libao2/ao_dsound.c

   1 /*
   2  * Windows DirectSound interface
   3  *
   4  * Copyright (c) 2004 Gabor Szecsi <deje@miki.hu>
   5  *
   6  * This file is part of MPlayer.
   7  *
   8  * MPlayer is free software; you can redistribute it and/or modify
   9  * it under the terms of the GNU General Public License as published by
  10  * the Free Software Foundation; either version 2 of the License, or
  11  * (at your option) any later version.
  12  *
  13  * MPlayer is distributed in the hope that it will be useful,
  14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  16  * GNU General Public License for more details.
  17  *
  18  * You should have received a copy of the GNU General Public License along
  19  * with MPlayer; if not, write to the Free Software Foundation, Inc.,
  20  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  21  */
  22
  23 /**
  24 \todo verify/extend multichannel support
  25 */
  26
  27
  28 #include <stdio.h>
  29 #include <stdlib.h>
  30 #include <windows.h>
  31 #define DIRECTSOUND_VERSION 0x0600
  32 #include <dsound.h>
  33 #include <math.h>
  34
  35 #include "config.h"
  36 #include "libaf/af_format.h"
  37 #include "audio_out.h"
  38 #include "audio_out_internal.h"
  39 #include "mp_msg.h"
  40 #include "libvo/fastmemcpy.h"
  41 #include "osdep/timer.h"
  42 #include "subopt-helper.h"
  43
  44
  45 static const ao_info_t info =
  46 {
  47         "Windows DirectSound audio output",
  48         "dsound",
  49         "Gabor Szecsi <deje@miki.hu>",
  50         ""
  51 };
  52
  53 LIBAO_EXTERN(dsound)
  54
  55 /**
  56 \todo use the definitions from the win32 api headers when they define these
  57 */
  58 #define WAVE_FORMAT_IEEE_FLOAT 0x0003
  59 #define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092
  60 #define WAVE_FORMAT_EXTENSIBLE 0xFFFE
  61
  62 static const GUID KSDATAFORMAT_SUBTYPE_PCM = {0x1,0x0000,0x0010, {0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}};
  63
  64 #define SPEAKER_FRONT_LEFT             0x1
  65 #define SPEAKER_FRONT_RIGHT            0x2
  66 #define SPEAKER_FRONT_CENTER           0x4
  67 #define SPEAKER_LOW_FREQUENCY          0x8
  68 #define SPEAKER_BACK_LEFT              0x10
  69 #define SPEAKER_BACK_RIGHT             0x20
  70 #define SPEAKER_FRONT_LEFT_OF_CENTER   0x40
  71 #define SPEAKER_FRONT_RIGHT_OF_CENTER  0x80
  72 #define SPEAKER_BACK_CENTER            0x100
  73 #define SPEAKER_SIDE_LEFT              0x200
  74 #define SPEAKER_SIDE_RIGHT             0x400
  75 #define SPEAKER_TOP_CENTER             0x800
  76 #define SPEAKER_TOP_FRONT_LEFT         0x1000
  77 #define SPEAKER_TOP_FRONT_CENTER       0x2000
  78 #define SPEAKER_TOP_FRONT_RIGHT        0x4000
  79 #define SPEAKER_TOP_BACK_LEFT          0x8000
  80 #define SPEAKER_TOP_BACK_CENTER        0x10000
  81 #define SPEAKER_TOP_BACK_RIGHT         0x20000
  82 #define SPEAKER_RESERVED               0x80000000
  83
  84 #define DSSPEAKER_HEADPHONE         0x00000001
  85 #define DSSPEAKER_MONO              0x00000002
  86 #define DSSPEAKER_QUAD              0x00000003
  87 #define DSSPEAKER_STEREO            0x00000004
  88 #define DSSPEAKER_SURROUND          0x00000005
  89 #define DSSPEAKER_5POINT1           0x00000006
  90
  91 #ifndef _WAVEFORMATEXTENSIBLE_
  92 typedef struct {
  93     WAVEFORMATEX    Format;
  94     union {
  95         WORD wValidBitsPerSample;       /* bits of precision  */
  96         WORD wSamplesPerBlock;          /* valid if wBitsPerSample==0 */
  97         WORD wReserved;                 /* If neither applies, set to zero. */
  98     } Samples;
  99     DWORD           dwChannelMask;      /* which channels are */
 100                                         /* present in stream  */
 101     GUID            SubFormat;
 102 } WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
 103 #endif
 104
 105 static const int channel_mask[] = {
 106   SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT  | SPEAKER_LOW_FREQUENCY,
 107   SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT,
 108   SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT   | SPEAKER_LOW_FREQUENCY,
 109   SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT     | SPEAKER_LOW_FREQUENCY
 110 };
 111
 112 static HINSTANCE hdsound_dll = NULL;      ///handle to the dll
 113 static LPDIRECTSOUND hds = NULL;          ///direct sound object
 114 static LPDIRECTSOUNDBUFFER hdspribuf = NULL; ///primary direct sound buffer
 115 static LPDIRECTSOUNDBUFFER hdsbuf = NULL; ///secondary direct sound buffer (stream buffer)
 116 static int buffer_size = 0;               ///size in bytes of the direct sound buffer
 117 static int write_offset = 0;              ///offset of the write cursor in the direct sound buffer
 118 static int min_free_space = 0;            ///if the free space is below this value get_space() will return 0
 119                                           ///there will always be at least this amout of free space to prevent
 120                                           ///get_space() from returning wrong values when buffer is 100% full.
 121                                           ///will be replaced with nBlockAlign in init()
 122 static int device_num = 0;                ///wanted device number
 123 static GUID device;                       ///guid of the device
 124
 125 /***************************************************************************************/
 126
 127 /**
 128 \brief output error message
 129 \param err error code
 130 \return string with the error message
 131 */
 132 static char * dserr2str(int err)
 133 {
 134         switch (err) {
 135                 case DS_OK: return "DS_OK";
 136                 case DS_NO_VIRTUALIZATION: return "DS_NO_VIRTUALIZATION";
 137                 case DSERR_ALLOCATED: return "DS_NO_VIRTUALIZATION";
 138                 case DSERR_CONTROLUNAVAIL: return "DSERR_CONTROLUNAVAIL";
 139                 case DSERR_INVALIDPARAM: return "DSERR_INVALIDPARAM";
 140                 case DSERR_INVALIDCALL: return "DSERR_INVALIDCALL";
 141                 case DSERR_GENERIC: return "DSERR_GENERIC";
 142                 case DSERR_PRIOLEVELNEEDED: return "DSERR_PRIOLEVELNEEDED";
 143                 case DSERR_OUTOFMEMORY: return "DSERR_OUTOFMEMORY";
 144                 case DSERR_BADFORMAT: return "DSERR_BADFORMAT";
 145                 case DSERR_UNSUPPORTED: return "DSERR_UNSUPPORTED";
 146                 case DSERR_NODRIVER: return "DSERR_NODRIVER";
 147                 case DSERR_ALREADYINITIALIZED: return "DSERR_ALREADYINITIALIZED";
 148                 case DSERR_NOAGGREGATION: return "DSERR_NOAGGREGATION";
 149                 case DSERR_BUFFERLOST: return "DSERR_BUFFERLOST";
 150                 case DSERR_OTHERAPPHASPRIO: return "DSERR_OTHERAPPHASPRIO";
 151                 case DSERR_UNINITIALIZED: return "DSERR_UNINITIALIZED";
 152                 case DSERR_NOINTERFACE: return "DSERR_NOINTERFACE";
 153                 case DSERR_ACCESSDENIED: return "DSERR_ACCESSDENIED";
 154                 default: return "unknown";
 155         }
 156 }
 157
 158 /**
 159 \brief uninitialize direct sound
 160 */
 161 static void UninitDirectSound(void)
 162 {
 163     // finally release the DirectSound object
 164     if (hds) {
 165         IDirectSound_Release(hds);
 166         hds = NULL;
 167     }
 168     // free DSOUND.DLL
 169     if (hdsound_dll) {
 170         FreeLibrary(hdsound_dll);
 171         hdsound_dll = NULL;
 172     }
 173         mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound uninitialized\n");
 174 }
 175
 176 /**
 177 \brief print the commandline help
 178 */
 179 static void print_help(void)
 180 {
 181   mp_msg(MSGT_AO, MSGL_FATAL,
 182            "\n-ao dsound commandline help:\n"
 183            "Example: mplayer -ao dsound:device=1\n"
 184            "  sets 1st device\n"
 185            "\nOptions:\n"
 186            "  device=<device-number>\n"
 187            "    Sets device number, use -v to get a list\n");
 188 }
 189
 190
 191 /**
 192 \brief enumerate direct sound devices
 193 \return TRUE to continue with the enumeration
 194 */
 195 static BOOL CALLBACK DirectSoundEnum(LPGUID guid,LPCSTR desc,LPCSTR module,LPVOID context)
 196 {
 197     int* device_index=context;
 198     mp_msg(MSGT_AO, MSGL_V,"%i %s ",*device_index,desc);
 199     if(device_num==*device_index){
 200         mp_msg(MSGT_AO, MSGL_V,"<--");
 201         if(guid){
 202             memcpy(&device,guid,sizeof(GUID));
 203         }
 204     }
 205     mp_msg(MSGT_AO, MSGL_V,"\n");
 206     (*device_index)++;
 207     return TRUE;
 208 }
 209
 210
 211 /**
 212 \brief initilize direct sound
 213 \return 0 if error, 1 if ok
 214 */
 215 static int InitDirectSound(void)
 216 {
 217         DSCAPS dscaps;
 218
 219         // initialize directsound
 220     HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
 221         HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
 222         int device_index=0;
 223         const opt_t subopts[] = {
 224           {"device", OPT_ARG_INT, &device_num,NULL},
 225           {NULL}
 226         };
 227         if (subopt_parse(ao_subdevice, subopts) != 0) {
 228                 print_help();
 229                 return 0;
 230         }
 231
 232         hdsound_dll = LoadLibrary("DSOUND.DLL");
 233         if (hdsound_dll == NULL) {
 234                 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot load DSOUND.DLL\n");
 235                 return 0;
 236         }
 237         OurDirectSoundCreate = (void*)GetProcAddress(hdsound_dll, "DirectSoundCreate");
 238         OurDirectSoundEnumerate = (void*)GetProcAddress(hdsound_dll, "DirectSoundEnumerateA");
 239
 240         if (OurDirectSoundCreate == NULL || OurDirectSoundEnumerate == NULL) {
 241                 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: GetProcAddress FAILED\n");
 242                 FreeLibrary(hdsound_dll);
 243                 return 0;
 244         }
 245
 246         // Enumerate all directsound devices
 247         mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Output Devices:\n");
 248         OurDirectSoundEnumerate(DirectSoundEnum,&device_index);
 249
 250         // Create the direct sound object
 251         if FAILED(OurDirectSoundCreate((device_num)?&device:NULL, &hds, NULL )) {
 252                 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create a DirectSound device\n");
 253                 FreeLibrary(hdsound_dll);
 254                 return 0;
 255         }
 256
 257         /* Set DirectSound Cooperative level, ie what control we want over Windows
 258          * sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the
 259          * settings of the primary buffer, but also that only the sound of our
 260          * application will be hearable when it will have the focus.
 261          * !!! (this is not really working as intended yet because to set the
 262          * cooperative level you need the window handle of your application, and
 263          * I don't know of any easy way to get it. Especially since we might play
 264          * sound without any video, and so what window handle should we use ???
 265          * The hack for now is to use the Desktop window handle - it seems to be
 266          * working */
 267         if (IDirectSound_SetCooperativeLevel(hds, GetDesktopWindow(), DSSCL_EXCLUSIVE)) {
 268                 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot set direct sound cooperative level\n");
 269                 IDirectSound_Release(hds);
 270                 FreeLibrary(hdsound_dll);
 271                 return 0;
 272         }
 273         mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound initialized\n");
 274
 275         memset(&dscaps, 0, sizeof(DSCAPS));
 276         dscaps.dwSize = sizeof(DSCAPS);
 277         if (DS_OK == IDirectSound_GetCaps(hds, &dscaps)) {
 278                 if (dscaps.dwFlags & DSCAPS_EMULDRIVER) mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound is emulated, waveOut may give better performance\n");
 279         } else {
 280                 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: cannot get device capabilities\n");
 281         }
 282
 283         return 1;
 284 }
 285
 286 /**
 287 \brief destroy the direct sound buffer
 288 */
 289 static void DestroyBuffer(void)
 290 {
 291         if (hdsbuf) {
 292                 IDirectSoundBuffer_Release(hdsbuf);
 293                 hdsbuf = NULL;
 294         }
 295         if (hdspribuf) {
 296                 IDirectSoundBuffer_Release(hdspribuf);
 297                 hdspribuf = NULL;
 298         }
 299 }
 300
 301 /**
 302 \brief fill sound buffer
 303 \param data pointer to the sound data to copy
 304 \param len length of the data to copy in bytes
 305 \return number of copyed bytes
 306 */
 307 static int write_buffer(unsigned char *data, int len)
 308 {
 309   HRESULT res;
 310   LPVOID lpvPtr1;
 311   DWORD dwBytes1;
 312   LPVOID lpvPtr2;
 313   DWORD dwBytes2;
 314
 315   // Lock the buffer
 316   res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
 317   // If the buffer was lost, restore and retry lock.
 318   if (DSERR_BUFFERLOST == res)
 319   {
 320     IDirectSoundBuffer_Restore(hdsbuf);
 321         res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
 322   }
 323
 324
 325   if (SUCCEEDED(res))
 326   {
 327         if( (ao_data.channels == 6) && !AF_FORMAT_IS_AC3(ao_data.format) ) {
 328             // reorder channels while writing to pointers.
 329             // it's this easy because buffer size and len are always
 330             // aligned to multiples of channels*bytespersample
 331             // there's probably some room for speed improvements here
 332             const int chantable[6] = {0, 1, 4, 5, 2, 3}; // reorder "matrix"
 333             int i, j;
 334             int numsamp,sampsize;
 335
 336             sampsize = af_fmt2bits(ao_data.format)>>3; // bytes per sample
 337             numsamp = dwBytes1 / (ao_data.channels * sampsize);  // number of samples for each channel in this buffer
 338
 339             for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) {
 340                 memcpy(lpvPtr1+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize);
 341             }
 342
 343             if (NULL != lpvPtr2 )
 344             {
 345                 numsamp = dwBytes2 / (ao_data.channels * sampsize);
 346                 for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) {
 347                     memcpy(lpvPtr2+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+dwBytes1+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize);
 348                 }
 349             }
 350
 351             write_offset+=dwBytes1+dwBytes2;
 352             if(write_offset>=buffer_size)write_offset=dwBytes2;
 353         } else {
 354             // Write to pointers without reordering.
 355         fast_memcpy(lpvPtr1,data,dwBytes1);
 356     if (NULL != lpvPtr2 )fast_memcpy(lpvPtr2,data+dwBytes1,dwBytes2);
 357         write_offset+=dwBytes1+dwBytes2;
 358     if(write_offset>=buffer_size)write_offset=dwBytes2;
 359         }
 360
 361    // Release the data back to DirectSound.
 362     res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2);
 363     if (SUCCEEDED(res))
 364     {
 365           // Success.
 366           DWORD status;
 367           IDirectSoundBuffer_GetStatus(hdsbuf, &status);
 368       if (!(status & DSBSTATUS_PLAYING)){
 369             res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
 370           }
 371           return dwBytes1+dwBytes2;
 372     }
 373   }
 374   // Lock, Unlock, or Restore failed.
 375   return 0;
 376 }
 377
 378 /***************************************************************************************/
 379
 380 /**
 381 \brief handle control commands
 382 \param cmd command
 383 \param arg argument
 384 \return CONTROL_OK or -1 in case the command can't be handled
 385 */
 386 static int control(int cmd, void *arg)
 387 {
 388         DWORD volume;
 389         switch (cmd) {
 390                 case AOCONTROL_GET_VOLUME: {
 391                         ao_control_vol_t* vol = (ao_control_vol_t*)arg;
 392                         IDirectSoundBuffer_GetVolume(hdsbuf, &volume);
 393                         vol->left = vol->right = pow(10.0, (float)(volume+10000) / 5000.0);
 394                         //printf("ao_dsound: volume: %f\n",vol->left);
 395                         return CONTROL_OK;
 396                 }
 397                 case AOCONTROL_SET_VOLUME: {
 398                         ao_control_vol_t* vol = (ao_control_vol_t*)arg;
 399                         volume = (DWORD)(log10(vol->right) * 5000.0) - 10000;
 400                         IDirectSoundBuffer_SetVolume(hdsbuf, volume);
 401                         //printf("ao_dsound: volume: %f\n",vol->left);
 402                         return CONTROL_OK;
 403                 }
 404         }
 405         return -1;
 406 }
 407
 408 /**
 409 \brief setup sound device
 410 \param rate samplerate
 411 \param channels number of channels
 412 \param format format
 413 \param flags unused
 414 \return 1=success 0=fail
 415 */
 416 static int init(int rate, int channels, int format, int flags)
 417 {
 418     int res;
 419         if (!InitDirectSound()) return 0;
 420
 421         // ok, now create the buffers
 422         WAVEFORMATEXTENSIBLE wformat;
 423         DSBUFFERDESC dsbpridesc;
 424         DSBUFFERDESC dsbdesc;
 425
 426         //check if the channel count and format is supported in general
 427         if (channels > 6) {
 428                 UninitDirectSound();
 429                 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: 8 channel audio not yet supported\n");
 430                 return 0;
 431         }
 432
 433         if (AF_FORMAT_IS_AC3(format))
 434                 format = AF_FORMAT_AC3_NE;
 435         switch(format){
 436                 case AF_FORMAT_AC3_NE:
 437                 case AF_FORMAT_S24_LE:
 438                 case AF_FORMAT_S16_LE:
 439                 case AF_FORMAT_U8:
 440                         break;
 441                 default:
 442                         mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
 443                         format=AF_FORMAT_S16_LE;
 444         }
 445         //fill global ao_data
 446         ao_data.channels = channels;
 447         ao_data.samplerate = rate;
 448         ao_data.format = format;
 449         ao_data.bps = channels * rate * (af_fmt2bits(format)>>3);
 450         if(ao_data.buffersize==-1) ao_data.buffersize = ao_data.bps; // space for 1 sec
 451         mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, channels, af_fmt2str_short(format));
 452         mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Buffersize:%d bytes (%d msec)\n", ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000);
 453
 454         //fill waveformatex
 455         ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE));
 456         wformat.Format.cbSize          = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0;
 457         wformat.Format.nChannels       = channels;
 458         wformat.Format.nSamplesPerSec  = rate;
 459         if (AF_FORMAT_IS_AC3(format)) {
 460                 wformat.Format.wFormatTag      = WAVE_FORMAT_DOLBY_AC3_SPDIF;
 461                 wformat.Format.wBitsPerSample  = 16;
 462                 wformat.Format.nBlockAlign     = 4;
 463         } else {
 464                 wformat.Format.wFormatTag      = (channels > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
 465                 wformat.Format.wBitsPerSample  = af_fmt2bits(format);
 466                 wformat.Format.nBlockAlign     = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
 467         }
 468
 469         // fill in primary sound buffer descriptor
 470         memset(&dsbpridesc, 0, sizeof(DSBUFFERDESC));
 471         dsbpridesc.dwSize = sizeof(DSBUFFERDESC);
 472         dsbpridesc.dwFlags       = DSBCAPS_PRIMARYBUFFER;
 473         dsbpridesc.dwBufferBytes = 0;
 474         dsbpridesc.lpwfxFormat   = NULL;
 475
 476
 477         // fill in the secondary sound buffer (=stream buffer) descriptor
 478         memset(&dsbdesc, 0, sizeof(DSBUFFERDESC));
 479         dsbdesc.dwSize = sizeof(DSBUFFERDESC);
 480         dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 /** Better position accuracy */
 481                         | DSBCAPS_GLOBALFOCUS         /** Allows background playing */
 482                         | DSBCAPS_CTRLVOLUME;         /** volume control enabled */
 483
 484         if (channels > 2) {
 485                 wformat.dwChannelMask = channel_mask[channels - 3];
 486                 wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
 487                 wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample;
 488                 // Needed for 5.1 on emu101k - shit soundblaster
 489                 dsbdesc.dwFlags |= DSBCAPS_LOCHARDWARE;
 490         }
 491         wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
 492
 493         dsbdesc.dwBufferBytes = ao_data.buffersize;
 494         dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat;
 495         buffer_size = dsbdesc.dwBufferBytes;
 496         write_offset = 0;
 497         min_free_space = wformat.Format.nBlockAlign;
 498         ao_data.outburst = wformat.Format.nBlockAlign * 512;
 499
 500         // create primary buffer and set its format
 501
 502         res = IDirectSound_CreateSoundBuffer( hds, &dsbpridesc, &hdspribuf, NULL );
 503         if ( res != DS_OK ) {
 504                 UninitDirectSound();
 505                 mp_msg(MSGT_AO, MSGL_ERR,"ao_dsound: cannot create primary buffer (%s)\n", dserr2str(res));
 506                 return 0;
 507         }
 508         res = IDirectSoundBuffer_SetFormat( hdspribuf, (WAVEFORMATEX *)&wformat );
 509         if ( res != DS_OK ) mp_msg(MSGT_AO, MSGL_WARN,"ao_dsound: cannot set primary buffer format (%s), using standard setting (bad quality)", dserr2str(res));
 510
 511         mp_msg(MSGT_AO, MSGL_V, "ao_dsound: primary buffer created\n");
 512
 513         // now create the stream buffer
 514
 515         res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL);
 516         if (res != DS_OK) {
 517                 if (dsbdesc.dwFlags & DSBCAPS_LOCHARDWARE) {
 518                         // Try without DSBCAPS_LOCHARDWARE
 519                         dsbdesc.dwFlags &= ~DSBCAPS_LOCHARDWARE;
 520                         res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL);
 521                 }
 522                 if (res != DS_OK) {
 523                         UninitDirectSound();
 524                         mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create secondary (stream)buffer (%s)\n", dserr2str(res));
 525                         return 0;
 526                 }
 527         }
 528         mp_msg(MSGT_AO, MSGL_V, "ao_dsound: secondary (stream)buffer created\n");
 529         return 1;
 530 }
 531
 532
 533
 534 /**
 535 \brief stop playing and empty buffers (for seeking/pause)
 536 */
 537 static void reset(void)
 538 {
 539         IDirectSoundBuffer_Stop(hdsbuf);
 540         // reset directsound buffer
 541         IDirectSoundBuffer_SetCurrentPosition(hdsbuf, 0);
 542         write_offset=0;
 543 }
 544
 545 /**
 546 \brief stop playing, keep buffers (for pause)
 547 */
 548 static void audio_pause(void)
 549 {
 550         IDirectSoundBuffer_Stop(hdsbuf);
 551 }
 552
 553 /**
 554 \brief resume playing, after audio_pause()
 555 */
 556 static void audio_resume(void)
 557 {
 558         IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
 559 }
 560
 561 /**
 562 \brief close audio device
 563 \param immed stop playback immediately
 564 */
 565 static void uninit(int immed)
 566 {
 567         if(immed)reset();
 568         else{
 569                 DWORD status;
 570                 IDirectSoundBuffer_Play(hdsbuf, 0, 0, 0);
 571                 while(!IDirectSoundBuffer_GetStatus(hdsbuf,&status) && (status&DSBSTATUS_PLAYING))
 572                         usec_sleep(20000);
 573         }
 574         DestroyBuffer();
 575         UninitDirectSound();
 576 }
 577
 578 /**
 579 \brief find out how many bytes can be written into the audio buffer without
 580 \return free space in bytes, has to return 0 if the buffer is almost full
 581 */
 582 static int get_space(void)
 583 {
 584         int space;
 585         DWORD play_offset;
 586         IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
 587         space=buffer_size-(write_offset-play_offset);
 588         //                |                                                      | <-- const --> |                |                 |
 589         //                buffer start                                           play_cursor     write_cursor     write_offset      buffer end
 590         // play_cursor is the actual postion of the play cursor
 591         // write_cursor is the position after which it is assumed to be save to write data
 592         // write_offset is the postion where we actually write the data to
 593         if(space > buffer_size)space -= buffer_size; // write_offset < play_offset
 594         if(space < min_free_space)return 0;
 595         return space-min_free_space;
 596 }
 597
 598 /**
 599 \brief play 'len' bytes of 'data'
 600 \param data pointer to the data to play
 601 \param len size in bytes of the data buffer, gets rounded down to outburst*n
 602 \param flags currently unused
 603 \return number of played bytes
 604 */
 605 static int play(void* data, int len, int flags)
 606 {
 607         DWORD play_offset;
 608         int space;
 609
 610         // make sure we have enough space to write data
 611         IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
 612         space=buffer_size-(write_offset-play_offset);
 613         if(space > buffer_size)space -= buffer_size; // write_offset < play_offset
 614         if(space < len) len = space;
 615
 616         if (!(flags & AOPLAY_FINAL_CHUNK))
 617         len = (len / ao_data.outburst) * ao_data.outburst;
 618         return write_buffer(data, len);
 619 }
 620
 621 /**
 622 \brief get the delay between the first and last sample in the buffer
 623 \return delay in seconds
 624 */
 625 static float get_delay(void)
 626 {
 627         DWORD play_offset;
 628         int space;
 629         IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
 630         space=play_offset-write_offset;
 631         if(space <= 0)space += buffer_size;
 632         return (float)(buffer_size - space) / (float)ao_data.bps;
 633 }