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