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annotate Alc/backends/coreaudio.c @ 0:f9476ff7637e

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initial forking of open-al to create multiple listeners
author Robert McIntyre <rlm@mit.edu>
date Tue, 25 Oct 2011 13:02:31 -0700
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rev   line source
rlm@0 1 /**
rlm@0 2 * OpenAL cross platform audio library
rlm@0 3 * Copyright (C) 1999-2007 by authors.
rlm@0 4 * This library is free software; you can redistribute it and/or
rlm@0 5 * modify it under the terms of the GNU Library General Public
rlm@0 6 * License as published by the Free Software Foundation; either
rlm@0 7 * version 2 of the License, or (at your option) any later version.
rlm@0 8 *
rlm@0 9 * This library is distributed in the hope that it will be useful,
rlm@0 10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
rlm@0 11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
rlm@0 12 * Library General Public License for more details.
rlm@0 13 *
rlm@0 14 * You should have received a copy of the GNU Library General Public
rlm@0 15 * License along with this library; if not, write to the
rlm@0 16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
rlm@0 17 * Boston, MA 02111-1307, USA.
rlm@0 18 * Or go to http://www.gnu.org/copyleft/lgpl.html
rlm@0 19 */
rlm@0 20
rlm@0 21 #include "config.h"
rlm@0 22
rlm@0 23 #include <stdio.h>
rlm@0 24 #include <stdlib.h>
rlm@0 25 #include <string.h>
rlm@0 26
rlm@0 27 #include "alMain.h"
rlm@0 28 #include "AL/al.h"
rlm@0 29 #include "AL/alc.h"
rlm@0 30
rlm@0 31 #include <CoreServices/CoreServices.h>
rlm@0 32 #include <unistd.h>
rlm@0 33 #include <AudioUnit/AudioUnit.h>
rlm@0 34 #include <AudioToolbox/AudioToolbox.h>
rlm@0 35
rlm@0 36
rlm@0 37 typedef struct {
rlm@0 38 AudioUnit audioUnit;
rlm@0 39
rlm@0 40 ALuint frameSize;
rlm@0 41 ALdouble sampleRateRatio; // Ratio of hardware sample rate / requested sample rate
rlm@0 42 AudioStreamBasicDescription format; // This is the OpenAL format as a CoreAudio ASBD
rlm@0 43
rlm@0 44 AudioConverterRef audioConverter; // Sample rate converter if needed
rlm@0 45 AudioBufferList *bufferList; // Buffer for data coming from the input device
rlm@0 46 ALCvoid *resampleBuffer; // Buffer for returned RingBuffer data when resampling
rlm@0 47
rlm@0 48 RingBuffer *ring;
rlm@0 49 } ca_data;
rlm@0 50
rlm@0 51 static const ALCchar ca_device[] = "CoreAudio Default";
rlm@0 52
rlm@0 53
rlm@0 54 static void destroy_buffer_list(AudioBufferList* list)
rlm@0 55 {
rlm@0 56 if(list)
rlm@0 57 {
rlm@0 58 UInt32 i;
rlm@0 59 for(i = 0;i < list->mNumberBuffers;i++)
rlm@0 60 free(list->mBuffers[i].mData);
rlm@0 61 free(list);
rlm@0 62 }
rlm@0 63 }
rlm@0 64
rlm@0 65 static AudioBufferList* allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
rlm@0 66 {
rlm@0 67 AudioBufferList *list;
rlm@0 68
rlm@0 69 list = calloc(1, sizeof(AudioBufferList) + sizeof(AudioBuffer));
rlm@0 70 if(list)
rlm@0 71 {
rlm@0 72 list->mNumberBuffers = 1;
rlm@0 73
rlm@0 74 list->mBuffers[0].mNumberChannels = channelCount;
rlm@0 75 list->mBuffers[0].mDataByteSize = byteSize;
rlm@0 76 list->mBuffers[0].mData = malloc(byteSize);
rlm@0 77 if(list->mBuffers[0].mData == NULL)
rlm@0 78 {
rlm@0 79 free(list);
rlm@0 80 list = NULL;
rlm@0 81 }
rlm@0 82 }
rlm@0 83 return list;
rlm@0 84 }
rlm@0 85
rlm@0 86 static OSStatus ca_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp,
rlm@0 87 UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
rlm@0 88 {
rlm@0 89 ALCdevice *device = (ALCdevice*)inRefCon;
rlm@0 90 ca_data *data = (ca_data*)device->ExtraData;
rlm@0 91
rlm@0 92 aluMixData(device, ioData->mBuffers[0].mData,
rlm@0 93 ioData->mBuffers[0].mDataByteSize / data->frameSize);
rlm@0 94
rlm@0 95 return noErr;
rlm@0 96 }
rlm@0 97
rlm@0 98 static OSStatus ca_capture_conversion_callback(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets,
rlm@0 99 AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void* inUserData)
rlm@0 100 {
rlm@0 101 ALCdevice *device = (ALCdevice*)inUserData;
rlm@0 102 ca_data *data = (ca_data*)device->ExtraData;
rlm@0 103
rlm@0 104 // Read from the ring buffer and store temporarily in a large buffer
rlm@0 105 ReadRingBuffer(data->ring, data->resampleBuffer, (ALsizei)(*ioNumberDataPackets));
rlm@0 106
rlm@0 107 // Set the input data
rlm@0 108 ioData->mNumberBuffers = 1;
rlm@0 109 ioData->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
rlm@0 110 ioData->mBuffers[0].mData = data->resampleBuffer;
rlm@0 111 ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * data->format.mBytesPerFrame;
rlm@0 112
rlm@0 113 return noErr;
rlm@0 114 }
rlm@0 115
rlm@0 116 static OSStatus ca_capture_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags,
rlm@0 117 const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
rlm@0 118 UInt32 inNumberFrames, AudioBufferList *ioData)
rlm@0 119 {
rlm@0 120 ALCdevice *device = (ALCdevice*)inRefCon;
rlm@0 121 ca_data *data = (ca_data*)device->ExtraData;
rlm@0 122 AudioUnitRenderActionFlags flags = 0;
rlm@0 123 OSStatus err;
rlm@0 124
rlm@0 125 // fill the bufferList with data from the input device
rlm@0 126 err = AudioUnitRender(data->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, data->bufferList);
rlm@0 127 if(err != noErr)
rlm@0 128 {
rlm@0 129 ERR("AudioUnitRender error: %d\n", err);
rlm@0 130 return err;
rlm@0 131 }
rlm@0 132
rlm@0 133 WriteRingBuffer(data->ring, data->bufferList->mBuffers[0].mData, inNumberFrames);
rlm@0 134
rlm@0 135 return noErr;
rlm@0 136 }
rlm@0 137
rlm@0 138 static ALCboolean ca_open_playback(ALCdevice *device, const ALCchar *deviceName)
rlm@0 139 {
rlm@0 140 ComponentDescription desc;
rlm@0 141 Component comp;
rlm@0 142 ca_data *data;
rlm@0 143 OSStatus err;
rlm@0 144
rlm@0 145 if(!deviceName)
rlm@0 146 deviceName = ca_device;
rlm@0 147 else if(strcmp(deviceName, ca_device) != 0)
rlm@0 148 return ALC_FALSE;
rlm@0 149
rlm@0 150 /* open the default output unit */
rlm@0 151 desc.componentType = kAudioUnitType_Output;
rlm@0 152 desc.componentSubType = kAudioUnitSubType_DefaultOutput;
rlm@0 153 desc.componentManufacturer = kAudioUnitManufacturer_Apple;
rlm@0 154 desc.componentFlags = 0;
rlm@0 155 desc.componentFlagsMask = 0;
rlm@0 156
rlm@0 157 comp = FindNextComponent(NULL, &desc);
rlm@0 158 if(comp == NULL)
rlm@0 159 {
rlm@0 160 ERR("FindNextComponent failed\n");
rlm@0 161 return ALC_FALSE;
rlm@0 162 }
rlm@0 163
rlm@0 164 data = calloc(1, sizeof(*data));
rlm@0 165 device->ExtraData = data;
rlm@0 166
rlm@0 167 err = OpenAComponent(comp, &data->audioUnit);
rlm@0 168 if(err != noErr)
rlm@0 169 {
rlm@0 170 ERR("OpenAComponent failed\n");
rlm@0 171 free(data);
rlm@0 172 device->ExtraData = NULL;
rlm@0 173 return ALC_FALSE;
rlm@0 174 }
rlm@0 175
rlm@0 176 return ALC_TRUE;
rlm@0 177 }
rlm@0 178
rlm@0 179 static void ca_close_playback(ALCdevice *device)
rlm@0 180 {
rlm@0 181 ca_data *data = (ca_data*)device->ExtraData;
rlm@0 182
rlm@0 183 CloseComponent(data->audioUnit);
rlm@0 184
rlm@0 185 free(data);
rlm@0 186 device->ExtraData = NULL;
rlm@0 187 }
rlm@0 188
rlm@0 189 static ALCboolean ca_reset_playback(ALCdevice *device)
rlm@0 190 {
rlm@0 191 ca_data *data = (ca_data*)device->ExtraData;
rlm@0 192 AudioStreamBasicDescription streamFormat;
rlm@0 193 AURenderCallbackStruct input;
rlm@0 194 OSStatus err;
rlm@0 195 UInt32 size;
rlm@0 196
rlm@0 197 /* init and start the default audio unit... */
rlm@0 198 err = AudioUnitInitialize(data->audioUnit);
rlm@0 199 if(err != noErr)
rlm@0 200 {
rlm@0 201 ERR("AudioUnitInitialize failed\n");
rlm@0 202 return ALC_FALSE;
rlm@0 203 }
rlm@0 204
rlm@0 205 err = AudioOutputUnitStart(data->audioUnit);
rlm@0 206 if(err != noErr)
rlm@0 207 {
rlm@0 208 ERR("AudioOutputUnitStart failed\n");
rlm@0 209 return ALC_FALSE;
rlm@0 210 }
rlm@0 211
rlm@0 212 /* retrieve default output unit's properties (output side) */
rlm@0 213 size = sizeof(AudioStreamBasicDescription);
rlm@0 214 err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
rlm@0 215 if(err != noErr || size != sizeof(AudioStreamBasicDescription))
rlm@0 216 {
rlm@0 217 ERR("AudioUnitGetProperty failed\n");
rlm@0 218 return ALC_FALSE;
rlm@0 219 }
rlm@0 220
rlm@0 221 #if 0
rlm@0 222 TRACE("Output streamFormat of default output unit -\n");
rlm@0 223 TRACE(" streamFormat.mFramesPerPacket = %d\n", streamFormat.mFramesPerPacket);
rlm@0 224 TRACE(" streamFormat.mChannelsPerFrame = %d\n", streamFormat.mChannelsPerFrame);
rlm@0 225 TRACE(" streamFormat.mBitsPerChannel = %d\n", streamFormat.mBitsPerChannel);
rlm@0 226 TRACE(" streamFormat.mBytesPerPacket = %d\n", streamFormat.mBytesPerPacket);
rlm@0 227 TRACE(" streamFormat.mBytesPerFrame = %d\n", streamFormat.mBytesPerFrame);
rlm@0 228 TRACE(" streamFormat.mSampleRate = %5.0f\n", streamFormat.mSampleRate);
rlm@0 229 #endif
rlm@0 230
rlm@0 231 /* set default output unit's input side to match output side */
rlm@0 232 err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
rlm@0 233 if(err != noErr)
rlm@0 234 {
rlm@0 235 ERR("AudioUnitSetProperty failed\n");
rlm@0 236 return ALC_FALSE;
rlm@0 237 }
rlm@0 238
rlm@0 239 if(device->Frequency != streamFormat.mSampleRate)
rlm@0 240 {
rlm@0 241 if((device->Flags&DEVICE_FREQUENCY_REQUEST))
rlm@0 242 ERR("CoreAudio does not support changing sample rates (wanted %dhz, got %dhz)\n", device->Frequency, streamFormat.mSampleRate);
rlm@0 243 device->Flags &= ~DEVICE_FREQUENCY_REQUEST;
rlm@0 244
rlm@0 245 device->UpdateSize = (ALuint)((ALuint64)device->UpdateSize *
rlm@0 246 streamFormat.mSampleRate /
rlm@0 247 device->Frequency);
rlm@0 248 device->Frequency = streamFormat.mSampleRate;
rlm@0 249 }
rlm@0 250
rlm@0 251 /* FIXME: How to tell what channels are what in the output device, and how
rlm@0 252 * to specify what we're giving? eg, 6.0 vs 5.1 */
rlm@0 253 switch(streamFormat.mChannelsPerFrame)
rlm@0 254 {
rlm@0 255 case 1:
rlm@0 256 if((device->Flags&DEVICE_CHANNELS_REQUEST) &&
rlm@0 257 device->FmtChans != DevFmtMono)
rlm@0 258 {
rlm@0 259 ERR("Failed to set %s, got Mono instead\n", DevFmtChannelsString(device->FmtChans));
rlm@0 260 device->Flags &= ~DEVICE_CHANNELS_REQUEST;
rlm@0 261 }
rlm@0 262 device->FmtChans = DevFmtMono;
rlm@0 263 break;
rlm@0 264 case 2:
rlm@0 265 if((device->Flags&DEVICE_CHANNELS_REQUEST) &&
rlm@0 266 device->FmtChans != DevFmtStereo)
rlm@0 267 {
rlm@0 268 ERR("Failed to set %s, got Stereo instead\n", DevFmtChannelsString(device->FmtChans));
rlm@0 269 device->Flags &= ~DEVICE_CHANNELS_REQUEST;
rlm@0 270 }
rlm@0 271 device->FmtChans = DevFmtStereo;
rlm@0 272 break;
rlm@0 273 case 4:
rlm@0 274 if((device->Flags&DEVICE_CHANNELS_REQUEST) &&
rlm@0 275 device->FmtChans != DevFmtQuad)
rlm@0 276 {
rlm@0 277 ERR("Failed to set %s, got Quad instead\n", DevFmtChannelsString(device->FmtChans));
rlm@0 278 device->Flags &= ~DEVICE_CHANNELS_REQUEST;
rlm@0 279 }
rlm@0 280 device->FmtChans = DevFmtQuad;
rlm@0 281 break;
rlm@0 282 case 6:
rlm@0 283 if((device->Flags&DEVICE_CHANNELS_REQUEST) &&
rlm@0 284 device->FmtChans != DevFmtX51)
rlm@0 285 {
rlm@0 286 ERR("Failed to set %s, got 5.1 Surround instead\n", DevFmtChannelsString(device->FmtChans));
rlm@0 287 device->Flags &= ~DEVICE_CHANNELS_REQUEST;
rlm@0 288 }
rlm@0 289 device->FmtChans = DevFmtX51;
rlm@0 290 break;
rlm@0 291 case 7:
rlm@0 292 if((device->Flags&DEVICE_CHANNELS_REQUEST) &&
rlm@0 293 device->FmtChans != DevFmtX61)
rlm@0 294 {
rlm@0 295 ERR("Failed to set %s, got 6.1 Surround instead\n", DevFmtChannelsString(device->FmtChans));
rlm@0 296 device->Flags &= ~DEVICE_CHANNELS_REQUEST;
rlm@0 297 }
rlm@0 298 device->FmtChans = DevFmtX61;
rlm@0 299 break;
rlm@0 300 case 8:
rlm@0 301 if((device->Flags&DEVICE_CHANNELS_REQUEST) &&
rlm@0 302 device->FmtChans != DevFmtX71)
rlm@0 303 {
rlm@0 304 ERR("Failed to set %s, got 7.1 Surround instead\n", DevFmtChannelsString(device->FmtChans));
rlm@0 305 device->Flags &= ~DEVICE_CHANNELS_REQUEST;
rlm@0 306 }
rlm@0 307 device->FmtChans = DevFmtX71;
rlm@0 308 break;
rlm@0 309 default:
rlm@0 310 ERR("Unhandled channel count (%d), using Stereo\n", streamFormat.mChannelsPerFrame);
rlm@0 311 device->Flags &= ~DEVICE_CHANNELS_REQUEST;
rlm@0 312 device->FmtChans = DevFmtStereo;
rlm@0 313 streamFormat.mChannelsPerFrame = 2;
rlm@0 314 break;
rlm@0 315 }
rlm@0 316 SetDefaultWFXChannelOrder(device);
rlm@0 317
rlm@0 318 /* use channel count and sample rate from the default output unit's current
rlm@0 319 * parameters, but reset everything else */
rlm@0 320 streamFormat.mFramesPerPacket = 1;
rlm@0 321 switch(device->FmtType)
rlm@0 322 {
rlm@0 323 case DevFmtUByte:
rlm@0 324 device->FmtType = DevFmtByte;
rlm@0 325 /* fall-through */
rlm@0 326 case DevFmtByte:
rlm@0 327 streamFormat.mBitsPerChannel = 8;
rlm@0 328 streamFormat.mBytesPerPacket = streamFormat.mChannelsPerFrame;
rlm@0 329 streamFormat.mBytesPerFrame = streamFormat.mChannelsPerFrame;
rlm@0 330 break;
rlm@0 331 case DevFmtUShort:
rlm@0 332 case DevFmtFloat:
rlm@0 333 device->FmtType = DevFmtShort;
rlm@0 334 /* fall-through */
rlm@0 335 case DevFmtShort:
rlm@0 336 streamFormat.mBitsPerChannel = 16;
rlm@0 337 streamFormat.mBytesPerPacket = 2 * streamFormat.mChannelsPerFrame;
rlm@0 338 streamFormat.mBytesPerFrame = 2 * streamFormat.mChannelsPerFrame;
rlm@0 339 break;
rlm@0 340 }
rlm@0 341 streamFormat.mFormatID = kAudioFormatLinearPCM;
rlm@0 342 streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger |
rlm@0 343 kAudioFormatFlagsNativeEndian |
rlm@0 344 kLinearPCMFormatFlagIsPacked;
rlm@0 345
rlm@0 346 err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
rlm@0 347 if(err != noErr)
rlm@0 348 {
rlm@0 349 ERR("AudioUnitSetProperty failed\n");
rlm@0 350 return ALC_FALSE;
rlm@0 351 }
rlm@0 352
rlm@0 353 /* setup callback */
rlm@0 354 data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
rlm@0 355 input.inputProc = ca_callback;
rlm@0 356 input.inputProcRefCon = device;
rlm@0 357
rlm@0 358 err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
rlm@0 359 if(err != noErr)
rlm@0 360 {
rlm@0 361 ERR("AudioUnitSetProperty failed\n");
rlm@0 362 return ALC_FALSE;
rlm@0 363 }
rlm@0 364
rlm@0 365 return ALC_TRUE;
rlm@0 366 }
rlm@0 367
rlm@0 368 static void ca_stop_playback(ALCdevice *device)
rlm@0 369 {
rlm@0 370 ca_data *data = (ca_data*)device->ExtraData;
rlm@0 371 OSStatus err;
rlm@0 372
rlm@0 373 AudioOutputUnitStop(data->audioUnit);
rlm@0 374 err = AudioUnitUninitialize(data->audioUnit);
rlm@0 375 if(err != noErr)
rlm@0 376 ERR("-- AudioUnitUninitialize failed.\n");
rlm@0 377 }
rlm@0 378
rlm@0 379 static ALCboolean ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
rlm@0 380 {
rlm@0 381 AudioStreamBasicDescription requestedFormat; // The application requested format
rlm@0 382 AudioStreamBasicDescription hardwareFormat; // The hardware format
rlm@0 383 AudioStreamBasicDescription outputFormat; // The AudioUnit output format
rlm@0 384 AURenderCallbackStruct input;
rlm@0 385 ComponentDescription desc;
rlm@0 386 AudioDeviceID inputDevice;
rlm@0 387 UInt32 outputFrameCount;
rlm@0 388 UInt32 propertySize;
rlm@0 389 UInt32 enableIO;
rlm@0 390 Component comp;
rlm@0 391 ca_data *data;
rlm@0 392 OSStatus err;
rlm@0 393
rlm@0 394 desc.componentType = kAudioUnitType_Output;
rlm@0 395 desc.componentSubType = kAudioUnitSubType_HALOutput;
rlm@0 396 desc.componentManufacturer = kAudioUnitManufacturer_Apple;
rlm@0 397 desc.componentFlags = 0;
rlm@0 398 desc.componentFlagsMask = 0;
rlm@0 399
rlm@0 400 // Search for component with given description
rlm@0 401 comp = FindNextComponent(NULL, &desc);
rlm@0 402 if(comp == NULL)
rlm@0 403 {
rlm@0 404 ERR("FindNextComponent failed\n");
rlm@0 405 return ALC_FALSE;
rlm@0 406 }
rlm@0 407
rlm@0 408 data = calloc(1, sizeof(*data));
rlm@0 409 device->ExtraData = data;
rlm@0 410
rlm@0 411 // Open the component
rlm@0 412 err = OpenAComponent(comp, &data->audioUnit);
rlm@0 413 if(err != noErr)
rlm@0 414 {
rlm@0 415 ERR("OpenAComponent failed\n");
rlm@0 416 goto error;
rlm@0 417 }
rlm@0 418
rlm@0 419 // Turn off AudioUnit output
rlm@0 420 enableIO = 0;
rlm@0 421 err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
rlm@0 422 if(err != noErr)
rlm@0 423 {
rlm@0 424 ERR("AudioUnitSetProperty failed\n");
rlm@0 425 goto error;
rlm@0 426 }
rlm@0 427
rlm@0 428 // Turn on AudioUnit input
rlm@0 429 enableIO = 1;
rlm@0 430 err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
rlm@0 431 if(err != noErr)
rlm@0 432 {
rlm@0 433 ERR("AudioUnitSetProperty failed\n");
rlm@0 434 goto error;
rlm@0 435 }
rlm@0 436
rlm@0 437 // Get the default input device
rlm@0 438 propertySize = sizeof(AudioDeviceID);
rlm@0 439 err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propertySize, &inputDevice);
rlm@0 440 if(err != noErr)
rlm@0 441 {
rlm@0 442 ERR("AudioHardwareGetProperty failed\n");
rlm@0 443 goto error;
rlm@0 444 }
rlm@0 445
rlm@0 446 if(inputDevice == kAudioDeviceUnknown)
rlm@0 447 {
rlm@0 448 ERR("No input device found\n");
rlm@0 449 goto error;
rlm@0 450 }
rlm@0 451
rlm@0 452 // Track the input device
rlm@0 453 err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
rlm@0 454 if(err != noErr)
rlm@0 455 {
rlm@0 456 ERR("AudioUnitSetProperty failed\n");
rlm@0 457 goto error;
rlm@0 458 }
rlm@0 459
rlm@0 460 // set capture callback
rlm@0 461 input.inputProc = ca_capture_callback;
rlm@0 462 input.inputProcRefCon = device;
rlm@0 463
rlm@0 464 err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
rlm@0 465 if(err != noErr)
rlm@0 466 {
rlm@0 467 ERR("AudioUnitSetProperty failed\n");
rlm@0 468 goto error;
rlm@0 469 }
rlm@0 470
rlm@0 471 // Initialize the device
rlm@0 472 err = AudioUnitInitialize(data->audioUnit);
rlm@0 473 if(err != noErr)
rlm@0 474 {
rlm@0 475 ERR("AudioUnitInitialize failed\n");
rlm@0 476 goto error;
rlm@0 477 }
rlm@0 478
rlm@0 479 // Get the hardware format
rlm@0 480 propertySize = sizeof(AudioStreamBasicDescription);
rlm@0 481 err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
rlm@0 482 if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
rlm@0 483 {
rlm@0 484 ERR("AudioUnitGetProperty failed\n");
rlm@0 485 goto error;
rlm@0 486 }
rlm@0 487
rlm@0 488 // Set up the requested format description
rlm@0 489 switch(device->FmtType)
rlm@0 490 {
rlm@0 491 case DevFmtUByte:
rlm@0 492 requestedFormat.mBitsPerChannel = 8;
rlm@0 493 requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
rlm@0 494 break;
rlm@0 495 case DevFmtShort:
rlm@0 496 requestedFormat.mBitsPerChannel = 16;
rlm@0 497 requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
rlm@0 498 break;
rlm@0 499 case DevFmtFloat:
rlm@0 500 requestedFormat.mBitsPerChannel = 32;
rlm@0 501 requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
rlm@0 502 break;
rlm@0 503 case DevFmtByte:
rlm@0 504 case DevFmtUShort:
rlm@0 505 ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
rlm@0 506 goto error;
rlm@0 507 }
rlm@0 508
rlm@0 509 switch(device->FmtChans)
rlm@0 510 {
rlm@0 511 case DevFmtMono:
rlm@0 512 requestedFormat.mChannelsPerFrame = 1;
rlm@0 513 break;
rlm@0 514 case DevFmtStereo:
rlm@0 515 requestedFormat.mChannelsPerFrame = 2;
rlm@0 516 break;
rlm@0 517
rlm@0 518 case DevFmtQuad:
rlm@0 519 case DevFmtX51:
rlm@0 520 case DevFmtX51Side:
rlm@0 521 case DevFmtX61:
rlm@0 522 case DevFmtX71:
rlm@0 523 ERR("%s not supported\n", DevFmtChannelsString(device->FmtChans));
rlm@0 524 goto error;
rlm@0 525 }
rlm@0 526
rlm@0 527 requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8;
rlm@0 528 requestedFormat.mBytesPerPacket = requestedFormat.mBytesPerFrame;
rlm@0 529 requestedFormat.mSampleRate = device->Frequency;
rlm@0 530 requestedFormat.mFormatID = kAudioFormatLinearPCM;
rlm@0 531 requestedFormat.mReserved = 0;
rlm@0 532 requestedFormat.mFramesPerPacket = 1;
rlm@0 533
rlm@0 534 // save requested format description for later use
rlm@0 535 data->format = requestedFormat;
rlm@0 536 data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
rlm@0 537
rlm@0 538 // Use intermediate format for sample rate conversion (outputFormat)
rlm@0 539 // Set sample rate to the same as hardware for resampling later
rlm@0 540 outputFormat = requestedFormat;
rlm@0 541 outputFormat.mSampleRate = hardwareFormat.mSampleRate;
rlm@0 542
rlm@0 543 // Determine sample rate ratio for resampling
rlm@0 544 data->sampleRateRatio = outputFormat.mSampleRate / device->Frequency;
rlm@0 545
rlm@0 546 // The output format should be the requested format, but using the hardware sample rate
rlm@0 547 // This is because the AudioUnit will automatically scale other properties, except for sample rate
rlm@0 548 err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
rlm@0 549 if(err != noErr)
rlm@0 550 {
rlm@0 551 ERR("AudioUnitSetProperty failed\n");
rlm@0 552 goto error;
rlm@0 553 }
rlm@0 554
rlm@0 555 // Set the AudioUnit output format frame count
rlm@0 556 outputFrameCount = device->UpdateSize * data->sampleRateRatio;
rlm@0 557 err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
rlm@0 558 if(err != noErr)
rlm@0 559 {
rlm@0 560 ERR("AudioUnitSetProperty failed: %d\n", err);
rlm@0 561 goto error;
rlm@0 562 }
rlm@0 563
rlm@0 564 // Set up sample converter
rlm@0 565 err = AudioConverterNew(&outputFormat, &requestedFormat, &data->audioConverter);
rlm@0 566 if(err != noErr)
rlm@0 567 {
rlm@0 568 ERR("AudioConverterNew failed: %d\n", err);
rlm@0 569 goto error;
rlm@0 570 }
rlm@0 571
rlm@0 572 // Create a buffer for use in the resample callback
rlm@0 573 data->resampleBuffer = malloc(device->UpdateSize * data->frameSize * data->sampleRateRatio);
rlm@0 574
rlm@0 575 // Allocate buffer for the AudioUnit output
rlm@0 576 data->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * data->frameSize * data->sampleRateRatio);
rlm@0 577 if(data->bufferList == NULL)
rlm@0 578 {
rlm@0 579 alcSetError(device, ALC_OUT_OF_MEMORY);
rlm@0 580 goto error;
rlm@0 581 }
rlm@0 582
rlm@0 583 data->ring = CreateRingBuffer(data->frameSize, (device->UpdateSize * data->sampleRateRatio) * device->NumUpdates);
rlm@0 584 if(data->ring == NULL)
rlm@0 585 {
rlm@0 586 alcSetError(device, ALC_OUT_OF_MEMORY);
rlm@0 587 goto error;
rlm@0 588 }
rlm@0 589
rlm@0 590 return ALC_TRUE;
rlm@0 591
rlm@0 592 error:
rlm@0 593 DestroyRingBuffer(data->ring);
rlm@0 594 free(data->resampleBuffer);
rlm@0 595 destroy_buffer_list(data->bufferList);
rlm@0 596
rlm@0 597 if(data->audioConverter)
rlm@0 598 AudioConverterDispose(data->audioConverter);
rlm@0 599 if(data->audioUnit)
rlm@0 600 CloseComponent(data->audioUnit);
rlm@0 601
rlm@0 602 free(data);
rlm@0 603 device->ExtraData = NULL;
rlm@0 604
rlm@0 605 return ALC_FALSE;
rlm@0 606 }
rlm@0 607
rlm@0 608 static void ca_close_capture(ALCdevice *device)
rlm@0 609 {
rlm@0 610 ca_data *data = (ca_data*)device->ExtraData;
rlm@0 611
rlm@0 612 DestroyRingBuffer(data->ring);
rlm@0 613 free(data->resampleBuffer);
rlm@0 614 destroy_buffer_list(data->bufferList);
rlm@0 615
rlm@0 616 AudioConverterDispose(data->audioConverter);
rlm@0 617 CloseComponent(data->audioUnit);
rlm@0 618
rlm@0 619 free(data);
rlm@0 620 device->ExtraData = NULL;
rlm@0 621 }
rlm@0 622
rlm@0 623 static void ca_start_capture(ALCdevice *device)
rlm@0 624 {
rlm@0 625 ca_data *data = (ca_data*)device->ExtraData;
rlm@0 626 OSStatus err = AudioOutputUnitStart(data->audioUnit);
rlm@0 627 if(err != noErr)
rlm@0 628 ERR("AudioOutputUnitStart failed\n");
rlm@0 629 }
rlm@0 630
rlm@0 631 static void ca_stop_capture(ALCdevice *device)
rlm@0 632 {
rlm@0 633 ca_data *data = (ca_data*)device->ExtraData;
rlm@0 634 OSStatus err = AudioOutputUnitStop(data->audioUnit);
rlm@0 635 if(err != noErr)
rlm@0 636 ERR("AudioOutputUnitStop failed\n");
rlm@0 637 }
rlm@0 638
rlm@0 639 static ALCuint ca_available_samples(ALCdevice *device)
rlm@0 640 {
rlm@0 641 ca_data *data = device->ExtraData;
rlm@0 642 return RingBufferSize(data->ring) / data->sampleRateRatio;
rlm@0 643 }
rlm@0 644
rlm@0 645 static void ca_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples)
rlm@0 646 {
rlm@0 647 ca_data *data = (ca_data*)device->ExtraData;
rlm@0 648
rlm@0 649 if(samples <= ca_available_samples(device))
rlm@0 650 {
rlm@0 651 AudioBufferList *list;
rlm@0 652 UInt32 frameCount;
rlm@0 653 OSStatus err;
rlm@0 654
rlm@0 655 // If no samples are requested, just return
rlm@0 656 if(samples == 0)
rlm@0 657 return;
rlm@0 658
rlm@0 659 // Allocate a temporary AudioBufferList to use as the return resamples data
rlm@0 660 list = alloca(sizeof(AudioBufferList) + sizeof(AudioBuffer));
rlm@0 661
rlm@0 662 // Point the resampling buffer to the capture buffer
rlm@0 663 list->mNumberBuffers = 1;
rlm@0 664 list->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
rlm@0 665 list->mBuffers[0].mDataByteSize = samples * data->frameSize;
rlm@0 666 list->mBuffers[0].mData = buffer;
rlm@0 667
rlm@0 668 // Resample into another AudioBufferList
rlm@0 669 frameCount = samples;
rlm@0 670 err = AudioConverterFillComplexBuffer(data->audioConverter, ca_capture_conversion_callback, device,
rlm@0 671 &frameCount, list, NULL);
rlm@0 672 if(err != noErr)
rlm@0 673 {
rlm@0 674 ERR("AudioConverterFillComplexBuffer error: %d\n", err);
rlm@0 675 alcSetError(device, ALC_INVALID_VALUE);
rlm@0 676 }
rlm@0 677 }
rlm@0 678 else
rlm@0 679 alcSetError(device, ALC_INVALID_VALUE);
rlm@0 680 }
rlm@0 681
rlm@0 682 static const BackendFuncs ca_funcs = {
rlm@0 683 ca_open_playback,
rlm@0 684 ca_close_playback,
rlm@0 685 ca_reset_playback,
rlm@0 686 ca_stop_playback,
rlm@0 687 ca_open_capture,
rlm@0 688 ca_close_capture,
rlm@0 689 ca_start_capture,
rlm@0 690 ca_stop_capture,
rlm@0 691 ca_capture_samples,
rlm@0 692 ca_available_samples
rlm@0 693 };
rlm@0 694
rlm@0 695 ALCboolean alc_ca_init(BackendFuncs *func_list)
rlm@0 696 {
rlm@0 697 *func_list = ca_funcs;
rlm@0 698 return ALC_TRUE;
rlm@0 699 }
rlm@0 700
rlm@0 701 void alc_ca_deinit(void)
rlm@0 702 {
rlm@0 703 }
rlm@0 704
rlm@0 705 void alc_ca_probe(enum DevProbe type)
rlm@0 706 {
rlm@0 707 switch(type)
rlm@0 708 {
rlm@0 709 case DEVICE_PROBE:
rlm@0 710 AppendDeviceList(ca_device);
rlm@0 711 break;
rlm@0 712 case ALL_DEVICE_PROBE:
rlm@0 713 AppendAllDeviceList(ca_device);
rlm@0 714 break;
rlm@0 715 case CAPTURE_DEVICE_PROBE:
rlm@0 716 AppendCaptureDeviceList(ca_device);
rlm@0 717 break;
rlm@0 718 }
rlm@0 719 }