859 // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
\r
3860 // between HW and the user. The convertBufferWasapi function is used to perform this conversion
\r
3861 // between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
\r
3862 // This sample rate converter favors speed over quality, and works best with conversions between
\r
3863 // one rate and its multiple.
\r
3864 void convertBufferWasapi( char* outBuffer,
\r
3865 const char* inBuffer,
\r
3866 const unsigned int& channelCount,
\r
3867 const unsigned int& inSampleRate,
\r
3868 const unsigned int& outSampleRate,
\r
3869 const unsigned int& inSampleCount,
\r
3870 unsigned int& outSampleCount,
\r
3871 const RtAudioFormat& format )
\r
3873 // calculate the new outSampleCount and relative sampleStep
\r
3874 float sampleRatio = ( float ) outSampleRate / inSampleRate;
\r
3875 float sampleStep = 1.0f / sampleRatio;
\r
3876 float inSampleFraction = 0.0f;
\r
3878 outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );
\r
3880 // frame-by-frame, copy each relative input sample into it's corresponding output sample
\r
3881 for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
\r
3883 unsigned int inSample = ( unsigned int ) inSampleFraction;
\r
3887 case RTAUDIO_SINT8:
\r
3888 memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
\r
3890 case RTAUDIO_SINT16:
\r
3891 memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
\r
3893 case RTAUDIO_SINT24:
\r
3894 memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
\r
3896 case RTAUDIO_SINT32:
\r
3897 memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
\r
3899 case RTAUDIO_FLOAT32:
\r
3900 memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
\r
3902 case RTAUDIO_FLOAT64:
\r
3903 memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
\r
3907 // jump to next in sample
\r
3908 inSampleFraction += sampleStep;
\r
3912 //-----------------------------------------------------------------------------
\r
3914 // A structure to hold various information related to the WASAPI implementation.
\r
3915 struct WasapiHandle
\r
3917 IAudioClient* captureAudioClient;
\r
3918 IAudioClient* renderAudioClient;
\r
3919 IAudioCaptureClient* captureClient;
\r
3920 IAudioRenderClient* renderClient;
\r
3921 HANDLE captureEvent;
\r
3922 HANDLE renderEvent;
\r
3925 : captureAudioClient( NULL ),
\r
3926 renderAudioClient( NULL ),
\r
3927 captureClient( NULL ),
\r
3928 renderClient( NULL ),
\r
3929 captureEvent( NULL ),
\r
3930 renderEvent( NULL ) {}
\r
3933 //=============================================================================
\r
3935 RtApiWasapi::RtApiWasapi()
\r
3936 : coInitialized_( false ), deviceEnumerator_( NULL )
\r
3938 // WASAPI can run either apartment or multi-threaded
\r
3939 HRESULT hr = CoInitialize( NULL );
\r
3940 if ( !FAILED( hr ) )
\r
3941 coInitialized_ = true;
\r
3943 // Instantiate device enumerator
\r
3944 hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
\r
3945 CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
\r
3946 ( void** ) &deviceEnumerator_ );
\r
3948 if ( FAILED( hr ) ) {
\r
3949 errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
\r
3950 error( RtAudioError::DRIVER_ERROR );
\r
3954 //-----------------------------------------------------------------------------
\r
3956 RtApiWasapi::~RtApiWasapi()
\r
3958 if ( stream_.state != STREAM_CLOSED )
\r
3961 SAFE_RELEASE( deviceEnumerator_ );
\r
3963 // If this object previously called CoInitialize()
\r
3964 if ( coInitialized_ )
\r
3968 //=============================================================================
\r
3970 unsigned int RtApiWasapi::getDeviceCount( void )
\r
3972 unsigned int captureDeviceCount = 0;
\r
3973 unsigned int renderDeviceCount = 0;
\r
3975 IMMDeviceCollection* captureDevices = NULL;
\r
3976 IMMDeviceCollection* renderDevices = NULL;
\r
3978 // Count capture devices
\r
3979 errorText_.clear();
\r
3980 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
\r
3981 if ( FAILED( hr ) ) {
\r
3982 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
\r
3986 hr = captureDevices->GetCount( &captureDeviceCount );
\r
3987 if ( FAILED( hr ) ) {
\r
3988 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
\r
3992 // Count render devices
\r
3993 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
\r
3994 if ( FAILED( hr ) ) {
\r
3995 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
\r
3999 hr = renderDevices->GetCount( &renderDeviceCount );
\r
4000 if ( FAILED( hr ) ) {
\r
4001 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
\r
4006 // release all references
\r
4007 SAFE_RELEASE( captureDevices );
\r
4008 SAFE_RELEASE( renderDevices );
\r
4010 if ( errorText_.empty() )
\r
4011 return captureDeviceCount + renderDeviceCount;
\r
4013 error( RtAudioError::DRIVER_ERROR );
\r
4017 //-----------------------------------------------------------------------------
\r
4019 RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
\r
4021 RtAudio::DeviceInfo info;
\r
4022 unsigned int captureDeviceCount = 0;
\r
4023 unsigned int renderDeviceCount = 0;
\r
4024 std::string defaultDeviceName;
\r
4025 bool isCaptureDevice = false;
\r
4027 PROPVARIANT deviceNameProp;
\r
4028 PROPVARIANT defaultDeviceNameProp;
\r
4030 IMMDeviceCollection* captureDevices = NULL;
\r
4031 IMMDeviceCollection* renderDevices = NULL;
\r
4032 IMMDevice* devicePtr = NULL;
\r
4033 IMMDevice* defaultDevicePtr = NULL;
\r
4034 IAudioClient* audioClient = NULL;
\r
4035 IPropertyStore* devicePropStore = NULL;
\r
4036 IPropertyStore* defaultDevicePropStore = NULL;
\r
4038 WAVEFORMATEX* deviceFormat = NULL;
\r
4039 WAVEFORMATEX* closestMatchFormat = NULL;
\r
4042 info.probed = false;
\r
4044 // Count capture devices
\r
4045 errorText_.clear();
\r
4046 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
\r
4047 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
\r
4048 if ( FAILED( hr ) ) {
\r
4049 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
\r
4053 hr = captureDevices->GetCount( &captureDeviceCount );
\r
4054 if ( FAILED( hr ) ) {
\r
4055 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
\r
4059 // Count render devices
\r
4060 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
\r
4061 if ( FAILED( hr ) ) {
\r
4062 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
\r
4066 hr = renderDevices->GetCount( &renderDeviceCount );
\r
4067 if ( FAILED( hr ) ) {
\r
4068 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
\r
4072 // validate device index
\r
4073 if ( device >= captureDeviceCount + renderDeviceCount ) {
\r
4074 errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
\r
4075 errorType = RtAudioError::INVALID_USE;
\r
4079 // determine whether index falls within capture or render devices
\r
4080 if ( device >= renderDeviceCount ) {
\r
4081 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
\r
4082 if ( FAILED( hr ) ) {
\r
4083 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
\r
4086 isCaptureDevice = true;
\r
4089 hr = renderDevices->Item( device, &devicePtr );
\r
4090 if ( FAILED( hr ) ) {
\r
4091 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
\r
4094 isCaptureDevice = false;
\r
4097 // get default device name
\r
4098 if ( isCaptureDevice ) {
\r
4099 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
\r
4100 if ( FAILED( hr ) ) {
\r
4101 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
\r
4106 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
\r
4107 if ( FAILED( hr ) ) {
\r
4108 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
\r
4113 hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
\r
4114 if ( FAILED( hr ) ) {
\r
4115 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
\r
4118 PropVariantInit( &defaultDeviceNameProp );
\r
4120 hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
\r
4121 if ( FAILED( hr ) ) {
\r
4122 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
\r
4126 defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
\r
4129 hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
\r
4130 if ( FAILED( hr ) ) {
\r
4131 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
\r
4135 PropVariantInit( &deviceNameProp );
\r
4137 hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
\r
4138 if ( FAILED( hr ) ) {
\r
4139 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
\r
4143 info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
\r
4146 if ( isCaptureDevice ) {
\r
4147 info.isDefaultInput = info.name == defaultDeviceName;
\r
4148 info.isDefaultOutput = false;
\r
4151 info.isDefaultInput = false;
\r
4152 info.isDefaultOutput = info.name == defaultDeviceName;
\r
4156 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
\r
4157 if ( FAILED( hr ) ) {
\r
4158 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
\r
4162 hr = audioClient->GetMixFormat( &deviceFormat );
\r
4163 if ( FAILED( hr ) ) {
\r
4164 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
\r
4168 if ( isCaptureDevice ) {
\r
4169 info.inputChannels = deviceFormat->nChannels;
\r
4170 info.outputChannels = 0;
\r
4171 info.duplexChannels = 0;
\r
4174 info.inputChannels = 0;
\r
4175 info.outputChannels = deviceFormat->nChannels;
\r
4176 info.duplexChannels = 0;
\r
4180 info.sampleRates.clear();
\r
4182 // allow support for all sample rates as we have a built-in sample rate converter
\r
4183 for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
\r
4184 info.sampleRates.push_back( SAMPLE_RATES[i] );
\r
4186 info.preferredSampleRate = deviceFormat->nSamplesPerSec;
\r
4189 info.nativeFormats = 0;
\r
4191 if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
\r
4192 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
\r
4193 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
\r
4195 if ( deviceFormat->wBitsPerSample == 32 ) {
\r
4196 info.nativeFormats |= RTAUDIO_FLOAT32;
\r
4198 else if ( deviceFormat->wBitsPerSample == 64 ) {
\r
4199 info.nativeFormats |= RTAUDIO_FLOAT64;
\r
4202 else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
\r
4203 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
\r
4204 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
\r
4206 if ( deviceFormat->wBitsPerSample == 8 ) {
\r
4207 info.nativeFormats |= RTAUDIO_SINT8;
\r
4209 else if ( deviceFormat->wBitsPerSample == 16 ) {
\r
4210 info.nativeFormats |= RTAUDIO_SINT16;
\r
4212 else if ( deviceFormat->wBitsPerSample == 24 ) {
\r
4213 info.nativeFormats |= RTAUDIO_SINT24;
\r
4215 else if ( deviceFormat->wBitsPerSample == 32 ) {
\r
4216 info.nativeFormats |= RTAUDIO_SINT32;
\r
4221 info.probed = true;
\r
4224 // release all references
\r
4225 PropVariantClear( &deviceNameProp );
\r
4226 PropVariantClear( &defaultDeviceNameProp );
\r
4228 SAFE_RELEASE( captureDevices );
\r
4229 SAFE_RELEASE( renderDevices );
\r
4230 SAFE_RELEASE( devicePtr );
\r
4231 SAFE_RELEASE( defaultDevicePtr );
\r
4232 SAFE_RELEASE( audioClient );
\r
4233 SAFE_RELEASE( devicePropStore );
\r
4234 SAFE_RELEASE( defaultDevicePropStore );
\r
4236 CoTaskMemFree( deviceFormat );
\r
4237 CoTaskMemFree( closestMatchFormat );
\r
4239 if ( !errorText_.empty() )
\r
4240 error( errorType );
\r
4244 //-----------------------------------------------------------------------------
\r
4246 unsigned int RtApiWasapi::getDefaultOutputDevice( void )
\r
4248 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
\r
4249 if ( getDeviceInfo( i ).isDefaultOutput ) {
\r
4257 //-----------------------------------------------------------------------------
\r
4259 unsigned int RtApiWasapi::getDefaultInputDevice( void )
\r
4261 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
\r
4262 if ( getDeviceInfo( i ).isDefaultInput ) {
\r
4270 //-----------------------------------------------------------------------------
\r
4272 void RtApiWasapi::closeStream( void )
\r
4274 if ( stream_.state == STREAM_CLOSED ) {
\r
4275 errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
\r
4276 error( RtAudioError::WARNING );
\r
4280 if ( stream_.state != STREAM_STOPPED )
\r
4283 // clean up stream memory
\r
4284 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
\r
4285 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
\r
4287 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
\r
4288 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
\r
4290 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
\r
4291 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
\r
4293 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
\r
4294 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
\r
4296 delete ( WasapiHandle* ) stream_.apiHandle;
\r
4297 stream_.apiHandle = NULL;
\r
4299 for ( int i = 0; i < 2; i++ ) {
\r
4300 if ( stream_.userBuffer[i] ) {
\r
4301 free( stream_.userBuffer[i] );
\r
4302 stream_.userBuffer[i] = 0;
\r
4306 if ( stream_.deviceBuffer ) {
\r
4307 free( stream_.deviceBuffer );
\r
4308 stream_.deviceBuffer = 0;
\r
4311 // update stream state
\r
4312 stream_.state = STREAM_CLOSED;
\r
4315 //-----------------------------------------------------------------------------
\r
4317 void RtApiWasapi::startStream( void )
\r
4321 if ( stream_.state == STREAM_RUNNING ) {
\r
4322 errorText_ = "RtApiWasapi::startStream: The stream is already running.";
\r
4323 error( RtAudioError::WARNING );
\r
4327 // update stream state
\r
4328 stream_.state = STREAM_RUNNING;
\r
4330 // create WASAPI stream thread
\r
4331 stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
\r
4333 if ( !stream_.callbackInfo.thread ) {
\r
4334 errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
\r
4335 error( RtAudioError::THREAD_ERROR );
\r
4338 SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
\r
4339 ResumeThread( ( void* ) stream_.callbackInfo.thread );
\r
4343 //-----------------------------------------------------------------------------
\r
4345 void RtApiWasapi::stopStream( void )
\r
4349 if ( stream_.state == STREAM_STOPPED ) {
\r
4350 errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
\r
4351 error( RtAudioError::WARNING );
\r
4355 // inform stream thread by setting stream state to STREAM_STOPPING
\r
4356 stream_.state = STREAM_STOPPING;
\r
4358 // wait until stream thread is stopped
\r
4359 while( stream_.state != STREAM_STOPPED ) {
\r
4363 // Wait for the last buffer to play before stopping.
\r
4364 Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
\r
4366 // stop capture client if applicable
\r
4367 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
\r
4368 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
\r
4369 if ( FAILED( hr ) ) {
\r
4370 errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
\r
4371 error( RtAudioError::DRIVER_ERROR );
\r
4376 // stop render client if applicable
\r
4377 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
\r
4378 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
\r
4379 if ( FAILED( hr ) ) {
\r
4380 errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
\r
4381 error( RtAudioError::DRIVER_ERROR );
\r
4386 // close thread handle
\r
4387 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
\r
4388 errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
\r
4389 error( RtAudioError::THREAD_ERROR );
\r
4393 stream_.callbackInfo.thread = (ThreadHandle) NULL;
\r
4396 //-----------------------------------------------------------------------------
\r
4398 void RtApiWasapi::abortStream( void )
\r
4402 if ( stream_.state == STREAM_STOPPED ) {
\r
4403 errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
\r
4404 error( RtAudioError::WARNING );
\r
4408 // inform stream thread by setting stream state to STREAM_STOPPING
\r
4409 stream_.state = STREAM_STOPPING;
\r
4411 // wait until stream thread is stopped
\r
4412 while ( stream_.state != STREAM_STOPPED ) {
\r
4416 // stop capture client if applicable
\r
4417 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
\r
4418 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
\r
4419 if ( FAILED( hr ) ) {
\r
4420 errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
\r
4421 error( RtAudioError::DRIVER_ERROR );
\r
4426 // stop render client if applicable
\r
4427 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
\r
4428 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
\r
4429 if ( FAILED( hr ) ) {
\r
4430 errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
\r
4431 error( RtAudioError::DRIVER_ERROR );
\r
4436 // close thread handle
\r
4437 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
\r
4438 errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
\r
4439 error( RtAudioError::THREAD_ERROR );
\r
4443 stream_.callbackInfo.thread = (ThreadHandle) NULL;
\r
4446 //-----------------------------------------------------------------------------
\r
4448 bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
4449 unsigned int firstChannel, unsigned int sampleRate,
\r
4450 RtAudioFormat format, unsigned int* bufferSize,
\r
4451 RtAudio::StreamOptions* options )
\r
4453 bool methodResult = FAILURE;
\r
4454 unsigned int captureDeviceCount = 0;
\r
4455 unsigned int renderDeviceCount = 0;
\r
4457 IMMDeviceCollection* captureDevices = NULL;
\r
4458 IMMDeviceCollection* renderDevices = NULL;
\r
4459 IMMDevice* devicePtr = NULL;
\r
4460 WAVEFORMATEX* deviceFormat = NULL;
\r
4461 unsigned int bufferBytes;
\r
4462 stream_.state = STREAM_STOPPED;
\r
4464 // create API Handle if not already created
\r
4465 if ( !stream_.apiHandle )
\r
4466 stream_.apiHandle = ( void* ) new WasapiHandle();
\r
4468 // Count capture devices
\r
4469 errorText_.clear();
\r
4470 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
\r
4471 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
\r
4472 if ( FAILED( hr ) ) {
\r
4473 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
\r
4477 hr = captureDevices->GetCount( &captureDeviceCount );
\r
4478 if ( FAILED( hr ) ) {
\r
4479 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
\r
4483 // Count render devices
\r
4484 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
\r
4485 if ( FAILED( hr ) ) {
\r
4486 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
\r
4490 hr = renderDevices->GetCount( &renderDeviceCount );
\r
4491 if ( FAILED( hr ) ) {
\r
4492 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
\r
4496 // validate device index
\r
4497 if ( device >= captureDeviceCount + renderDeviceCount ) {
\r
4498 errorType = RtAudioError::INVALID_USE;
\r
4499 errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
\r
4503 // determine whether index falls within capture or render devices
\r
4504 if ( device >= renderDeviceCount ) {
\r
4505 if ( mode != INPUT ) {
\r
4506 errorType = RtAudioError::INVALID_USE;
\r
4507 errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
\r
4511 // retrieve captureAudioClient from devicePtr
\r
4512 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
\r
4514 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
\r
4515 if ( FAILED( hr ) ) {
\r
4516 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
\r
4520 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
\r
4521 NULL, ( void** ) &captureAudioClient );
\r
4522 if ( FAILED( hr ) ) {
\r
4523 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
\r
4527 hr = captureAudioClient->GetMixFormat( &deviceFormat );
\r
4528 if ( FAILED( hr ) ) {
\r
4529 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
\r
4533 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
\r
4534 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
\r
4537 if ( mode != OUTPUT ) {
\r
4538 errorType = RtAudioError::INVALID_USE;
\r
4539 errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
\r
4543 // retrieve renderAudioClient from devicePtr
\r
4544 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
\r
4546 hr = renderDevices->Item( device, &devicePtr );
\r
4547 if ( FAILED( hr ) ) {
\r
4548 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
\r
4552 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
\r
4553 NULL, ( void** ) &renderAudioClient );
\r
4554 if ( FAILED( hr ) ) {
\r
4555 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
\r
4559 hr = renderAudioClient->GetMixFormat( &deviceFormat );
\r
4560 if ( FAILED( hr ) ) {
\r
4561 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
\r
4565 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
\r
4566 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
\r
4569 // fill stream data
\r
4570 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
\r
4571 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
\r
4572 stream_.mode = DUPLEX;
\r
4575 stream_.mode = mode;
\r
4578 stream_.device[mode] = device;
\r
4579 stream_.doByteSwap[mode] = false;
\r
4580 stream_.sampleRate = sampleRate;
\r
4581 stream_.bufferSize = *bufferSize;
\r
4582 stream_.nBuffers = 1;
\r
4583 stream_.nUserChannels[mode] = channels;
\r
4584 stream_.channelOffset[mode] = firstChannel;
\r
4585 stream_.userFormat = format;
\r
4586 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
\r
4588 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
\r
4589 stream_.userInterleaved = false;
\r
4591 stream_.userInterleaved = true;
\r
4592 stream_.deviceInterleaved[mode] = true;
\r
4594 // Set flags for buffer conversion.
\r
4595 stream_.doConvertBuffer[mode] = false;
\r
4596 if ( stream_.userFormat != stream_.deviceFormat[mode] ||
\r
4597 stream_.nUserChannels != stream_.nDeviceChannels )
\r
4598 stream_.doConvertBuffer[mode] = true;
\r
4599 else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
\r
4600 stream_.nUserChannels[mode] > 1 )
\r
4601 stream_.doConvertBuffer[mode] = true;
\r
4603 if ( stream_.doConvertBuffer[mode] )
\r
4604 setConvertInfo( mode, 0 );
\r
4606 // Allocate necessary internal buffers
\r
4607 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
\r
4609 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
\r
4610 if ( !stream_.userBuffer[mode] ) {
\r
4611 errorType = RtAudioError::MEMORY_ERROR;
\r
4612 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
\r
4616 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
\r
4617 stream_.callbackInfo.priority = 15;
\r
4619 stream_.callbackInfo.priority = 0;
\r
4621 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
\r
4622 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
\r
4624 methodResult = SUCCESS;
\r
4628 SAFE_RELEASE( captureDevices );
\r
4629 SAFE_RELEASE( renderDevices );
\r
4630 SAFE_RELEASE( devicePtr );
\r
4631 CoTaskMemFree( deviceFormat );
\r
4633 // if method failed, close the stream
\r
4634 if ( methodResult == FAILURE )
\r
4637 if ( !errorText_.empty() )
\r
4638 error( errorType );
\r
4639 return methodResult;
\r
4642 //=============================================================================
\r
4644 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
\r
4647 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
\r
4652 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
\r
4655 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
\r
4660 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
\r
4663 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
\r
4668 //-----------------------------------------------------------------------------
\r
4670 void RtApiWasapi::wasapiThread()
\r
4672 // as this is a new thread, we must CoInitialize it
\r
4673 CoInitialize( NULL );
\r
4677 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
\r
4678 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
\r
4679 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
\r
4680 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
\r
4681 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
\r
4682 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
\r
4684 WAVEFORMATEX* captureFormat = NULL;
\r
4685 WAVEFORMATEX* renderFormat = NULL;
\r
4686 float captureSrRatio = 0.0f;
\r
4687 float renderSrRatio = 0.0f;
\r
4688 WasapiBuffer captureBuffer;
\r
4689 WasapiBuffer renderBuffer;
\r
4691 // declare local stream variables
\r
4692 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
\r
4693 BYTE* streamBuffer = NULL;
\r
4694 unsigned long captureFlags = 0;
\r
4695 unsigned int bufferFrameCount = 0;
\r
4696 unsigned int numFramesPadding = 0;
\r
4697 unsigned int convBufferSize = 0;
\r
4698 bool callbackPushed = false;
\r
4699 bool callbackPulled = false;
\r
4700 bool callbackStopped = false;
\r
4701 int callbackResult = 0;
\r
4703 // convBuffer is used to store converted buffers between WASAPI and the user
\r
4704 char* convBuffer = NULL;
\r
4705 unsigned int convBuffSize = 0;
\r
4706 unsigned int deviceBuffSize = 0;
\r
4708 errorText_.clear();
\r
4709 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
\r
4711 // Attempt to assign "Pro Audio" characteristic to thread
\r
4712 HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
\r
4714 DWORD taskIndex = 0;
\r
4715 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
\r
4716 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
\r
4717 FreeLibrary( AvrtDll );
\r
4720 // start capture stream if applicable
\r
4721 if ( captureAudioClient ) {
\r
4722 hr = captureAudioClient->GetMixFormat( &captureFormat );
\r
4723 if ( FAILED( hr ) ) {
\r
4724 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
\r
4728 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
\r
4730 // initialize capture stream according to desire buffer size
\r
4731 float desiredBufferSize = stream_.bufferSize * captureSrRatio;
\r
4732 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
\r
4734 if ( !captureClient ) {
\r
4735 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
\r
4736 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
\r
4737 desiredBufferPeriod,
\r
4738 desiredBufferPeriod,
\r
4741 if ( FAILED( hr ) ) {
\r
4742 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
\r
4746 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
\r
4747 ( void** ) &captureClient );
\r
4748 if ( FAILED( hr ) ) {
\r
4749 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
\r
4753 // configure captureEvent to trigger on every available capture buffer
\r
4754 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
\r
4755 if ( !captureEvent ) {
\r
4756 errorType = RtAudioError::SYSTEM_ERROR;
\r
4757 errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
\r
4761 hr = captureAudioClient->SetEventHandle( captureEvent );
\r
4762 if ( FAILED( hr ) ) {
\r
4763 errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
\r
4767 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
\r
4768 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
\r
4771 unsigned int inBufferSize = 0;
\r
4772 hr = captureAudioClient->GetBufferSize( &inBufferSize );
\r
4773 if ( FAILED( hr ) ) {
\r
4774 errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
\r
4778 // scale outBufferSize according to stream->user sample rate ratio
\r
4779 unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
\r
4780 inBufferSize *= stream_.nDeviceChannels[INPUT];
\r
4782 // set captureBuffer size
\r
4783 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
\r
4785 // reset the capture stream
\r
4786 hr = captureAudioClient->Reset();
\r
4787 if ( FAILED( hr ) ) {
\r
4788 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
\r
4792 // start the capture stream
\r
4793 hr = captureAudioClient->Start();
\r
4794 if ( FAILED( hr ) ) {
\r
4795 errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
\r
4800 // start render stream if applicable
\r
4801 if ( renderAudioClient ) {
\r
4802 hr = renderAudioClient->GetMixFormat( &renderFormat );
\r
4803 if ( FAILED( hr ) ) {
\r
4804 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
\r
4808 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
\r
4810 // initialize render stream according to desire buffer size
\r
4811 float desiredBufferSize = stream_.bufferSize * renderSrRatio;
\r
4812 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
\r
4814 if ( !renderClient ) {
\r
4815 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
\r
4816 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
\r
4817 desiredBufferPeriod,
\r
4818 desiredBufferPeriod,
\r
4821 if ( FAILED( hr ) ) {
\r
4822 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
\r
4826 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
\r
4827 ( void** ) &renderClient );
\r
4828 if ( FAILED( hr ) ) {
\r
4829 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
\r
4833 // configure renderEvent to trigger on every available render buffer
\r
4834 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
\r
4835 if ( !renderEvent ) {
\r
4836 errorType = RtAudioError::SYSTEM_ERROR;
\r
4837 errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
\r
4841 hr = renderAudioClient->SetEventHandle( renderEvent );
\r
4842 if ( FAILED( hr ) ) {
\r
4843 errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
\r
4847 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
\r
4848 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
\r
4851 unsigned int outBufferSize = 0;
\r
4852 hr = renderAudioClient->GetBufferSize( &outBufferSize );
\r
4853 if ( FAILED( hr ) ) {
\r
4854 errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
\r
4858 // scale inBufferSize according to user->stream sample rate ratio
\r
4859 unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
\r
4860 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
\r
4862 // set renderBuffer size
\r
4863 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
\r
4865 // reset the render stream
\r
4866 hr = renderAudioClient->Reset();
\r
4867 if ( FAILED( hr ) ) {
\r
4868 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
\r
4872 // start the render stream
\r
4873 hr = renderAudioClient->Start();
\r
4874 if ( FAILED( hr ) ) {
\r
4875 errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
\r
4880 if ( stream_.mode == INPUT ) {
\r
4881 convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
\r
4882 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
\r
4884 else if ( stream_.mode == OUTPUT ) {
\r
4885 convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
\r
4886 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
\r
4888 else if ( stream_.mode == DUPLEX ) {
\r
4889 convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
\r
4890 ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
\r
4891 deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
\r
4892 stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
\r
4895 convBuffer = ( char* ) malloc( convBuffSize );
\r
4896 stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
\r
4897 if ( !convBuffer || !stream_.deviceBuffer ) {
\r
4898 errorType = RtAudioError::MEMORY_ERROR;
\r
4899 errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
\r
4903 // stream process loop
\r
4904 while ( stream_.state != STREAM_STOPPING ) {
\r
4905 if ( !callbackPulled ) {
\r
4908 // 1. Pull callback buffer from inputBuffer
\r
4909 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
\r
4910 // Convert callback buffer to user format
\r
4912 if ( captureAudioClient ) {
\r
4913 // Pull callback buffer from inputBuffer
\r
4914 callbackPulled = captureBuffer.pullBuffer( convBuffer,
\r
4915 ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
\r
4916 stream_.deviceFormat[INPUT] );
\r
4918 if ( callbackPulled ) {
\r
4919 // Convert callback buffer to user sample rate
\r
4920 convertBufferWasapi( stream_.deviceBuffer,
\r
4922 stream_.nDeviceChannels[INPUT],
\r
4923 captureFormat->nSamplesPerSec,
\r
4924 stream_.sampleRate,
\r
4925 ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
\r
4927 stream_.deviceFormat[INPUT] );
\r
4929 if ( stream_.doConvertBuffer[INPUT] ) {
\r
4930 // Convert callback buffer to user format
\r
4931 convertBuffer( stream_.userBuffer[INPUT],
\r
4932 stream_.deviceBuffer,
\r
4933 stream_.convertInfo[INPUT] );
\r
4936 // no further conversion, simple copy deviceBuffer to userBuffer
\r
4937 memcpy( stream_.userBuffer[INPUT],
\r
4938 stream_.deviceBuffer,
\r
4939 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
\r
4944 // if there is no capture stream, set callbackPulled flag
\r
4945 callbackPulled = true;
\r
4948 // Execute Callback
\r
4949 // ================
\r
4950 // 1. Execute user callback method
\r
4951 // 2. Handle return value from callback
\r
4953 // if callback has not requested the stream to stop
\r
4954 if ( callbackPulled && !callbackStopped ) {
\r
4955 // Execute user callback method
\r
4956 callbackResult = callback( stream_.userBuffer[OUTPUT],
\r
4957 stream_.userBuffer[INPUT],
\r
4958 stream_.bufferSize,
\r
4960 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
\r
4961 stream_.callbackInfo.userData );
\r
4963 // Handle return value from callback
\r
4964 if ( callbackResult == 1 ) {
\r
4965 // instantiate a thread to stop this thread
\r
4966 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
\r
4967 if ( !threadHandle ) {
\r
4968 errorType = RtAudioError::THREAD_ERROR;
\r
4969 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
\r
4972 else if ( !CloseHandle( threadHandle ) ) {
\r
4973 errorType = RtAudioError::THREAD_ERROR;
\r
4974 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
\r
4978 callbackStopped = true;
\r
4980 else if ( callbackResult == 2 ) {
\r
4981 // instantiate a thread to stop this thread
\r
4982 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
\r
4983 if ( !threadHandle ) {
\r
4984 errorType = RtAudioError::THREAD_ERROR;
\r
4985 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
\r
4988 else if ( !CloseHandle( threadHandle ) ) {
\r
4989 errorType = RtAudioError::THREAD_ERROR;
\r
4990 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
\r
4994 callbackStopped = true;
\r
4999 // Callback Output
\r
5000 // ===============
\r
5001 // 1. Convert callback buffer to stream format
\r
5002 // 2. Convert callback buffer to stream sample rate and channel count
\r
5003 // 3. Push callback buffer into outputBuffer
\r
5005 if ( renderAudioClient && callbackPulled ) {
\r
5006 if ( stream_.doConvertBuffer[OUTPUT] ) {
\r
5007 // Convert callback buffer to stream format
\r
5008 convertBuffer( stream_.deviceBuffer,
\r
5009 stream_.userBuffer[OUTPUT],
\r
5010 stream_.convertInfo[OUTPUT] );
\r
5014 // Convert callback buffer to stream sample rate
\r
5015 convertBufferWasapi( convBuffer,
\r
5016 stream_.deviceBuffer,
\r
5017 stream_.nDeviceChannels[OUTPUT],
\r
5018 stream_.sampleRate,
\r
5019 renderFormat->nSamplesPerSec,
\r
5020 stream_.bufferSize,
\r
5022 stream_.deviceFormat[OUTPUT] );
\r
5024 // Push callback buffer into outputBuffer
\r
5025 callbackPushed = renderBuffer.pushBuffer( convBuffer,
\r
5026 convBufferSize * stream_.nDeviceChannels[OUTPUT],
\r
5027 stream_.deviceFormat[OUTPUT] );
\r
5030 // if there is no render stream, set callbackPushed flag
\r
5031 callbackPushed = true;
\r
5036 // 1. Get capture buffer from stream
\r
5037 // 2. Push capture buffer into inputBuffer
\r
5038 // 3. If 2. was successful: Release capture buffer
\r
5040 if ( captureAudioClient ) {
\r
5041 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
\r
5042 if ( !callbackPulled ) {
\r
5043 WaitForSingleObject( captureEvent, INFINITE );
\r
5046 // Get capture buffer from stream
\r
5047 hr = captureClient->GetBuffer( &streamBuffer,
\r
5048 &bufferFrameCount,
\r
5049 &captureFlags, NULL, NULL );
\r
5050 if ( FAILED( hr ) ) {
\r
5051 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
\r
5055 if ( bufferFrameCount != 0 ) {
\r
5056 // Push capture buffer into inputBuffer
\r
5057 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
\r
5058 bufferFrameCount * stream_.nDeviceChannels[INPUT],
\r
5059 stream_.deviceFormat[INPUT] ) )
\r
5061 // Release capture buffer
\r
5062 hr = captureClient->ReleaseBuffer( bufferFrameCount );
\r
5063 if ( FAILED( hr ) ) {
\r
5064 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
\r
5070 // Inform WASAPI that capture was unsuccessful
\r
5071 hr = captureClient->ReleaseBuffer( 0 );
\r
5072 if ( FAILED( hr ) ) {
\r
5073 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
\r
5080 // Inform WASAPI that capture was unsuccessful
\r
5081 hr = captureClient->ReleaseBuffer( 0 );
\r
5082 if ( FAILED( hr ) ) {
\r
5083 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
\r
5091 // 1. Get render buffer from stream
\r
5092 // 2. Pull next buffer from outputBuffer
\r
5093 // 3. If 2. was successful: Fill render buffer with next buffer
\r
5094 // Release render buffer
\r
5096 if ( renderAudioClient ) {
\r
5097 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
\r
5098 if ( callbackPulled && !callbackPushed ) {
\r
5099 WaitForSingleObject( renderEvent, INFINITE );
\r
5102 // Get render buffer from stream
\r
5103 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
\r
5104 if ( FAILED( hr ) ) {
\r
5105 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
\r
5109 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
\r
5110 if ( FAILED( hr ) ) {
\r
5111 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
\r
5115 bufferFrameCount -= numFramesPadding;
\r
5117 if ( bufferFrameCount != 0 ) {
\r
5118 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
\r
5119 if ( FAILED( hr ) ) {
\r
5120 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
\r
5124 // Pull next buffer from outputBuffer
\r
5125 // Fill render buffer with next buffer
\r
5126 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
\r
5127 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
\r
5128 stream_.deviceFormat[OUTPUT] ) )
\r
5130 // Release render buffer
\r
5131 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
\r
5132 if ( FAILED( hr ) ) {
\r
5133 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
\r
5139 // Inform WASAPI that render was unsuccessful
\r
5140 hr = renderClient->ReleaseBuffer( 0, 0 );
\r
5141 if ( FAILED( hr ) ) {
\r
5142 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
\r
5149 // Inform WASAPI that render was unsuccessful
\r
5150 hr = renderClient->ReleaseBuffer( 0, 0 );
\r
5151 if ( FAILED( hr ) ) {
\r
5152 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
\r
5158 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
\r
5159 if ( callbackPushed ) {
\r
5160 callbackPulled = false;
\r
5161 // tick stream time
\r
5162 RtApi::tickStreamTime();
\r
5169 CoTaskMemFree( captureFormat );
\r
5170 CoTaskMemFree( renderFormat );
\r
5172 free ( convBuffer );
\r
5176 // update stream state
\r
5177 stream_.state = STREAM_STOPPED;
\r
5179 if ( errorText_.empty() )
\r
5182 error( errorType );
\r
5185 //******************** End of __WINDOWS_WASAPI__ *********************//
\r
5189 #if defined(__WINDOWS_DS__) // Windows DirectSound API
\r
5191 // Modified by Robin Davies, October 2005
\r
5192 // - Improvements to DirectX pointer chasing.
\r
5193 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
\r
5194 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
\r
5195 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
\r
5196 // Changed device query structure for RtAudio 4.0.7, January 2010
\r
5198 #include <mmsystem.h>
\r
5199 #include <mmreg.h>
\r
5200 #include <dsound.h>
\r
5201 #include <assert.h>
\r
5202 #include <algorithm>
\r
5204 #if defined(__MINGW32__)
\r
5205 // missing from latest mingw winapi
\r
5206 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
\r
5207 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
\r
5208 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
\r
5209 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
\r
5212 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
\r
5214 #ifdef _MSC_VER // if Microsoft Visual C++
\r
5215 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
\r
5218 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
\r
5220 if ( pointer > bufferSize ) pointer -= bufferSize;
\r
5221 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
\r
5222 if ( pointer < earlierPointer ) pointer += bufferSize;
\r
5223 return pointer >= earlierPointer && pointer < laterPointer;
\r
5226 // A structure to hold various information related to the DirectSound
\r
5227 // API implementation.
\r
5229 unsigned int drainCounter; // Tracks callback counts when draining
\r
5230 bool internalDrain; // Indicates if stop is initiated from callback or not.
\r
5234 UINT bufferPointer[2];
\r
5235 DWORD dsBufferSize[2];
\r
5236 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
\r
5240 :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
\r
5243 // Declarations for utility functions, callbacks, and structures
\r
5244 // specific to the DirectSound implementation.
\r
5245 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
\r
5246 LPCTSTR description,
\r
5248 LPVOID lpContext );
\r
5250 static const char* getErrorString( int code );
\r
5252 static unsigned __stdcall callbackHandler( void *ptr );
\r
5261 : found(false) { validId[0] = false; validId[1] = false; }
\r
5264 struct DsProbeData {
\r
5266 std::vector<struct DsDevice>* dsDevices;
\r
5269 RtApiDs :: RtApiDs()
\r
5271 // Dsound will run both-threaded. If CoInitialize fails, then just
\r
5272 // accept whatever the mainline chose for a threading model.
\r
5273 coInitialized_ = false;
\r
5274 HRESULT hr = CoInitialize( NULL );
\r
5275 if ( !FAILED( hr ) ) coInitialized_ = true;
\r
5278 RtApiDs :: ~RtApiDs()
\r
5280 if ( coInitialized_ ) CoUninitialize(); // balanced call.
\r
5281 if ( stream_.state != STREAM_CLOSED ) closeStream();
\r
5284 // The DirectSound default output is always the first device.
\r
5285 unsigned int RtApiDs :: getDefaultOutputDevice( void )
\r
5290 // The DirectSound default input is always the first input device,
\r
5291 // which is the first capture device enumerated.
\r
5292 unsigned int RtApiDs :: getDefaultInputDevice( void )
\r
5297 unsigned int RtApiDs :: getDeviceCount( void )
\r
5299 // Set query flag for previously found devices to false, so that we
\r
5300 // can check for any devices that have disappeared.
\r
5301 for ( unsigned int i=0; i<dsDevices.size(); i++ )
\r
5302 dsDevices[i].found = false;
\r
5304 // Query DirectSound devices.
\r
5305 struct DsProbeData probeInfo;
\r
5306 probeInfo.isInput = false;
\r
5307 probeInfo.dsDevices = &dsDevices;
\r
5308 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
\r
5309 if ( FAILED( result ) ) {
\r
5310 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
\r
5311 errorText_ = errorStream_.str();
\r
5312 error( RtAudioError::WARNING );
\r
5315 // Query DirectSoundCapture devices.
\r
5316 probeInfo.isInput = true;
\r
5317 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
\r
5318 if ( FAILED( result ) ) {
\r
5319 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
\r
5320 errorText_ = errorStream_.str();
\r
5321 error( RtAudioError::WARNING );
\r
5324 // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
\r
5325 for ( unsigned int i=0; i<dsDevices.size(); ) {
\r
5326 if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
\r
5330 return static_cast<unsigned int>(dsDevices.size());
\r
5333 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
\r
5335 RtAudio::DeviceInfo info;
\r
5336 info.probed = false;
\r
5338 if ( dsDevices.size() == 0 ) {
\r
5339 // Force a query of all devices
\r
5341 if ( dsDevices.size() == 0 ) {
\r
5342 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
\r
5343 error( RtAudioError::INVALID_USE );
\r
5348 if ( device >= dsDevices.size() ) {
\r
5349 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
\r
5350 error( RtAudioError::INVALID_USE );
\r
5355 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
\r
5357 LPDIRECTSOUND output;
\r
5359 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
\r
5360 if ( FAILED( result ) ) {
\r
5361 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
\r
5362 errorText_ = errorStream_.str();
\r
5363 error( RtAudioError::WARNING );
\r
5367 outCaps.dwSize = sizeof( outCaps );
\r
5368 result = output->GetCaps( &outCaps );
\r
5369 if ( FAILED( result ) ) {
\r
5370 output->Release();
\r
5371 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
\r
5372 errorText_ = errorStream_.str();
\r
5373 error( RtAudioError::WARNING );
\r
5377 // Get output channel information.
\r
5378 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
\r
5380 // Get sample rate information.
\r
5381 info.sampleRates.clear();
\r
5382 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
\r
5383 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
\r
5384 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
\r
5385 info.sampleRates.push_back( SAMPLE_RATES[k] );
\r
5387 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
\r
5388 info.preferredSampleRate = SAMPLE_RATES[k];
\r
5392 // Get format information.
\r
5393 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5394 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5396 output->Release();
\r
5398 if ( getDefaultOutputDevice() == device )
\r
5399 info.isDefaultOutput = true;
\r
5401 if ( dsDevices[ device ].validId[1] == false ) {
\r
5402 info.name = dsDevices[ device ].name;
\r
5403 info.probed = true;
\r
5409 LPDIRECTSOUNDCAPTURE input;
\r
5410 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
\r
5411 if ( FAILED( result ) ) {
\r
5412 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
\r
5413 errorText_ = errorStream_.str();
\r
5414 error( RtAudioError::WARNING );
\r
5419 inCaps.dwSize = sizeof( inCaps );
\r
5420 result = input->GetCaps( &inCaps );
\r
5421 if ( FAILED( result ) ) {
\r
5423 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
\r
5424 errorText_ = errorStream_.str();
\r
5425 error( RtAudioError::WARNING );
\r
5429 // Get input channel information.
\r
5430 info.inputChannels = inCaps.dwChannels;
\r
5432 // Get sample rate and format information.
\r
5433 std::vector<unsigned int> rates;
\r
5434 if ( inCaps.dwChannels >= 2 ) {
\r
5435 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5436 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5437 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5438 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5439 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5440 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5441 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5442 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5444 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
\r
5445 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
\r
5446 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
\r
5447 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
\r
5448 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
\r
5450 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
\r
5451 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
\r
5452 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
\r
5453 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
\r
5454 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
\r
5457 else if ( inCaps.dwChannels == 1 ) {
\r
5458 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5459 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5460 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5461 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5462 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5463 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5464 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5465 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5467 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
\r
5468 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
\r
5469 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
\r
5470 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
\r
5471 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
\r
5473 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
\r
5474 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
\r
5475 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
\r
5476 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
\r
5477 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
\r
5480 else info.inputChannels = 0; // technically, this would be an error
\r
5484 if ( info.inputChannels == 0 ) return info;
\r
5486 // Copy the supported rates to the info structure but avoid duplication.
\r
5488 for ( unsigned int i=0; i<rates.size(); i++ ) {
\r
5490 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
\r
5491 if ( rates[i] == info.sampleRates[j] ) {
\r
5496 if ( found == false ) info.sampleRates.push_back( rates[i] );
\r
5498 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
\r
5500 // If device opens for both playback and capture, we determine the channels.
\r
5501 if ( info.outputChannels > 0 && info.inputChannels > 0 )
\r
5502 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
\r
5504 if ( device == 0 ) info.isDefaultInput = true;
\r
5506 // Copy name and return.
\r
5507 info.name = dsDevices[ device ].name;
\r
5508 info.probed = true;
\r
5512 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
5513 unsigned int firstChannel, unsigned int sampleRate,
\r
5514 RtAudioFormat format, unsigned int *bufferSize,
\r
5515 RtAudio::StreamOptions *options )
\r
5517 if ( channels + firstChannel > 2 ) {
\r
5518 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
\r
5522 size_t nDevices = dsDevices.size();
\r
5523 if ( nDevices == 0 ) {
\r
5524 // This should not happen because a check is made before this function is called.
\r
5525 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
\r
5529 if ( device >= nDevices ) {
\r
5530 // This should not happen because a check is made before this function is called.
\r
5531 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
\r
5535 if ( mode == OUTPUT ) {
\r
5536 if ( dsDevices[ device ].validId[0] == false ) {
\r
5537 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
\r
5538 errorText_ = errorStream_.str();
\r
5542 else { // mode == INPUT
\r
5543 if ( dsDevices[ device ].validId[1] == false ) {
\r
5544 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
\r
5545 errorText_ = errorStream_.str();
\r
5550 // According to a note in PortAudio, using GetDesktopWindow()
\r
5551 // instead of GetForegroundWindow() is supposed to avoid problems
\r
5552 // that occur when the application's window is not the foreground
\r
5553 // window. Also, if the application window closes before the
\r
5554 // DirectSound buffer, DirectSound can crash. In the past, I had
\r
5555 // problems when using GetDesktopWindow() but it seems fine now
\r
5556 // (January 2010). I'll leave it commented here.
\r
5557 // HWND hWnd = GetForegroundWindow();
\r
5558 HWND hWnd = GetDesktopWindow();
\r
5560 // Check the numberOfBuffers parameter and limit the lowest value to
\r
5561 // two. This is a judgement call and a value of two is probably too
\r
5562 // low for capture, but it should work for playback.
\r
5564 if ( options ) nBuffers = options->numberOfBuffers;
\r
5565 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
\r
5566 if ( nBuffers < 2 ) nBuffers = 3;
\r
5568 // Check the lower range of the user-specified buffer size and set
\r
5569 // (arbitrarily) to a lower bound of 32.
\r
5570 if ( *bufferSize < 32 ) *bufferSize = 32;
\r
5572 // Create the wave format structure. The data format setting will
\r
5573 // be determined later.
\r
5574 WAVEFORMATEX waveFormat;
\r
5575 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
\r
5576 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
\r
5577 waveFormat.nChannels = channels + firstChannel;
\r
5578 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
\r
5580 // Determine the device buffer size. By default, we'll use the value
\r
5581 // defined above (32K), but we will grow it to make allowances for
\r
5582 // very large software buffer sizes.
\r
5583 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
\r
5584 DWORD dsPointerLeadTime = 0;
\r
5586 void *ohandle = 0, *bhandle = 0;
\r
5588 if ( mode == OUTPUT ) {
\r
5590 LPDIRECTSOUND output;
\r
5591 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
\r
5592 if ( FAILED( result ) ) {
\r
5593 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
\r
5594 errorText_ = errorStream_.str();
\r
5599 outCaps.dwSize = sizeof( outCaps );
\r
5600 result = output->GetCaps( &outCaps );
\r
5601 if ( FAILED( result ) ) {
\r
5602 output->Release();
\r
5603 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
\r
5604 errorText_ = errorStream_.str();
\r
5608 // Check channel information.
\r
5609 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
\r
5610 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
\r
5611 errorText_ = errorStream_.str();
\r
5615 // Check format information. Use 16-bit format unless not
\r
5616 // supported or user requests 8-bit.
\r
5617 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
\r
5618 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
\r
5619 waveFormat.wBitsPerSample = 16;
\r
5620 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
5623 waveFormat.wBitsPerSample = 8;
\r
5624 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
\r
5626 stream_.userFormat = format;
\r
5628 // Update wave format structure and buffer information.
\r
5629 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
\r
5630 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
\r
5631 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
\r
5633 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
\r
5634 while ( dsPointerLeadTime * 2U > dsBufferSize )
\r
5635 dsBufferSize *= 2;
\r
5637 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
\r
5638 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
\r
5639 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
\r
5640 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
\r
5641 if ( FAILED( result ) ) {
\r
5642 output->Release();
\r
5643 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
\r
5644 errorText_ = errorStream_.str();
\r
5648 // Even though we will write to the secondary buffer, we need to
\r
5649 // access the primary buffer to set the correct output format
\r
5650 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
\r
5651 // buffer description.
\r
5652 DSBUFFERDESC bufferDescription;
\r
5653 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
\r
5654 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
\r
5655 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
\r
5657 // Obtain the primary buffer
\r
5658 LPDIRECTSOUNDBUFFER buffer;
\r
5659 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
\r
5660 if ( FAILED( result ) ) {
\r
5661 output->Release();
\r
5662 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
\r
5663 errorText_ = errorStream_.str();
\r
5667 // Set the primary DS buffer sound format.
\r
5668 result = buffer->SetFormat( &waveFormat );
\r
5669 if ( FAILED( result ) ) {
\r
5670 output->Release();
\r
5671 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
\r
5672 errorText_ = errorStream_.str();
\r
5676 // Setup the secondary DS buffer description.
\r
5677 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
\r
5678 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
\r
5679 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
\r
5680 DSBCAPS_GLOBALFOCUS |
\r
5681 DSBCAPS_GETCURRENTPOSITION2 |
\r
5682 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
\r
5683 bufferDescription.dwBufferBytes = dsBufferSize;
\r
5684 bufferDescription.lpwfxFormat = &waveFormat;
\r
5686 // Try to create the secondary DS buffer. If that doesn't work,
\r
5687 // try to use software mixing. Otherwise, there's a problem.
\r
5688 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
\r
5689 if ( FAILED( result ) ) {
\r
5690 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
\r
5691 DSBCAPS_GLOBALFOCUS |
\r
5692 DSBCAPS_GETCURRENTPOSITION2 |
\r
5693 DSBCAPS_LOCSOFTWARE ); // Force software mixing
\r
5694 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
\r
5695 if ( FAILED( result ) ) {
\r
5696 output->Release();
\r
5697 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
\r
5698 errorText_ = errorStream_.str();
\r
5703 // Get the buffer size ... might be different from what we specified.
\r
5705 dsbcaps.dwSize = sizeof( DSBCAPS );
\r
5706 result = buffer->GetCaps( &dsbcaps );
\r
5707 if ( FAILED( result ) ) {
\r
5708 output->Release();
\r
5709 buffer->Release();
\r
5710 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
\r
5711 errorText_ = errorStream_.str();
\r
5715 dsBufferSize = dsbcaps.dwBufferBytes;
\r
5717 // Lock the DS buffer
\r
5720 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
\r
5721 if ( FAILED( result ) ) {
\r
5722 output->Release();
\r
5723 buffer->Release();
\r
5724 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
\r
5725 errorText_ = errorStream_.str();
\r
5729 // Zero the DS buffer
\r
5730 ZeroMemory( audioPtr, dataLen );
\r
5732 // Unlock the DS buffer
\r
5733 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
\r
5734 if ( FAILED( result ) ) {
\r
5735 output->Release();
\r
5736 buffer->Release();
\r
5737 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
\r
5738 errorText_ = errorStream_.str();
\r
5742 ohandle = (void *) output;
\r
5743 bhandle = (void *) buffer;
\r
5746 if ( mode == INPUT ) {
\r
5748 LPDIRECTSOUNDCAPTURE input;
\r
5749 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
\r
5750 if ( FAILED( result ) ) {
\r
5751 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
\r
5752 errorText_ = errorStream_.str();
\r
5757 inCaps.dwSize = sizeof( inCaps );
\r
5758 result = input->GetCaps( &inCaps );
\r
5759 if ( FAILED( result ) ) {
\r
5761 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
\r
5762 errorText_ = errorStream_.str();
\r
5766 // Check channel information.
\r
5767 if ( inCaps.dwChannels < channels + firstChannel ) {
\r
5768 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
\r
5772 // Check format information. Use 16-bit format unless user
\r
5773 // requests 8-bit.
\r
5774 DWORD deviceFormats;
\r
5775 if ( channels + firstChannel == 2 ) {
\r
5776 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
\r
5777 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
\r
5778 waveFormat.wBitsPerSample = 8;
\r
5779 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
\r
5781 else { // assume 16-bit is supported
\r
5782 waveFormat.wBitsPerSample = 16;
\r
5783 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
5786 else { // channel == 1
\r
5787 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
\r
5788 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
\r
5789 waveFormat.wBitsPerSample = 8;
\r
5790 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
\r
5792 else { // assume 16-bit is supported
\r
5793 waveFormat.wBitsPerSample = 16;
\r
5794 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
5797 stream_.userFormat = format;
\r
5799 // Update wave format structure and buffer information.
\r
5800 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
\r
5801 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
\r
5802 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
\r
5804 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
\r
5805 while ( dsPointerLeadTime * 2U > dsBufferSize )
\r
5806 dsBufferSize *= 2;
\r
5808 // Setup the secondary DS buffer description.
\r
5809 DSCBUFFERDESC bufferDescription;
\r
5810 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
\r
5811 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
\r
5812 bufferDescription.dwFlags = 0;
\r
5813 bufferDescription.dwReserved = 0;
\r
5814 bufferDescription.dwBufferBytes = dsBufferSize;
\r
5815 bufferDescription.lpwfxFormat = &waveFormat;
\r
5817 // Create the capture buffer.
\r
5818 LPDIRECTSOUNDCAPTUREBUFFER buffer;
\r
5819 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
\r
5820 if ( FAILED( result ) ) {
\r
5822 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
\r
5823 errorText_ = errorStream_.str();
\r
5827 // Get the buffer size ... might be different from what we specified.
\r
5828 DSCBCAPS dscbcaps;
\r
5829 dscbcaps.dwSize = sizeof( DSCBCAPS );
\r
5830 result = buffer->GetCaps( &dscbcaps );
\r
5831 if ( FAILED( result ) ) {
\r
5833 buffer->Release();
\r
5834 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
\r
5835 errorText_ = errorStream_.str();
\r
5839 dsBufferSize = dscbcaps.dwBufferBytes;
\r
5841 // NOTE: We could have a problem here if this is a duplex stream
\r
5842 // and the play and capture hardware buffer sizes are different
\r
5843 // (I'm actually not sure if that is a problem or not).
\r
5844 // Currently, we are not verifying that.
\r
5846 // Lock the capture buffer
\r
5849 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
\r
5850 if ( FAILED( result ) ) {
\r
5852 buffer->Release();
\r
5853 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
\r
5854 errorText_ = errorStream_.str();
\r
5858 // Zero the buffer
\r
5859 ZeroMemory( audioPtr, dataLen );
\r
5861 // Unlock the buffer
\r
5862 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
\r
5863 if ( FAILED( result ) ) {
\r
5865 buffer->Release();
\r
5866 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
\r
5867 errorText_ = errorStream_.str();
\r
5871 ohandle = (void *) input;
\r
5872 bhandle = (void *) buffer;
\r
5875 // Set various stream parameters
\r
5876 DsHandle *handle = 0;
\r
5877 stream_.nDeviceChannels[mode] = channels + firstChannel;
\r
5878 stream_.nUserChannels[mode] = channels;
\r
5879 stream_.bufferSize = *bufferSize;
\r
5880 stream_.channelOffset[mode] = firstChannel;
\r
5881 stream_.deviceInterleaved[mode] = true;
\r
5882 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
\r
5883 else stream_.userInterleaved = true;
\r
5885 // Set flag for buffer conversion
\r
5886 stream_.doConvertBuffer[mode] = false;
\r
5887 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
\r
5888 stream_.doConvertBuffer[mode] = true;
\r
5889 if (stream_.userFormat != stream_.deviceFormat[mode])
\r
5890 stream_.doConvertBuffer[mode] = true;
\r
5891 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
\r
5892 stream_.nUserChannels[mode] > 1 )
\r
5893 stream_.doConvertBuffer[mode] = true;
\r
5895 // Allocate necessary internal buffers
\r
5896 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
5897 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
5898 if ( stream_.userBuffer[mode] == NULL ) {
\r
5899 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
\r
5903 if ( stream_.doConvertBuffer[mode] ) {
\r
5905 bool makeBuffer = true;
\r
5906 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
\r
5907 if ( mode == INPUT ) {
\r
5908 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
5909 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
5910 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
\r
5914 if ( makeBuffer ) {
\r
5915 bufferBytes *= *bufferSize;
\r
5916 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
5917 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
5918 if ( stream_.deviceBuffer == NULL ) {
\r
5919 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
\r
5925 // Allocate our DsHandle structures for the stream.
\r
5926 if ( stream_.apiHandle == 0 ) {
\r
5928 handle = new DsHandle;
\r
5930 catch ( std::bad_alloc& ) {
\r
5931 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
\r
5935 // Create a manual-reset event.
\r
5936 handle->condition = CreateEvent( NULL, // no security
\r
5937 TRUE, // manual-reset
\r
5938 FALSE, // non-signaled initially
\r
5939 NULL ); // unnamed
\r
5940 stream_.apiHandle = (void *) handle;
\r
5943 handle = (DsHandle *) stream_.apiHandle;
\r
5944 handle->id[mode] = ohandle;
\r
5945 handle->buffer[mode] = bhandle;
\r
5946 handle->dsBufferSize[mode] = dsBufferSize;
\r
5947 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
\r
5949 stream_.device[mode] = device;
\r
5950 stream_.state = STREAM_STOPPED;
\r
5951 if ( stream_.mode == OUTPUT && mode == INPUT )
\r
5952 // We had already set up an output stream.
\r
5953 stream_.mode = DUPLEX;
\r
5955 stream_.mode = mode;
\r
5956 stream_.nBuffers = nBuffers;
\r
5957 stream_.sampleRate = sampleRate;
\r
5959 // Setup the buffer conversion information structure.
\r
5960 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
\r
5962 // Setup the callback thread.
\r
5963 if ( stream_.callbackInfo.isRunning == false ) {
\r
5964 unsigned threadId;
\r
5965 stream_.callbackInfo.isRunning = true;
\r
5966 stream_.callbackInfo.object = (void *) this;
\r
5967 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
\r
5968 &stream_.callbackInfo, 0, &threadId );
\r
5969 if ( stream_.callbackInfo.thread == 0 ) {
\r
5970 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
\r
5974 // Boost DS thread priority
\r
5975 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
\r
5981 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
\r
5982 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
\r
5983 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
5984 if ( buffer ) buffer->Release();
\r
5985 object->Release();
\r
5987 if ( handle->buffer[1] ) {
\r
5988 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
\r
5989 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
\r
5990 if ( buffer ) buffer->Release();
\r
5991 object->Release();
\r
5993 CloseHandle( handle->condition );
\r
5995 stream_.apiHandle = 0;
\r
5998 for ( int i=0; i<2; i++ ) {
\r
5999 if ( stream_.userBuffer[i] ) {
\r
6000 free( stream_.userBuffer[i] );
\r
6001 stream_.userBuffer[i] = 0;
\r
6005 if ( stream_.deviceBuffer ) {
\r
6006 free( stream_.deviceBuffer );
\r
6007 stream_.deviceBuffer = 0;
\r
6010 stream_.state = STREAM_CLOSED;
\r
6014 void RtApiDs :: closeStream()
\r
6016 if ( stream_.state == STREAM_CLOSED ) {
\r
6017 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
\r
6018 error( RtAudioError::WARNING );
\r
6022 // Stop the callback thread.
\r
6023 stream_.callbackInfo.isRunning = false;
\r
6024 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
\r
6025 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
\r
6027 DsHandle *handle = (DsHandle *) stream_.apiHandle;
\r
6029 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
\r
6030 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
\r
6031 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
6034 buffer->Release();
\r
6036 object->Release();
\r
6038 if ( handle->buffer[1] ) {
\r
6039 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
\r
6040 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
\r
6043 buffer->Release();
\r
6045 object->Release();
\r
6047 CloseHandle( handle->condition );
\r
6049 stream_.apiHandle = 0;
\r
6052 for ( int i=0; i<2; i++ ) {
\r
6053 if ( stream_.userBuffer[i] ) {
\r
6054 free( stream_.userBuffer[i] );
\r
6055 stream_.userBuffer[i] = 0;
\r
6059 if ( stream_.deviceBuffer ) {
\r
6060 free( stream_.deviceBuffer );
\r
6061 stream_.deviceBuffer = 0;
\r
6064 stream_.mode = UNINITIALIZED;
\r
6065 stream_.state = STREAM_CLOSED;
\r
6068 void RtApiDs :: startStream()
\r
6071 if ( stream_.state == STREAM_RUNNING ) {
\r
6072 errorText_ = "RtApiDs::startStream(): the stream is already running!";
\r
6073 error( RtAudioError::WARNING );
\r
6077 DsHandle *handle = (DsHandle *) stream_.apiHandle;
\r
6079 // Increase scheduler frequency on lesser windows (a side-effect of
\r
6080 // increasing timer accuracy). On greater windows (Win2K or later),
\r
6081 // this is already in effect.
\r
6082 timeBeginPeriod( 1 );
\r
6084 buffersRolling = false;
\r
6085 duplexPrerollBytes = 0;
\r
6087 if ( stream_.mode == DUPLEX ) {
\r
6088 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
\r
6089 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
\r
6092 HRESULT result = 0;
\r
6093 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
6095 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
6096 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
\r
6097 if ( FAILED( result ) ) {
\r
6098 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
\r
6099 errorText_ = errorStream_.str();
\r
6104 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
6106 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
\r
6107 result = buffer->Start( DSCBSTART_LOOPING );
\r
6108 if ( FAILED( result ) ) {
\r
6109 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
\r
6110 errorText_ = errorStream_.str();
\r
6115 handle->drainCounter = 0;
\r
6116 handle->internalDrain = false;
\r
6117 ResetEvent( handle->condition );
\r
6118 stream_.state = STREAM_RUNNING;
\r
6121 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
\r
6124 void RtApiDs :: stopStream()
\r
6127 if ( stream_.state == STREAM_STOPPED ) {
\r
6128 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
\r
6129 error( RtAudioError::WARNING );
\r
6133 HRESULT result = 0;
\r
6136 DsHandle *handle = (DsHandle *) stream_.apiHandle;
\r
6137 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
6138 if ( handle->drainCounter == 0 ) {
\r
6139 handle->drainCounter = 2;
\r
6140 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
\r
6143 stream_.state = STREAM_STOPPED;
\r
6145 MUTEX_LOCK( &stream_.mutex );
\r
6147 // Stop the buffer and clear memory
\r
6148 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
6149 result = buffer->Stop();
\r
6150 if ( FAILED( result ) ) {
\r
6151 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
\r
6152 errorText_ = errorStream_.str();
\r
6156 // Lock the buffer and clear it so that if we start to play again,
\r
6157 // we won't have old data playing.
\r
6158 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
\r
6159 if ( FAILED( result ) ) {
\r
6160 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
\r
6161 errorText_ = errorStream_.str();
\r
6165 // Zero the DS buffer
\r
6166 ZeroMemory( audioPtr, dataLen );
\r
6168 // Unlock the DS buffer
\r
6169 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
\r
6170 if ( FAILED( result ) ) {
\r
6171 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
\r
6172 errorText_ = errorStream_.str();
\r
6176 // If we start playing again, we must begin at beginning of buffer.
\r
6177 handle->bufferPointer[0] = 0;
\r
6180 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
6181 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
\r
6185 stream_.state = STREAM_STOPPED;
\r
6187 if ( stream_.mode != DUPLEX )
\r
6188 MUTEX_LOCK( &stream_.mutex );
\r
6190 result = buffer->Stop();
\r
6191 if ( FAILED( result ) ) {
\r
6192 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
\r
6193 errorText_ = errorStream_.str();
\r
6197 // Lock the buffer and clear it so that if we start to play again,
\r
6198 // we won't have old data playing.
\r
6199 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
\r
6200 if ( FAILED( result ) ) {
\r
6201 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
\r
6202 errorText_ = errorStream_.str();
\r
6206 // Zero the DS buffer
\r
6207 ZeroMemory( audioPtr, dataLen );
\r
6209 // Unlock the DS buffer
\r
6210 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
\r
6211 if ( FAILED( result ) ) {
\r
6212 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
\r
6213 errorText_ = errorStream_.str();
\r
6217 // If we start recording again, we must begin at beginning of buffer.
\r
6218 handle->bufferPointer[1] = 0;
\r
6222 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
\r
6223 MUTEX_UNLOCK( &stream_.mutex );
\r
6225 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
\r
6228 void RtApiDs :: abortStream()
\r
6231 if ( stream_.state == STREAM_STOPPED ) {
\r
6232 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
\r
6233 error( RtAudioError::WARNING );
\r
6237 DsHandle *handle = (DsHandle *) stream_.apiHandle;
\r
6238 handle->drainCounter = 2;
\r
6243 void RtApiDs :: callbackEvent()
\r
6245 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
\r
6246 Sleep( 50 ); // sleep 50 milliseconds
\r
6250 if ( stream_.state == STREAM_CLOSED ) {
\r
6251 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
\r
6252 error( RtAudioError::WARNING );
\r
6256 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
\r
6257 DsHandle *handle = (DsHandle *) stream_.apiHandle;
\r
6259 // Check if we were draining the stream and signal is finished.
\r
6260 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
\r
6262 stream_.state = STREAM_STOPPING;
\r
6263 if ( handle->internalDrain == false )
\r
6264 SetEvent( handle->condition );
\r
6270 // Invoke user callback to get fresh output data UNLESS we are
\r
6271 // draining stream.
\r
6272 if ( handle->drainCounter == 0 ) {
\r
6273 RtAudioCallback callback = (RtAudioCallback) info->callback;
\r
6274 double streamTime = getStreamTime();
\r
6275 RtAudioStreamStatus status = 0;
\r
6276 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
\r
6277 status |= RTAUDIO_OUTPUT_UNDERFLOW;
\r
6278 handle->xrun[0] = false;
\r
6280 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
\r
6281 status |= RTAUDIO_INPUT_OVERFLOW;
\r
6282 handle->xrun[1] = false;
\r
6284 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
\r
6285 stream_.bufferSize, streamTime, status, info->userData );
\r
6286 if ( cbReturnValue == 2 ) {
\r
6287 stream_.state = STREAM_STOPPING;
\r
6288 handle->drainCounter = 2;
\r
6292 else if ( cbReturnValue == 1 ) {
\r
6293 handle->drainCounter = 1;
\r
6294 handle->internalDrain = true;
\r
6299 DWORD currentWritePointer, safeWritePointer;
\r
6300 DWORD currentReadPointer, safeReadPointer;
\r
6301 UINT nextWritePointer;
\r
6303 LPVOID buffer1 = NULL;
\r
6304 LPVOID buffer2 = NULL;
\r
6305 DWORD bufferSize1 = 0;
\r
6306 DWORD bufferSize2 = 0;
\r
6311 MUTEX_LOCK( &stream_.mutex );
\r
6312 if ( stream_.state == STREAM_STOPPED ) {
\r
6313 MUTEX_UNLOCK( &stream_.mutex );
\r
6317 if ( buffersRolling == false ) {
\r
6318 if ( stream_.mode == DUPLEX ) {
\r
6319 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
\r
6321 // It takes a while for the devices to get rolling. As a result,
\r
6322 // there's no guarantee that the capture and write device pointers
\r
6323 // will move in lockstep. Wait here for both devices to start
\r
6324 // rolling, and then set our buffer pointers accordingly.
\r
6325 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
\r
6326 // bytes later than the write buffer.
\r
6328 // Stub: a serious risk of having a pre-emptive scheduling round
\r
6329 // take place between the two GetCurrentPosition calls... but I'm
\r
6330 // really not sure how to solve the problem. Temporarily boost to
\r
6331 // Realtime priority, maybe; but I'm not sure what priority the
\r
6332 // DirectSound service threads run at. We *should* be roughly
\r
6333 // within a ms or so of correct.
\r
6335 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
6336 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
\r
6338 DWORD startSafeWritePointer, startSafeReadPointer;
\r
6340 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
\r
6341 if ( FAILED( result ) ) {
\r
6342 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
\r
6343 errorText_ = errorStream_.str();
\r
6344 MUTEX_UNLOCK( &stream_.mutex );
\r
6345 error( RtAudioError::SYSTEM_ERROR );
\r
6348 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
\r
6349 if ( FAILED( result ) ) {
\r
6350 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
\r
6351 errorText_ = errorStream_.str();
\r
6352 MUTEX_UNLOCK( &stream_.mutex );
\r
6353 error( RtAudioError::SYSTEM_ERROR );
\r
6357 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
\r
6358 if ( FAILED( result ) ) {
\r
6359 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
\r
6360 errorText_ = errorStream_.str();
\r
6361 MUTEX_UNLOCK( &stream_.mutex );
\r
6362 error( RtAudioError::SYSTEM_ERROR );
\r
6365 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
\r
6366 if ( FAILED( result ) ) {
\r
6367 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
\r
6368 errorText_ = errorStream_.str();
\r
6369 MUTEX_UNLOCK( &stream_.mutex );
\r
6370 error( RtAudioError::SYSTEM_ERROR );
\r
6373 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
\r
6377 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
\r
6379 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
\r
6380 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
\r
6381 handle->bufferPointer[1] = safeReadPointer;
\r
6383 else if ( stream_.mode == OUTPUT ) {
\r
6385 // Set the proper nextWritePosition after initial startup.
\r
6386 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
6387 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
\r
6388 if ( FAILED( result ) ) {
\r
6389 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
\r
6390 errorText_ = errorStream_.str();
\r
6391 MUTEX_UNLOCK( &stream_.mutex );
\r
6392 error( RtAudioError::SYSTEM_ERROR );
\r
6395 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
\r
6396 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
\r
6399 buffersRolling = true;
\r
6402 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
6404 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
6406 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
\r
6407 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
\r
6408 bufferBytes *= formatBytes( stream_.userFormat );
\r
6409 memset( stream_.userBuffer[0], 0, bufferBytes );
\r
6412 // Setup parameters and do buffer conversion if necessary.
\r
6413 if ( stream_.doConvertBuffer[0] ) {
\r
6414 buffer = stream_.deviceBuffer;
\r
6415 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
\r
6416 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
\r
6417 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
\r
6420 buffer = stream_.userBuffer[0];
\r
6421 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
\r
6422 bufferBytes *= formatBytes( stream_.userFormat );
\r
6425 // No byte swapping necessary in DirectSound implementation.
\r
6427 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
\r
6428 // unsigned. So, we need to convert our signed 8-bit data here to
\r
6430 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
\r
6431 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
\r
6433 DWORD dsBufferSize = handle->dsBufferSize[0];
\r
6434 nextWritePointer = handle->bufferPointer[0];
\r
6436 DWORD endWrite, leadPointer;
\r
6438 // Find out where the read and "safe write" pointers are.
\r
6439 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
\r
6440 if ( FAILED( result ) ) {
\r
6441 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
\r
6442 errorText_ = errorStream_.str();
\r
6443 MUTEX_UNLOCK( &stream_.mutex );
\r
6444 error( RtAudioError::SYSTEM_ERROR );
\r
6448 // We will copy our output buffer into the region between
\r
6449 // safeWritePointer and leadPointer. If leadPointer is not
\r
6450 // beyond the next endWrite position, wait until it is.
\r
6451 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
\r
6452 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
\r
6453 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
\r
6454 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
\r
6455 endWrite = nextWritePointer + bufferBytes;
\r
6457 // Check whether the entire write region is behind the play pointer.
\r
6458 if ( leadPointer >= endWrite ) break;
\r
6460 // If we are here, then we must wait until the leadPointer advances
\r
6461 // beyond the end of our next write region. We use the
\r
6462 // Sleep() function to suspend operation until that happens.
\r
6463 double millis = ( endWrite - leadPointer ) * 1000.0;
\r
6464 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
\r
6465 if ( millis < 1.0 ) millis = 1.0;
\r
6466 Sleep( (DWORD) millis );
\r
6469 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
\r
6470 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
\r
6471 // We've strayed into the forbidden zone ... resync the read pointer.
\r
6472 handle->xrun[0] = true;
\r
6473 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
\r
6474 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
\r
6475 handle->bufferPointer[0] = nextWritePointer;
\r
6476 endWrite = nextWritePointer + bufferBytes;
\r
6479 // Lock free space in the buffer
\r
6480 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
\r
6481 &bufferSize1, &buffer2, &bufferSize2, 0 );
\r
6482 if ( FAILED( result ) ) {
\r
6483 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
\r
6484 errorText_ = errorStream_.str();
\r
6485 MUTEX_UNLOCK( &stream_.mutex );
\r
6486 error( RtAudioError::SYSTEM_ERROR );
\r
6490 // Copy our buffer into the DS buffer
\r
6491 CopyMemory( buffer1, buffer, bufferSize1 );
\r
6492 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
\r
6494 // Update our buffer offset and unlock sound buffer
\r
6495 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
\r
6496 if ( FAILED( result ) ) {
\r
6497 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
\r
6498 errorText_ = errorStream_.str();
\r
6499 MUTEX_UNLOCK( &stream_.mutex );
\r
6500 error( RtAudioError::SYSTEM_ERROR );
\r
6503 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
\r
6504 handle->bufferPointer[0] = nextWritePointer;
\r
6507 // Don't bother draining input
\r
6508 if ( handle->drainCounter ) {
\r
6509 handle->drainCounter++;
\r
6513 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
6515 // Setup parameters.
\r
6516 if ( stream_.doConvertBuffer[1] ) {
\r
6517 buffer = stream_.deviceBuffer;
\r
6518 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
\r
6519 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
\r
6522 buffer = stream_.userBuffer[1];
\r
6523 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
\r
6524 bufferBytes *= formatBytes( stream_.userFormat );
\r
6527 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
\r
6528 long nextReadPointer = handle->bufferPointer[1];
\r
6529 DWORD dsBufferSize = handle->dsBufferSize[1];
\r
6531 // Find out where the write and "safe read" pointers are.
\r
6532 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
\r
6533 if ( FAILED( result ) ) {
\r
6534 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
\r
6535 errorText_ = errorStream_.str();
\r
6536 MUTEX_UNLOCK( &stream_.mutex );
\r
6537 error( RtAudioError::SYSTEM_ERROR );
\r
6541 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
\r
6542 DWORD endRead = nextReadPointer + bufferBytes;
\r
6544 // Handling depends on whether we are INPUT or DUPLEX.
\r
6545 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
\r
6546 // then a wait here will drag the write pointers into the forbidden zone.
\r
6548 // In DUPLEX mode, rather than wait, we will back off the read pointer until
\r
6549 // it's in a safe position. This causes dropouts, but it seems to be the only
\r
6550 // practical way to sync up the read and write pointers reliably, given the
\r
6551 // the very complex relationship between phase and increment of the read and write
\r
6554 // In order to minimize audible dropouts in DUPLEX mode, we will
\r
6555 // provide a pre-roll period of 0.5 seconds in which we return
\r
6556 // zeros from the read buffer while the pointers sync up.
\r
6558 if ( stream_.mode == DUPLEX ) {
\r
6559 if ( safeReadPointer < endRead ) {
\r
6560 if ( duplexPrerollBytes <= 0 ) {
\r
6561 // Pre-roll time over. Be more agressive.
\r
6562 int adjustment = endRead-safeReadPointer;
\r
6564 handle->xrun[1] = true;
\r
6566 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
\r
6567 // and perform fine adjustments later.
\r
6568 // - small adjustments: back off by twice as much.
\r
6569 if ( adjustment >= 2*bufferBytes )
\r
6570 nextReadPointer = safeReadPointer-2*bufferBytes;
\r
6572 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
\r
6574 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
\r
6578 // In pre=roll time. Just do it.
\r
6579 nextReadPointer = safeReadPointer - bufferBytes;
\r
6580 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
\r
6582 endRead = nextReadPointer + bufferBytes;
\r
6585 else { // mode == INPUT
\r
6586 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
\r
6587 // See comments for playback.
\r
6588 double millis = (endRead - safeReadPointer) * 1000.0;
\r
6589 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
\r
6590 if ( millis < 1.0 ) millis = 1.0;
\r
6591 Sleep( (DWORD) millis );
\r
6593 // Wake up and find out where we are now.
\r
6594 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
\r
6595 if ( FAILED( result ) ) {
\r
6596 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
\r
6597 errorText_ = errorStream_.str();
\r
6598 MUTEX_UNLOCK( &stream_.mutex );
\r
6599 error( RtAudioError::SYSTEM_ERROR );
\r
6603 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
\r
6607 // Lock free space in the buffer
\r
6608 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
\r
6609 &bufferSize1, &buffer2, &bufferSize2, 0 );
\r
6610 if ( FAILED( result ) ) {
\r
6611 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
\r
6612 errorText_ = errorStream_.str();
\r
6613 MUTEX_UNLOCK( &stream_.mutex );
\r
6614 error( RtAudioError::SYSTEM_ERROR );
\r
6618 if ( duplexPrerollBytes <= 0 ) {
\r
6619 // Copy our buffer into the DS buffer
\r
6620 CopyMemory( buffer, buffer1, bufferSize1 );
\r
6621 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
\r
6624 memset( buffer, 0, bufferSize1 );
\r
6625 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
\r
6626 duplexPrerollBytes -= bufferSize1 + bufferSize2;
\r
6629 // Update our buffer offset and unlock sound buffer
\r
6630 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
\r
6631 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
\r
6632 if ( FAILED( result ) ) {
\r
6633 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
\r
6634 errorText_ = errorStream_.str();
\r
6635 MUTEX_UNLOCK( &stream_.mutex );
\r
6636 error( RtAudioError::SYSTEM_ERROR );
\r
6639 handle->bufferPointer[1] = nextReadPointer;
\r
6641 // No byte swapping necessary in DirectSound implementation.
\r
6643 // If necessary, convert 8-bit data from unsigned to signed.
\r
6644 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
\r
6645 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
\r
6647 // Do buffer conversion if necessary.
\r
6648 if ( stream_.doConvertBuffer[1] )
\r
6649 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
\r
6653 MUTEX_UNLOCK( &stream_.mutex );
\r
6654 RtApi::tickStreamTime();
\r
6657 // Definitions for utility functions and callbacks
\r
6658 // specific to the DirectSound implementation.
\r
6660 static unsigned __stdcall callbackHandler( void *ptr )
\r
6662 CallbackInfo *info = (CallbackInfo *) ptr;
\r
6663 RtApiDs *object = (RtApiDs *) info->object;
\r
6664 bool* isRunning = &info->isRunning;
\r
6666 while ( *isRunning == true ) {
\r
6667 object->callbackEvent();
\r
6670 _endthreadex( 0 );
\r
6674 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
\r
6675 LPCTSTR description,
\r
6676 LPCTSTR /*module*/,
\r
6677 LPVOID lpContext )
\r
6679 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
\r
6680 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
\r
6683 bool validDevice = false;
\r
6684 if ( probeInfo.isInput == true ) {
\r
6686 LPDIRECTSOUNDCAPTURE object;
\r
6688 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
\r
6689 if ( hr != DS_OK ) return TRUE;
\r
6691 caps.dwSize = sizeof(caps);
\r
6692 hr = object->GetCaps( &caps );
\r
6693 if ( hr == DS_OK ) {
\r
6694 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
\r
6695 validDevice = true;
\r
6697 object->Release();
\r
6701 LPDIRECTSOUND object;
\r
6702 hr = DirectSoundCreate( lpguid, &object, NULL );
\r
6703 if ( hr != DS_OK ) return TRUE;
\r
6705 caps.dwSize = sizeof(caps);
\r
6706 hr = object->GetCaps( &caps );
\r
6707 if ( hr == DS_OK ) {
\r
6708 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
\r
6709 validDevice = true;
\r
6711 object->Release();
\r
6714 // If good device, then save its name and guid.
\r
6715 std::string name = convertCharPointerToStdString( description );
\r
6716 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
\r
6717 if ( lpguid == NULL )
\r
6718 name = "Default Device";
\r
6719 if ( validDevice ) {
\r
6720 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
\r
6721 if ( dsDevices[i].name == name ) {
\r
6722 dsDevices[i].found = true;
\r
6723 if ( probeInfo.isInput ) {
\r
6724 dsDevices[i].id[1] = lpguid;
\r
6725 dsDevices[i].validId[1] = true;
\r
6728 dsDevices[i].id[0] = lpguid;
\r
6729 dsDevices[i].validId[0] = true;
\r
6736 device.name = name;
\r
6737 device.found = true;
\r
6738 if ( probeInfo.isInput ) {
\r
6739 device.id[1] = lpguid;
\r
6740 device.validId[1] = true;
\r
6743 device.id[0] = lpguid;
\r
6744 device.validId[0] = true;
\r
6746 dsDevices.push_back( device );
\r
6752 static const char* getErrorString( int code )
\r
6756 case DSERR_ALLOCATED:
\r
6757 return "Already allocated";
\r
6759 case DSERR_CONTROLUNAVAIL:
\r
6760 return "Control unavailable";
\r
6762 case DSERR_INVALIDPARAM:
\r
6763 return "Invalid parameter";
\r
6765 case DSERR_INVALIDCALL:
\r
6766 return "Invalid call";
\r
6768 case DSERR_GENERIC:
\r
6769 return "Generic error";
\r
6771 case DSERR_PRIOLEVELNEEDED:
\r
6772 return "Priority level needed";
\r
6774 case DSERR_OUTOFMEMORY:
\r
6775 return "Out of memory";
\r
6777 case DSERR_BADFORMAT:
\r
6778 return "The sample rate or the channel format is not supported";
\r
6780 case DSERR_UNSUPPORTED:
\r
6781 return "Not supported";
\r
6783 case DSERR_NODRIVER:
\r
6784 return "No driver";
\r
6786 case DSERR_ALREADYINITIALIZED:
\r
6787 return "Already initialized";
\r
6789 case DSERR_NOAGGREGATION:
\r
6790 return "No aggregation";
\r
6792 case DSERR_BUFFERLOST:
\r
6793 return "Buffer lost";
\r
6795 case DSERR_OTHERAPPHASPRIO:
\r
6796 return "Another application already has priority";
\r
6798 case DSERR_UNINITIALIZED:
\r
6799 return "Uninitialized";
\r
6802 return "DirectSound unknown error";
\r
6805 //******************** End of __WINDOWS_DS__ *********************//
\r
6809 #if defined(__LINUX_ALSA__)
\r
6811 #include <alsa/asoundlib.h>
\r
6812 #include <unistd.h>
\r
6814 // A structure to hold various information related to the ALSA API
\r
6815 // implementation.
\r
6816 struct AlsaHandle {
\r
6817 snd_pcm_t *handles[2];
\r
6818 bool synchronized;
\r
6820 pthread_cond_t runnable_cv;
\r
6824 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
\r
6827 static void *alsaCallbackHandler( void * ptr );
\r
6829 RtApiAlsa :: RtApiAlsa()
\r
6831 // Nothing to do here.
\r
6834 RtApiAlsa :: ~RtApiAlsa()
\r
6836 if ( stream_.state != STREAM_CLOSED ) closeStream();
\r
6839 unsigned int RtApiAlsa :: getDeviceCount( void )
\r
6841 unsigned nDevices = 0;
\r
6842 int result, subdevice, card;
\r
6844 snd_ctl_t *handle;
\r
6846 // Count cards and devices
\r
6848 snd_card_next( &card );
\r
6849 while ( card >= 0 ) {
\r
6850 sprintf( name, "hw:%d", card );
\r
6851 result = snd_ctl_open( &handle, name, 0 );
\r
6852 if ( result < 0 ) {
\r
6853 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
\r
6854 errorText_ = errorStream_.str();
\r
6855 error( RtAudioError::WARNING );
\r
6860 result = snd_ctl_pcm_next_device( handle, &subdevice );
\r
6861 if ( result < 0 ) {
\r
6862 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
\r
6863 errorText_ = errorStream_.str();
\r
6864 error( RtAudioError::WARNING );
\r
6867 if ( subdevice < 0 )
\r
6872 snd_ctl_close( handle );
\r
6873 snd_card_next( &card );
\r
6876 result = snd_ctl_open( &handle, "default", 0 );
\r
6877 if (result == 0) {
\r
6879 snd_ctl_close( handle );
\r
6885 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
\r
6887 RtAudio::DeviceInfo info;
\r
6888 info.probed = false;
\r
6890 unsigned nDevices = 0;
\r
6891 int result, subdevice, card;
\r
6893 snd_ctl_t *chandle;
\r
6895 // Count cards and devices
\r
6898 snd_card_next( &card );
\r
6899 while ( card >= 0 ) {
\r
6900 sprintf( name, "hw:%d", card );
\r
6901 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
\r
6902 if ( result < 0 ) {
\r
6903 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
\r
6904 errorText_ = errorStream_.str();
\r
6905 error( RtAudioError::WARNING );
\r
6910 result = snd_ctl_pcm_next_device( chandle, &subdevice );
\r
6911 if ( result < 0 ) {
\r
6912 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
\r
6913 errorText_ = errorStream_.str();
\r
6914 error( RtAudioError::WARNING );
\r
6917 if ( subdevice < 0 ) break;
\r
6918 if ( nDevices == device ) {
\r
6919 sprintf( name, "hw:%d,%d", card, subdevice );
\r
6925 snd_ctl_close( chandle );
\r
6926 snd_card_next( &card );
\r
6929 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
\r
6930 if ( result == 0 ) {
\r
6931 if ( nDevices == device ) {
\r
6932 strcpy( name, "default" );
\r
6938 if ( nDevices == 0 ) {
\r
6939 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
\r
6940 error( RtAudioError::INVALID_USE );
\r
6944 if ( device >= nDevices ) {
\r
6945 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
\r
6946 error( RtAudioError::INVALID_USE );
\r
6952 // If a stream is already open, we cannot probe the stream devices.
\r
6953 // Thus, use the saved results.
\r
6954 if ( stream_.state != STREAM_CLOSED &&
\r
6955 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
\r
6956 snd_ctl_close( chandle );
\r
6957 if ( device >= devices_.size() ) {
\r
6958 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
\r
6959 error( RtAudioError::WARNING );
\r
6962 return devices_[ device ];
\r
6965 int openMode = SND_PCM_ASYNC;
\r
6966 snd_pcm_stream_t stream;
\r
6967 snd_pcm_info_t *pcminfo;
\r
6968 snd_pcm_info_alloca( &pcminfo );
\r
6969 snd_pcm_t *phandle;
\r
6970 snd_pcm_hw_params_t *params;
\r
6971 snd_pcm_hw_params_alloca( ¶ms );
\r
6973 // First try for playback unless default device (which has subdev -1)
\r
6974 stream = SND_PCM_STREAM_PLAYBACK;
\r
6975 snd_pcm_info_set_stream( pcminfo, stream );
\r
6976 if ( subdevice != -1 ) {
\r
6977 snd_pcm_info_set_device( pcminfo, subdevice );
\r
6978 snd_pcm_info_set_subdevice( pcminfo, 0 );
\r
6980 result = snd_ctl_pcm_info( chandle, pcminfo );
\r
6981 if ( result < 0 ) {
\r
6982 // Device probably doesn't support playback.
\r
6983 goto captureProbe;
\r
6987 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
\r
6988 if ( result < 0 ) {
\r
6989 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
\r
6990 errorText_ = errorStream_.str();
\r
6991 error( RtAudioError::WARNING );
\r
6992 goto captureProbe;
\r
6995 // The device is open ... fill the parameter structure.
\r
6996 result = snd_pcm_hw_params_any( phandle, params );
\r
6997 if ( result < 0 ) {
\r
6998 snd_pcm_close( phandle );
\r
6999 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7000 errorText_ = errorStream_.str();
\r
7001 error( RtAudioError::WARNING );
\r
7002 goto captureProbe;
\r
7005 // Get output channel information.
\r
7006 unsigned int value;
\r
7007 result = snd_pcm_hw_params_get_channels_max( params, &value );
\r
7008 if ( result < 0 ) {
\r
7009 snd_pcm_close( phandle );
\r
7010 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
\r
7011 errorText_ = errorStream_.str();
\r
7012 error( RtAudioError::WARNING );
\r
7013 goto captureProbe;
\r
7015 info.outputChannels = value;
\r
7016 snd_pcm_close( phandle );
\r
7019 stream = SND_PCM_STREAM_CAPTURE;
\r
7020 snd_pcm_info_set_stream( pcminfo, stream );
\r
7022 // Now try for capture unless default device (with subdev = -1)
\r
7023 if ( subdevice != -1 ) {
\r
7024 result = snd_ctl_pcm_info( chandle, pcminfo );
\r
7025 snd_ctl_close( chandle );
\r
7026 if ( result < 0 ) {
\r
7027 // Device probably doesn't support capture.
\r
7028 if ( info.outputChannels == 0 ) return info;
\r
7029 goto probeParameters;
\r
7033 snd_ctl_close( chandle );
\r
7035 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
\r
7036 if ( result < 0 ) {
\r
7037 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7038 errorText_ = errorStream_.str();
\r
7039 error( RtAudioError::WARNING );
\r
7040 if ( info.outputChannels == 0 ) return info;
\r
7041 goto probeParameters;
\r
7044 // The device is open ... fill the parameter structure.
\r
7045 result = snd_pcm_hw_params_any( phandle, params );
\r
7046 if ( result < 0 ) {
\r
7047 snd_pcm_close( phandle );
\r
7048 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7049 errorText_ = errorStream_.str();
\r
7050 error( RtAudioError::WARNING );
\r
7051 if ( info.outputChannels == 0 ) return info;
\r
7052 goto probeParameters;
\r
7055 result = snd_pcm_hw_params_get_channels_max( params, &value );
\r
7056 if ( result < 0 ) {
\r
7057 snd_pcm_close( phandle );
\r
7058 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
\r
7059 errorText_ = errorStream_.str();
\r
7060 error( RtAudioError::WARNING );
\r
7061 if ( info.outputChannels == 0 ) return info;
\r
7062 goto probeParameters;
\r
7064 info.inputChannels = value;
\r
7065 snd_pcm_close( phandle );
\r
7067 // If device opens for both playback and capture, we determine the channels.
\r
7068 if ( info.outputChannels > 0 && info.inputChannels > 0 )
\r
7069 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
\r
7071 // ALSA doesn't provide default devices so we'll use the first available one.
\r
7072 if ( device == 0 && info.outputChannels > 0 )
\r
7073 info.isDefaultOutput = true;
\r
7074 if ( device == 0 && info.inputChannels > 0 )
\r
7075 info.isDefaultInput = true;
\r
7078 // At this point, we just need to figure out the supported data
\r
7079 // formats and sample rates. We'll proceed by opening the device in
\r
7080 // the direction with the maximum number of channels, or playback if
\r
7081 // they are equal. This might limit our sample rate options, but so
\r
7084 if ( info.outputChannels >= info.inputChannels )
\r
7085 stream = SND_PCM_STREAM_PLAYBACK;
\r
7087 stream = SND_PCM_STREAM_CAPTURE;
\r
7088 snd_pcm_info_set_stream( pcminfo, stream );
\r
7090 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
\r
7091 if ( result < 0 ) {
\r
7092 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7093 errorText_ = errorStream_.str();
\r
7094 error( RtAudioError::WARNING );
\r
7098 // The device is open ... fill the parameter structure.
\r
7099 result = snd_pcm_hw_params_any( phandle, params );
\r
7100 if ( result < 0 ) {
\r
7101 snd_pcm_close( phandle );
\r
7102 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7103 errorText_ = errorStream_.str();
\r
7104 error( RtAudioError::WARNING );
\r
7108 // Test our discrete set of sample rate values.
\r
7109 info.sampleRates.clear();
\r
7110 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
\r
7111 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
\r
7112 info.sampleRates.push_back( SAMPLE_RATES[i] );
\r
7114 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
\r
7115 info.preferredSampleRate = SAMPLE_RATES[i];
\r
7118 if ( info.sampleRates.size() == 0 ) {
\r
7119 snd_pcm_close( phandle );
\r
7120 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
\r
7121 errorText_ = errorStream_.str();
\r
7122 error( RtAudioError::WARNING );
\r
7126 // Probe the supported data formats ... we don't care about endian-ness just yet
\r
7127 snd_pcm_format_t format;
\r
7128 info.nativeFormats = 0;
\r
7129 format = SND_PCM_FORMAT_S8;
\r
7130 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
\r
7131 info.nativeFormats |= RTAUDIO_SINT8;
\r
7132 format = SND_PCM_FORMAT_S16;
\r
7133 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
\r
7134 info.nativeFormats |= RTAUDIO_SINT16;
\r
7135 format = SND_PCM_FORMAT_S24;
\r
7136 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
\r
7137 info.nativeFormats |= RTAUDIO_SINT24;
\r
7138 format = SND_PCM_FORMAT_S32;
\r
7139 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
\r
7140 info.nativeFormats |= RTAUDIO_SINT32;
\r
7141 format = SND_PCM_FORMAT_FLOAT;
\r
7142 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
\r
7143 info.nativeFormats |= RTAUDIO_FLOAT32;
\r
7144 format = SND_PCM_FORMAT_FLOAT64;
\r
7145 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
\r
7146 info.nativeFormats |= RTAUDIO_FLOAT64;
\r
7148 // Check that we have at least one supported format
\r
7149 if ( info.nativeFormats == 0 ) {
\r
7150 snd_pcm_close( phandle );
\r
7151 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
\r
7152 errorText_ = errorStream_.str();
\r
7153 error( RtAudioError::WARNING );
\r
7157 // Get the device name
\r
7159 result = snd_card_get_name( card, &cardname );
\r
7160 if ( result >= 0 ) {
\r
7161 sprintf( name, "hw:%s,%d", cardname, subdevice );
\r
7166 // That's all ... close the device and return
\r
7167 snd_pcm_close( phandle );
\r
7168 info.probed = true;
\r
7172 void RtApiAlsa :: saveDeviceInfo( void )
\r
7176 unsigned int nDevices = getDeviceCount();
\r
7177 devices_.resize( nDevices );
\r
7178 for ( unsigned int i=0; i<nDevices; i++ )
\r
7179 devices_[i] = getDeviceInfo( i );
\r
7182 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
7183 unsigned int firstChannel, unsigned int sampleRate,
\r
7184 RtAudioFormat format, unsigned int *bufferSize,
\r
7185 RtAudio::StreamOptions *options )
\r
7188 #if defined(__RTAUDIO_DEBUG__)
\r
7189 snd_output_t *out;
\r
7190 snd_output_stdio_attach(&out, stderr, 0);
\r
7193 // I'm not using the "plug" interface ... too much inconsistent behavior.
\r
7195 unsigned nDevices = 0;
\r
7196 int result, subdevice, card;
\r
7198 snd_ctl_t *chandle;
\r
7200 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
\r
7201 snprintf(name, sizeof(name), "%s", "default");
\r
7203 // Count cards and devices
\r
7205 snd_card_next( &card );
\r
7206 while ( card >= 0 ) {
\r
7207 sprintf( name, "hw:%d", card );
\r
7208 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
\r
7209 if ( result < 0 ) {
\r
7210 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
\r
7211 errorText_ = errorStream_.str();
\r
7216 result = snd_ctl_pcm_next_device( chandle, &subdevice );
\r
7217 if ( result < 0 ) break;
\r
7218 if ( subdevice < 0 ) break;
\r
7219 if ( nDevices == device ) {
\r
7220 sprintf( name, "hw:%d,%d", card, subdevice );
\r
7221 snd_ctl_close( chandle );
\r
7226 snd_ctl_close( chandle );
\r
7227 snd_card_next( &card );
\r
7230 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
\r
7231 if ( result == 0 ) {
\r
7232 if ( nDevices == device ) {
\r
7233 strcpy( name, "default" );
\r
7239 if ( nDevices == 0 ) {
\r
7240 // This should not happen because a check is made before this function is called.
\r
7241 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
\r
7245 if ( device >= nDevices ) {
\r
7246 // This should not happen because a check is made before this function is called.
\r
7247 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
\r
7254 // The getDeviceInfo() function will not work for a device that is
\r
7255 // already open. Thus, we'll probe the system before opening a
\r
7256 // stream and save the results for use by getDeviceInfo().
\r
7257 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
\r
7258 this->saveDeviceInfo();
\r
7260 snd_pcm_stream_t stream;
\r
7261 if ( mode == OUTPUT )
\r
7262 stream = SND_PCM_STREAM_PLAYBACK;
\r
7264 stream = SND_PCM_STREAM_CAPTURE;
\r
7266 snd_pcm_t *phandle;
\r
7267 int openMode = SND_PCM_ASYNC;
\r
7268 result = snd_pcm_open( &phandle, name, stream, openMode );
\r
7269 if ( result < 0 ) {
\r
7270 if ( mode == OUTPUT )
\r
7271 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
\r
7273 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
\r
7274 errorText_ = errorStream_.str();
\r
7278 // Fill the parameter structure.
\r
7279 snd_pcm_hw_params_t *hw_params;
\r
7280 snd_pcm_hw_params_alloca( &hw_params );
\r
7281 result = snd_pcm_hw_params_any( phandle, hw_params );
\r
7282 if ( result < 0 ) {
\r
7283 snd_pcm_close( phandle );
\r
7284 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
\r
7285 errorText_ = errorStream_.str();
\r
7289 #if defined(__RTAUDIO_DEBUG__)
\r
7290 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
\r
7291 snd_pcm_hw_params_dump( hw_params, out );
\r
7294 // Set access ... check user preference.
\r
7295 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
\r
7296 stream_.userInterleaved = false;
\r
7297 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
\r
7298 if ( result < 0 ) {
\r
7299 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
\r
7300 stream_.deviceInterleaved[mode] = true;
\r
7303 stream_.deviceInterleaved[mode] = false;
\r
7306 stream_.userInterleaved = true;
\r
7307 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
\r
7308 if ( result < 0 ) {
\r
7309 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
\r
7310 stream_.deviceInterleaved[mode] = false;
\r
7313 stream_.deviceInterleaved[mode] = true;
\r
7316 if ( result < 0 ) {
\r
7317 snd_pcm_close( phandle );
\r
7318 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
\r
7319 errorText_ = errorStream_.str();
\r
7323 // Determine how to set the device format.
\r
7324 stream_.userFormat = format;
\r
7325 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
\r
7327 if ( format == RTAUDIO_SINT8 )
\r
7328 deviceFormat = SND_PCM_FORMAT_S8;
\r
7329 else if ( format == RTAUDIO_SINT16 )
\r
7330 deviceFormat = SND_PCM_FORMAT_S16;
\r
7331 else if ( format == RTAUDIO_SINT24 )
\r
7332 deviceFormat = SND_PCM_FORMAT_S24;
\r
7333 else if ( format == RTAUDIO_SINT32 )
\r
7334 deviceFormat = SND_PCM_FORMAT_S32;
\r
7335 else if ( format == RTAUDIO_FLOAT32 )
\r
7336 deviceFormat = SND_PCM_FORMAT_FLOAT;
\r
7337 else if ( format == RTAUDIO_FLOAT64 )
\r
7338 deviceFormat = SND_PCM_FORMAT_FLOAT64;
\r
7340 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
\r
7341 stream_.deviceFormat[mode] = format;
\r
7345 // The user requested format is not natively supported by the device.
\r
7346 deviceFormat = SND_PCM_FORMAT_FLOAT64;
\r
7347 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
\r
7348 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
\r
7352 deviceFormat = SND_PCM_FORMAT_FLOAT;
\r
7353 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
\r
7354 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
\r
7358 deviceFormat = SND_PCM_FORMAT_S32;
\r
7359 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
\r
7360 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
\r
7364 deviceFormat = SND_PCM_FORMAT_S24;
\r
7365 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
\r
7366 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
\r
7370 deviceFormat = SND_PCM_FORMAT_S16;
\r
7371 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
\r
7372 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
7376 deviceFormat = SND_PCM_FORMAT_S8;
\r
7377 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
\r
7378 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
\r
7382 // If we get here, no supported format was found.
\r
7383 snd_pcm_close( phandle );
\r
7384 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
\r
7385 errorText_ = errorStream_.str();
\r
7389 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
\r
7390 if ( result < 0 ) {
\r
7391 snd_pcm_close( phandle );
\r
7392 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
\r
7393 errorText_ = errorStream_.str();
\r
7397 // Determine whether byte-swaping is necessary.
\r
7398 stream_.doByteSwap[mode] = false;
\r
7399 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
\r
7400 result = snd_pcm_format_cpu_endian( deviceFormat );
\r
7401 if ( result == 0 )
\r
7402 stream_.doByteSwap[mode] = true;
\r
7403 else if (result < 0) {
\r
7404 snd_pcm_close( phandle );
\r
7405 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
\r
7406 errorText_ = errorStream_.str();
\r
7411 // Set the sample rate.
\r
7412 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
\r
7413 if ( result < 0 ) {
\r
7414 snd_pcm_close( phandle );
\r
7415 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
\r
7416 errorText_ = errorStream_.str();
\r
7420 // Determine the number of channels for this device. We support a possible
\r
7421 // minimum device channel number > than the value requested by the user.
\r
7422 stream_.nUserChannels[mode] = channels;
\r
7423 unsigned int value;
\r
7424 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
\r
7425 unsigned int deviceChannels = value;
\r
7426 if ( result < 0 || deviceChannels < channels + firstChannel ) {
\r
7427 snd_pcm_close( phandle );
\r
7428 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
\r
7429 errorText_ = errorStream_.str();
\r
7433 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
\r
7434 if ( result < 0 ) {
\r
7435 snd_pcm_close( phandle );
\r
7436 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7437 errorText_ = errorStream_.str();
\r
7440 deviceChannels = value;
\r
7441 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
\r
7442 stream_.nDeviceChannels[mode] = deviceChannels;
\r
7444 // Set the device channels.
\r
7445 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
\r
7446 if ( result < 0 ) {
\r
7447 snd_pcm_close( phandle );
\r
7448 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7449 errorText_ = errorStream_.str();
\r
7453 // Set the buffer (or period) size.
\r
7455 snd_pcm_uframes_t periodSize = *bufferSize;
\r
7456 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
\r
7457 if ( result < 0 ) {
\r
7458 snd_pcm_close( phandle );
\r
7459 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7460 errorText_ = errorStream_.str();
\r
7463 *bufferSize = periodSize;
\r
7465 // Set the buffer number, which in ALSA is referred to as the "period".
\r
7466 unsigned int periods = 0;
\r
7467 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
\r
7468 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
\r
7469 if ( periods < 2 ) periods = 4; // a fairly safe default value
\r
7470 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
\r
7471 if ( result < 0 ) {
\r
7472 snd_pcm_close( phandle );
\r
7473 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7474 errorText_ = errorStream_.str();
\r
7478 // If attempting to setup a duplex stream, the bufferSize parameter
\r
7479 // MUST be the same in both directions!
\r
7480 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
\r
7481 snd_pcm_close( phandle );
\r
7482 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
\r
7483 errorText_ = errorStream_.str();
\r
7487 stream_.bufferSize = *bufferSize;
\r
7489 // Install the hardware configuration
\r
7490 result = snd_pcm_hw_params( phandle, hw_params );
\r
7491 if ( result < 0 ) {
\r
7492 snd_pcm_close( phandle );
\r
7493 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
\r
7494 errorText_ = errorStream_.str();
\r
7498 #if defined(__RTAUDIO_DEBUG__)
\r
7499 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
\r
7500 snd_pcm_hw_params_dump( hw_params, out );
\r
7503 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
\r
7504 snd_pcm_sw_params_t *sw_params = NULL;
\r
7505 snd_pcm_sw_params_alloca( &sw_params );
\r
7506 snd_pcm_sw_params_current( phandle, sw_params );
\r
7507 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
\r
7508 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
\r
7509 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
\r
7511 // The following two settings were suggested by Theo Veenker
\r
7512 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
\r
7513 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
\r
7515 // here are two options for a fix
\r
7516 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
\r
7517 snd_pcm_uframes_t val;
\r
7518 snd_pcm_sw_params_get_boundary( sw_params, &val );
\r
7519 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
\r
7521 result = snd_pcm_sw_params( phandle, sw_params );
\r
7522 if ( result < 0 ) {
\r
7523 snd_pcm_close( phandle );
\r
7524 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
\r
7525 errorText_ = errorStream_.str();
\r
7529 #if defined(__RTAUDIO_DEBUG__)
\r
7530 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
\r
7531 snd_pcm_sw_params_dump( sw_params, out );
\r
7534 // Set flags for buffer conversion
\r
7535 stream_.doConvertBuffer[mode] = false;
\r
7536 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
7537 stream_.doConvertBuffer[mode] = true;
\r
7538 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
\r
7539 stream_.doConvertBuffer[mode] = true;
\r
7540 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
\r
7541 stream_.nUserChannels[mode] > 1 )
\r
7542 stream_.doConvertBuffer[mode] = true;
\r
7544 // Allocate the ApiHandle if necessary and then save.
\r
7545 AlsaHandle *apiInfo = 0;
\r
7546 if ( stream_.apiHandle == 0 ) {
\r
7548 apiInfo = (AlsaHandle *) new AlsaHandle;
\r
7550 catch ( std::bad_alloc& ) {
\r
7551 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
\r
7555 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
\r
7556 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
\r
7560 stream_.apiHandle = (void *) apiInfo;
\r
7561 apiInfo->handles[0] = 0;
\r
7562 apiInfo->handles[1] = 0;
\r
7565 apiInfo = (AlsaHandle *) stream_.apiHandle;
\r
7567 apiInfo->handles[mode] = phandle;
\r
7570 // Allocate necessary internal buffers.
\r
7571 unsigned long bufferBytes;
\r
7572 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
7573 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
7574 if ( stream_.userBuffer[mode] == NULL ) {
\r
7575 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
\r
7579 if ( stream_.doConvertBuffer[mode] ) {
\r
7581 bool makeBuffer = true;
\r
7582 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
\r
7583 if ( mode == INPUT ) {
\r
7584 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
7585 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
7586 if ( bufferBytes <= bytesOut ) makeBuffer = false;
\r
7590 if ( makeBuffer ) {
\r
7591 bufferBytes *= *bufferSize;
\r
7592 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
7593 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
7594 if ( stream_.deviceBuffer == NULL ) {
\r
7595 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
\r
7601 stream_.sampleRate = sampleRate;
\r
7602 stream_.nBuffers = periods;
\r
7603 stream_.device[mode] = device;
\r
7604 stream_.state = STREAM_STOPPED;
\r
7606 // Setup the buffer conversion information structure.
\r
7607 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
\r
7609 // Setup thread if necessary.
\r
7610 if ( stream_.mode == OUTPUT && mode == INPUT ) {
\r
7611 // We had already set up an output stream.
\r
7612 stream_.mode = DUPLEX;
\r
7613 // Link the streams if possible.
\r
7614 apiInfo->synchronized = false;
\r
7615 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
\r
7616 apiInfo->synchronized = true;
\r
7618 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
\r
7619 error( RtAudioError::WARNING );
\r
7623 stream_.mode = mode;
\r
7625 // Setup callback thread.
\r
7626 stream_.callbackInfo.object = (void *) this;
\r
7628 // Set the thread attributes for joinable and realtime scheduling
\r
7629 // priority (optional). The higher priority will only take affect
\r
7630 // if the program is run as root or suid. Note, under Linux
\r
7631 // processes with CAP_SYS_NICE privilege, a user can change
\r
7632 // scheduling policy and priority (thus need not be root). See
\r
7633 // POSIX "capabilities".
\r
7634 pthread_attr_t attr;
\r
7635 pthread_attr_init( &attr );
\r
7636 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
\r
7638 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
\r
7639 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
\r
7640 // We previously attempted to increase the audio callback priority
\r
7641 // to SCHED_RR here via the attributes. However, while no errors
\r
7642 // were reported in doing so, it did not work. So, now this is
\r
7643 // done in the alsaCallbackHandler function.
\r
7644 stream_.callbackInfo.doRealtime = true;
\r
7645 int priority = options->priority;
\r
7646 int min = sched_get_priority_min( SCHED_RR );
\r
7647 int max = sched_get_priority_max( SCHED_RR );
\r
7648 if ( priority < min ) priority = min;
\r
7649 else if ( priority > max ) priority = max;
\r
7650 stream_.callbackInfo.priority = priority;
\r
7654 stream_.callbackInfo.isRunning = true;
\r
7655 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
\r
7656 pthread_attr_destroy( &attr );
\r
7658 stream_.callbackInfo.isRunning = false;
\r
7659 errorText_ = "RtApiAlsa::error creating callback thread!";
\r
7668 pthread_cond_destroy( &apiInfo->runnable_cv );
\r
7669 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
\r
7670 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
\r
7672 stream_.apiHandle = 0;
\r
7675 if ( phandle) snd_pcm_close( phandle );
\r
7677 for ( int i=0; i<2; i++ ) {
\r
7678 if ( stream_.userBuffer[i] ) {
\r
7679 free( stream_.userBuffer[i] );
\r
7680 stream_.userBuffer[i] = 0;
\r
7684 if ( stream_.deviceBuffer ) {
\r
7685 free( stream_.deviceBuffer );
\r
7686 stream_.deviceBuffer = 0;
\r
7689 stream_.state = STREAM_CLOSED;
\r
7693 void RtApiAlsa :: closeStream()
\r
7695 if ( stream_.state == STREAM_CLOSED ) {
\r
7696 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
\r
7697 error( RtAudioError::WARNING );
\r
7701 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
\r
7702 stream_.callbackInfo.isRunning = false;
\r
7703 MUTEX_LOCK( &stream_.mutex );
\r
7704 if ( stream_.state == STREAM_STOPPED ) {
\r
7705 apiInfo->runnable = true;
\r
7706 pthread_cond_signal( &apiInfo->runnable_cv );
\r
7708 MUTEX_UNLOCK( &stream_.mutex );
\r
7709 pthread_join( stream_.callbackInfo.thread, NULL );
\r
7711 if ( stream_.state == STREAM_RUNNING ) {
\r
7712 stream_.state = STREAM_STOPPED;
\r
7713 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
\r
7714 snd_pcm_drop( apiInfo->handles[0] );
\r
7715 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
\r
7716 snd_pcm_drop( apiInfo->handles[1] );
\r
7720 pthread_cond_destroy( &apiInfo->runnable_cv );
\r
7721 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
\r
7722 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
\r
7724 stream_.apiHandle = 0;
\r
7727 for ( int i=0; i<2; i++ ) {
\r
7728 if ( stream_.userBuffer[i] ) {
\r
7729 free( stream_.userBuffer[i] );
\r
7730 stream_.userBuffer[i] = 0;
\r
7734 if ( stream_.deviceBuffer ) {
\r
7735 free( stream_.deviceBuffer );
\r
7736 stream_.deviceBuffer = 0;
\r
7739 stream_.mode = UNINITIALIZED;
\r
7740 stream_.state = STREAM_CLOSED;
\r
7743 void RtApiAlsa :: startStream()
\r
7745 // This method calls snd_pcm_prepare if the device isn't already in that state.
\r
7748 if ( stream_.state == STREAM_RUNNING ) {
\r
7749 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
\r
7750 error( RtAudioError::WARNING );
\r
7754 MUTEX_LOCK( &stream_.mutex );
\r
7757 snd_pcm_state_t state;
\r
7758 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
\r
7759 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
\r
7760 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
7761 state = snd_pcm_state( handle[0] );
\r
7762 if ( state != SND_PCM_STATE_PREPARED ) {
\r
7763 result = snd_pcm_prepare( handle[0] );
\r
7764 if ( result < 0 ) {
\r
7765 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
\r
7766 errorText_ = errorStream_.str();
\r
7772 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
\r
7773 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
\r
7774 state = snd_pcm_state( handle[1] );
\r
7775 if ( state != SND_PCM_STATE_PREPARED ) {
\r
7776 result = snd_pcm_prepare( handle[1] );
\r
7777 if ( result < 0 ) {
\r
7778 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
\r
7779 errorText_ = errorStream_.str();
\r
7785 stream_.state = STREAM_RUNNING;
\r
7788 apiInfo->runnable = true;
\r
7789 pthread_cond_signal( &apiInfo->runnable_cv );
\r
7790 MUTEX_UNLOCK( &stream_.mutex );
\r
7792 if ( result >= 0 ) return;
\r
7793 error( RtAudioError::SYSTEM_ERROR );
\r
7796 void RtApiAlsa :: stopStream()
\r
7799 if ( stream_.state == STREAM_STOPPED ) {
\r
7800 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
\r
7801 error( RtAudioError::WARNING );
\r
7805 stream_.state = STREAM_STOPPED;
\r
7806 MUTEX_LOCK( &stream_.mutex );
\r
7809 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
\r
7810 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
\r
7811 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
7812 if ( apiInfo->synchronized )
\r
7813 result = snd_pcm_drop( handle[0] );
\r
7815 result = snd_pcm_drain( handle[0] );
\r
7816 if ( result < 0 ) {
\r
7817 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
\r
7818 errorText_ = errorStream_.str();
\r
7823 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
\r
7824 result = snd_pcm_drop( handle[1] );
\r
7825 if ( result < 0 ) {
\r
7826 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
\r
7827 errorText_ = errorStream_.str();
\r
7833 apiInfo->runnable = false; // fixes high CPU usage when stopped
\r
7834 MUTEX_UNLOCK( &stream_.mutex );
\r
7836 if ( result >= 0 ) return;
\r
7837 error( RtAudioError::SYSTEM_ERROR );
\r
7840 void RtApiAlsa :: abortStream()
\r
7843 if ( stream_.state == STREAM_STOPPED ) {
\r
7844 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
\r
7845 error( RtAudioError::WARNING );
\r
7849 stream_.state = STREAM_STOPPED;
\r
7850 MUTEX_LOCK( &stream_.mutex );
\r
7853 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
\r
7854 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
\r
7855 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
7856 result = snd_pcm_drop( handle[0] );
\r
7857 if ( result < 0 ) {
\r
7858 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
\r
7859 errorText_ = errorStream_.str();
\r
7864 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
\r
7865 result = snd_pcm_drop( handle[1] );
\r
7866 if ( result < 0 ) {
\r
7867 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
\r
7868 errorText_ = errorStream_.str();
\r
7874 apiInfo->runnable = false; // fixes high CPU usage when stopped
\r
7875 MUTEX_UNLOCK( &stream_.mutex );
\r
7877 if ( result >= 0 ) return;
\r
7878 error( RtAudioError::SYSTEM_ERROR );
\r
7881 void RtApiAlsa :: callbackEvent()
\r
7883 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
\r
7884 if ( stream_.state == STREAM_STOPPED ) {
\r
7885 MUTEX_LOCK( &stream_.mutex );
\r
7886 while ( !apiInfo->runnable )
\r
7887 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
\r
7889 if ( stream_.state != STREAM_RUNNING ) {
\r
7890 MUTEX_UNLOCK( &stream_.mutex );
\r
7893 MUTEX_UNLOCK( &stream_.mutex );
\r
7896 if ( stream_.state == STREAM_CLOSED ) {
\r
7897 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
\r
7898 error( RtAudioError::WARNING );
\r
7902 int doStopStream = 0;
\r
7903 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
\r
7904 double streamTime = getStreamTime();
\r
7905 RtAudioStreamStatus status = 0;
\r
7906 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
\r
7907 status |= RTAUDIO_OUTPUT_UNDERFLOW;
\r
7908 apiInfo->xrun[0] = false;
\r
7910 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
\r
7911 status |= RTAUDIO_INPUT_OVERFLOW;
\r
7912 apiInfo->xrun[1] = false;
\r
7914 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
\r
7915 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
\r
7917 if ( doStopStream == 2 ) {
\r
7922 MUTEX_LOCK( &stream_.mutex );
\r
7924 // The state might change while waiting on a mutex.
\r
7925 if ( stream_.state == STREAM_STOPPED ) goto unlock;
\r
7930 snd_pcm_t **handle;
\r
7931 snd_pcm_sframes_t frames;
\r
7932 RtAudioFormat format;
\r
7933 handle = (snd_pcm_t **) apiInfo->handles;
\r
7935 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
7937 // Setup parameters.
\r
7938 if ( stream_.doConvertBuffer[1] ) {
\r
7939 buffer = stream_.deviceBuffer;
\r
7940 channels = stream_.nDeviceChannels[1];
\r
7941 format = stream_.deviceFormat[1];
\r
7944 buffer = stream_.userBuffer[1];
\r
7945 channels = stream_.nUserChannels[1];
\r
7946 format = stream_.userFormat;
\r
7949 // Read samples from device in interleaved/non-interleaved format.
\r
7950 if ( stream_.deviceInterleaved[1] )
\r
7951 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
\r
7953 void *bufs[channels];
\r
7954 size_t offset = stream_.bufferSize * formatBytes( format );
\r
7955 for ( int i=0; i<channels; i++ )
\r
7956 bufs[i] = (void *) (buffer + (i * offset));
\r
7957 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
\r
7960 if ( result < (int) stream_.bufferSize ) {
\r
7961 // Either an error or overrun occured.
\r
7962 if ( result == -EPIPE ) {
\r
7963 snd_pcm_state_t state = snd_pcm_state( handle[1] );
\r
7964 if ( state == SND_PCM_STATE_XRUN ) {
\r
7965 apiInfo->xrun[1] = true;
\r
7966 result = snd_pcm_prepare( handle[1] );
\r
7967 if ( result < 0 ) {
\r
7968 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
\r
7969 errorText_ = errorStream_.str();
\r
7973 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
\r
7974 errorText_ = errorStream_.str();
\r
7978 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
\r
7979 errorText_ = errorStream_.str();
\r
7981 error( RtAudioError::WARNING );
\r
7985 // Do byte swapping if necessary.
\r
7986 if ( stream_.doByteSwap[1] )
\r
7987 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
\r
7989 // Do buffer conversion if necessary.
\r
7990 if ( stream_.doConvertBuffer[1] )
\r
7991 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
\r
7993 // Check stream latency
\r
7994 result = snd_pcm_delay( handle[1], &frames );
\r
7995 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
\r
8000 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
8002 // Setup parameters and do buffer conversion if necessary.
\r
8003 if ( stream_.doConvertBuffer[0] ) {
\r
8004 buffer = stream_.deviceBuffer;
\r
8005 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
\r
8006 channels = stream_.nDeviceChannels[0];
\r
8007 format = stream_.deviceFormat[0];
\r
8010 buffer = stream_.userBuffer[0];
\r
8011 channels = stream_.nUserChannels[0];
\r
8012 format = stream_.userFormat;
\r
8015 // Do byte swapping if necessary.
\r
8016 if ( stream_.doByteSwap[0] )
\r
8017 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
\r
8019 // Write samples to device in interleaved/non-interleaved format.
\r
8020 if ( stream_.deviceInterleaved[0] )
\r
8021 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
\r
8023 void *bufs[channels];
\r
8024 size_t offset = stream_.bufferSize * formatBytes( format );
\r
8025 for ( int i=0; i<channels; i++ )
\r
8026 bufs[i] = (void *) (buffer + (i * offset));
\r
8027 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
\r
8030 if ( result < (int) stream_.bufferSize ) {
\r
8031 // Either an error or underrun occured.
\r
8032 if ( result == -EPIPE ) {
\r
8033 snd_pcm_state_t state = snd_pcm_state( handle[0] );
\r
8034 if ( state == SND_PCM_STATE_XRUN ) {
\r
8035 apiInfo->xrun[0] = true;
\r
8036 result = snd_pcm_prepare( handle[0] );
\r
8037 if ( result < 0 ) {
\r
8038 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
\r
8039 errorText_ = errorStream_.str();
\r
8042 errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
\r
8045 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
\r
8046 errorText_ = errorStream_.str();
\r
8050 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
\r
8051 errorText_ = errorStream_.str();
\r
8053 error( RtAudioError::WARNING );
\r
8057 // Check stream latency
\r
8058 result = snd_pcm_delay( handle[0], &frames );
\r
8059 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
\r
8063 MUTEX_UNLOCK( &stream_.mutex );
\r
8065 RtApi::tickStreamTime();
\r
8066 if ( doStopStream == 1 ) this->stopStream();
\r
8069 static void *alsaCallbackHandler( void *ptr )
\r
8071 CallbackInfo *info = (CallbackInfo *) ptr;
\r
8072 RtApiAlsa *object = (RtApiAlsa *) info->object;
\r
8073 bool *isRunning = &info->isRunning;
\r
8075 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
\r
8076 if ( info->doRealtime ) {
\r
8077 pthread_t tID = pthread_self(); // ID of this thread
\r
8078 sched_param prio = { info->priority }; // scheduling priority of thread
\r
8079 pthread_setschedparam( tID, SCHED_RR, &prio );
\r
8083 while ( *isRunning == true ) {
\r
8084 pthread_testcancel();
\r
8085 object->callbackEvent();
\r
8088 pthread_exit( NULL );
\r
8091 //******************** End of __LINUX_ALSA__ *********************//
\r
8094 #if defined(__LINUX_PULSE__)
\r
8096 // Code written by Peter Meerwald, pmeerw@pmeerw.net
\r
8097 // and Tristan Matthews.
\r
8099 #include <pulse/error.h>
\r
8100 #include <pulse/simple.h>
\r
8103 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
\r
8104 44100, 48000, 96000, 0};
\r
8106 struct rtaudio_pa_format_mapping_t {
\r
8107 RtAudioFormat rtaudio_format;
\r
8108 pa_sample_format_t pa_format;
\r
8111 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
\r
8112 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
\r
8113 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
\r
8114 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
\r
8115 {0, PA_SAMPLE_INVALID}};
\r
8117 struct PulseAudioHandle {
\r
8118 pa_simple *s_play;
\r
8121 pthread_cond_t runnable_cv;
\r
8123 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
\r
8126 RtApiPulse::~RtApiPulse()
\r
8128 if ( stream_.state != STREAM_CLOSED )
\r
8132 unsigned int RtApiPulse::getDeviceCount( void )
\r
8137 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
\r
8139 RtAudio::DeviceInfo info;
\r
8140 info.probed = true;
\r
8141 info.name = "PulseAudio";
\r
8142 info.outputChannels = 2;
\r
8143 info.inputChannels = 2;
\r
8144 info.duplexChannels = 2;
\r
8145 info.isDefaultOutput = true;
\r
8146 info.isDefaultInput = true;
\r
8148 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
\r
8149 info.sampleRates.push_back( *sr );
\r
8151 info.preferredSampleRate = 48000;
\r
8152 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
\r
8157 static void *pulseaudio_callback( void * user )
\r
8159 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
\r
8160 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
\r
8161 volatile bool *isRunning = &cbi->isRunning;
\r
8163 while ( *isRunning ) {
\r
8164 pthread_testcancel();
\r
8165 context->callbackEvent();
\r
8168 pthread_exit( NULL );
\r
8171 void RtApiPulse::closeStream( void )
\r
8173 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
\r
8175 stream_.callbackInfo.isRunning = false;
\r
8177 MUTEX_LOCK( &stream_.mutex );
\r
8178 if ( stream_.state == STREAM_STOPPED ) {
\r
8179 pah->runnable = true;
\r
8180 pthread_cond_signal( &pah->runnable_cv );
\r
8182 MUTEX_UNLOCK( &stream_.mutex );
\r
8184 pthread_join( pah->thread, 0 );
\r
8185 if ( pah->s_play ) {
\r
8186 pa_simple_flush( pah->s_play, NULL );
\r
8187 pa_simple_free( pah->s_play );
\r
8190 pa_simple_free( pah->s_rec );
\r
8192 pthread_cond_destroy( &pah->runnable_cv );
\r
8194 stream_.apiHandle = 0;
\r
8197 if ( stream_.userBuffer[0] ) {
\r
8198 free( stream_.userBuffer[0] );
\r
8199 stream_.userBuffer[0] = 0;
\r
8201 if ( stream_.userBuffer[1] ) {
\r
8202 free( stream_.userBuffer[1] );
\r
8203 stream_.userBuffer[1] = 0;
\r
8206 stream_.state = STREAM_CLOSED;
\r
8207 stream_.mode = UNINITIALIZED;
\r
8210 void RtApiPulse::callbackEvent( void )
\r
8212 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
\r
8214 if ( stream_.state == STREAM_STOPPED ) {
\r
8215 MUTEX_LOCK( &stream_.mutex );
\r
8216 while ( !pah->runnable )
\r
8217 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
\r
8219 if ( stream_.state != STREAM_RUNNING ) {
\r
8220 MUTEX_UNLOCK( &stream_.mutex );
\r
8223 MUTEX_UNLOCK( &stream_.mutex );
\r
8226 if ( stream_.state == STREAM_CLOSED ) {
\r
8227 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
\r
8228 "this shouldn't happen!";
\r
8229 error( RtAudioError::WARNING );
\r
8233 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
\r
8234 double streamTime = getStreamTime();
\r
8235 RtAudioStreamStatus status = 0;
\r
8236 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
\r
8237 stream_.bufferSize, streamTime, status,
\r
8238 stream_.callbackInfo.userData );
\r
8240 if ( doStopStream == 2 ) {
\r
8245 MUTEX_LOCK( &stream_.mutex );
\r
8246 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
\r
8247 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
\r
8249 if ( stream_.state != STREAM_RUNNING )
\r
8254 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
8255 if ( stream_.doConvertBuffer[OUTPUT] ) {
\r
8256 convertBuffer( stream_.deviceBuffer,
\r
8257 stream_.userBuffer[OUTPUT],
\r
8258 stream_.convertInfo[OUTPUT] );
\r
8259 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
\r
8260 formatBytes( stream_.deviceFormat[OUTPUT] );
\r
8262 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
\r
8263 formatBytes( stream_.userFormat );
\r
8265 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
\r
8266 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
\r
8267 pa_strerror( pa_error ) << ".";
\r
8268 errorText_ = errorStream_.str();
\r
8269 error( RtAudioError::WARNING );
\r
8273 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
\r
8274 if ( stream_.doConvertBuffer[INPUT] )
\r
8275 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
\r
8276 formatBytes( stream_.deviceFormat[INPUT] );
\r
8278 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
\r
8279 formatBytes( stream_.userFormat );
\r
8281 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
\r
8282 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
\r
8283 pa_strerror( pa_error ) << ".";
\r
8284 errorText_ = errorStream_.str();
\r
8285 error( RtAudioError::WARNING );
\r
8287 if ( stream_.doConvertBuffer[INPUT] ) {
\r
8288 convertBuffer( stream_.userBuffer[INPUT],
\r
8289 stream_.deviceBuffer,
\r
8290 stream_.convertInfo[INPUT] );
\r
8295 MUTEX_UNLOCK( &stream_.mutex );
\r
8296 RtApi::tickStreamTime();
\r
8298 if ( doStopStream == 1 )
\r
8302 void RtApiPulse::startStream( void )
\r
8304 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
\r
8306 if ( stream_.state == STREAM_CLOSED ) {
\r
8307 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
\r
8308 error( RtAudioError::INVALID_USE );
\r
8311 if ( stream_.state == STREAM_RUNNING ) {
\r
8312 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
\r
8313 error( RtAudioError::WARNING );
\r
8317 MUTEX_LOCK( &stream_.mutex );
\r
8319 stream_.state = STREAM_RUNNING;
\r
8321 pah->runnable = true;
\r
8322 pthread_cond_signal( &pah->runnable_cv );
\r
8323 MUTEX_UNLOCK( &stream_.mutex );
\r
8326 void RtApiPulse::stopStream( void )
\r
8328 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
\r
8330 if ( stream_.state == STREAM_CLOSED ) {
\r
8331 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
\r
8332 error( RtAudioError::INVALID_USE );
\r
8335 if ( stream_.state == STREAM_STOPPED ) {
\r
8336 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
\r
8337 error( RtAudioError::WARNING );
\r
8341 stream_.state = STREAM_STOPPED;
\r
8342 MUTEX_LOCK( &stream_.mutex );
\r
8344 if ( pah && pah->s_play ) {
\r
8346 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
\r
8347 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
\r
8348 pa_strerror( pa_error ) << ".";
\r
8349 errorText_ = errorStream_.str();
\r
8350 MUTEX_UNLOCK( &stream_.mutex );
\r
8351 error( RtAudioError::SYSTEM_ERROR );
\r
8356 stream_.state = STREAM_STOPPED;
\r
8357 MUTEX_UNLOCK( &stream_.mutex );
\r
8360 void RtApiPulse::abortStream( void )
\r
8362 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
\r
8364 if ( stream_.state == STREAM_CLOSED ) {
\r
8365 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
\r
8366 error( RtAudioError::INVALID_USE );
\r
8369 if ( stream_.state == STREAM_STOPPED ) {
\r
8370 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
\r
8371 error( RtAudioError::WARNING );
\r
8375 stream_.state = STREAM_STOPPED;
\r
8376 MUTEX_LOCK( &stream_.mutex );
\r
8378 if ( pah && pah->s_play ) {
\r
8380 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
\r
8381 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
\r
8382 pa_strerror( pa_error ) << ".";
\r
8383 errorText_ = errorStream_.str();
\r
8384 MUTEX_UNLOCK( &stream_.mutex );
\r
8385 error( RtAudioError::SYSTEM_ERROR );
\r
8390 stream_.state = STREAM_STOPPED;
\r
8391 MUTEX_UNLOCK( &stream_.mutex );
\r
8394 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
\r
8395 unsigned int channels, unsigned int firstChannel,
\r
8396 unsigned int sampleRate, RtAudioFormat format,
\r
8397 unsigned int *bufferSize, RtAudio::StreamOptions *options )
\r
8399 PulseAudioHandle *pah = 0;
\r
8400 unsigned long bufferBytes = 0;
\r
8401 pa_sample_spec ss;
\r
8403 if ( device != 0 ) return false;
\r
8404 if ( mode != INPUT && mode != OUTPUT ) return false;
\r
8405 if ( channels != 1 && channels != 2 ) {
\r
8406 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
\r
8409 ss.channels = channels;
\r
8411 if ( firstChannel != 0 ) return false;
\r
8413 bool sr_found = false;
\r
8414 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
\r
8415 if ( sampleRate == *sr ) {
\r
8417 stream_.sampleRate = sampleRate;
\r
8418 ss.rate = sampleRate;
\r
8422 if ( !sr_found ) {
\r
8423 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
\r
8427 bool sf_found = 0;
\r
8428 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
\r
8429 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
\r
8430 if ( format == sf->rtaudio_format ) {
\r
8432 stream_.userFormat = sf->rtaudio_format;
\r
8433 stream_.deviceFormat[mode] = stream_.userFormat;
\r
8434 ss.format = sf->pa_format;
\r
8438 if ( !sf_found ) { // Use internal data format conversion.
\r
8439 stream_.userFormat = format;
\r
8440 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
\r
8441 ss.format = PA_SAMPLE_FLOAT32LE;
\r
8444 // Set other stream parameters.
\r
8445 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
\r
8446 else stream_.userInterleaved = true;
\r
8447 stream_.deviceInterleaved[mode] = true;
\r
8448 stream_.nBuffers = 1;
\r
8449 stream_.doByteSwap[mode] = false;
\r
8450 stream_.nUserChannels[mode] = channels;
\r
8451 stream_.nDeviceChannels[mode] = channels + firstChannel;
\r
8452 stream_.channelOffset[mode] = 0;
\r
8453 std::string streamName = "RtAudio";
\r
8455 // Set flags for buffer conversion.
\r
8456 stream_.doConvertBuffer[mode] = false;
\r
8457 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
8458 stream_.doConvertBuffer[mode] = true;
\r
8459 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
\r
8460 stream_.doConvertBuffer[mode] = true;
\r
8462 // Allocate necessary internal buffers.
\r
8463 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
8464 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
8465 if ( stream_.userBuffer[mode] == NULL ) {
\r
8466 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
\r
8469 stream_.bufferSize = *bufferSize;
\r
8471 if ( stream_.doConvertBuffer[mode] ) {
\r
8473 bool makeBuffer = true;
\r
8474 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
\r
8475 if ( mode == INPUT ) {
\r
8476 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
8477 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
8478 if ( bufferBytes <= bytesOut ) makeBuffer = false;
\r
8482 if ( makeBuffer ) {
\r
8483 bufferBytes *= *bufferSize;
\r
8484 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
8485 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
8486 if ( stream_.deviceBuffer == NULL ) {
\r
8487 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
\r
8493 stream_.device[mode] = device;
\r
8495 // Setup the buffer conversion information structure.
\r
8496 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
\r
8498 if ( !stream_.apiHandle ) {
\r
8499 PulseAudioHandle *pah = new PulseAudioHandle;
\r
8501 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
\r
8505 stream_.apiHandle = pah;
\r
8506 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
\r
8507 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
\r
8511 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
\r
8514 if ( options && !options->streamName.empty() ) streamName = options->streamName;
\r
8517 pa_buffer_attr buffer_attr;
\r
8518 buffer_attr.fragsize = bufferBytes;
\r
8519 buffer_attr.maxlength = -1;
\r
8521 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
\r
8522 if ( !pah->s_rec ) {
\r
8523 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
\r
8528 pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
\r
8529 if ( !pah->s_play ) {
\r
8530 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
\r
8538 if ( stream_.mode == UNINITIALIZED )
\r
8539 stream_.mode = mode;
\r
8540 else if ( stream_.mode == mode )
\r
8543 stream_.mode = DUPLEX;
\r
8545 if ( !stream_.callbackInfo.isRunning ) {
\r
8546 stream_.callbackInfo.object = this;
\r
8547 stream_.callbackInfo.isRunning = true;
\r
8548 if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {
\r
8549 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
\r
8554 stream_.state = STREAM_STOPPED;
\r
8558 if ( pah && stream_.callbackInfo.isRunning ) {
\r
8559 pthread_cond_destroy( &pah->runnable_cv );
\r
8561 stream_.apiHandle = 0;
\r
8564 for ( int i=0; i<2; i++ ) {
\r
8565 if ( stream_.userBuffer[i] ) {
\r
8566 free( stream_.userBuffer[i] );
\r
8567 stream_.userBuffer[i] = 0;
\r
8571 if ( stream_.deviceBuffer ) {
\r
8572 free( stream_.deviceBuffer );
\r
8573 stream_.deviceBuffer = 0;
\r
8579 //******************** End of __LINUX_PULSE__ *********************//
\r
8582 #if defined(__LINUX_OSS__)
\r
8584 #include <unistd.h>
\r
8585 #include <sys/ioctl.h>
\r
8586 #include <unistd.h>
\r
8587 #include <fcntl.h>
\r
8588 #include <sys/soundcard.h>
\r
8589 #include <errno.h>
\r
8592 static void *ossCallbackHandler(void * ptr);
\r
8594 // A structure to hold various information related to the OSS API
\r
8595 // implementation.
\r
8596 struct OssHandle {
\r
8597 int id[2]; // device ids
\r
8600 pthread_cond_t runnable;
\r
8603 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
\r
8606 RtApiOss :: RtApiOss()
\r
8608 // Nothing to do here.
\r
8611 RtApiOss :: ~RtApiOss()
\r
8613 if ( stream_.state != STREAM_CLOSED ) closeStream();
\r
8616 unsigned int RtApiOss :: getDeviceCount( void )
\r
8618 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
\r
8619 if ( mixerfd == -1 ) {
\r
8620 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
\r
8621 error( RtAudioError::WARNING );
\r
8625 oss_sysinfo sysinfo;
\r
8626 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
\r
8628 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
\r
8629 error( RtAudioError::WARNING );
\r
8634 return sysinfo.numaudios;
\r
8637 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
\r
8639 RtAudio::DeviceInfo info;
\r
8640 info.probed = false;
\r
8642 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
\r
8643 if ( mixerfd == -1 ) {
\r
8644 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
\r
8645 error( RtAudioError::WARNING );
\r
8649 oss_sysinfo sysinfo;
\r
8650 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
\r
8651 if ( result == -1 ) {
\r
8653 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
\r
8654 error( RtAudioError::WARNING );
\r
8658 unsigned nDevices = sysinfo.numaudios;
\r
8659 if ( nDevices == 0 ) {
\r
8661 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
\r
8662 error( RtAudioError::INVALID_USE );
\r
8666 if ( device >= nDevices ) {
\r
8668 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
\r
8669 error( RtAudioError::INVALID_USE );
\r
8673 oss_audioinfo ainfo;
\r
8674 ainfo.dev = device;
\r
8675 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
\r
8677 if ( result == -1 ) {
\r
8678 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
\r
8679 errorText_ = errorStream_.str();
\r
8680 error( RtAudioError::WARNING );
\r
8685 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
\r
8686 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
\r
8687 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
\r
8688 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
\r
8689 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
\r
8692 // Probe data formats ... do for input
\r
8693 unsigned long mask = ainfo.iformats;
\r
8694 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
\r
8695 info.nativeFormats |= RTAUDIO_SINT16;
\r
8696 if ( mask & AFMT_S8 )
\r
8697 info.nativeFormats |= RTAUDIO_SINT8;
\r
8698 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
\r
8699 info.nativeFormats |= RTAUDIO_SINT32;
\r
8700 if ( mask & AFMT_FLOAT )
\r
8701 info.nativeFormats |= RTAUDIO_FLOAT32;
\r
8702 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
\r
8703 info.nativeFormats |= RTAUDIO_SINT24;
\r
8705 // Check that we have at least one supported format
\r
8706 if ( info.nativeFormats == 0 ) {
\r
8707 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
\r
8708 errorText_ = errorStream_.str();
\r
8709 error( RtAudioError::WARNING );
\r
8713 // Probe the supported sample rates.
\r
8714 info.sampleRates.clear();
\r
8715 if ( ainfo.nrates ) {
\r
8716 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
\r
8717 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
\r
8718 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
\r
8719 info.sampleRates.push_back( SAMPLE_RATES[k] );
\r
8721 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
\r
8722 info.preferredSampleRate = SAMPLE_RATES[k];
\r
8730 // Check min and max rate values;
\r
8731 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
\r
8732 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
\r
8733 info.sampleRates.push_back( SAMPLE_RATES[k] );
\r
8735 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
\r
8736 info.preferredSampleRate = SAMPLE_RATES[k];
\r
8741 if ( info.sampleRates.size() == 0 ) {
\r
8742 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
\r
8743 errorText_ = errorStream_.str();
\r
8744 error( RtAudioError::WARNING );
\r
8747 info.probed = true;
\r
8748 info.name = ainfo.name;
\r
8755 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
8756 unsigned int firstChannel, unsigned int sampleRate,
\r
8757 RtAudioFormat format, unsigned int *bufferSize,
\r
8758 RtAudio::StreamOptions *options )
\r
8760 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
\r
8761 if ( mixerfd == -1 ) {
\r
8762 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
\r
8766 oss_sysinfo sysinfo;
\r
8767 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
\r
8768 if ( result == -1 ) {
\r
8770 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
\r
8774 unsigned nDevices = sysinfo.numaudios;
\r
8775 if ( nDevices == 0 ) {
\r
8776 // This should not happen because a check is made before this function is called.
\r
8778 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
\r
8782 if ( device >= nDevices ) {
\r
8783 // This should not happen because a check is made before this function is called.
\r
8785 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
\r
8789 oss_audioinfo ainfo;
\r
8790 ainfo.dev = device;
\r
8791 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
\r
8793 if ( result == -1 ) {
\r
8794 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
\r
8795 errorText_ = errorStream_.str();
\r
8799 // Check if device supports input or output
\r
8800 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
\r
8801 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
\r
8802 if ( mode == OUTPUT )
\r
8803 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
\r
8805 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
\r
8806 errorText_ = errorStream_.str();
\r
8811 OssHandle *handle = (OssHandle *) stream_.apiHandle;
\r
8812 if ( mode == OUTPUT )
\r
8813 flags |= O_WRONLY;
\r
8814 else { // mode == INPUT
\r
8815 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
\r
8816 // We just set the same device for playback ... close and reopen for duplex (OSS only).
\r
8817 close( handle->id[0] );
\r
8818 handle->id[0] = 0;
\r
8819 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
\r
8820 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
\r
8821 errorText_ = errorStream_.str();
\r
8824 // Check that the number previously set channels is the same.
\r
8825 if ( stream_.nUserChannels[0] != channels ) {
\r
8826 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
\r
8827 errorText_ = errorStream_.str();
\r
8833 flags |= O_RDONLY;
\r
8836 // Set exclusive access if specified.
\r
8837 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
\r
8839 // Try to open the device.
\r
8841 fd = open( ainfo.devnode, flags, 0 );
\r
8843 if ( errno == EBUSY )
\r
8844 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
\r
8846 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
\r
8847 errorText_ = errorStream_.str();
\r
8851 // For duplex operation, specifically set this mode (this doesn't seem to work).
\r
8853 if ( flags | O_RDWR ) {
\r
8854 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
\r
8855 if ( result == -1) {
\r
8856 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
\r
8857 errorText_ = errorStream_.str();
\r
8863 // Check the device channel support.
\r
8864 stream_.nUserChannels[mode] = channels;
\r
8865 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
\r
8867 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
\r
8868 errorText_ = errorStream_.str();
\r
8872 // Set the number of channels.
\r
8873 int deviceChannels = channels + firstChannel;
\r
8874 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
\r
8875 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
\r
8877 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
\r
8878 errorText_ = errorStream_.str();
\r
8881 stream_.nDeviceChannels[mode] = deviceChannels;
\r
8883 // Get the data format mask
\r
8885 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
\r
8886 if ( result == -1 ) {
\r
8888 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
\r
8889 errorText_ = errorStream_.str();
\r
8893 // Determine how to set the device format.
\r
8894 stream_.userFormat = format;
\r
8895 int deviceFormat = -1;
\r
8896 stream_.doByteSwap[mode] = false;
\r
8897 if ( format == RTAUDIO_SINT8 ) {
\r
8898 if ( mask & AFMT_S8 ) {
\r
8899 deviceFormat = AFMT_S8;
\r
8900 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
\r
8903 else if ( format == RTAUDIO_SINT16 ) {
\r
8904 if ( mask & AFMT_S16_NE ) {
\r
8905 deviceFormat = AFMT_S16_NE;
\r
8906 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
8908 else if ( mask & AFMT_S16_OE ) {
\r
8909 deviceFormat = AFMT_S16_OE;
\r
8910 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
8911 stream_.doByteSwap[mode] = true;
\r
8914 else if ( format == RTAUDIO_SINT24 ) {
\r
8915 if ( mask & AFMT_S24_NE ) {
\r
8916 deviceFormat = AFMT_S24_NE;
\r
8917 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
\r
8919 else if ( mask & AFMT_S24_OE ) {
\r
8920 deviceFormat = AFMT_S24_OE;
\r
8921 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
\r
8922 stream_.doByteSwap[mode] = true;
\r
8925 else if ( format == RTAUDIO_SINT32 ) {
\r
8926 if ( mask & AFMT_S32_NE ) {
\r
8927 deviceFormat = AFMT_S32_NE;
\r
8928 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
\r
8930 else if ( mask & AFMT_S32_OE ) {
\r
8931 deviceFormat = AFMT_S32_OE;
\r
8932 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
\r
8933 stream_.doByteSwap[mode] = true;
\r
8937 if ( deviceFormat == -1 ) {
\r
8938 // The user requested format is not natively supported by the device.
\r
8939 if ( mask & AFMT_S16_NE ) {
\r
8940 deviceFormat = AFMT_S16_NE;
\r
8941 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
8943 else if ( mask & AFMT_S32_NE ) {
\r
8944 deviceFormat = AFMT_S32_NE;
\r
8945 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
\r
8947 else if ( mask & AFMT_S24_NE ) {
\r
8948 deviceFormat = AFMT_S24_NE;
\r
8949 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
\r
8951 else if ( mask & AFMT_S16_OE ) {
\r
8952 deviceFormat = AFMT_S16_OE;
\r
8953 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
8954 stream_.doByteSwap[mode] = true;
\r
8956 else if ( mask & AFMT_S32_OE ) {
\r
8957 deviceFormat = AFMT_S32_OE;
\r
8958 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
\r
8959 stream_.doByteSwap[mode] = true;
\r
8961 else if ( mask & AFMT_S24_OE ) {
\r
8962 deviceFormat = AFMT_S24_OE;
\r
8963 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
\r
8964 stream_.doByteSwap[mode] = true;
\r
8966 else if ( mask & AFMT_S8) {
\r
8967 deviceFormat = AFMT_S8;
\r
8968 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
\r
8972 if ( stream_.deviceFormat[mode] == 0 ) {
\r
8973 // This really shouldn't happen ...
\r
8975 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
\r
8976 errorText_ = errorStream_.str();
\r
8980 // Set the data format.
\r
8981 int temp = deviceFormat;
\r
8982 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
\r
8983 if ( result == -1 || deviceFormat != temp ) {
\r
8985 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
\r
8986 errorText_ = errorStream_.str();
\r
8990 // Attempt to set the buffer size. According to OSS, the minimum
\r
8991 // number of buffers is two. The supposed minimum buffer size is 16
\r
8992 // bytes, so that will be our lower bound. The argument to this
\r
8993 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
\r
8994 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
\r
8995 // We'll check the actual value used near the end of the setup
\r
8997 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
\r
8998 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
\r
9000 if ( options ) buffers = options->numberOfBuffers;
\r
9001 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
\r
9002 if ( buffers < 2 ) buffers = 3;
\r
9003 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
\r
9004 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
\r
9005 if ( result == -1 ) {
\r
9007 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
\r
9008 errorText_ = errorStream_.str();
\r
9011 stream_.nBuffers = buffers;
\r
9013 // Save buffer size (in sample frames).
\r
9014 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
\r
9015 stream_.bufferSize = *bufferSize;
\r
9017 // Set the sample rate.
\r
9018 int srate = sampleRate;
\r
9019 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
\r
9020 if ( result == -1 ) {
\r
9022 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
\r
9023 errorText_ = errorStream_.str();
\r
9027 // Verify the sample rate setup worked.
\r
9028 if ( abs( srate - sampleRate ) > 100 ) {
\r
9030 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
\r
9031 errorText_ = errorStream_.str();
\r
9034 stream_.sampleRate = sampleRate;
\r
9036 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
\r
9037 // We're doing duplex setup here.
\r
9038 stream_.deviceFormat[0] = stream_.deviceFormat[1];
\r
9039 stream_.nDeviceChannels[0] = deviceChannels;
\r
9042 // Set interleaving parameters.
\r
9043 stream_.userInterleaved = true;
\r
9044 stream_.deviceInterleaved[mode] = true;
\r
9045 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
\r
9046 stream_.userInterleaved = false;
\r
9048 // Set flags for buffer conversion
\r
9049 stream_.doConvertBuffer[mode] = false;
\r
9050 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
9051 stream_.doConvertBuffer[mode] = true;
\r
9052 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
\r
9053 stream_.doConvertBuffer[mode] = true;
\r
9054 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
\r
9055 stream_.nUserChannels[mode] > 1 )
\r
9056 stream_.doConvertBuffer[mode] = true;
\r
9058 // Allocate the stream handles if necessary and then save.
\r
9059 if ( stream_.apiHandle == 0 ) {
\r
9061 handle = new OssHandle;
\r
9063 catch ( std::bad_alloc& ) {
\r
9064 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
\r
9068 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
\r
9069 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
\r
9073 stream_.apiHandle = (void *) handle;
\r
9076 handle = (OssHandle *) stream_.apiHandle;
\r
9078 handle->id[mode] = fd;
\r
9080 // Allocate necessary internal buffers.
\r
9081 unsigned long bufferBytes;
\r
9082 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
9083 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
9084 if ( stream_.userBuffer[mode] == NULL ) {
\r
9085 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
\r
9089 if ( stream_.doConvertBuffer[mode] ) {
\r
9091 bool makeBuffer = true;
\r
9092 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
\r
9093 if ( mode == INPUT ) {
\r
9094 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
9095 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
9096 if ( bufferBytes <= bytesOut ) makeBuffer = false;
\r
9100 if ( makeBuffer ) {
\r
9101 bufferBytes *= *bufferSize;
\r
9102 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
9103 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
9104 if ( stream_.deviceBuffer == NULL ) {
\r
9105 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
\r
9111 stream_.device[mode] = device;
\r
9112 stream_.state = STREAM_STOPPED;
\r
9114 // Setup the buffer conversion information structure.
\r
9115 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
\r
9117 // Setup thread if necessary.
\r
9118 if ( stream_.mode == OUTPUT && mode == INPUT ) {
\r
9119 // We had already set up an output stream.
\r
9120 stream_.mode = DUPLEX;
\r
9121 if ( stream_.device[0] == device ) handle->id[0] = fd;
\r
9124 stream_.mode = mode;
\r
9126 // Setup callback thread.
\r
9127 stream_.callbackInfo.object = (void *) this;
\r
9129 // Set the thread attributes for joinable and realtime scheduling
\r
9130 // priority. The higher priority will only take affect if the
\r
9131 // program is run as root or suid.
\r
9132 pthread_attr_t attr;
\r
9133 pthread_attr_init( &attr );
\r
9134 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
\r
9135 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
\r
9136 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
\r
9137 struct sched_param param;
\r
9138 int priority = options->priority;
\r
9139 int min = sched_get_priority_min( SCHED_RR );
\r
9140 int max = sched_get_priority_max( SCHED_RR );
\r
9141 if ( priority < min ) priority = min;
\r
9142 else if ( priority > max ) priority = max;
\r
9143 param.sched_priority = priority;
\r
9144 pthread_attr_setschedparam( &attr, ¶m );
\r
9145 pthread_attr_setschedpolicy( &attr, SCHED_RR );
\r
9148 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
\r
9150 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
\r
9153 stream_.callbackInfo.isRunning = true;
\r
9154 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
\r
9155 pthread_attr_destroy( &attr );
\r
9157 stream_.callbackInfo.isRunning = false;
\r
9158 errorText_ = "RtApiOss::error creating callback thread!";
\r
9167 pthread_cond_destroy( &handle->runnable );
\r
9168 if ( handle->id[0] ) close( handle->id[0] );
\r
9169 if ( handle->id[1] ) close( handle->id[1] );
\r
9171 stream_.apiHandle = 0;
\r
9174 for ( int i=0; i<2; i++ ) {
\r
9175 if ( stream_.userBuffer[i] ) {
\r
9176 free( stream_.userBuffer[i] );
\r
9177 stream_.userBuffer[i] = 0;
\r
9181 if ( stream_.deviceBuffer ) {
\r
9182 free( stream_.deviceBuffer );
\r
9183 stream_.deviceBuffer = 0;
\r
9189 void RtApiOss :: closeStream()
\r
9191 if ( stream_.state == STREAM_CLOSED ) {
\r
9192 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
\r
9193 error( RtAudioError::WARNING );
\r
9197 OssHandle *handle = (OssHandle *) stream_.apiHandle;
\r
9198 stream_.callbackInfo.isRunning = false;
\r
9199 MUTEX_LOCK( &stream_.mutex );
\r
9200 if ( stream_.state == STREAM_STOPPED )
\r
9201 pthread_cond_signal( &handle->runnable );
\r
9202 MUTEX_UNLOCK( &stream_.mutex );
\r
9203 pthread_join( stream_.callbackInfo.thread, NULL );
\r
9205 if ( stream_.state == STREAM_RUNNING ) {
\r
9206 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
\r
9207 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
\r
9209 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
\r
9210 stream_.state = STREAM_STOPPED;
\r
9214 pthread_cond_destroy( &handle->runnable );
\r
9215 if ( handle->id[0] ) close( handle->id[0] );
\r
9216 if ( handle->id[1] ) close( handle->id[1] );
\r
9218 stream_.apiHandle = 0;
\r
9221 for ( int i=0; i<2; i++ ) {
\r
9222 if ( stream_.userBuffer[i] ) {
\r
9223 free( stream_.userBuffer[i] );
\r
9224 stream_.userBuffer[i] = 0;
\r
9228 if ( stream_.deviceBuffer ) {
\r
9229 free( stream_.deviceBuffer );
\r
9230 stream_.deviceBuffer = 0;
\r
9233 stream_.mode = UNINITIALIZED;
\r
9234 stream_.state = STREAM_CLOSED;
\r
9237 void RtApiOss :: startStream()
\r
9240 if ( stream_.state == STREAM_RUNNING ) {
\r
9241 errorText_ = "RtApiOss::startStream(): the stream is already running!";
\r
9242 error( RtAudioError::WARNING );
\r
9246 MUTEX_LOCK( &stream_.mutex );
\r
9248 stream_.state = STREAM_RUNNING;
\r
9250 // No need to do anything else here ... OSS automatically starts
\r
9251 // when fed samples.
\r
9253 MUTEX_UNLOCK( &stream_.mutex );
\r
9255 OssHandle *handle = (OssHandle *) stream_.apiHandle;
\r
9256 pthread_cond_signal( &handle->runnable );
\r
9259 void RtApiOss :: stopStream()
\r
9262 if ( stream_.state == STREAM_STOPPED ) {
\r
9263 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
\r
9264 error( RtAudioError::WARNING );
\r
9268 MUTEX_LOCK( &stream_.mutex );
\r
9270 // The state might change while waiting on a mutex.
\r
9271 if ( stream_.state == STREAM_STOPPED ) {
\r
9272 MUTEX_UNLOCK( &stream_.mutex );
\r
9277 OssHandle *handle = (OssHandle *) stream_.apiHandle;
\r
9278 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
9280 // Flush the output with zeros a few times.
\r
9283 RtAudioFormat format;
\r
9285 if ( stream_.doConvertBuffer[0] ) {
\r
9286 buffer = stream_.deviceBuffer;
\r
9287 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
\r
9288 format = stream_.deviceFormat[0];
\r
9291 buffer = stream_.userBuffer[0];
\r
9292 samples = stream_.bufferSize * stream_.nUserChannels[0];
\r
9293 format = stream_.userFormat;
\r
9296 memset( buffer, 0, samples * formatBytes(format) );
\r
9297 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
\r
9298 result = write( handle->id[0], buffer, samples * formatBytes(format) );
\r
9299 if ( result == -1 ) {
\r
9300 errorText_ = "RtApiOss::stopStream: audio write error.";
\r
9301 error( RtAudioError::WARNING );
\r
9305 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
\r
9306 if ( result == -1 ) {
\r
9307 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
\r
9308 errorText_ = errorStream_.str();
\r
9311 handle->triggered = false;
\r
9314 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
\r
9315 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
\r
9316 if ( result == -1 ) {
\r
9317 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
\r
9318 errorText_ = errorStream_.str();
\r
9324 stream_.state = STREAM_STOPPED;
\r
9325 MUTEX_UNLOCK( &stream_.mutex );
\r
9327 if ( result != -1 ) return;
\r
9328 error( RtAudioError::SYSTEM_ERROR );
\r
9331 void RtApiOss :: abortStream()
\r
9334 if ( stream_.state == STREAM_STOPPED ) {
\r
9335 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
\r
9336 error( RtAudioError::WARNING );
\r
9340 MUTEX_LOCK( &stream_.mutex );
\r
9342 // The state might change while waiting on a mutex.
\r
9343 if ( stream_.state == STREAM_STOPPED ) {
\r
9344 MUTEX_UNLOCK( &stream_.mutex );
\r
9349 OssHandle *handle = (OssHandle *) stream_.apiHandle;
\r
9350 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
9351 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
\r
9352 if ( result == -1 ) {
\r
9353 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
\r
9354 errorText_ = errorStream_.str();
\r
9357 handle->triggered = false;
\r
9360 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
\r
9361 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
\r
9362 if ( result == -1 ) {
\r
9363 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
\r
9364 errorText_ = errorStream_.str();
\r
9370 stream_.state = STREAM_STOPPED;
\r
9371 MUTEX_UNLOCK( &stream_.mutex );
\r
9373 if ( result != -1 ) return;
\r
9374 error( RtAudioError::SYSTEM_ERROR );
\r
9377 void RtApiOss :: callbackEvent()
\r
9379 OssHandle *handle = (OssHandle *) stream_.apiHandle;
\r
9380 if ( stream_.state == STREAM_STOPPED ) {
\r
9381 MUTEX_LOCK( &stream_.mutex );
\r
9382 pthread_cond_wait( &handle->runnable, &stream_.mutex );
\r
9383 if ( stream_.state != STREAM_RUNNING ) {
\r
9384 MUTEX_UNLOCK( &stream_.mutex );
\r
9387 MUTEX_UNLOCK( &stream_.mutex );
\r
9390 if ( stream_.state == STREAM_CLOSED ) {
\r
9391 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
\r
9392 error( RtAudioError::WARNING );
\r
9396 // Invoke user callback to get fresh output data.
\r
9397 int doStopStream = 0;
\r
9398 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
\r
9399 double streamTime = getStreamTime();
\r
9400 RtAudioStreamStatus status = 0;
\r
9401 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
\r
9402 status |= RTAUDIO_OUTPUT_UNDERFLOW;
\r
9403 handle->xrun[0] = false;
\r
9405 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
\r
9406 status |= RTAUDIO_INPUT_OVERFLOW;
\r
9407 handle->xrun[1] = false;
\r
9409 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
\r
9410 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
\r
9411 if ( doStopStream == 2 ) {
\r
9412 this->abortStream();
\r
9416 MUTEX_LOCK( &stream_.mutex );
\r
9418 // The state might change while waiting on a mutex.
\r
9419 if ( stream_.state == STREAM_STOPPED ) goto unlock;
\r
9424 RtAudioFormat format;
\r
9426 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
9428 // Setup parameters and do buffer conversion if necessary.
\r
9429 if ( stream_.doConvertBuffer[0] ) {
\r
9430 buffer = stream_.deviceBuffer;
\r
9431 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
\r
9432 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
\r
9433 format = stream_.deviceFormat[0];
\r
9436 buffer = stream_.userBuffer[0];
\r
9437 samples = stream_.bufferSize * stream_.nUserChannels[0];
\r
9438 format = stream_.userFormat;
\r
9441 // Do byte swapping if necessary.
\r
9442 if ( stream_.doByteSwap[0] )
\r
9443 byteSwapBuffer( buffer, samples, format );
\r
9445 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
\r
9447 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
\r
9448 result = write( handle->id[0], buffer, samples * formatBytes(format) );
\r
9449 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
\r
9450 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
\r
9451 handle->triggered = true;
\r
9454 // Write samples to device.
\r
9455 result = write( handle->id[0], buffer, samples * formatBytes(format) );
\r
9457 if ( result == -1 ) {
\r
9458 // We'll assume this is an underrun, though there isn't a
\r
9459 // specific means for determining that.
\r
9460 handle->xrun[0] = true;
\r
9461 errorText_ = "RtApiOss::callbackEvent: audio write error.";
\r
9462 error( RtAudioError::WARNING );
\r
9463 // Continue on to input section.
\r
9467 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
9469 // Setup parameters.
\r
9470 if ( stream_.doConvertBuffer[1] ) {
\r
9471 buffer = stream_.deviceBuffer;
\r
9472 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
\r
9473 format = stream_.deviceFormat[1];
\r
9476 buffer = stream_.userBuffer[1];
\r
9477 samples = stream_.bufferSize * stream_.nUserChannels[1];
\r
9478 format = stream_.userFormat;
\r
9481 // Read samples from device.
\r
9482 result = read( handle->id[1], buffer, samples * formatBytes(format) );
\r
9484 if ( result == -1 ) {
\r
9485 // We'll assume this is an overrun, though there isn't a
\r
9486 // specific means for determining that.
\r
9487 handle->xrun[1] = true;
\r
9488 errorText_ = "RtApiOss::callbackEvent: audio read error.";
\r
9489 error( RtAudioError::WARNING );
\r
9493 // Do byte swapping if necessary.
\r
9494 if ( stream_.doByteSwap[1] )
\r
9495 byteSwapBuffer( buffer, samples, format );
\r
9497 // Do buffer conversion if necessary.
\r
9498 if ( stream_.doConvertBuffer[1] )
\r
9499 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
\r
9503 MUTEX_UNLOCK( &stream_.mutex );
\r
9505 RtApi::tickStreamTime();
\r
9506 if ( doStopStream == 1 ) this->stopStream();
\r
9509 static void *ossCallbackHandler( void *ptr )
\r
9511 CallbackInfo *info = (CallbackInfo *) ptr;
\r
9512 RtApiOss *object = (RtApiOss *) info->object;
\r
9513 bool *isRunning = &info->isRunning;
\r
9515 while ( *isRunning == true ) {
\r
9516 pthread_testcancel();
\r
9517 object->callbackEvent();
\r
9520 pthread_exit( NULL );
\r
9523 //******************** End of __LINUX_OSS__ *********************//
\r
9527 // *************************************************** //
\r
9529 // Protected common (OS-independent) RtAudio methods.
\r
9531 // *************************************************** //
\r
9533 // This method can be modified to control the behavior of error
\r
9534 // message printing.
\r
9535 void RtApi :: error( RtAudioError::Type type )
\r
9537 errorStream_.str(""); // clear the ostringstream
\r
9539 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
\r
9540 if ( errorCallback ) {
\r
9541 // abortStream() can generate new error messages. Ignore them. Just keep original one.
\r
9543 if ( firstErrorOccurred_ )
\r
9546 firstErrorOccurred_ = true;
\r
9547 const std::string errorMessage = errorText_;
\r
9549 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
\r
9550 stream_.callbackInfo.isRunning = false; // exit from the thread
\r
9554 errorCallback( type, errorMessage );
\r
9555 firstErrorOccurred_ = false;
\r
9559 if ( type == RtAudioError::WARNING && showWarnings_ == true )
\r
9560 std::cerr << '\n' << errorText_ << "\n\n";
\r
9561 else if ( type != RtAudioError::WARNING )
\r
9562 throw( RtAudioError( errorText_, type ) );
\r
9565 void RtApi :: verifyStream()
\r
9567 if ( stream_.state == STREAM_CLOSED ) {
\r
9568 errorText_ = "RtApi:: a stream is not open!";
\r
9569 error( RtAudioError::INVALID_USE );
\r
9573 void RtApi :: clearStreamInfo()
\r
9575 stream_.mode = UNINITIALIZED;
\r
9576 stream_.state = STREAM_CLOSED;
\r
9577 stream_.sampleRate = 0;
\r
9578 stream_.bufferSize = 0;
\r
9579 stream_.nBuffers = 0;
\r
9580 stream_.userFormat = 0;
\r
9581 stream_.userInterleaved = true;
\r
9582 stream_.streamTime = 0.0;
\r
9583 stream_.apiHandle = 0;
\r
9584 stream_.deviceBuffer = 0;
\r
9585 stream_.callbackInfo.callback = 0;
\r
9586 stream_.callbackInfo.userData = 0;
\r
9587 stream_.callbackInfo.isRunning = false;
\r
9588 stream_.callbackInfo.errorCallback = 0;
\r
9589 for ( int i=0; i<2; i++ ) {
\r
9590 stream_.device[i] = 11111;
\r
9591 stream_.doConvertBuffer[i] = false;
\r
9592 stream_.deviceInterleaved[i] = true;
\r
9593 stream_.doByteSwap[i] = false;
\r
9594 stream_.nUserChannels[i] = 0;
\r
9595 stream_.nDeviceChannels[i] = 0;
\r
9596 stream_.channelOffset[i] = 0;
\r
9597 stream_.deviceFormat[i] = 0;
\r
9598 stream_.latency[i] = 0;
\r
9599 stream_.userBuffer[i] = 0;
\r
9600 stream_.convertInfo[i].channels = 0;
\r
9601 stream_.convertInfo[i].inJump = 0;
\r
9602 stream_.convertInfo[i].outJump = 0;
\r
9603 stream_.convertInfo[i].inFormat = 0;
\r
9604 stream_.convertInfo[i].outFormat = 0;
\r
9605 stream_.convertInfo[i].inOffset.clear();
\r
9606 stream_.convertInfo[i].outOffset.clear();
\r
9610 unsigned int RtApi :: formatBytes( RtAudioFormat format )
\r
9612 if ( format == RTAUDIO_SINT16 )
\r
9614 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
\r
9616 else if ( format == RTAUDIO_FLOAT64 )
\r
9618 else if ( format == RTAUDIO_SINT24 )
\r
9620 else if ( format == RTAUDIO_SINT8 )
\r
9623 errorText_ = "RtApi::formatBytes: undefined format.";
\r
9624 error( RtAudioError::WARNING );
\r
9629 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
\r
9631 if ( mode == INPUT ) { // convert device to user buffer
\r
9632 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
\r
9633 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
\r
9634 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
\r
9635 stream_.convertInfo[mode].outFormat = stream_.userFormat;
\r
9637 else { // convert user to device buffer
\r
9638 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
\r
9639 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
\r
9640 stream_.convertInfo[mode].inFormat = stream_.userFormat;
\r
9641 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
\r
9644 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
\r
9645 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
\r
9647 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
\r
9649 // Set up the interleave/deinterleave offsets.
\r
9650 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
\r
9651 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
\r
9652 ( mode == INPUT && stream_.userInterleaved ) ) {
\r
9653 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
\r
9654 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
\r
9655 stream_.convertInfo[mode].outOffset.push_back( k );
\r
9656 stream_.convertInfo[mode].inJump = 1;
\r
9660 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
\r
9661 stream_.convertInfo[mode].inOffset.push_back( k );
\r
9662 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
\r
9663 stream_.convertInfo[mode].outJump = 1;
\r
9667 else { // no (de)interleaving
\r
9668 if ( stream_.userInterleaved ) {
\r
9669 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
\r
9670 stream_.convertInfo[mode].inOffset.push_back( k );
\r
9671 stream_.convertInfo[mode].outOffset.push_back( k );
\r
9675 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
\r
9676 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
\r
9677 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
\r
9678 stream_.convertInfo[mode].inJump = 1;
\r
9679 stream_.convertInfo[mode].outJump = 1;
\r
9684 // Add channel offset.
\r
9685 if ( firstChannel > 0 ) {
\r
9686 if ( stream_.deviceInterleaved[mode] ) {
\r
9687 if ( mode == OUTPUT ) {
\r
9688 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
\r
9689 stream_.convertInfo[mode].outOffset[k] += firstChannel;
\r
9692 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
\r
9693 stream_.convertInfo[mode].inOffset[k] += firstChannel;
\r
9697 if ( mode == OUTPUT ) {
\r
9698 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
\r
9699 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
\r
9702 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
\r
9703 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
\r
9709 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
\r
9711 // This function does format conversion, input/output channel compensation, and
\r
9712 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
\r
9713 // the lower three bytes of a 32-bit integer.
\r
9715 // Clear our device buffer when in/out duplex device channels are different
\r
9716 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
\r
9717 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
\r
9718 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
\r
9721 if (info.outFormat == RTAUDIO_FLOAT64) {
\r
9723 Float64 *out = (Float64 *)outBuffer;
\r
9725 if (info.inFormat == RTAUDIO_SINT8) {
\r
9726 signed char *in = (signed char *)inBuffer;
\r
9727 scale = 1.0 / 127.5;
\r
9728 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9729 for (j=0; j<info.channels; j++) {
\r
9730 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
\r
9731 out[info.outOffset[j]] += 0.5;
\r
9732 out[info.outOffset[j]] *= scale;
\r
9734 in += info.inJump;
\r
9735 out += info.outJump;
\r
9738 else if (info.inFormat == RTAUDIO_SINT16) {
\r
9739 Int16 *in = (Int16 *)inBuffer;
\r
9740 scale = 1.0 / 32767.5;
\r
9741 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9742 for (j=0; j<info.channels; j++) {
\r
9743 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
\r
9744 out[info.outOffset[j]] += 0.5;
\r
9745 out[info.outOffset[j]] *= scale;
\r
9747 in += info.inJump;
\r
9748 out += info.outJump;
\r
9751 else if (info.inFormat == RTAUDIO_SINT24) {
\r
9752 Int24 *in = (Int24 *)inBuffer;
\r
9753 scale = 1.0 / 8388607.5;
\r
9754 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9755 for (j=0; j<info.channels; j++) {
\r
9756 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
\r
9757 out[info.outOffset[j]] += 0.5;
\r
9758 out[info.outOffset[j]] *= scale;
\r
9760 in += info.inJump;
\r
9761 out += info.outJump;
\r
9764 else if (info.inFormat == RTAUDIO_SINT32) {
\r
9765 Int32 *in = (Int32 *)inBuffer;
\r
9766 scale = 1.0 / 2147483647.5;
\r
9767 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9768 for (j=0; j<info.channels; j++) {
\r
9769 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
\r
9770 out[info.outOffset[j]] += 0.5;
\r
9771 out[info.outOffset[j]] *= scale;
\r
9773 in += info.inJump;
\r
9774 out += info.outJump;
\r
9777 else if (info.inFormat == RTAUDIO_FLOAT32) {
\r
9778 Float32 *in = (Float32 *)inBuffer;
\r
9779 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9780 for (j=0; j<info.channels; j++) {
\r
9781 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
\r
9783 in += info.inJump;
\r
9784 out += info.outJump;
\r
9787 else if (info.inFormat == RTAUDIO_FLOAT64) {
\r
9788 // Channel compensation and/or (de)interleaving only.
\r
9789 Float64 *in = (Float64 *)inBuffer;
\r
9790 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9791 for (j=0; j<info.channels; j++) {
\r
9792 out[info.outOffset[j]] = in[info.inOffset[j]];
\r
9794 in += info.inJump;
\r
9795 out += info.outJump;
\r
9799 else if (info.outFormat == RTAUDIO_FLOAT32) {
\r
9801 Float32 *out = (Float32 *)outBuffer;
\r
9803 if (info.inFormat == RTAUDIO_SINT8) {
\r
9804 signed char *in = (signed char *)inBuffer;
\r
9805 scale = (Float32) ( 1.0 / 127.5 );
\r
9806 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9807 for (j=0; j<info.channels; j++) {
\r
9808 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
\r
9809 out[info.outOffset[j]] += 0.5;
\r
9810 out[info.outOffset[j]] *= scale;
\r
9812 in += info.inJump;
\r
9813 out += info.outJump;
\r
9816 else if (info.inFormat == RTAUDIO_SINT16) {
\r
9817 Int16 *in = (Int16 *)inBuffer;
\r
9818 scale = (Float32) ( 1.0 / 32767.5 );
\r
9819 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9820 for (j=0; j<info.channels; j++) {
\r
9821 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
\r
9822 out[info.outOffset[j]] += 0.5;
\r
9823 out[info.outOffset[j]] *= scale;
\r
9825 in += info.inJump;
\r
9826 out += info.outJump;
\r
9829 else if (info.inFormat == RTAUDIO_SINT24) {
\r
9830 Int24 *in = (Int24 *)inBuffer;
\r
9831 scale = (Float32) ( 1.0 / 8388607.5 );
\r
9832 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9833 for (j=0; j<info.channels; j++) {
\r
9834 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
\r
9835 out[info.outOffset[j]] += 0.5;
\r
9836 out[info.outOffset[j]] *= scale;
\r
9838 in += info.inJump;
\r
9839 out += info.outJump;
\r
9842 else if (info.inFormat == RTAUDIO_SINT32) {
\r
9843 Int32 *in = (Int32 *)inBuffer;
\r
9844 scale = (Float32) ( 1.0 / 2147483647.5 );
\r
9845 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9846 for (j=0; j<info.channels; j++) {
\r
9847 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
\r
9848 out[info.outOffset[j]] += 0.5;
\r
9849 out[info.outOffset[j]] *= scale;
\r
9851 in += info.inJump;
\r
9852 out += info.outJump;
\r
9855 else if (info.inFormat == RTAUDIO_FLOAT32) {
\r
9856 // Channel compensation and/or (de)interleaving only.
\r
9857 Float32 *in = (Float32 *)inBuffer;
\r
9858 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9859 for (j=0; j<info.channels; j++) {
\r
9860 out[info.outOffset[j]] = in[info.inOffset[j]];
\r
9862 in += info.inJump;
\r
9863 out += info.outJump;
\r
9866 else if (info.inFormat == RTAUDIO_FLOAT64) {
\r
9867 Float64 *in = (Float64 *)inBuffer;
\r
9868 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9869 for (j=0; j<info.channels; j++) {
\r
9870 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
\r
9872 in += info.inJump;
\r
9873 out += info.outJump;
\r
9877 else if (info.outFormat == RTAUDIO_SINT32) {
\r
9878 Int32 *out = (Int32 *)outBuffer;
\r
9879 if (info.inFormat == RTAUDIO_SINT8) {
\r
9880 signed char *in = (signed char *)inBuffer;
\r
9881 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9882 for (j=0; j<info.channels; j++) {
\r
9883 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
\r
9884 out[info.outOffset[j]] <<= 24;
\r
9886 in += info.inJump;
\r
9887 out += info.outJump;
\r
9890 else if (info.inFormat == RTAUDIO_SINT16) {
\r
9891 Int16 *in = (Int16 *)inBuffer;
\r
9892 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9893 for (j=0; j<info.channels; j++) {
\r
9894 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
\r
9895 out[info.outOffset[j]] <<= 16;
\r
9897 in += info.inJump;
\r
9898 out += info.outJump;
\r
9901 else if (info.inFormat == RTAUDIO_SINT24) {
\r
9902 Int24 *in = (Int24 *)inBuffer;
\r
9903 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9904 for (j=0; j<info.channels; j++) {
\r
9905 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
\r
9906 out[info.outOffset[j]] <<= 8;
\r
9908 in += info.inJump;
\r
9909 out += info.outJump;
\r
9912 else if (info.inFormat == RTAUDIO_SINT32) {
\r
9913 // Channel compensation and/or (de)interleaving only.
\r
9914 Int32 *in = (Int32 *)inBuffer;
\r
9915 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9916 for (j=0; j<info.channels; j++) {
\r
9917 out[info.outOffset[j]] = in[info.inOffset[j]];
\r
9919 in += info.inJump;
\r
9920 out += info.outJump;
\r
9923 else if (info.inFormat == RTAUDIO_FLOAT32) {
\r
9924 Float32 *in = (Float32 *)inBuffer;
\r
9925 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9926 for (j=0; j<info.channels; j++) {
\r
9927 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
\r
9929 in += info.inJump;
\r
9930 out += info.outJump;
\r
9933 else if (info.inFormat == RTAUDIO_FLOAT64) {
\r
9934 Float64 *in = (Float64 *)inBuffer;
\r
9935 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9936 for (j=0; j<info.channels; j++) {
\r
9937 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
\r
9939 in += info.inJump;
\r
9940 out += info.outJump;
\r
9944 else if (info.outFormat == RTAUDIO_SINT24) {
\r
9945 Int24 *out = (Int24 *)outBuffer;
\r
9946 if (info.inFormat == RTAUDIO_SINT8) {
\r
9947 signed char *in = (signed char *)inBuffer;
\r
9948 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9949 for (j=0; j<info.channels; j++) {
\r
9950 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
\r
9951 //out[info.outOffset[j]] <<= 16;
\r
9953 in += info.inJump;
\r
9954 out += info.outJump;
\r
9957 else if (info.inFormat == RTAUDIO_SINT16) {
\r
9958 Int16 *in = (Int16 *)inBuffer;
\r
9959 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9960 for (j=0; j<info.channels; j++) {
\r
9961 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
\r
9962 //out[info.outOffset[j]] <<= 8;
\r
9964 in += info.inJump;
\r
9965 out += info.outJump;
\r
9968 else if (info.inFormat == RTAUDIO_SINT24) {
\r
9969 // Channel compensation and/or (de)interleaving only.
\r
9970 Int24 *in = (Int24 *)inBuffer;
\r
9971 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9972 for (j=0; j<info.channels; j++) {
\r
9973 out[info.outOffset[j]] = in[info.inOffset[j]];
\r
9975 in += info.inJump;
\r
9976 out += info.outJump;
\r
9979 else if (info.inFormat == RTAUDIO_SINT32) {
\r
9980 Int32 *in = (Int32 *)inBuffer;
\r
9981 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9982 for (j=0; j<info.channels; j++) {
\r
9983 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
\r
9984 //out[info.outOffset[j]] >>= 8;
\r
9986 in += info.inJump;
\r
9987 out += info.outJump;
\r
9990 else if (info.inFormat == RTAUDIO_FLOAT32) {
\r
9991 Float32 *in = (Float32 *)inBuffer;
\r
9992 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9993 for (j=0; j<info.channels; j++) {
\r
9994 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
\r
9996 in += info.inJump;
\r
9997 out += info.outJump;
\r
10000 else if (info.inFormat == RTAUDIO_FLOAT64) {
\r
10001 Float64 *in = (Float64 *)inBuffer;
\r
10002 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10003 for (j=0; j<info.channels; j++) {
\r
10004 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
\r
10006 in += info.inJump;
\r
10007 out += info.outJump;
\r
10011 else if (info.outFormat == RTAUDIO_SINT16) {
\r
10012 Int16 *out = (Int16 *)outBuffer;
\r
10013 if (info.inFormat == RTAUDIO_SINT8) {
\r
10014 signed char *in = (signed char *)inBuffer;
\r
10015 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10016 for (j=0; j<info.channels; j++) {
\r
10017 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
\r
10018 out[info.outOffset[j]] <<= 8;
\r
10020 in += info.inJump;
\r
10021 out += info.outJump;
\r
10024 else if (info.inFormat == RTAUDIO_SINT16) {
\r
10025 // Channel compensation and/or (de)interleaving only.
\r
10026 Int16 *in = (Int16 *)inBuffer;
\r
10027 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10028 for (j=0; j<info.channels; j++) {
\r
10029 out[info.outOffset[j]] = in[info.inOffset[j]];
\r
10031 in += info.inJump;
\r
10032 out += info.outJump;
\r
10035 else if (info.inFormat == RTAUDIO_SINT24) {
\r
10036 Int24 *in = (Int24 *)inBuffer;
\r
10037 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10038 for (j=0; j<info.channels; j++) {
\r
10039 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
\r
10041 in += info.inJump;
\r
10042 out += info.outJump;
\r
10045 else if (info.inFormat == RTAUDIO_SINT32) {
\r
10046 Int32 *in = (Int32 *)inBuffer;
\r
10047 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10048 for (j=0; j<info.channels; j++) {
\r
10049 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
\r
10051 in += info.inJump;
\r
10052 out += info.outJump;
\r
10055 else if (info.inFormat == RTAUDIO_FLOAT32) {
\r
10056 Float32 *in = (Float32 *)inBuffer;
\r
10057 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10058 for (j=0; j<info.channels; j++) {
\r
10059 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
\r
10061 in += info.inJump;
\r
10062 out += info.outJump;
\r
10065 else if (info.inFormat == RTAUDIO_FLOAT64) {
\r
10066 Float64 *in = (Float64 *)inBuffer;
\r
10067 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10068 for (j=0; j<info.channels; j++) {
\r
10069 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
\r
10071 in += info.inJump;
\r
10072 out += info.outJump;
\r
10076 else if (info.outFormat == RTAUDIO_SINT8) {
\r
10077 signed char *out = (signed char *)outBuffer;
\r
10078 if (info.inFormat == RTAUDIO_SINT8) {
\r
10079 // Channel compensation and/or (de)interleaving only.
\r
10080 signed char *in = (signed char *)inBuffer;
\r
10081 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10082 for (j=0; j<info.channels; j++) {
\r
10083 out[info.outOffset[j]] = in[info.inOffset[j]];
\r
10085 in += info.inJump;
\r
10086 out += info.outJump;
\r
10089 if (info.inFormat == RTAUDIO_SINT16) {
\r
10090 Int16 *in = (Int16 *)inBuffer;
\r
10091 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10092 for (j=0; j<info.channels; j++) {
\r
10093 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
\r
10095 in += info.inJump;
\r
10096 out += info.outJump;
\r
10099 else if (info.inFormat == RTAUDIO_SINT24) {
\r
10100 Int24 *in = (Int24 *)inBuffer;
\r
10101 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10102 for (j=0; j<info.channels; j++) {
\r
10103 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
\r
10105 in += info.inJump;
\r
10106 out += info.outJump;
\r
10109 else if (info.inFormat == RTAUDIO_SINT32) {
\r
10110 Int32 *in = (Int32 *)inBuffer;
\r
10111 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10112 for (j=0; j<info.channels; j++) {
\r
10113 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
\r
10115 in += info.inJump;
\r
10116 out += info.outJump;
\r
10119 else if (info.inFormat == RTAUDIO_FLOAT32) {
\r
10120 Float32 *in = (Float32 *)inBuffer;
\r
10121 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10122 for (j=0; j<info.channels; j++) {
\r
10123 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
\r
10125 in += info.inJump;
\r
10126 out += info.outJump;
\r
10129 else if (info.inFormat == RTAUDIO_FLOAT64) {
\r
10130 Float64 *in = (Float64 *)inBuffer;
\r
10131 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10132 for (j=0; j<info.channels; j++) {
\r
10133 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
\r
10135 in += info.inJump;
\r
10136 out += info.outJump;
\r
10142 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
\r
10143 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
\r
10144 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
\r
10146 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
\r
10152 if ( format == RTAUDIO_SINT16 ) {
\r
10153 for ( unsigned int i=0; i<samples; i++ ) {
\r
10154 // Swap 1st and 2nd bytes.
\r
10156 *(ptr) = *(ptr+1);
\r
10159 // Increment 2 bytes.
\r
10163 else if ( format == RTAUDIO_SINT32 ||
\r
10164 format == RTAUDIO_FLOAT32 ) {
\r
10165 for ( unsigned int i=0; i<samples; i++ ) {
\r
10166 // Swap 1st and 4th bytes.
\r
10168 *(ptr) = *(ptr+3);
\r
10171 // Swap 2nd and 3rd bytes.
\r
10174 *(ptr) = *(ptr+1);
\r
10177 // Increment 3 more bytes.
\r
10181 else if ( format == RTAUDIO_SINT24 ) {
\r
10182 for ( unsigned int i=0; i<samples; i++ ) {
\r
10183 // Swap 1st and 3rd bytes.
\r
10185 *(ptr) = *(ptr+2);
\r
10188 // Increment 2 more bytes.
\r
10192 else if ( format == RTAUDIO_FLOAT64 ) {
\r
10193 for ( unsigned int i=0; i<samples; i++ ) {
\r
10194 // Swap 1st and 8th bytes
\r
10196 *(ptr) = *(ptr+7);
\r
10199 // Swap 2nd and 7th bytes
\r
10202 *(ptr) = *(ptr+5);
\r
10205 // Swap 3rd and 6th bytes
\r
10208 *(ptr) = *(ptr+3);
\r
10211 // Swap 4th and 5th bytes
\r
10214 *(ptr) = *(ptr+1);
\r
10217 // Increment 5 more bytes.
\r
10223 // Indentation settings for Vim and Emacs
\r
10225 // Local Variables:
\r
10226 // c-basic-offset: 2
\r
10227 // indent-tabs-mode: nil
\r
10230 // vim: et sts=2 sw=2
\r
projects / rtaudio.git / blob