libao2/ao_coreaudio.c

   1 /*
   2  * CoreAudio audio output driver for Mac OS X
   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 #include "libavutil/fifo.h"
  56
  57 static const ao_info_t info =
  58   {
  59     "Darwin/Mac OS X native audio output",
  60     "coreaudio",
  61     "Timothy J. Wood & Dan Christiansen & Chris Roccati",
  62     ""
  63   };
  64
  65 LIBAO_EXTERN(coreaudio)
  66
  67 /* Prefix for all mp_msg() calls */
  68 #define ao_msg(a, b, c...) mp_msg(a, b, "AO: [coreaudio] " c)
  69
  70 typedef struct ao_coreaudio_s
  71 {
  72   AudioDeviceID i_selected_dev;             /* Keeps DeviceID of the selected device. */
  73   int b_supports_digital;                   /* Does the currently selected device support digital mode? */
  74   int b_digital;                            /* Are we running in digital mode? */
  75   int b_muted;                              /* Are we muted in digital mode? */
  76
  77   /* AudioUnit */
  78   AudioUnit theOutputUnit;
  79
  80   /* CoreAudio SPDIF mode specific */
  81   pid_t i_hog_pid;                          /* Keeps the pid of our hog status. */
  82   AudioStreamID i_stream_id;                /* The StreamID that has a cac3 streamformat */
  83   int i_stream_index;                       /* The index of i_stream_id in an AudioBufferList */
  84   AudioStreamBasicDescription stream_format;/* The format we changed the stream to */
  85   AudioStreamBasicDescription sfmt_revert;  /* The original format of the stream */
  86   int b_revert;                             /* Whether we need to revert the stream format */
  87   int b_changed_mixing;                     /* Whether we need to set the mixing mode back */
  88   int b_stream_format_changed;              /* Flag for main thread to reset stream's format to digital and reset buffer */
  89
  90   /* Original common part */
  91   int packetSize;
  92   int paused;
  93
  94   /* Ring-buffer */
  95   AVFifoBuffer *buffer;
  96   unsigned int buffer_len; ///< must always be num_chunks * chunk_size
  97   unsigned int num_chunks;
  98   unsigned int chunk_size;
  99 } ao_coreaudio_t;
 100
 101 static ao_coreaudio_t *ao = NULL;
 102
 103 /**
 104  * \brief add data to ringbuffer
 105  */
 106 static int write_buffer(unsigned char* data, int len){
 107   int free = ao->buffer_len - av_fifo_size(ao->buffer);
 108   if (len > free) len = free;
 109   return av_fifo_generic_write(ao->buffer, data, len, NULL);
 110 }
 111
 112 /**
 113  * \brief remove data from ringbuffer
 114  */
 115 static int read_buffer(unsigned char* data,int len){
 116   int buffered = av_fifo_size(ao->buffer);
 117   if (len > buffered) len = buffered;
 118   av_fifo_generic_read(ao->buffer, data, len, NULL);
 119   return len;
 120 }
 121
 122 OSStatus theRenderProc(void *inRefCon, AudioUnitRenderActionFlags *inActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumFrames, AudioBufferList *ioData)
 123 {
 124 int amt=av_fifo_size(ao->buffer);
 125 int req=(inNumFrames)*ao->packetSize;
 126
 127         if(amt>req)
 128                 amt=req;
 129
 130         if(amt)
 131                 read_buffer((unsigned char *)ioData->mBuffers[0].mData, amt);
 132         else audio_pause();
 133         ioData->mBuffers[0].mDataByteSize = amt;
 134
 135         return noErr;
 136 }
 137
 138 static int control(int cmd,void *arg){
 139 ao_control_vol_t *control_vol;
 140 OSStatus err;
 141 Float32 vol;
 142         switch (cmd) {
 143         case AOCONTROL_GET_VOLUME:
 144                 control_vol = (ao_control_vol_t*)arg;
 145                 if (ao->b_digital) {
 146                         // Digital output has no volume adjust.
 147                         return CONTROL_FALSE;
 148                 }
 149                 err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol);
 150
 151                 if(err==0) {
 152                         // printf("GET VOL=%f\n", vol);
 153                         control_vol->left=control_vol->right=vol*100.0/4.0;
 154                         return CONTROL_TRUE;
 155                 }
 156                 else {
 157                         ao_msg(MSGT_AO, MSGL_WARN, "could not get HAL output volume: [%4.4s]\n", (char *)&err);
 158                         return CONTROL_FALSE;
 159                 }
 160
 161         case AOCONTROL_SET_VOLUME:
 162                 control_vol = (ao_control_vol_t*)arg;
 163
 164                 if (ao->b_digital) {
 165                         // Digital output can not set volume. Here we have to return true
 166                         // to make mixer forget it. Else mixer will add a soft filter,
 167                         // that's not we expected and the filter not support ac3 stream
 168                         // will cause mplayer die.
 169
 170                         // Although not support set volume, but at least we support mute.
 171                         // MPlayer set mute by set volume to zero, we handle it.
 172                         if (control_vol->left == 0 && control_vol->right == 0)
 173                                 ao->b_muted = 1;
 174                         else
 175                                 ao->b_muted = 0;
 176                         return CONTROL_TRUE;
 177                 }
 178
 179                 vol=(control_vol->left+control_vol->right)*4.0/200.0;
 180                 err = AudioUnitSetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, vol, 0);
 181                 if(err==0) {
 182                         // printf("SET VOL=%f\n", vol);
 183                         return CONTROL_TRUE;
 184                 }
 185                 else {
 186                         ao_msg(MSGT_AO, MSGL_WARN, "could not set HAL output volume: [%4.4s]\n", (char *)&err);
 187                         return CONTROL_FALSE;
 188                 }
 189           /* Everything is currently unimplemented */
 190         default:
 191           return CONTROL_FALSE;
 192         }
 193
 194 }
 195
 196
 197 static void print_format(int lev, const char* str, const AudioStreamBasicDescription *f){
 198     uint32_t flags=(uint32_t) f->mFormatFlags;
 199     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",
 200             str, f->mSampleRate, f->mBitsPerChannel,
 201             (int)(f->mFormatID & 0xff000000) >> 24,
 202             (int)(f->mFormatID & 0x00ff0000) >> 16,
 203             (int)(f->mFormatID & 0x0000ff00) >>  8,
 204             (int)(f->mFormatID & 0x000000ff) >>  0,
 205             f->mFormatFlags, f->mBytesPerPacket,
 206             f->mFramesPerPacket, f->mBytesPerFrame,
 207             f->mChannelsPerFrame,
 208             (flags&kAudioFormatFlagIsFloat) ? "float" : "int",
 209             (flags&kAudioFormatFlagIsBigEndian) ? "BE" : "LE",
 210             (flags&kAudioFormatFlagIsSignedInteger) ? "S" : "U",
 211             (flags&kAudioFormatFlagIsPacked) ? " packed" : "",
 212             (flags&kAudioFormatFlagIsAlignedHigh) ? " aligned" : "",
 213             (flags&kAudioFormatFlagIsNonInterleaved) ? " ni" : "" );
 214 }
 215
 216
 217 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id );
 218 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id );
 219 static int OpenSPDIF(void);
 220 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format );
 221 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
 222                                     const AudioTimeStamp * inNow,
 223                                     const void * inInputData,
 224                                     const AudioTimeStamp * inInputTime,
 225                                     AudioBufferList * outOutputData,
 226                                     const AudioTimeStamp * inOutputTime,
 227                                     void * threadGlobals );
 228 static OSStatus StreamListener( AudioStreamID inStream,
 229                                 UInt32 inChannel,
 230                                 AudioDevicePropertyID inPropertyID,
 231                                 void * inClientData );
 232 static OSStatus DeviceListener( AudioDeviceID inDevice,
 233                                 UInt32 inChannel,
 234                                 Boolean isInput,
 235                                 AudioDevicePropertyID inPropertyID,
 236                                 void* inClientData );
 237
 238 static int init(int rate,int channels,int format,int flags)
 239 {
 240 AudioStreamBasicDescription inDesc;
 241 ComponentDescription desc;
 242 Component comp;
 243 AURenderCallbackStruct renderCallback;
 244 OSStatus err;
 245 UInt32 size, maxFrames, i_param_size;
 246 char *psz_name;
 247 AudioDeviceID devid_def = 0;
 248 int b_alive;
 249
 250     ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags);
 251
 252     ao = calloc(1, sizeof(ao_coreaudio_t));
 253
 254     ao->i_selected_dev = 0;
 255     ao->b_supports_digital = 0;
 256     ao->b_digital = 0;
 257     ao->b_muted = 0;
 258     ao->b_stream_format_changed = 0;
 259     ao->i_hog_pid = -1;
 260     ao->i_stream_id = 0;
 261     ao->i_stream_index = -1;
 262     ao->b_revert = 0;
 263     ao->b_changed_mixing = 0;
 264
 265     /* Probe whether device support S/PDIF stream output if input is AC3 stream. */
 266     if (AF_FORMAT_IS_AC3(format))
 267     {
 268         /* Find the ID of the default Device. */
 269         i_param_size = sizeof(AudioDeviceID);
 270         err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
 271                                        &i_param_size, &devid_def);
 272         if (err != noErr)
 273         {
 274             ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device: [%4.4s]\n", (char *)&err);
 275             goto err_out;
 276         }
 277
 278         /* Retrieve the length of the device name. */
 279         i_param_size = 0;
 280         err = AudioDeviceGetPropertyInfo(devid_def, 0, 0,
 281                                          kAudioDevicePropertyDeviceName,
 282                                          &i_param_size, NULL);
 283         if (err != noErr)
 284         {
 285             ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name length: [%4.4s]\n", (char *)&err);
 286             goto err_out;
 287         }
 288
 289         /* Retrieve the name of the device. */
 290         psz_name = (char *)malloc(i_param_size);
 291         err = AudioDeviceGetProperty(devid_def, 0, 0,
 292                                      kAudioDevicePropertyDeviceName,
 293                                      &i_param_size, psz_name);
 294         if (err != noErr)
 295         {
 296             ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err);
 297             free( psz_name);
 298             goto err_out;
 299         }
 300
 301         ao_msg(MSGT_AO,MSGL_V, "got default audio output device ID: %#lx Name: %s\n", devid_def, psz_name );
 302
 303         if (AudioDeviceSupportsDigital(devid_def))
 304         {
 305             ao->b_supports_digital = 1;
 306             ao->i_selected_dev = devid_def;
 307         }
 308         ao_msg(MSGT_AO,MSGL_V, "probe default audio output device whether support digital s/pdif output:%d\n", ao->b_supports_digital );
 309
 310         free( psz_name);
 311     }
 312
 313         // Build Description for the input format
 314         inDesc.mSampleRate=rate;
 315         inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
 316         inDesc.mChannelsPerFrame=channels;
 317         inDesc.mBitsPerChannel=af_fmt2bits(format);
 318
 319     if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) {
 320         // float
 321                 inDesc.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked;
 322     }
 323     else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI) {
 324         // signed int
 325                 inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
 326     }
 327     else {
 328         // unsigned int
 329                 inDesc.mFormatFlags = kAudioFormatFlagIsPacked;
 330     }
 331     if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
 332         inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
 333
 334     inDesc.mFramesPerPacket = 1;
 335     ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8);
 336     print_format(MSGL_V, "source:",&inDesc);
 337
 338     if (ao->b_supports_digital)
 339     {
 340         b_alive = 1;
 341         i_param_size = sizeof(b_alive);
 342         err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
 343                                      kAudioDevicePropertyDeviceIsAlive,
 344                                      &i_param_size, &b_alive);
 345         if (err != noErr)
 346             ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err);
 347         if (!b_alive)
 348             ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" );
 349         /* S/PDIF output need device in HogMode. */
 350         i_param_size = sizeof(ao->i_hog_pid);
 351         err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
 352                                      kAudioDevicePropertyHogMode,
 353                                      &i_param_size, &ao->i_hog_pid);
 354
 355         if (err != noErr)
 356         {
 357             /* This is not a fatal error. Some drivers simply don't support this property. */
 358             ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n",
 359                      (char *)&err);
 360             ao->i_hog_pid = -1;
 361         }
 362
 363         if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid())
 364         {
 365             ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" );
 366             goto err_out;
 367         }
 368         ao->stream_format = inDesc;
 369         return OpenSPDIF();
 370     }
 371
 372         /* original analog output code */
 373         desc.componentType = kAudioUnitType_Output;
 374         desc.componentSubType = kAudioUnitSubType_DefaultOutput;
 375         desc.componentManufacturer = kAudioUnitManufacturer_Apple;
 376         desc.componentFlags = 0;
 377         desc.componentFlagsMask = 0;
 378
 379         comp = FindNextComponent(NULL, &desc);  //Finds an component that meets the desc spec's
 380         if (comp == NULL) {
 381                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n");
 382                 goto err_out;
 383         }
 384
 385         err = OpenAComponent(comp, &(ao->theOutputUnit));  //gains access to the services provided by the component
 386         if (err) {
 387                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err);
 388                 goto err_out;
 389         }
 390
 391         // Initialize AudioUnit
 392         err = AudioUnitInitialize(ao->theOutputUnit);
 393         if (err) {
 394                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err);
 395                 goto err_out1;
 396         }
 397
 398         size =  sizeof(AudioStreamBasicDescription);
 399         err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size);
 400
 401         if (err) {
 402                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err);
 403                 goto err_out2;
 404         }
 405
 406         size = sizeof(UInt32);
 407         err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size);
 408
 409         if (err)
 410         {
 411                 ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err);
 412                 goto err_out2;
 413         }
 414
 415         ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame;
 416
 417         ao_data.samplerate = inDesc.mSampleRate;
 418         ao_data.channels = inDesc.mChannelsPerFrame;
 419     ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame;
 420     ao_data.outburst = ao->chunk_size;
 421         ao_data.buffersize = ao_data.bps;
 422
 423         ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
 424     ao->buffer_len = ao->num_chunks * ao->chunk_size;
 425     ao->buffer = av_fifo_alloc(ao->buffer_len);
 426
 427         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);
 428
 429     renderCallback.inputProc = theRenderProc;
 430     renderCallback.inputProcRefCon = 0;
 431     err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct));
 432         if (err) {
 433                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err);
 434                 goto err_out2;
 435         }
 436
 437         reset();
 438
 439     return CONTROL_OK;
 440
 441 err_out2:
 442     AudioUnitUninitialize(ao->theOutputUnit);
 443 err_out1:
 444     CloseComponent(ao->theOutputUnit);
 445 err_out:
 446     av_fifo_free(ao->buffer);
 447     free(ao);
 448     ao = NULL;
 449     return CONTROL_FALSE;
 450 }
 451
 452 /*****************************************************************************
 453  * Setup a encoded digital stream (SPDIF)
 454  *****************************************************************************/
 455 static int OpenSPDIF(void)
 456 {
 457     OSStatus                err = noErr;
 458     UInt32                  i_param_size, b_mix = 0;
 459     Boolean                 b_writeable = 0;
 460     AudioStreamID           *p_streams = NULL;
 461     int                     i, i_streams = 0;
 462
 463     /* Start doing the SPDIF setup process. */
 464     ao->b_digital = 1;
 465
 466     /* Hog the device. */
 467     i_param_size = sizeof(ao->i_hog_pid);
 468     ao->i_hog_pid = getpid() ;
 469
 470     err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
 471                                  kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid);
 472
 473     if (err != noErr)
 474     {
 475         ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err);
 476         ao->i_hog_pid = -1;
 477         goto err_out;
 478     }
 479
 480     /* Set mixable to false if we are allowed to. */
 481     err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE,
 482                                      kAudioDevicePropertySupportsMixing,
 483                                      &i_param_size, &b_writeable);
 484     err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
 485                                  kAudioDevicePropertySupportsMixing,
 486                                  &i_param_size, &b_mix);
 487     if (err != noErr && b_writeable)
 488     {
 489         b_mix = 0;
 490         err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
 491                                      kAudioDevicePropertySupportsMixing,
 492                                      i_param_size, &b_mix);
 493         ao->b_changed_mixing = 1;
 494     }
 495     if (err != noErr)
 496     {
 497         ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
 498         goto err_out;
 499     }
 500
 501     /* Get a list of all the streams on this device. */
 502     err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE,
 503                                      kAudioDevicePropertyStreams,
 504                                      &i_param_size, NULL);
 505     if (err != noErr)
 506     {
 507         ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err);
 508         goto err_out;
 509     }
 510
 511     i_streams = i_param_size / sizeof(AudioStreamID);
 512     p_streams = (AudioStreamID *)malloc(i_param_size);
 513     if (p_streams == NULL)
 514     {
 515         ao_msg(MSGT_AO, MSGL_WARN, "out of memory\n" );
 516         goto err_out;
 517     }
 518
 519     err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
 520                                  kAudioDevicePropertyStreams,
 521                                  &i_param_size, p_streams);
 522     if (err != noErr)
 523     {
 524         ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err);
 525         if (p_streams) free(p_streams);
 526         goto err_out;
 527     }
 528
 529     ao_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams);
 530
 531     for (i = 0; i < i_streams && ao->i_stream_index < 0; ++i)
 532     {
 533         /* Find a stream with a cac3 stream. */
 534         AudioStreamBasicDescription *p_format_list = NULL;
 535         int i_formats = 0, j = 0, b_digital = 0;
 536
 537         /* Retrieve all the stream formats supported by each output stream. */
 538         err = AudioStreamGetPropertyInfo(p_streams[i], 0,
 539                                          kAudioStreamPropertyPhysicalFormats,
 540                                          &i_param_size, NULL);
 541         if (err != noErr)
 542         {
 543             ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streamformats: [%4.4s]\n", (char *)&err);
 544             continue;
 545         }
 546
 547         i_formats = i_param_size / sizeof(AudioStreamBasicDescription);
 548         p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size);
 549         if (p_format_list == NULL)
 550         {
 551             ao_msg(MSGT_AO, MSGL_WARN, "could not malloc the memory\n" );
 552             continue;
 553         }
 554
 555         err = AudioStreamGetProperty(p_streams[i], 0,
 556                                      kAudioStreamPropertyPhysicalFormats,
 557                                      &i_param_size, p_format_list);
 558         if (err != noErr)
 559         {
 560             ao_msg(MSGT_AO, MSGL_WARN, "could not get the list of streamformats: [%4.4s]\n", (char *)&err);
 561             if (p_format_list) free(p_format_list);
 562             continue;
 563         }
 564
 565         /* Check if one of the supported formats is a digital format. */
 566         for (j = 0; j < i_formats; ++j)
 567         {
 568             if (p_format_list[j].mFormatID == 'IAC3' ||
 569                   p_format_list[j].mFormatID == kAudioFormat60958AC3)
 570             {
 571                 b_digital = 1;
 572                 break;
 573             }
 574         }
 575
 576         if (b_digital)
 577         {
 578             /* If this stream supports a digital (cac3) format, then set it. */
 579             int i_requested_rate_format = -1;
 580             int i_current_rate_format = -1;
 581             int i_backup_rate_format = -1;
 582
 583             ao->i_stream_id = p_streams[i];
 584             ao->i_stream_index = i;
 585
 586             if (ao->b_revert == 0)
 587             {
 588                 /* Retrieve the original format of this stream first if not done so already. */
 589                 i_param_size = sizeof(ao->sfmt_revert);
 590                 err = AudioStreamGetProperty(ao->i_stream_id, 0,
 591                                              kAudioStreamPropertyPhysicalFormat,
 592                                              &i_param_size,
 593                                              &ao->sfmt_revert);
 594                 if (err != noErr)
 595                 {
 596                     ao_msg(MSGT_AO, MSGL_WARN, "could not retrieve the original streamformat: [%4.4s]\n", (char *)&err);
 597                     if (p_format_list) free(p_format_list);
 598                     continue;
 599                 }
 600                 ao->b_revert = 1;
 601             }
 602
 603             for (j = 0; j < i_formats; ++j)
 604                 if (p_format_list[j].mFormatID == 'IAC3' ||
 605                       p_format_list[j].mFormatID == kAudioFormat60958AC3)
 606                 {
 607                     if (p_format_list[j].mSampleRate == ao->stream_format.mSampleRate)
 608                     {
 609                         i_requested_rate_format = j;
 610                         break;
 611                     }
 612                     if (p_format_list[j].mSampleRate == ao->sfmt_revert.mSampleRate)
 613                         i_current_rate_format = j;
 614                     else if (i_backup_rate_format < 0 || p_format_list[j].mSampleRate > p_format_list[i_backup_rate_format].mSampleRate)
 615                         i_backup_rate_format = j;
 616                 }
 617
 618             if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio. */
 619                 ao->stream_format = p_format_list[i_requested_rate_format];
 620             else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device. */
 621                 ao->stream_format = p_format_list[i_current_rate_format];
 622             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). */
 623         }
 624         if (p_format_list) free(p_format_list);
 625     }
 626     if (p_streams) free(p_streams);
 627
 628     if (ao->i_stream_index < 0)
 629     {
 630         ao_msg(MSGT_AO, MSGL_WARN, "can not find any digital output stream format when OpenSPDIF().\n");
 631         goto err_out;
 632     }
 633
 634     print_format(MSGL_V, "original stream format:", &ao->sfmt_revert);
 635
 636     if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
 637         goto err_out;
 638
 639     err = AudioDeviceAddPropertyListener(ao->i_selected_dev,
 640                                          kAudioPropertyWildcardChannel,
 641                                          0,
 642                                          kAudioDevicePropertyDeviceHasChanged,
 643                                          DeviceListener,
 644                                          NULL);
 645     if (err != noErr)
 646         ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n", (char *)&err);
 647
 648
 649     /* FIXME: If output stream is not native byte-order, we need change endian somewhere. */
 650     /*        Although there's no such case reported.                                     */
 651 #if HAVE_BIGENDIAN
 652     if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian))
 653 #else
 654     if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
 655 #endif
 656         ao_msg(MSGT_AO, MSGL_WARN, "output stream has a no-native byte-order, digital output may failed.\n");
 657
 658     /* For ac3/dts, just use packet size 6144 bytes as chunk size. */
 659     ao->chunk_size = ao->stream_format.mBytesPerPacket;
 660
 661     ao_data.samplerate = ao->stream_format.mSampleRate;
 662     ao_data.channels = ao->stream_format.mChannelsPerFrame;
 663     ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket);
 664     ao_data.outburst = ao->chunk_size;
 665     ao_data.buffersize = ao_data.bps;
 666
 667     ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
 668     ao->buffer_len = ao->num_chunks * ao->chunk_size;
 669     ao->buffer = av_fifo_alloc(ao->buffer_len);
 670
 671     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);
 672
 673
 674     /* Add IOProc callback. */
 675     err = AudioDeviceAddIOProc(ao->i_selected_dev,
 676                                (AudioDeviceIOProc)RenderCallbackSPDIF,
 677                                (void *)ao);
 678     if (err != noErr)
 679     {
 680         ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err);
 681         goto err_out1;
 682     }
 683
 684     reset();
 685
 686     return CONTROL_TRUE;
 687
 688 err_out1:
 689     if (ao->b_revert)
 690         AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
 691 err_out:
 692     if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
 693     {
 694         int b_mix = 1;
 695         err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
 696                                      kAudioDevicePropertySupportsMixing,
 697                                      i_param_size, &b_mix);
 698         if (err != noErr)
 699             ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n",
 700                    (char *)&err);
 701     }
 702     if (ao->i_hog_pid == getpid())
 703     {
 704         ao->i_hog_pid = -1;
 705         i_param_size = sizeof(ao->i_hog_pid);
 706         err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
 707                                      kAudioDevicePropertyHogMode,
 708                                      i_param_size, &ao->i_hog_pid);
 709         if (err != noErr)
 710             ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n",
 711                    (char *)&err);
 712     }
 713     av_fifo_free(ao->buffer);
 714     free(ao);
 715     ao = NULL;
 716     return CONTROL_FALSE;
 717 }
 718
 719 /*****************************************************************************
 720  * AudioDeviceSupportsDigital: Check i_dev_id for digital stream support.
 721  *****************************************************************************/
 722 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id )
 723 {
 724     OSStatus                    err = noErr;
 725     UInt32                      i_param_size = 0;
 726     AudioStreamID               *p_streams = NULL;
 727     int                         i = 0, i_streams = 0;
 728     int                         b_return = CONTROL_FALSE;
 729
 730     /* Retrieve all the output streams. */
 731     err = AudioDeviceGetPropertyInfo(i_dev_id, 0, FALSE,
 732                                      kAudioDevicePropertyStreams,
 733                                      &i_param_size, NULL);
 734     if (err != noErr)
 735     {
 736         ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err);
 737         return CONTROL_FALSE;
 738     }
 739
 740     i_streams = i_param_size / sizeof(AudioStreamID);
 741     p_streams = (AudioStreamID *)malloc(i_param_size);
 742     if (p_streams == NULL)
 743     {
 744         ao_msg(MSGT_AO,MSGL_V, "out of memory\n");
 745         return CONTROL_FALSE;
 746     }
 747
 748     err = AudioDeviceGetProperty(i_dev_id, 0, FALSE,
 749                                  kAudioDevicePropertyStreams,
 750                                  &i_param_size, p_streams);
 751
 752     if (err != noErr)
 753     {
 754         ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err);
 755         free(p_streams);
 756         return CONTROL_FALSE;
 757     }
 758
 759     for (i = 0; i < i_streams; ++i)
 760     {
 761         if (AudioStreamSupportsDigital(p_streams[i]))
 762             b_return = CONTROL_OK;
 763     }
 764
 765     free(p_streams);
 766     return b_return;
 767 }
 768
 769 /*****************************************************************************
 770  * AudioStreamSupportsDigital: Check i_stream_id for digital stream support.
 771  *****************************************************************************/
 772 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id )
 773 {
 774     OSStatus err = noErr;
 775     UInt32 i_param_size;
 776     AudioStreamBasicDescription *p_format_list = NULL;
 777     int i, i_formats, b_return = CONTROL_FALSE;
 778
 779     /* Retrieve all the stream formats supported by each output stream. */
 780     err = AudioStreamGetPropertyInfo(i_stream_id, 0,
 781                                      kAudioStreamPropertyPhysicalFormats,
 782                                      &i_param_size, NULL);
 783     if (err != noErr)
 784     {
 785         ao_msg(MSGT_AO,MSGL_V, "could not get number of streamformats: [%4.4s]\n", (char *)&err);
 786         return CONTROL_FALSE;
 787     }
 788
 789     i_formats = i_param_size / sizeof(AudioStreamBasicDescription);
 790     p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size);
 791     if (p_format_list == NULL)
 792     {
 793         ao_msg(MSGT_AO,MSGL_V, "could not malloc the memory\n" );
 794         return CONTROL_FALSE;
 795     }
 796
 797     err = AudioStreamGetProperty(i_stream_id, 0,
 798                                  kAudioStreamPropertyPhysicalFormats,
 799                                  &i_param_size, p_format_list);
 800     if (err != noErr)
 801     {
 802         ao_msg(MSGT_AO,MSGL_V, "could not get the list of streamformats: [%4.4s]\n", (char *)&err);
 803         free(p_format_list);
 804         return CONTROL_FALSE;
 805     }
 806
 807     for (i = 0; i < i_formats; ++i)
 808     {
 809         print_format(MSGL_V, "supported format:", &p_format_list[i]);
 810
 811         if (p_format_list[i].mFormatID == 'IAC3' ||
 812                   p_format_list[i].mFormatID == kAudioFormat60958AC3)
 813             b_return = CONTROL_OK;
 814     }
 815
 816     free(p_format_list);
 817     return b_return;
 818 }
 819
 820 /*****************************************************************************
 821  * AudioStreamChangeFormat: Change i_stream_id to change_format
 822  *****************************************************************************/
 823 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format )
 824 {
 825     OSStatus err = noErr;
 826     UInt32 i_param_size = 0;
 827     int i;
 828
 829     static volatile int stream_format_changed;
 830     stream_format_changed = 0;
 831
 832     print_format(MSGL_V, "setting stream format:", &change_format);
 833
 834     /* Install the callback. */
 835     err = AudioStreamAddPropertyListener(i_stream_id, 0,
 836                                          kAudioStreamPropertyPhysicalFormat,
 837                                          StreamListener,
 838                                          (void *)&stream_format_changed);
 839     if (err != noErr)
 840     {
 841         ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamAddPropertyListener failed: [%4.4s]\n", (char *)&err);
 842         return CONTROL_FALSE;
 843     }
 844
 845     /* Change the format. */
 846     err = AudioStreamSetProperty(i_stream_id, 0, 0,
 847                                  kAudioStreamPropertyPhysicalFormat,
 848                                  sizeof(AudioStreamBasicDescription),
 849                                  &change_format);
 850     if (err != noErr)
 851     {
 852         ao_msg(MSGT_AO, MSGL_WARN, "could not set the stream format: [%4.4s]\n", (char *)&err);
 853         return CONTROL_FALSE;
 854     }
 855
 856     /* The AudioStreamSetProperty is not only asynchronious,
 857      * it is also not Atomic, in its behaviour.
 858      * Therefore we check 5 times before we really give up.
 859      * FIXME: failing isn't actually implemented yet. */
 860     for (i = 0; i < 5; ++i)
 861     {
 862         AudioStreamBasicDescription actual_format;
 863         int j;
 864         for (j = 0; !stream_format_changed && j < 50; ++j)
 865             usec_sleep(10000);
 866         if (stream_format_changed)
 867             stream_format_changed = 0;
 868         else
 869             ao_msg(MSGT_AO, MSGL_V, "reached timeout\n" );
 870
 871         i_param_size = sizeof(AudioStreamBasicDescription);
 872         err = AudioStreamGetProperty(i_stream_id, 0,
 873                                      kAudioStreamPropertyPhysicalFormat,
 874                                      &i_param_size,
 875                                      &actual_format);
 876
 877         print_format(MSGL_V, "actual format in use:", &actual_format);
 878         if (actual_format.mSampleRate == change_format.mSampleRate &&
 879             actual_format.mFormatID == change_format.mFormatID &&
 880             actual_format.mFramesPerPacket == change_format.mFramesPerPacket)
 881         {
 882             /* The right format is now active. */
 883             break;
 884         }
 885         /* We need to check again. */
 886     }
 887
 888     /* Removing the property listener. */
 889     err = AudioStreamRemovePropertyListener(i_stream_id, 0,
 890                                             kAudioStreamPropertyPhysicalFormat,
 891                                             StreamListener);
 892     if (err != noErr)
 893     {
 894         ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamRemovePropertyListener failed: [%4.4s]\n", (char *)&err);
 895         return CONTROL_FALSE;
 896     }
 897
 898     return CONTROL_TRUE;
 899 }
 900
 901 /*****************************************************************************
 902  * RenderCallbackSPDIF: callback for SPDIF audio output
 903  *****************************************************************************/
 904 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
 905                                     const AudioTimeStamp * inNow,
 906                                     const void * inInputData,
 907                                     const AudioTimeStamp * inInputTime,
 908                                     AudioBufferList * outOutputData,
 909                                     const AudioTimeStamp * inOutputTime,
 910                                     void * threadGlobals )
 911 {
 912     int amt = av_fifo_size(ao->buffer);
 913     int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize;
 914
 915     if (amt > req)
 916         amt = req;
 917     if (amt)
 918         read_buffer(ao->b_muted ? NULL : (unsigned char *)outOutputData->mBuffers[ao->i_stream_index].mData, amt);
 919
 920     return noErr;
 921 }
 922
 923
 924 static int play(void* output_samples,int num_bytes,int flags)
 925 {
 926     int wrote, b_digital;
 927
 928     // Check whether we need to reset the digital output stream.
 929     if (ao->b_digital && ao->b_stream_format_changed)
 930     {
 931         ao->b_stream_format_changed = 0;
 932         b_digital = AudioStreamSupportsDigital(ao->i_stream_id);
 933         if (b_digital)
 934         {
 935             /* Current stream support digital format output, let's set it. */
 936             ao_msg(MSGT_AO, MSGL_V, "detected current stream support digital, try to restore digital output...\n");
 937
 938             if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
 939             {
 940                 ao_msg(MSGT_AO, MSGL_WARN, "restore digital output failed.\n");
 941             }
 942             else
 943             {
 944                 ao_msg(MSGT_AO, MSGL_WARN, "restore digital output succeed.\n");
 945                 reset();
 946             }
 947         }
 948         else
 949             ao_msg(MSGT_AO, MSGL_V, "detected current stream do not support digital.\n");
 950     }
 951
 952     wrote=write_buffer(output_samples, num_bytes);
 953     audio_resume();
 954     return wrote;
 955 }
 956
 957 /* set variables and buffer to initial state */
 958 static void reset(void)
 959 {
 960   audio_pause();
 961   av_fifo_reset(ao->buffer);
 962 }
 963
 964
 965 /* return available space */
 966 static int get_space(void)
 967 {
 968   return ao->buffer_len - av_fifo_size(ao->buffer);
 969 }
 970
 971
 972 /* return delay until audio is played */
 973 static float get_delay(void)
 974 {
 975   // inaccurate, should also contain the data buffered e.g. by the OS
 976   return (float)av_fifo_size(ao->buffer)/(float)ao_data.bps;
 977 }
 978
 979
 980 /* unload plugin and deregister from coreaudio */
 981 static void uninit(int immed)
 982 {
 983   OSStatus            err = noErr;
 984   UInt32              i_param_size = 0;
 985
 986   if (!immed) {
 987     long long timeleft=(1000000LL*av_fifo_size(ao->buffer))/ao_data.bps;
 988     ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", av_fifo_size(ao->buffer), ao_data.bps, (int)timeleft);
 989     usec_sleep((int)timeleft);
 990   }
 991
 992   if (!ao->b_digital) {
 993       AudioOutputUnitStop(ao->theOutputUnit);
 994       AudioUnitUninitialize(ao->theOutputUnit);
 995       CloseComponent(ao->theOutputUnit);
 996   }
 997   else {
 998       /* Stop device. */
 999       err = AudioDeviceStop(ao->i_selected_dev,
1000                             (AudioDeviceIOProc)RenderCallbackSPDIF);
1001       if (err != noErr)
1002           ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
1003
1004       /* Remove IOProc callback. */
1005       err = AudioDeviceRemoveIOProc(ao->i_selected_dev,
1006                                     (AudioDeviceIOProc)RenderCallbackSPDIF);
1007       if (err != noErr)
1008           ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceRemoveIOProc failed: [%4.4s]\n", (char *)&err);
1009
1010       if (ao->b_revert)
1011           AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
1012
1013       if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
1014       {
1015           int b_mix;
1016           Boolean b_writeable;
1017           /* Revert mixable to true if we are allowed to. */
1018           err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing,
1019                                            &i_param_size, &b_writeable);
1020           err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing,
1021                                        &i_param_size, &b_mix);
1022           if (err != noErr && b_writeable)
1023           {
1024               b_mix = 1;
1025               err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
1026                                            kAudioDevicePropertySupportsMixing, i_param_size, &b_mix);
1027           }
1028           if (err != noErr)
1029               ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
1030       }
1031       if (ao->i_hog_pid == getpid())
1032       {
1033           ao->i_hog_pid = -1;
1034           i_param_size = sizeof(ao->i_hog_pid);
1035           err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
1036                                        kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid);
1037           if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err);
1038       }
1039   }
1040
1041   av_fifo_free(ao->buffer);
1042   free(ao);
1043   ao = NULL;
1044 }
1045
1046
1047 /* stop playing, keep buffers (for pause) */
1048 static void audio_pause(void)
1049 {
1050     OSErr err=noErr;
1051
1052     /* Stop callback. */
1053     if (!ao->b_digital)
1054     {
1055         err=AudioOutputUnitStop(ao->theOutputUnit);
1056         if (err != noErr)
1057             ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStop returned [%4.4s]\n", (char *)&err);
1058     }
1059     else
1060     {
1061         err = AudioDeviceStop(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF);
1062         if (err != noErr)
1063             ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
1064     }
1065     ao->paused = 1;
1066 }
1067
1068
1069 /* resume playing, after audio_pause() */
1070 static void audio_resume(void)
1071 {
1072     OSErr err=noErr;
1073
1074     if (!ao->paused)
1075         return;
1076
1077     /* Start callback. */
1078     if (!ao->b_digital)
1079     {
1080         err = AudioOutputUnitStart(ao->theOutputUnit);
1081         if (err != noErr)
1082             ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStart returned [%4.4s]\n", (char *)&err);
1083     }
1084     else
1085     {
1086         err = AudioDeviceStart(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF);
1087         if (err != noErr)
1088             ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err);
1089     }
1090     ao->paused = 0;
1091 }
1092
1093 /*****************************************************************************
1094  * StreamListener
1095  *****************************************************************************/
1096 static OSStatus StreamListener( AudioStreamID inStream,
1097                                 UInt32 inChannel,
1098                                 AudioDevicePropertyID inPropertyID,
1099                                 void * inClientData )
1100 {
1101     switch (inPropertyID)
1102     {
1103         case kAudioStreamPropertyPhysicalFormat:
1104             ao_msg(MSGT_AO, MSGL_V, "got notify kAudioStreamPropertyPhysicalFormat changed.\n");
1105             if (inClientData)
1106                 *(volatile int *)inClientData = 1;
1107         default:
1108             break;
1109     }
1110     return noErr;
1111 }
1112
1113 static OSStatus DeviceListener( AudioDeviceID inDevice,
1114                                 UInt32 inChannel,
1115                                 Boolean isInput,
1116                                 AudioDevicePropertyID inPropertyID,
1117                                 void* inClientData )
1118 {
1119     switch (inPropertyID)
1120     {
1121         case kAudioDevicePropertyDeviceHasChanged:
1122             ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioDevicePropertyDeviceHasChanged.\n");
1123             ao->b_stream_format_changed = 1;
1124         default:
1125             break;
1126     }
1127     return noErr;
1128 }