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 ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3)
 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         switch(format&AF_FORMAT_BITS_MASK){
 318         case AF_FORMAT_8BIT:
 319                 inDesc.mBitsPerChannel=8;
 320                 break;
 321         case AF_FORMAT_16BIT:
 322                 inDesc.mBitsPerChannel=16;
 323                 break;
 324         case AF_FORMAT_24BIT:
 325                 inDesc.mBitsPerChannel=24;
 326                 break;
 327         case AF_FORMAT_32BIT:
 328                 inDesc.mBitsPerChannel=32;
 329                 break;
 330         default:
 331                 ao_msg(MSGT_AO, MSGL_WARN, "Unsupported format (0x%08x)\n", format);
 332                 goto err_out;
 333         }
 334
 335     if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) {
 336         // float
 337                 inDesc.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked;
 338     }
 339     else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI) {
 340         // signed int
 341                 inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
 342     }
 343     else {
 344         // unsigned int
 345                 inDesc.mFormatFlags = kAudioFormatFlagIsPacked;
 346     }
 347     if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3) {
 348         // Currently ac3 input (comes from hwac3) is always in native byte-order.
 349 #if HAVE_BIGENDIAN
 350         inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
 351 #endif
 352     }
 353     else if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
 354         inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
 355
 356     inDesc.mFramesPerPacket = 1;
 357     ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8);
 358     print_format(MSGL_V, "source:",&inDesc);
 359
 360     if (ao->b_supports_digital)
 361     {
 362         b_alive = 1;
 363         i_param_size = sizeof(b_alive);
 364         err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
 365                                      kAudioDevicePropertyDeviceIsAlive,
 366                                      &i_param_size, &b_alive);
 367         if (err != noErr)
 368             ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err);
 369         if (!b_alive)
 370             ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" );
 371         /* S/PDIF output need device in HogMode. */
 372         i_param_size = sizeof(ao->i_hog_pid);
 373         err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
 374                                      kAudioDevicePropertyHogMode,
 375                                      &i_param_size, &ao->i_hog_pid);
 376
 377         if (err != noErr)
 378         {
 379             /* This is not a fatal error. Some drivers simply don't support this property. */
 380             ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n",
 381                      (char *)&err);
 382             ao->i_hog_pid = -1;
 383         }
 384
 385         if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid())
 386         {
 387             ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" );
 388             goto err_out;
 389         }
 390         ao->stream_format = inDesc;
 391         return OpenSPDIF();
 392     }
 393
 394         /* original analog output code */
 395         desc.componentType = kAudioUnitType_Output;
 396         desc.componentSubType = kAudioUnitSubType_DefaultOutput;
 397         desc.componentManufacturer = kAudioUnitManufacturer_Apple;
 398         desc.componentFlags = 0;
 399         desc.componentFlagsMask = 0;
 400
 401         comp = FindNextComponent(NULL, &desc);  //Finds an component that meets the desc spec's
 402         if (comp == NULL) {
 403                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n");
 404                 goto err_out;
 405         }
 406
 407         err = OpenAComponent(comp, &(ao->theOutputUnit));  //gains access to the services provided by the component
 408         if (err) {
 409                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err);
 410                 goto err_out;
 411         }
 412
 413         // Initialize AudioUnit
 414         err = AudioUnitInitialize(ao->theOutputUnit);
 415         if (err) {
 416                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err);
 417                 goto err_out1;
 418         }
 419
 420         size =  sizeof(AudioStreamBasicDescription);
 421         err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size);
 422
 423         if (err) {
 424                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err);
 425                 goto err_out2;
 426         }
 427
 428         size = sizeof(UInt32);
 429         err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size);
 430
 431         if (err)
 432         {
 433                 ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err);
 434                 goto err_out2;
 435         }
 436
 437         ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame;
 438
 439         ao_data.samplerate = inDesc.mSampleRate;
 440         ao_data.channels = inDesc.mChannelsPerFrame;
 441     ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame;
 442     ao_data.outburst = ao->chunk_size;
 443         ao_data.buffersize = ao_data.bps;
 444
 445         ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
 446     ao->buffer_len = ao->num_chunks * ao->chunk_size;
 447     ao->buffer = av_fifo_alloc(ao->buffer_len);
 448
 449         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);
 450
 451     renderCallback.inputProc = theRenderProc;
 452     renderCallback.inputProcRefCon = 0;
 453     err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct));
 454         if (err) {
 455                 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err);
 456                 goto err_out2;
 457         }
 458
 459         reset();
 460
 461     return CONTROL_OK;
 462
 463 err_out2:
 464     AudioUnitUninitialize(ao->theOutputUnit);
 465 err_out1:
 466     CloseComponent(ao->theOutputUnit);
 467 err_out:
 468     av_fifo_free(ao->buffer);
 469     free(ao);
 470     ao = NULL;
 471     return CONTROL_FALSE;
 472 }
 473
 474 /*****************************************************************************
 475  * Setup a encoded digital stream (SPDIF)
 476  *****************************************************************************/
 477 static int OpenSPDIF(void)
 478 {
 479     OSStatus                err = noErr;
 480     UInt32                  i_param_size, b_mix = 0;
 481     Boolean                 b_writeable = 0;
 482     AudioStreamID           *p_streams = NULL;
 483     int                     i, i_streams = 0;
 484
 485     /* Start doing the SPDIF setup process. */
 486     ao->b_digital = 1;
 487
 488     /* Hog the device. */
 489     i_param_size = sizeof(ao->i_hog_pid);
 490     ao->i_hog_pid = getpid() ;
 491
 492     err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
 493                                  kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid);
 494
 495     if (err != noErr)
 496     {
 497         ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err);
 498         ao->i_hog_pid = -1;
 499         goto err_out;
 500     }
 501
 502     /* Set mixable to false if we are allowed to. */
 503     err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE,
 504                                      kAudioDevicePropertySupportsMixing,
 505                                      &i_param_size, &b_writeable);
 506     err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
 507                                  kAudioDevicePropertySupportsMixing,
 508                                  &i_param_size, &b_mix);
 509     if (err != noErr && b_writeable)
 510     {
 511         b_mix = 0;
 512         err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
 513                                      kAudioDevicePropertySupportsMixing,
 514                                      i_param_size, &b_mix);
 515         ao->b_changed_mixing = 1;
 516     }
 517     if (err != noErr)
 518     {
 519         ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
 520         goto err_out;
 521     }
 522
 523     /* Get a list of all the streams on this device. */
 524     err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE,
 525                                      kAudioDevicePropertyStreams,
 526                                      &i_param_size, NULL);
 527     if (err != noErr)
 528     {
 529         ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err);
 530         goto err_out;
 531     }
 532
 533     i_streams = i_param_size / sizeof(AudioStreamID);
 534     p_streams = (AudioStreamID *)malloc(i_param_size);
 535     if (p_streams == NULL)
 536     {
 537         ao_msg(MSGT_AO, MSGL_WARN, "out of memory\n" );
 538         goto err_out;
 539     }
 540
 541     err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
 542                                  kAudioDevicePropertyStreams,
 543                                  &i_param_size, p_streams);
 544     if (err != noErr)
 545     {
 546         ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err);
 547         if (p_streams) free(p_streams);
 548         goto err_out;
 549     }
 550
 551     ao_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams);
 552
 553     for (i = 0; i < i_streams && ao->i_stream_index < 0; ++i)
 554     {
 555         /* Find a stream with a cac3 stream. */
 556         AudioStreamBasicDescription *p_format_list = NULL;
 557         int i_formats = 0, j = 0, b_digital = 0;
 558
 559         /* Retrieve all the stream formats supported by each output stream. */
 560         err = AudioStreamGetPropertyInfo(p_streams[i], 0,
 561                                          kAudioStreamPropertyPhysicalFormats,
 562                                          &i_param_size, NULL);
 563         if (err != noErr)
 564         {
 565             ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streamformats: [%4.4s]\n", (char *)&err);
 566             continue;
 567         }
 568
 569         i_formats = i_param_size / sizeof(AudioStreamBasicDescription);
 570         p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size);
 571         if (p_format_list == NULL)
 572         {
 573             ao_msg(MSGT_AO, MSGL_WARN, "could not malloc the memory\n" );
 574             continue;
 575         }
 576
 577         err = AudioStreamGetProperty(p_streams[i], 0,
 578                                      kAudioStreamPropertyPhysicalFormats,
 579                                      &i_param_size, p_format_list);
 580         if (err != noErr)
 581         {
 582             ao_msg(MSGT_AO, MSGL_WARN, "could not get the list of streamformats: [%4.4s]\n", (char *)&err);
 583             if (p_format_list) free(p_format_list);
 584             continue;
 585         }
 586
 587         /* Check if one of the supported formats is a digital format. */
 588         for (j = 0; j < i_formats; ++j)
 589         {
 590             if (p_format_list[j].mFormatID == 'IAC3' ||
 591                   p_format_list[j].mFormatID == kAudioFormat60958AC3)
 592             {
 593                 b_digital = 1;
 594                 break;
 595             }
 596         }
 597
 598         if (b_digital)
 599         {
 600             /* If this stream supports a digital (cac3) format, then set it. */
 601             int i_requested_rate_format = -1;
 602             int i_current_rate_format = -1;
 603             int i_backup_rate_format = -1;
 604
 605             ao->i_stream_id = p_streams[i];
 606             ao->i_stream_index = i;
 607
 608             if (ao->b_revert == 0)
 609             {
 610                 /* Retrieve the original format of this stream first if not done so already. */
 611                 i_param_size = sizeof(ao->sfmt_revert);
 612                 err = AudioStreamGetProperty(ao->i_stream_id, 0,
 613                                              kAudioStreamPropertyPhysicalFormat,
 614                                              &i_param_size,
 615                                              &ao->sfmt_revert);
 616                 if (err != noErr)
 617                 {
 618                     ao_msg(MSGT_AO, MSGL_WARN, "could not retrieve the original streamformat: [%4.4s]\n", (char *)&err);
 619                     if (p_format_list) free(p_format_list);
 620                     continue;
 621                 }
 622                 ao->b_revert = 1;
 623             }
 624
 625             for (j = 0; j < i_formats; ++j)
 626                 if (p_format_list[j].mFormatID == 'IAC3' ||
 627                       p_format_list[j].mFormatID == kAudioFormat60958AC3)
 628                 {
 629                     if (p_format_list[j].mSampleRate == ao->stream_format.mSampleRate)
 630                     {
 631                         i_requested_rate_format = j;
 632                         break;
 633                     }
 634                     if (p_format_list[j].mSampleRate == ao->sfmt_revert.mSampleRate)
 635                         i_current_rate_format = j;
 636                     else if (i_backup_rate_format < 0 || p_format_list[j].mSampleRate > p_format_list[i_backup_rate_format].mSampleRate)
 637                         i_backup_rate_format = j;
 638                 }
 639
 640             if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio. */
 641                 ao->stream_format = p_format_list[i_requested_rate_format];
 642             else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device. */
 643                 ao->stream_format = p_format_list[i_current_rate_format];
 644             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). */
 645         }
 646         if (p_format_list) free(p_format_list);
 647     }
 648     if (p_streams) free(p_streams);
 649
 650     if (ao->i_stream_index < 0)
 651     {
 652         ao_msg(MSGT_AO, MSGL_WARN, "can not find any digital output stream format when OpenSPDIF().\n");
 653         goto err_out;
 654     }
 655
 656     print_format(MSGL_V, "original stream format:", &ao->sfmt_revert);
 657
 658     if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
 659         goto err_out;
 660
 661     err = AudioDeviceAddPropertyListener(ao->i_selected_dev,
 662                                          kAudioPropertyWildcardChannel,
 663                                          0,
 664                                          kAudioDevicePropertyDeviceHasChanged,
 665                                          DeviceListener,
 666                                          NULL);
 667     if (err != noErr)
 668         ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n", (char *)&err);
 669
 670
 671     /* FIXME: If output stream is not native byte-order, we need change endian somewhere. */
 672     /*        Although there's no such case reported.                                     */
 673 #if HAVE_BIGENDIAN
 674     if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian))
 675 #else
 676     if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
 677 #endif
 678         ao_msg(MSGT_AO, MSGL_WARN, "output stream has a no-native byte-order, digital output may failed.\n");
 679
 680     /* For ac3/dts, just use packet size 6144 bytes as chunk size. */
 681     ao->chunk_size = ao->stream_format.mBytesPerPacket;
 682
 683     ao_data.samplerate = ao->stream_format.mSampleRate;
 684     ao_data.channels = ao->stream_format.mChannelsPerFrame;
 685     ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket);
 686     ao_data.outburst = ao->chunk_size;
 687     ao_data.buffersize = ao_data.bps;
 688
 689     ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
 690     ao->buffer_len = ao->num_chunks * ao->chunk_size;
 691     ao->buffer = av_fifo_alloc(ao->buffer_len);
 692
 693     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);
 694
 695
 696     /* Add IOProc callback. */
 697     err = AudioDeviceAddIOProc(ao->i_selected_dev,
 698                                (AudioDeviceIOProc)RenderCallbackSPDIF,
 699                                (void *)ao);
 700     if (err != noErr)
 701     {
 702         ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err);
 703         goto err_out1;
 704     }
 705
 706     reset();
 707
 708     return CONTROL_TRUE;
 709
 710 err_out1:
 711     if (ao->b_revert)
 712         AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
 713 err_out:
 714     if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
 715     {
 716         int b_mix = 1;
 717         err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
 718                                      kAudioDevicePropertySupportsMixing,
 719                                      i_param_size, &b_mix);
 720         if (err != noErr)
 721             ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n",
 722                    (char *)&err);
 723     }
 724     if (ao->i_hog_pid == getpid())
 725     {
 726         ao->i_hog_pid = -1;
 727         i_param_size = sizeof(ao->i_hog_pid);
 728         err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
 729                                      kAudioDevicePropertyHogMode,
 730                                      i_param_size, &ao->i_hog_pid);
 731         if (err != noErr)
 732             ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n",
 733                    (char *)&err);
 734     }
 735     av_fifo_free(ao->buffer);
 736     free(ao);
 737     ao = NULL;
 738     return CONTROL_FALSE;
 739 }
 740
 741 /*****************************************************************************
 742  * AudioDeviceSupportsDigital: Check i_dev_id for digital stream support.
 743  *****************************************************************************/
 744 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id )
 745 {
 746     OSStatus                    err = noErr;
 747     UInt32                      i_param_size = 0;
 748     AudioStreamID               *p_streams = NULL;
 749     int                         i = 0, i_streams = 0;
 750     int                         b_return = CONTROL_FALSE;
 751
 752     /* Retrieve all the output streams. */
 753     err = AudioDeviceGetPropertyInfo(i_dev_id, 0, FALSE,
 754                                      kAudioDevicePropertyStreams,
 755                                      &i_param_size, NULL);
 756     if (err != noErr)
 757     {
 758         ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err);
 759         return CONTROL_FALSE;
 760     }
 761
 762     i_streams = i_param_size / sizeof(AudioStreamID);
 763     p_streams = (AudioStreamID *)malloc(i_param_size);
 764     if (p_streams == NULL)
 765     {
 766         ao_msg(MSGT_AO,MSGL_V, "out of memory\n");
 767         return CONTROL_FALSE;
 768     }
 769
 770     err = AudioDeviceGetProperty(i_dev_id, 0, FALSE,
 771                                  kAudioDevicePropertyStreams,
 772                                  &i_param_size, p_streams);
 773
 774     if (err != noErr)
 775     {
 776         ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err);
 777         free(p_streams);
 778         return CONTROL_FALSE;
 779     }
 780
 781     for (i = 0; i < i_streams; ++i)
 782     {
 783         if (AudioStreamSupportsDigital(p_streams[i]))
 784             b_return = CONTROL_OK;
 785     }
 786
 787     free(p_streams);
 788     return b_return;
 789 }
 790
 791 /*****************************************************************************
 792  * AudioStreamSupportsDigital: Check i_stream_id for digital stream support.
 793  *****************************************************************************/
 794 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id )
 795 {
 796     OSStatus err = noErr;
 797     UInt32 i_param_size;
 798     AudioStreamBasicDescription *p_format_list = NULL;
 799     int i, i_formats, b_return = CONTROL_FALSE;
 800
 801     /* Retrieve all the stream formats supported by each output stream. */
 802     err = AudioStreamGetPropertyInfo(i_stream_id, 0,
 803                                      kAudioStreamPropertyPhysicalFormats,
 804                                      &i_param_size, NULL);
 805     if (err != noErr)
 806     {
 807         ao_msg(MSGT_AO,MSGL_V, "could not get number of streamformats: [%4.4s]\n", (char *)&err);
 808         return CONTROL_FALSE;
 809     }
 810
 811     i_formats = i_param_size / sizeof(AudioStreamBasicDescription);
 812     p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size);
 813     if (p_format_list == NULL)
 814     {
 815         ao_msg(MSGT_AO,MSGL_V, "could not malloc the memory\n" );
 816         return CONTROL_FALSE;
 817     }
 818
 819     err = AudioStreamGetProperty(i_stream_id, 0,
 820                                  kAudioStreamPropertyPhysicalFormats,
 821                                  &i_param_size, p_format_list);
 822     if (err != noErr)
 823     {
 824         ao_msg(MSGT_AO,MSGL_V, "could not get the list of streamformats: [%4.4s]\n", (char *)&err);
 825         free(p_format_list);
 826         return CONTROL_FALSE;
 827     }
 828
 829     for (i = 0; i < i_formats; ++i)
 830     {
 831         print_format(MSGL_V, "supported format:", &p_format_list[i]);
 832
 833         if (p_format_list[i].mFormatID == 'IAC3' ||
 834                   p_format_list[i].mFormatID == kAudioFormat60958AC3)
 835             b_return = CONTROL_OK;
 836     }
 837
 838     free(p_format_list);
 839     return b_return;
 840 }
 841
 842 /*****************************************************************************
 843  * AudioStreamChangeFormat: Change i_stream_id to change_format
 844  *****************************************************************************/
 845 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format )
 846 {
 847     OSStatus err = noErr;
 848     UInt32 i_param_size = 0;
 849     int i;
 850
 851     static volatile int stream_format_changed;
 852     stream_format_changed = 0;
 853
 854     print_format(MSGL_V, "setting stream format:", &change_format);
 855
 856     /* Install the callback. */
 857     err = AudioStreamAddPropertyListener(i_stream_id, 0,
 858                                          kAudioStreamPropertyPhysicalFormat,
 859                                          StreamListener,
 860                                          (void *)&stream_format_changed);
 861     if (err != noErr)
 862     {
 863         ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamAddPropertyListener failed: [%4.4s]\n", (char *)&err);
 864         return CONTROL_FALSE;
 865     }
 866
 867     /* Change the format. */
 868     err = AudioStreamSetProperty(i_stream_id, 0, 0,
 869                                  kAudioStreamPropertyPhysicalFormat,
 870                                  sizeof(AudioStreamBasicDescription),
 871                                  &change_format);
 872     if (err != noErr)
 873     {
 874         ao_msg(MSGT_AO, MSGL_WARN, "could not set the stream format: [%4.4s]\n", (char *)&err);
 875         return CONTROL_FALSE;
 876     }
 877
 878     /* The AudioStreamSetProperty is not only asynchronious,
 879      * it is also not Atomic, in its behaviour.
 880      * Therefore we check 5 times before we really give up.
 881      * FIXME: failing isn't actually implemented yet. */
 882     for (i = 0; i < 5; ++i)
 883     {
 884         AudioStreamBasicDescription actual_format;
 885         int j;
 886         for (j = 0; !stream_format_changed && j < 50; ++j)
 887             usec_sleep(10000);
 888         if (stream_format_changed)
 889             stream_format_changed = 0;
 890         else
 891             ao_msg(MSGT_AO, MSGL_V, "reached timeout\n" );
 892
 893         i_param_size = sizeof(AudioStreamBasicDescription);
 894         err = AudioStreamGetProperty(i_stream_id, 0,
 895                                      kAudioStreamPropertyPhysicalFormat,
 896                                      &i_param_size,
 897                                      &actual_format);
 898
 899         print_format(MSGL_V, "actual format in use:", &actual_format);
 900         if (actual_format.mSampleRate == change_format.mSampleRate &&
 901             actual_format.mFormatID == change_format.mFormatID &&
 902             actual_format.mFramesPerPacket == change_format.mFramesPerPacket)
 903         {
 904             /* The right format is now active. */
 905             break;
 906         }
 907         /* We need to check again. */
 908     }
 909
 910     /* Removing the property listener. */
 911     err = AudioStreamRemovePropertyListener(i_stream_id, 0,
 912                                             kAudioStreamPropertyPhysicalFormat,
 913                                             StreamListener);
 914     if (err != noErr)
 915     {
 916         ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamRemovePropertyListener failed: [%4.4s]\n", (char *)&err);
 917         return CONTROL_FALSE;
 918     }
 919
 920     return CONTROL_TRUE;
 921 }
 922
 923 /*****************************************************************************
 924  * RenderCallbackSPDIF: callback for SPDIF audio output
 925  *****************************************************************************/
 926 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
 927                                     const AudioTimeStamp * inNow,
 928                                     const void * inInputData,
 929                                     const AudioTimeStamp * inInputTime,
 930                                     AudioBufferList * outOutputData,
 931                                     const AudioTimeStamp * inOutputTime,
 932                                     void * threadGlobals )
 933 {
 934     int amt = av_fifo_size(ao->buffer);
 935     int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize;
 936
 937     if (amt > req)
 938         amt = req;
 939     if (amt)
 940         read_buffer(ao->b_muted ? NULL : (unsigned char *)outOutputData->mBuffers[ao->i_stream_index].mData, amt);
 941
 942     return noErr;
 943 }
 944
 945
 946 static int play(void* output_samples,int num_bytes,int flags)
 947 {
 948     int wrote, b_digital;
 949
 950     // Check whether we need to reset the digital output stream.
 951     if (ao->b_digital && ao->b_stream_format_changed)
 952     {
 953         ao->b_stream_format_changed = 0;
 954         b_digital = AudioStreamSupportsDigital(ao->i_stream_id);
 955         if (b_digital)
 956         {
 957             /* Current stream support digital format output, let's set it. */
 958             ao_msg(MSGT_AO, MSGL_V, "detected current stream support digital, try to restore digital output...\n");
 959
 960             if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
 961             {
 962                 ao_msg(MSGT_AO, MSGL_WARN, "restore digital output failed.\n");
 963             }
 964             else
 965             {
 966                 ao_msg(MSGT_AO, MSGL_WARN, "restore digital output succeed.\n");
 967                 reset();
 968             }
 969         }
 970         else
 971             ao_msg(MSGT_AO, MSGL_V, "detected current stream do not support digital.\n");
 972     }
 973
 974     wrote=write_buffer(output_samples, num_bytes);
 975     audio_resume();
 976     return wrote;
 977 }
 978
 979 /* set variables and buffer to initial state */
 980 static void reset(void)
 981 {
 982   audio_pause();
 983   av_fifo_reset(ao->buffer);
 984 }
 985
 986
 987 /* return available space */
 988 static int get_space(void)
 989 {
 990   return ao->buffer_len - av_fifo_size(ao->buffer);
 991 }
 992
 993
 994 /* return delay until audio is played */
 995 static float get_delay(void)
 996 {
 997   // inaccurate, should also contain the data buffered e.g. by the OS
 998   return (float)av_fifo_size(ao->buffer)/(float)ao_data.bps;
 999 }
1000
1001
1002 /* unload plugin and deregister from coreaudio */
1003 static void uninit(int immed)
1004 {
1005   OSStatus            err = noErr;
1006   UInt32              i_param_size = 0;
1007
1008   if (!immed) {
1009     long long timeleft=(1000000LL*av_fifo_size(ao->buffer))/ao_data.bps;
1010     ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", av_fifo_size(ao->buffer), ao_data.bps, (int)timeleft);
1011     usec_sleep((int)timeleft);
1012   }
1013
1014   if (!ao->b_digital) {
1015       AudioOutputUnitStop(ao->theOutputUnit);
1016       AudioUnitUninitialize(ao->theOutputUnit);
1017       CloseComponent(ao->theOutputUnit);
1018   }
1019   else {
1020       /* Stop device. */
1021       err = AudioDeviceStop(ao->i_selected_dev,
1022                             (AudioDeviceIOProc)RenderCallbackSPDIF);
1023       if (err != noErr)
1024           ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
1025
1026       /* Remove IOProc callback. */
1027       err = AudioDeviceRemoveIOProc(ao->i_selected_dev,
1028                                     (AudioDeviceIOProc)RenderCallbackSPDIF);
1029       if (err != noErr)
1030           ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceRemoveIOProc failed: [%4.4s]\n", (char *)&err);
1031
1032       if (ao->b_revert)
1033           AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
1034
1035       if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
1036       {
1037           int b_mix;
1038           Boolean b_writeable;
1039           /* Revert mixable to true if we are allowed to. */
1040           err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing,
1041                                            &i_param_size, &b_writeable);
1042           err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing,
1043                                        &i_param_size, &b_mix);
1044           if (err != noErr && b_writeable)
1045           {
1046               b_mix = 1;
1047               err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
1048                                            kAudioDevicePropertySupportsMixing, i_param_size, &b_mix);
1049           }
1050           if (err != noErr)
1051               ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
1052       }
1053       if (ao->i_hog_pid == getpid())
1054       {
1055           ao->i_hog_pid = -1;
1056           i_param_size = sizeof(ao->i_hog_pid);
1057           err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
1058                                        kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid);
1059           if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err);
1060       }
1061   }
1062
1063   av_fifo_free(ao->buffer);
1064   free(ao);
1065   ao = NULL;
1066 }
1067
1068
1069 /* stop playing, keep buffers (for pause) */
1070 static void audio_pause(void)
1071 {
1072     OSErr err=noErr;
1073
1074     /* Stop callback. */
1075     if (!ao->b_digital)
1076     {
1077         err=AudioOutputUnitStop(ao->theOutputUnit);
1078         if (err != noErr)
1079             ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStop returned [%4.4s]\n", (char *)&err);
1080     }
1081     else
1082     {
1083         err = AudioDeviceStop(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF);
1084         if (err != noErr)
1085             ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
1086     }
1087     ao->paused = 1;
1088 }
1089
1090
1091 /* resume playing, after audio_pause() */
1092 static void audio_resume(void)
1093 {
1094     OSErr err=noErr;
1095
1096     if (!ao->paused)
1097         return;
1098
1099     /* Start callback. */
1100     if (!ao->b_digital)
1101     {
1102         err = AudioOutputUnitStart(ao->theOutputUnit);
1103         if (err != noErr)
1104             ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStart returned [%4.4s]\n", (char *)&err);
1105     }
1106     else
1107     {
1108         err = AudioDeviceStart(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF);
1109         if (err != noErr)
1110             ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err);
1111     }
1112     ao->paused = 0;
1113 }
1114
1115 /*****************************************************************************
1116  * StreamListener
1117  *****************************************************************************/
1118 static OSStatus StreamListener( AudioStreamID inStream,
1119                                 UInt32 inChannel,
1120                                 AudioDevicePropertyID inPropertyID,
1121                                 void * inClientData )
1122 {
1123     switch (inPropertyID)
1124     {
1125         case kAudioStreamPropertyPhysicalFormat:
1126             ao_msg(MSGT_AO, MSGL_V, "got notify kAudioStreamPropertyPhysicalFormat changed.\n");
1127             if (inClientData)
1128                 *(volatile int *)inClientData = 1;
1129         default:
1130             break;
1131     }
1132     return noErr;
1133 }
1134
1135 static OSStatus DeviceListener( AudioDeviceID inDevice,
1136                                 UInt32 inChannel,
1137                                 Boolean isInput,
1138                                 AudioDevicePropertyID inPropertyID,
1139                                 void* inClientData )
1140 {
1141     switch (inPropertyID)
1142     {
1143         case kAudioDevicePropertyDeviceHasChanged:
1144             ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioDevicePropertyDeviceHasChanged.\n");
1145             ao->b_stream_format_changed = 1;
1146         default:
1147             break;
1148     }
1149     return noErr;
1150 }