Alc/backends/coreaudio.c

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