libao2/ao_macosx.c

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