libao2/ao_macosx.c

   1 /*
   2  *
   3  *  ao_macosx.c
   4  *
   5  *      Original Copyright (C) Timothy J. Wood - Aug 2000
   6  *
   7  *      The S/PDIF part of the code is based on the auhal audio output
   8  *      module from VideoLAN:
   9  *      Copyright (c) 2006 Derk-Jan Hartman <hartman at videolan dot org>
  10  *
  11  *  This file is part of libao, a cross-platform library.  See
  12  *  README for a history of this source code.
  13  *
  14  *  libao is free software; you can redistribute it and/or modify
  15  *  it under the terms of the GNU General Public License as published by
  16  *  the Free Software Foundation; either version 2, or (at your option)
  17  *  any later version.
  18  *
  19  *  libao is distributed in the hope that it will be useful,
  20  *  but WITHOUT ANY WARRANTY; without even the implied warranty of
  21  *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  22  *  GNU General Public License for more details.
  23  *
  24  *  You should have received a copy of the GNU General Public License along
  25  *  with libao; if not, write to the Free Software Foundation, Inc.,
  26  *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  27  */
  28
  29 /*
  30  * The MacOS X CoreAudio framework doesn't mesh as simply as some
  31  * simpler frameworks do.  This is due to the fact that CoreAudio pulls
  32  * audio samples rather than having them pushed at it (which is nice
  33  * when you are wanting to do good buffering of audio).
  34  */
  35
  36 /* Change log:
  37  *
  38  * 14/5-2003: Ported to MPlayer libao2 by Dan Christiansen
  39  *
  40  *            AC-3 and MPEG audio passthrough is possible, but I don't have
  41  *            access to a sound card that supports it.
  42  */
  43
  44 #include <CoreServices/CoreServices.h>
  45 #include <AudioUnit/AudioUnit.h>
  46 #include <AudioToolbox/AudioToolbox.h>
  47 #include <stdio.h>
  48 #include <string.h>
  49 #include <stdlib.h>
  50 #include <inttypes.h>
  51 #include <sys/types.h>
  52 #include <unistd.h>
  53
  54 #include "config.h"
  55 #include "mp_msg.h"
  56
  57 #include "audio_out.h"
  58 #include "audio_out_internal.h"
  59 #include "libaf/af_format.h"
  60 #include "osdep/timer.h"
  61
  62 static ao_info_t info =
  63   {
  64     "Darwin/Mac OS X native audio output",
  65     "macosx",
  66     "Timothy J. Wood & Dan Christiansen & Chris Roccati",
  67     ""
  68   };
  69
  70 LIBAO_EXTERN(macosx)
  71
  72 /* Prefix for all mp_msg() calls */
  73 #define ao_msg(a, b, c...) mp_msg(a, b, "AO: [macosx] " c)
  74
  75 typedef struct ao_macosx_s
  76 {
  77   AudioDeviceID i_selected_dev;             /* Keeps DeviceID of the selected device. */
  78   int b_supports_digital;                   /* Does the currently selected device support digital mode? */
  79   int b_digital;                            /* Are we running in digital mode? */
  80   int b_muted;                              /* Are we muted in digital mode? */
  81
  82   /* AudioUnit */
  83   AudioUnit theOutputUnit;
  84
  85   /* CoreAudio SPDIF mode specific */
  86   pid_t i_hog_pid;                          /* Keeps the pid of our hog status. */
  87   AudioStreamID i_stream_id;                /* The StreamID that has a cac3 streamformat */
  88   int i_stream_index;                       /* The index of i_stream_id in an AudioBufferList */
  89   AudioStreamBasicDescription stream_format;/* The format we changed the stream to */
  90   AudioStreamBasicDescription sfmt_revert;  /* The original format of the stream */
  91   int b_revert;                             /* Whether we need to revert the stream format */
  92   int b_changed_mixing;                     /* Whether we need to set the mixing mode back */
  93   int b_stream_format_changed;              /* Flag for main thread to reset stream's format to digital and reset buffer */
  94
  95   /* Original common part */
  96   int packetSize;
  97   int paused;
  98
  99   /* Ring-buffer */
 100   /* does not need explicit synchronization, but needs to allocate
 101    * (num_chunks + 1) * chunk_size memory to store num_chunks * chunk_size
 102    * data */
 103   unsigned char *buffer;
 104   unsigned int buffer_len; ///< must always be (num_chunks + 1) * chunk_size
 105   unsigned int num_chunks;
 106   unsigned int chunk_size;
 107
 108   unsigned int buf_read_pos;
 109   unsigned int buf_write_pos;
 110 } ao_macosx_t;
 111
 112 static ao_macosx_t *ao = NULL;
 113
 114 /**
 115  * \brief return number of free bytes in the buffer
 116  *    may only be called by mplayer's thread
 117  * \return minimum number of free bytes in buffer, value may change between
 118  *    two immediately following calls, and the real number of free bytes
 119  *    might actually be larger!
 120  */
 121 static int buf_free(void) {
 122   int free = ao->buf_read_pos - ao->buf_write_pos - ao->chunk_size;
 123   if (free < 0) free += ao->buffer_len;
 124   return free;
 125 }
 126
 127 /**
 128  * \brief return number of buffered bytes
 129  *    may only be called by playback thread
 130  * \return minimum number of buffered bytes, value may change between
 131  *    two immediately following calls, and the real number of buffered bytes
 132  *    might actually be larger!
 133  */
 134 static int buf_used(void) {
 135   int used = ao->buf_write_pos - ao->buf_read_pos;
 136   if (used < 0) used += ao->buffer_len;
 137   return used;
 138 }
 139
 140 /**
 141  * \brief add data to ringbuffer
 142  */
 143 static int write_buffer(unsigned char* data, int len){
 144   int first_len = ao->buffer_len - ao->buf_write_pos;
 145   int free = buf_free();
 146   if (len > free) len = free;
 147   if (first_len > len) first_len = len;
 148   // till end of buffer
 149   memcpy (&ao->buffer[ao->buf_write_pos], data, first_len);
 150   if (len > first_len) { // we have to wrap around
 151     // remaining part from beginning of buffer
 152     memcpy (ao->buffer, &data[first_len], len - first_len);
 153   }
 154   ao->buf_write_pos = (ao->buf_write_pos + len) % ao->buffer_len;
 155   return len;
 156 }
 157
 158 /**
 159  * \brief remove data from ringbuffer
 160  */
 161 static int read_buffer(unsigned char* data,int len){
 162   int first_len = ao->buffer_len - ao->buf_read_pos;
 163   int buffered = buf_used();
 164   if (len > buffered) len = buffered;
 165   if (first_len > len) first_len = len;
 166   // till end of buffer
 167   if (data) {
 168     memcpy (data, &ao->buffer[ao->buf_read_pos], first_len);
 169     if (len > first_len) { // we have to wrap around
 170       // remaining part from beginning of buffer
 171       memcpy (&data[first_len], ao->buffer, len - first_len);
 172     }
 173   }
 174   ao->buf_read_pos = (ao->buf_read_pos + len) % ao->buffer_len;
 175   return len;
 176 }
 177
 178 OSStatus theRenderProc(void *inRefCon, AudioUnitRenderActionFlags *inActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumFrames, AudioBufferList *ioData)
 179 {
 180 int amt=buf_used();
 181 int req=(inNumFrames)*ao->packetSize;
 182
 183         if(amt>req)
 184                 amt=req;
 185
 186         if(amt)
 187                 read_buffer((unsigned char *)ioData->mBuffers[0].mData, amt);
 188         else audio_pause();
 189         ioData->mBuffers[0].mDataByteSize = amt;
 190
 191         return noErr;
 192 }
 193
 194 static int control(int cmd,void *arg){
 195 ao_control_vol_t *control_vol;
 196 OSStatus err;
 197 Float32 vol;
 198         switch (cmd) {
 199         case AOCONTROL_GET_VOLUME:
 200                 control_vol = (ao_control_vol_t*)arg;
 201                 if (ao->b_digital) {
 202                         // Digital output has no volume adjust.
 203                         return CONTROL_FALSE;
 204                 }
 205                 err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol);
 206
 207                 if(err==0) {
 208                         // printf("GET VOL=%f\n", vol);
 209                         control_vol->left=control_vol->right=vol*100.0/4.0;
 210                         return CONTROL_TRUE;
 211                 }
 212                 else {
 213                         ao_msg(MSGT_AO, MSGL_WARN, "could not get HAL output volume: [%4.4s]\n", (char *)&err);
 214                         return CONTROL_FALSE;
 215                 }
 216
 217         case AOCONTROL_SET_VOLUME:
 218                 control_vol = (ao_control_vol_t*)arg;
 219
 220                 if (ao->b_digital) {
 221                         // Digital output can not set volume. Here we have to return true
 222                         // to make mixer forget it. Else mixer will add a soft filter,
 223                         // that's not we expected and the filter not support ac3 stream
 224                         // will cause mplayer die.
 225
 226                         // Although not support set volume, but at least we support mute.
 227                         // MPlayer set mute by set volume to zero, we handle it.
 228                         if (control_vol->left == 0 && control_vol->right == 0)
 229                                 ao->b_muted = 1;
 230                         else
 231                                 ao->b_muted = 0;
 232                         return CONTROL_TRUE;
 233                 }
 234
 235                 vol=(control_vol->left+control_vol->right)*4.0/200.0;
 236                 err = AudioUnitSetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, vol, 0);
 237                 if(err==0) {
 238                         // printf("SET VOL=%f\n", vol);
 239                         return CONTROL_TRUE;
 240                 }
 241                 else {
 242                         ao_msg(MSGT_AO, MSGL_WARN, "could not set HAL output volume: [%4.4s]\n", (char *)&err);
 243                         return CONTROL_FALSE;
 244                 }
 245           /* Everything is currently unimplemented */
 246         default:
 247           return CONTROL_FALSE;
 248         }
 249
 250 }
 251
 252
 253 static void print_format(int lev, const char* str, const AudioStreamBasicDescription *f){
 254     uint32_t flags=(uint32_t) f->mFormatFlags;
 255     ao_msg(MSGT_AO,lev, "%s %7.1fHz %lubit [%c%c%c%c][%lu][%lu][%lu][%lu][%lu] %s %s %s%s%s%s\n",
 256             str, f->mSampleRate, f->mBitsPerChannel,
 257             (int)(f->mFormatID & 0xff000000) >> 24,
 258             (int)(f->mFormatID & 0x00ff0000) >> 16,
 259             (int)(f->mFormatID & 0x0000ff00) >>  8,
 260             (int)(f->mFormatID & 0x000000ff) >>  0,
 261             f->mFormatFlags, f->mBytesPerPacket,
 262             f->mFramesPerPacket, f->mBytesPerFrame,
 263             f->mChannelsPerFrame,
 264             (flags&kAudioFormatFlagIsFloat) ? "float" : "int",
 265             (flags&kAudioFormatFlagIsBigEndian) ? "BE" : "LE",
 266             (flags&kAudioFormatFlagIsSignedInteger) ? "S" : "U",
 267             (flags&kAudioFormatFlagIsPacked) ? " packed" : "",
 268             (flags&kAudioFormatFlagIsAlignedHigh) ? " aligned" : "",
 269             (flags&kAudioFormatFlagIsNonInterleaved) ? " ni" : "" );
 270 }
 271
 272
 273 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id );
 274 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id );
 275 static int OpenSPDIF();
 276 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format );
 277 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
 278                                     const AudioTimeStamp * inNow,
 279                                     const void * inInputData,
 280                                     const AudioTimeStamp * inInputTime,
 281                                     AudioBufferList * outOutputData,
 282                                     const AudioTimeStamp * inOutputTime,
 283                                     void * threadGlobals );
 284 static OSStatus StreamListener( AudioStreamID inStream,
 285                                 UInt32 inChannel,
 286                                 AudioDevicePropertyID inPropertyID,
 287                                 void * inClientData );
 288 static OSStatus DeviceListener( AudioDeviceID inDevice,
 289                                 UInt32 inChannel,
 290                                 Boolean isInput,
 291                                 AudioDevicePropertyID inPropertyID,
 292                                 void* inClientData );
 293
 294 static int init(int rate,int channels,int format,int flags)
 295 {
 296 AudioStreamBasicDescription inDesc;
 297 ComponentDescription desc;
 298 Component comp;
 299 AURenderCallbackStruct renderCallback;
 300 OSStatus err;
 301 UInt32 size, maxFrames, i_param_size;
 302 char *psz_name;
 303 AudioDeviceID devid_def = 0;
 304 int b_alive;
 305
 306     ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags);
 307
 308     ao = calloc(1, sizeof(ao_macosx_t));
 309
 310     ao->i_selected_dev = 0;
 311     ao->b_supports_digital = 0;
 312     ao->b_digital = 0;
 313     ao->b_muted = 0;
 314     ao->b_stream_format_changed = 0;
 315     ao->i_hog_pid = -1;
 316     ao->i_stream_id = 0;
 317     ao->i_stream_index = -1;
 318     ao->b_revert = 0;
 319     ao->b_changed_mixing = 0;
 320
 321     /* Probe whether device support S/PDIF stream output if input is AC3 stream. */
 322     if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3)
 323     {
 324         /* Find the ID of the default Device. */
 325         i_param_size = sizeof(AudioDeviceID);
 326         err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
 327                                        &i_param_size, &devid_def);
 328         if (err != noErr)
 329         {
 330             ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device: [%4.4s]\n", (char *)&err);
 331             goto err_out;
 332         }
 333
 334         /* Retrieve the length of the device name. */
 335         i_param_size = 0;
 336         err = AudioDeviceGetPropertyInfo(devid_def, 0, 0,
 337                                          kAudioDevicePropertyDeviceName,
 338                                          &i_param_size, NULL);
 339         if (err != noErr)
 340         {
 341             ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name length: [%4.4s]\n", (char *)&err);
 342             goto err_out;
 343         }
 344
 345         /* Retrieve the name of the device. */
 346         psz_name = (char *)malloc(i_param_size);
 347         err = AudioDeviceGetProperty(devid_def, 0, 0,
 348                                      kAudioDevicePropertyDeviceName,
 349                                      &i_param_size, psz_name);
 350         if (err != noErr)
 351         {
 352             ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err);
 353             free( psz_name);
 354             goto err_out;
 355         }
 356
 357         ao_msg(MSGT_AO,MSGL_V, "got default audio output device ID: %#lx Name: %s\n", devid_def, psz_name );
 358
 359         if (AudioDeviceSupportsDigital(devid_def))
 360         {
 361             ao->b_supports_digital = 1;
 362             ao->i_selected_dev = devid_def;
 363         }
 364         ao_msg(MSGT_AO,MSGL_V, "probe default audio output device whether support digital s/pdif output:%d\n", ao->b_supports_digital );
 365
 366         free( psz_name);
 367     }
 368
 369         // Build Description for the input format
 370         inDesc.mSampleRate=rate;
 371         inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
 372         inDesc.mChannelsPerFrame=channels;
 373         switch(format&AF_FORMAT_BITS_MASK){
 374         case AF_FORMAT_8BIT:
 375                 inDesc.mBitsPerChannel=8;
 376                 break;
 377         case AF_FORMAT_16BIT:
 378                 inDesc.mBitsPerChannel=16;
 379                 break;
 380         case AF_FORMAT_24BIT:
 381                 inDesc.mBitsPerChannel=24;
 382                 break;
 383         case AF_FORMAT_32BIT:
 384                 inDesc.mBitsPerChannel=32;
 385                 break;
 386         default:
 387                 ao_msg(MSGT_AO, MSGL_WARN, "Unsupported format (0x%08x)\n", format);
 388                 goto err_out;
 389         }
 390
 391     if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) {
 392         // float
 393                 inDesc.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked;
 394     }
 395     else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI) {
 396         // signed int
 397                 inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
 398     }
 399     else {
 400         // unsigned int
 401                 inDesc.mFormatFlags = kAudioFormatFlagIsPacked;
 402     }
 403     if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3) {
 404         // Currently ac3 input (comes from hwac3) is always in native byte-order.
 405 #ifdef WORDS_BIGENDIAN
 406         inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
 407 #endif
 408     }
 409     else if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
 410         inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
 411
 412     inDesc.mFramesPerPacket = 1;
 413     ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8);
 414     print_format(MSGL_V, "source:",&inDesc);
 415
 416     if (ao->b_supports_digital)
 417     {
 418         b_alive = 1;
 419         i_param_size = sizeof(b_alive);
 420         err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
 421                                      kAudioDevicePropertyDeviceIsAlive,
 422                                      &i_param_size, &b_alive);
 423         if (err != noErr)
 424             ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err);
 425         if (!b_alive)
 426             ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" );
 427         /* S/PDIF output need device in HogMode. */
 428         i_param_size = sizeof(ao->i_hog_pid);
 429         err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
 430                                      kAudioDevicePropertyHogMode,
 431                                      &i_param_size, &ao->i_hog_pid);
 432
 433         if (err != noErr)
 434         {
 435             /* This is not a fatal error. Some drivers simply don't support this property. */
 436             ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n",
 437                      (char *)&err);
 438             ao->i_hog_pid = -1;
 439         }
 440
 441         if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid())
 442         {
 443             ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" );
 444             goto err_out;
 445         }
 446         ao->stream_format = inDesc;
 447         return OpenSPDIF();
 448     }
 449
 450         /* original analog output code */
 451         desc.componentType = kAudioUnitType_Output;
 452         desc.componentSubType = kAudioUnitSubType_DefaultOutput;
 453         desc.componentManufacturer = kAudioUnitManufacturer_Apple;
 454         desc.componentFlags = 0;
 455         desc.componentFlagsMask = 0;
 456
 457         comp = FindNextComponent(NULL, &desc);  //Finds an component that meets the desc spec's
 458         if (comp == NULL) {
 459                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n");
 460                 goto err_out;
 461         }
 462
 463         err = OpenAComponent(comp, &(ao->theOutputUnit));  //gains access to the services provided by the component
 464         if (err) {
 465                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err);
 466                 goto err_out;
 467         }
 468
 469         // Initialize AudioUnit
 470         err = AudioUnitInitialize(ao->theOutputUnit);
 471         if (err) {
 472                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err);
 473                 goto err_out1;
 474         }
 475
 476         size =  sizeof(AudioStreamBasicDescription);
 477         err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size);
 478
 479         if (err) {
 480                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err);
 481                 goto err_out2;
 482         }
 483
 484         size = sizeof(UInt32);
 485         err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size);
 486
 487         if (err)
 488         {
 489                 ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err);
 490                 goto err_out2;
 491         }
 492
 493         ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame;
 494
 495         ao_data.samplerate = inDesc.mSampleRate;
 496         ao_data.channels = inDesc.mChannelsPerFrame;
 497     ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame;
 498     ao_data.outburst = ao->chunk_size;
 499         ao_data.buffersize = ao_data.bps;
 500
 501         ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
 502     ao->buffer_len = (ao->num_chunks + 1) * ao->chunk_size;
 503     ao->buffer = calloc(ao->num_chunks + 1, ao->chunk_size);
 504
 505         ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
 506
 507     renderCallback.inputProc = theRenderProc;
 508     renderCallback.inputProcRefCon = 0;
 509     err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct));
 510         if (err) {
 511                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err);
 512                 goto err_out2;
 513         }
 514
 515         reset();
 516
 517     return CONTROL_OK;
 518
 519 err_out2:
 520     AudioUnitUninitialize(ao->theOutputUnit);
 521 err_out1:
 522     CloseComponent(ao->theOutputUnit);
 523 err_out:
 524     free(ao->buffer);
 525     free(ao);
 526     ao = NULL;
 527     return CONTROL_FALSE;
 528 }
 529
 530 /*****************************************************************************
 531  * Setup a encoded digital stream (SPDIF)
 532  *****************************************************************************/
 533 static int OpenSPDIF()
 534 {
 535     OSStatus                err = noErr;
 536     UInt32                  i_param_size, b_mix = 0;
 537     Boolean                 b_writeable = 0;
 538     AudioStreamID           *p_streams = NULL;
 539     int                     i, i_streams = 0;
 540
 541     /* Start doing the SPDIF setup process. */
 542     ao->b_digital = 1;
 543
 544     /* Hog the device. */
 545     i_param_size = sizeof(ao->i_hog_pid);
 546     ao->i_hog_pid = getpid() ;
 547
 548     err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
 549                                  kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid);
 550
 551     if (err != noErr)
 552     {
 553         ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err);
 554         ao->i_hog_pid = -1;
 555         goto err_out;
 556     }
 557
 558     /* Set mixable to false if we are allowed to. */
 559     err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE,
 560                                      kAudioDevicePropertySupportsMixing,
 561                                      &i_param_size, &b_writeable);
 562     err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
 563                                  kAudioDevicePropertySupportsMixing,
 564                                  &i_param_size, &b_mix);
 565     if (err != noErr && b_writeable)
 566     {
 567         b_mix = 0;
 568         err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
 569                                      kAudioDevicePropertySupportsMixing,
 570                                      i_param_size, &b_mix);
 571         ao->b_changed_mixing = 1;
 572     }
 573     if (err != noErr)
 574     {
 575         ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
 576         goto err_out;
 577     }
 578
 579     /* Get a list of all the streams on this device. */
 580     err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE,
 581                                      kAudioDevicePropertyStreams,
 582                                      &i_param_size, NULL);
 583     if (err != noErr)
 584     {
 585         ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err);
 586         goto err_out;
 587     }
 588
 589     i_streams = i_param_size / sizeof(AudioStreamID);
 590     p_streams = (AudioStreamID *)malloc(i_param_size);
 591     if (p_streams == NULL)
 592     {
 593         ao_msg(MSGT_AO, MSGL_WARN, "out of memory\n" );
 594         goto err_out;
 595     }
 596
 597     err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
 598                                  kAudioDevicePropertyStreams,
 599                                  &i_param_size, p_streams);
 600     if (err != noErr)
 601     {
 602         ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err);
 603         if (p_streams) free(p_streams);
 604         goto err_out;
 605     }
 606
 607     ao_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams);
 608
 609     for (i = 0; i < i_streams && ao->i_stream_index < 0; ++i)
 610     {
 611         /* Find a stream with a cac3 stream. */
 612         AudioStreamBasicDescription *p_format_list = NULL;
 613         int i_formats = 0, j = 0, b_digital = 0;
 614
 615         /* Retrieve all the stream formats supported by each output stream. */
 616         err = AudioStreamGetPropertyInfo(p_streams[i], 0,
 617                                          kAudioStreamPropertyPhysicalFormats,
 618                                          &i_param_size, NULL);
 619         if (err != noErr)
 620         {
 621             ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streamformats: [%4.4s]\n", (char *)&err);
 622             continue;
 623         }
 624
 625         i_formats = i_param_size / sizeof(AudioStreamBasicDescription);
 626         p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size);
 627         if (p_format_list == NULL)
 628         {
 629             ao_msg(MSGT_AO, MSGL_WARN, "could not malloc the memory\n" );
 630             continue;
 631         }
 632
 633         err = AudioStreamGetProperty(p_streams[i], 0,
 634                                      kAudioStreamPropertyPhysicalFormats,
 635                                      &i_param_size, p_format_list);
 636         if (err != noErr)
 637         {
 638             ao_msg(MSGT_AO, MSGL_WARN, "could not get the list of streamformats: [%4.4s]\n", (char *)&err);
 639             if (p_format_list) free(p_format_list);
 640             continue;
 641         }
 642
 643         /* Check if one of the supported formats is a digital format. */
 644         for (j = 0; j < i_formats; ++j)
 645         {
 646             if (p_format_list[j].mFormatID == 'IAC3' ||
 647                   p_format_list[j].mFormatID == kAudioFormat60958AC3)
 648             {
 649                 b_digital = 1;
 650                 break;
 651             }
 652         }
 653
 654         if (b_digital)
 655         {
 656             /* If this stream supports a digital (cac3) format, then set it. */
 657             int i_requested_rate_format = -1;
 658             int i_current_rate_format = -1;
 659             int i_backup_rate_format = -1;
 660
 661             ao->i_stream_id = p_streams[i];
 662             ao->i_stream_index = i;
 663
 664             if (ao->b_revert == 0)
 665             {
 666                 /* Retrieve the original format of this stream first if not done so already. */
 667                 i_param_size = sizeof(ao->sfmt_revert);
 668                 err = AudioStreamGetProperty(ao->i_stream_id, 0,
 669                                              kAudioStreamPropertyPhysicalFormat,
 670                                              &i_param_size,
 671                                              &ao->sfmt_revert);
 672                 if (err != noErr)
 673                 {
 674                     ao_msg(MSGT_AO, MSGL_WARN, "could not retrieve the original streamformat: [%4.4s]\n", (char *)&err);
 675                     if (p_format_list) free(p_format_list);
 676                     continue;
 677                 }
 678                 ao->b_revert = 1;
 679             }
 680
 681             for (j = 0; j < i_formats; ++j)
 682                 if (p_format_list[j].mFormatID == 'IAC3' ||
 683                       p_format_list[j].mFormatID == kAudioFormat60958AC3)
 684                 {
 685                     if (p_format_list[j].mSampleRate == ao->stream_format.mSampleRate)
 686                     {
 687                         i_requested_rate_format = j;
 688                         break;
 689                     }
 690                     if (p_format_list[j].mSampleRate == ao->sfmt_revert.mSampleRate)
 691                         i_current_rate_format = j;
 692                     else if (i_backup_rate_format < 0 || p_format_list[j].mSampleRate > p_format_list[i_backup_rate_format].mSampleRate)
 693                         i_backup_rate_format = j;
 694                 }
 695
 696             if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio. */
 697                 ao->stream_format = p_format_list[i_requested_rate_format];
 698             else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device. */
 699                 ao->stream_format = p_format_list[i_current_rate_format];
 700             else ao->stream_format = p_format_list[i_backup_rate_format]; /* And if we have to, any digital format will be just fine (highest rate possible). */
 701         }
 702         if (p_format_list) free(p_format_list);
 703     }
 704     if (p_streams) free(p_streams);
 705
 706     if (ao->i_stream_index < 0)
 707     {
 708         ao_msg(MSGT_AO, MSGL_WARN, "can not find any digital output stream format when OpenSPDIF().\n");
 709         goto err_out;
 710     }
 711
 712     print_format(MSGL_V, "original stream format:", &ao->sfmt_revert);
 713
 714     if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
 715         goto err_out;
 716
 717     err = AudioDeviceAddPropertyListener(ao->i_selected_dev,
 718                                          kAudioPropertyWildcardChannel,
 719                                          0,
 720                                          kAudioDevicePropertyDeviceHasChanged,
 721                                          DeviceListener,
 722                                          NULL);
 723     if (err != noErr)
 724         ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n", (char *)&err);
 725
 726
 727     /* FIXME: If output stream is not native byte-order, we need change endian somewhere. */
 728     /*        Although there's no such case reported.                                     */
 729 #ifdef WORDS_BIGENDIAN
 730     if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian))
 731 #else
 732     if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
 733 #endif
 734         ao_msg(MSGT_AO, MSGL_WARN, "output stream has a no-native byte-order, digital output may failed.\n");
 735
 736     /* For ac3/dts, just use packet size 6144 bytes as chunk size. */
 737     ao->chunk_size = ao->stream_format.mBytesPerPacket;
 738
 739     ao_data.samplerate = ao->stream_format.mSampleRate;
 740     ao_data.channels = ao->stream_format.mChannelsPerFrame;
 741     ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket);
 742     ao_data.outburst = ao->chunk_size;
 743     ao_data.buffersize = ao_data.bps;
 744
 745     ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
 746     ao->buffer_len = (ao->num_chunks + 1) * ao->chunk_size;
 747     ao->buffer = calloc(ao->num_chunks + 1, ao->chunk_size);
 748
 749     ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
 750
 751
 752     /* Add IOProc callback. */
 753     err = AudioDeviceAddIOProc(ao->i_selected_dev,
 754                                (AudioDeviceIOProc)RenderCallbackSPDIF,
 755                                (void *)ao);
 756     if (err != noErr)
 757     {
 758         ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err);
 759         goto err_out1;
 760     }
 761
 762     reset();
 763
 764     return CONTROL_TRUE;
 765
 766 err_out1:
 767     if (ao->b_revert)
 768         AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
 769 err_out:
 770     if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
 771     {
 772         int b_mix = 1;
 773         err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
 774                                      kAudioDevicePropertySupportsMixing,
 775                                      i_param_size, &b_mix);
 776         if (err != noErr)
 777             ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n",
 778                    (char *)&err);
 779     }
 780     if (ao->i_hog_pid == getpid())
 781     {
 782         ao->i_hog_pid = -1;
 783         i_param_size = sizeof(ao->i_hog_pid);
 784         err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
 785                                      kAudioDevicePropertyHogMode,
 786                                      i_param_size, &ao->i_hog_pid);
 787         if (err != noErr)
 788             ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n",
 789                    (char *)&err);
 790     }
 791     free(ao->buffer);
 792     free(ao);
 793     ao = NULL;
 794     return CONTROL_FALSE;
 795 }
 796
 797 /*****************************************************************************
 798  * AudioDeviceSupportsDigital: Check i_dev_id for digital stream support.
 799  *****************************************************************************/
 800 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id )
 801 {
 802     OSStatus                    err = noErr;
 803     UInt32                      i_param_size = 0;
 804     AudioStreamID               *p_streams = NULL;
 805     int                         i = 0, i_streams = 0;
 806     int                         b_return = CONTROL_FALSE;
 807
 808     /* Retrieve all the output streams. */
 809     err = AudioDeviceGetPropertyInfo(i_dev_id, 0, FALSE,
 810                                      kAudioDevicePropertyStreams,
 811                                      &i_param_size, NULL);
 812     if (err != noErr)
 813     {
 814         ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err);
 815         return CONTROL_FALSE;
 816     }
 817
 818     i_streams = i_param_size / sizeof(AudioStreamID);
 819     p_streams = (AudioStreamID *)malloc(i_param_size);
 820     if (p_streams == NULL)
 821     {
 822         ao_msg(MSGT_AO,MSGL_V, "out of memory\n");
 823         return CONTROL_FALSE;
 824     }
 825
 826     err = AudioDeviceGetProperty(i_dev_id, 0, FALSE,
 827                                  kAudioDevicePropertyStreams,
 828                                  &i_param_size, p_streams);
 829
 830     if (err != noErr)
 831     {
 832         ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err);
 833         free(p_streams);
 834         return CONTROL_FALSE;
 835     }
 836
 837     for (i = 0; i < i_streams; ++i)
 838     {
 839         if (AudioStreamSupportsDigital(p_streams[i]))
 840             b_return = CONTROL_OK;
 841     }
 842
 843     free(p_streams);
 844     return b_return;
 845 }
 846
 847 /*****************************************************************************
 848  * AudioStreamSupportsDigital: Check i_stream_id for digital stream support.
 849  *****************************************************************************/
 850 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id )
 851 {
 852     OSStatus err = noErr;
 853     UInt32 i_param_size;
 854     AudioStreamBasicDescription *p_format_list = NULL;
 855     int i, i_formats, b_return = CONTROL_FALSE;
 856
 857     /* Retrieve all the stream formats supported by each output stream. */
 858     err = AudioStreamGetPropertyInfo(i_stream_id, 0,
 859                                      kAudioStreamPropertyPhysicalFormats,
 860                                      &i_param_size, NULL);
 861     if (err != noErr)
 862     {
 863         ao_msg(MSGT_AO,MSGL_V, "could not get number of streamformats: [%4.4s]\n", (char *)&err);
 864         return CONTROL_FALSE;
 865     }
 866
 867     i_formats = i_param_size / sizeof(AudioStreamBasicDescription);
 868     p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size);
 869     if (p_format_list == NULL)
 870     {
 871         ao_msg(MSGT_AO,MSGL_V, "could not malloc the memory\n" );
 872         return CONTROL_FALSE;
 873     }
 874
 875     err = AudioStreamGetProperty(i_stream_id, 0,
 876                                  kAudioStreamPropertyPhysicalFormats,
 877                                  &i_param_size, p_format_list);
 878     if (err != noErr)
 879     {
 880         ao_msg(MSGT_AO,MSGL_V, "could not get the list of streamformats: [%4.4s]\n", (char *)&err);
 881         free(p_format_list);
 882         return CONTROL_FALSE;
 883     }
 884
 885     for (i = 0; i < i_formats; ++i)
 886     {
 887         print_format(MSGL_V, "supported format:", &p_format_list[i]);
 888
 889         if (p_format_list[i].mFormatID == 'IAC3' ||
 890                   p_format_list[i].mFormatID == kAudioFormat60958AC3)
 891             b_return = CONTROL_OK;
 892     }
 893
 894     free(p_format_list);
 895     return b_return;
 896 }
 897
 898 /*****************************************************************************
 899  * AudioStreamChangeFormat: Change i_stream_id to change_format
 900  *****************************************************************************/
 901 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format )
 902 {
 903     OSStatus err = noErr;
 904     UInt32 i_param_size = 0;
 905     int i;
 906
 907     static volatile int stream_format_changed;
 908     stream_format_changed = 0;
 909
 910     print_format(MSGL_V, "setting stream format:", &change_format);
 911
 912     /* Install the callback. */
 913     err = AudioStreamAddPropertyListener(i_stream_id, 0,
 914                                          kAudioStreamPropertyPhysicalFormat,
 915                                          StreamListener,
 916                                          (void *)&stream_format_changed);
 917     if (err != noErr)
 918     {
 919         ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamAddPropertyListener failed: [%4.4s]\n", (char *)&err);
 920         return CONTROL_FALSE;
 921     }
 922
 923     /* Change the format. */
 924     err = AudioStreamSetProperty(i_stream_id, 0, 0,
 925                                  kAudioStreamPropertyPhysicalFormat,
 926                                  sizeof(AudioStreamBasicDescription),
 927                                  &change_format);
 928     if (err != noErr)
 929     {
 930         ao_msg(MSGT_AO, MSGL_WARN, "could not set the stream format: [%4.4s]\n", (char *)&err);
 931         return CONTROL_FALSE;
 932     }
 933
 934     /* The AudioStreamSetProperty is not only asynchronious,
 935      * it is also not Atomic, in its behaviour.
 936      * Therefore we check 5 times before we really give up.
 937      * FIXME: failing isn't actually implemented yet. */
 938     for (i = 0; i < 5; ++i)
 939     {
 940         AudioStreamBasicDescription actual_format;
 941         int j;
 942         for (j = 0; !stream_format_changed && j < 50; ++j)
 943             usec_sleep(10000);
 944         if (stream_format_changed)
 945             stream_format_changed = 0;
 946         else
 947             ao_msg(MSGT_AO, MSGL_V, "reached timeout\n" );
 948
 949         i_param_size = sizeof(AudioStreamBasicDescription);
 950         err = AudioStreamGetProperty(i_stream_id, 0,
 951                                      kAudioStreamPropertyPhysicalFormat,
 952                                      &i_param_size,
 953                                      &actual_format);
 954
 955         print_format(MSGL_V, "actual format in use:", &actual_format);
 956         if (actual_format.mSampleRate == change_format.mSampleRate &&
 957             actual_format.mFormatID == change_format.mFormatID &&
 958             actual_format.mFramesPerPacket == change_format.mFramesPerPacket)
 959         {
 960             /* The right format is now active. */
 961             break;
 962         }
 963         /* We need to check again. */
 964     }
 965
 966     /* Removing the property listener. */
 967     err = AudioStreamRemovePropertyListener(i_stream_id, 0,
 968                                             kAudioStreamPropertyPhysicalFormat,
 969                                             StreamListener);
 970     if (err != noErr)
 971     {
 972         ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamRemovePropertyListener failed: [%4.4s]\n", (char *)&err);
 973         return CONTROL_FALSE;
 974     }
 975
 976     return CONTROL_TRUE;
 977 }
 978
 979 /*****************************************************************************
 980  * RenderCallbackSPDIF: callback for SPDIF audio output
 981  *****************************************************************************/
 982 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
 983                                     const AudioTimeStamp * inNow,
 984                                     const void * inInputData,
 985                                     const AudioTimeStamp * inInputTime,
 986                                     AudioBufferList * outOutputData,
 987                                     const AudioTimeStamp * inOutputTime,
 988                                     void * threadGlobals )
 989 {
 990     int amt = buf_used();
 991     int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize;
 992
 993     if (amt > req)
 994         amt = req;
 995     if (amt)
 996         read_buffer(ao->b_muted ? NULL : (unsigned char *)outOutputData->mBuffers[ao->i_stream_index].mData, amt);
 997
 998     return noErr;
 999 }
1000
1001
1002 static int play(void* output_samples,int num_bytes,int flags)
1003 {
1004     int wrote, b_digital;
1005
1006     // Check whether we need to reset the digital output stream.
1007     if (ao->b_digital && ao->b_stream_format_changed)
1008     {
1009         ao->b_stream_format_changed = 0;
1010         b_digital = AudioStreamSupportsDigital(ao->i_stream_id);
1011         if (b_digital)
1012         {
1013             /* Current stream support digital format output, let's set it. */
1014             ao_msg(MSGT_AO, MSGL_V, "detected current stream support digital, try to restore digital output...\n");
1015
1016             if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
1017             {
1018                 ao_msg(MSGT_AO, MSGL_WARN, "restore digital output failed.\n");
1019             }
1020             else
1021             {
1022                 ao_msg(MSGT_AO, MSGL_WARN, "restore digital output succeed.\n");
1023                 reset();
1024             }
1025         }
1026         else
1027             ao_msg(MSGT_AO, MSGL_V, "detected current stream do not support digital.\n");
1028     }
1029
1030     wrote=write_buffer(output_samples, num_bytes);
1031     audio_resume();
1032     return wrote;
1033 }
1034
1035 /* set variables and buffer to initial state */
1036 static void reset(void)
1037 {
1038   audio_pause();
1039   /* reset ring-buffer state */
1040   ao->buf_read_pos=0;
1041   ao->buf_write_pos=0;
1042
1043   return;
1044 }
1045
1046
1047 /* return available space */
1048 static int get_space(void)
1049 {
1050   return buf_free();
1051 }
1052
1053
1054 /* return delay until audio is played */
1055 static float get_delay(void)
1056 {
1057   int buffered = ao->buffer_len - ao->chunk_size - buf_free(); // could be less
1058   // inaccurate, should also contain the data buffered e.g. by the OS
1059   return (float)(buffered)/(float)ao_data.bps;
1060 }
1061
1062
1063 /* unload plugin and deregister from coreaudio */
1064 static void uninit(int immed)
1065 {
1066   OSStatus            err = noErr;
1067   UInt32              i_param_size = 0;
1068
1069   if (!immed) {
1070     long long timeleft=(1000000LL*buf_used())/ao_data.bps;
1071     ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", buf_used(), ao_data.bps, (int)timeleft);
1072     usec_sleep((int)timeleft);
1073   }
1074
1075   if (!ao->b_digital) {
1076       AudioOutputUnitStop(ao->theOutputUnit);
1077       AudioUnitUninitialize(ao->theOutputUnit);
1078       CloseComponent(ao->theOutputUnit);
1079   }
1080   else {
1081       /* Stop device. */
1082       err = AudioDeviceStop(ao->i_selected_dev,
1083                             (AudioDeviceIOProc)RenderCallbackSPDIF);
1084       if (err != noErr)
1085           ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
1086
1087       /* Remove IOProc callback. */
1088       err = AudioDeviceRemoveIOProc(ao->i_selected_dev,
1089                                     (AudioDeviceIOProc)RenderCallbackSPDIF);
1090       if (err != noErr)
1091           ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceRemoveIOProc failed: [%4.4s]\n", (char *)&err);
1092
1093       if (ao->b_revert)
1094           AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
1095
1096       if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
1097       {
1098           int b_mix;
1099           Boolean b_writeable;
1100           /* Revert mixable to true if we are allowed to. */
1101           err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing,
1102                                            &i_param_size, &b_writeable);
1103           err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing,
1104                                        &i_param_size, &b_mix);
1105           if (err != noErr && b_writeable)
1106           {
1107               b_mix = 1;
1108               err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
1109                                            kAudioDevicePropertySupportsMixing, i_param_size, &b_mix);
1110           }
1111           if (err != noErr)
1112               ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
1113       }
1114       if (ao->i_hog_pid == getpid())
1115       {
1116           ao->i_hog_pid = -1;
1117           i_param_size = sizeof(ao->i_hog_pid);
1118           err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
1119                                        kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid);
1120           if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err);
1121       }
1122   }
1123
1124   free(ao->buffer);
1125   free(ao);
1126   ao = NULL;
1127 }
1128
1129
1130 /* stop playing, keep buffers (for pause) */
1131 static void audio_pause(void)
1132 {
1133     OSErr err=noErr;
1134
1135     /* Stop callback. */
1136     if (!ao->b_digital)
1137     {
1138         err=AudioOutputUnitStop(ao->theOutputUnit);
1139         if (err != noErr)
1140             ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStop returned [%4.4s]\n", (char *)&err);
1141     }
1142     else
1143     {
1144         err = AudioDeviceStop(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF);
1145         if (err != noErr)
1146             ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
1147     }
1148     ao->paused = 1;
1149 }
1150
1151
1152 /* resume playing, after audio_pause() */
1153 static void audio_resume(void)
1154 {
1155     OSErr err=noErr;
1156
1157     if (!ao->paused)
1158         return;
1159
1160     /* Start callback. */
1161     if (!ao->b_digital)
1162     {
1163         err = AudioOutputUnitStart(ao->theOutputUnit);
1164         if (err != noErr)
1165             ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStart returned [%4.4s]\n", (char *)&err);
1166     }
1167     else
1168     {
1169         err = AudioDeviceStart(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF);
1170         if (err != noErr)
1171             ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err);
1172     }
1173     ao->paused = 0;
1174 }
1175
1176 /*****************************************************************************
1177  * StreamListener
1178  *****************************************************************************/
1179 static OSStatus StreamListener( AudioStreamID inStream,
1180                                 UInt32 inChannel,
1181                                 AudioDevicePropertyID inPropertyID,
1182                                 void * inClientData )
1183 {
1184     switch (inPropertyID)
1185     {
1186         case kAudioStreamPropertyPhysicalFormat:
1187             ao_msg(MSGT_AO, MSGL_V, "got notify kAudioStreamPropertyPhysicalFormat changed.\n");
1188             if (inClientData)
1189                 *(volatile int *)inClientData = 1;
1190         default:
1191             break;
1192     }
1193     return noErr;
1194 }
1195
1196 static OSStatus DeviceListener( AudioDeviceID inDevice,
1197                                 UInt32 inChannel,
1198                                 Boolean isInput,
1199                                 AudioDevicePropertyID inPropertyID,
1200                                 void* inClientData )
1201 {
1202     switch (inPropertyID)
1203     {
1204         case kAudioDevicePropertyDeviceHasChanged:
1205             ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioDevicePropertyDeviceHasChanged.\n");
1206             ao->b_stream_format_changed = 1;
1207         default:
1208             break;
1209     }
1210     return noErr;
1211 }