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 #if 1
  59 #define WAVE_FORMAT_IEEE_FLOAT 0x0003
  60 #define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092
  61 #define WAVE_FORMAT_EXTENSIBLE 0xFFFE
  62
  63 static const GUID KSDATAFORMAT_SUBTYPE_PCM = {0x1,0x0000,0x0010, {0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}};
  64
  65 #define SPEAKER_FRONT_LEFT             0x1
  66 #define SPEAKER_FRONT_RIGHT            0x2
  67 #define SPEAKER_FRONT_CENTER           0x4
  68 #define SPEAKER_LOW_FREQUENCY          0x8
  69 #define SPEAKER_BACK_LEFT              0x10
  70 #define SPEAKER_BACK_RIGHT             0x20
  71 #define SPEAKER_FRONT_LEFT_OF_CENTER   0x40
  72 #define SPEAKER_FRONT_RIGHT_OF_CENTER  0x80
  73 #define SPEAKER_BACK_CENTER            0x100
  74 #define SPEAKER_SIDE_LEFT              0x200
  75 #define SPEAKER_SIDE_RIGHT             0x400
  76 #define SPEAKER_TOP_CENTER             0x800
  77 #define SPEAKER_TOP_FRONT_LEFT         0x1000
  78 #define SPEAKER_TOP_FRONT_CENTER       0x2000
  79 #define SPEAKER_TOP_FRONT_RIGHT        0x4000
  80 #define SPEAKER_TOP_BACK_LEFT          0x8000
  81 #define SPEAKER_TOP_BACK_CENTER        0x10000
  82 #define SPEAKER_TOP_BACK_RIGHT         0x20000
  83 #define SPEAKER_RESERVED               0x80000000
  84
  85 #define DSSPEAKER_HEADPHONE         0x00000001
  86 #define DSSPEAKER_MONO              0x00000002
  87 #define DSSPEAKER_QUAD              0x00000003
  88 #define DSSPEAKER_STEREO            0x00000004
  89 #define DSSPEAKER_SURROUND          0x00000005
  90 #define DSSPEAKER_5POINT1           0x00000006
  91
  92 #ifndef _WAVEFORMATEXTENSIBLE_
  93 typedef struct {
  94     WAVEFORMATEX    Format;
  95     union {
  96         WORD wValidBitsPerSample;       /* bits of precision  */
  97         WORD wSamplesPerBlock;          /* valid if wBitsPerSample==0 */
  98         WORD wReserved;                 /* If neither applies, set to zero. */
  99     } Samples;
 100     DWORD           dwChannelMask;      /* which channels are */
 101                                         /* present in stream  */
 102     GUID            SubFormat;
 103 } WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
 104 #endif
 105
 106 #endif
 107
 108 static const int channel_mask[] = {
 109   SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT  | SPEAKER_LOW_FREQUENCY,
 110   SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT,
 111   SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT   | SPEAKER_LOW_FREQUENCY,
 112   SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT     | SPEAKER_LOW_FREQUENCY
 113 };
 114
 115 static HINSTANCE hdsound_dll = NULL;      ///handle to the dll
 116 static LPDIRECTSOUND hds = NULL;          ///direct sound object
 117 static LPDIRECTSOUNDBUFFER hdspribuf = NULL; ///primary direct sound buffer
 118 static LPDIRECTSOUNDBUFFER hdsbuf = NULL; ///secondary direct sound buffer (stream buffer)
 119 static int buffer_size = 0;               ///size in bytes of the direct sound buffer
 120 static int write_offset = 0;              ///offset of the write cursor in the direct sound buffer
 121 static int min_free_space = 0;            ///if the free space is below this value get_space() will return 0
 122                                           ///there will always be at least this amout of free space to prevent
 123                                           ///get_space() from returning wrong values when buffer is 100% full.
 124                                           ///will be replaced with nBlockAlign in init()
 125 static int device_num = 0;                ///wanted device number
 126 static GUID device;                       ///guid of the device
 127
 128 /***************************************************************************************/
 129
 130 /**
 131 \brief output error message
 132 \param err error code
 133 \return string with the error message
 134 */
 135 static char * dserr2str(int err)
 136 {
 137         switch (err) {
 138                 case DS_OK: return "DS_OK";
 139                 case DS_NO_VIRTUALIZATION: return "DS_NO_VIRTUALIZATION";
 140                 case DSERR_ALLOCATED: return "DS_NO_VIRTUALIZATION";
 141                 case DSERR_CONTROLUNAVAIL: return "DSERR_CONTROLUNAVAIL";
 142                 case DSERR_INVALIDPARAM: return "DSERR_INVALIDPARAM";
 143                 case DSERR_INVALIDCALL: return "DSERR_INVALIDCALL";
 144                 case DSERR_GENERIC: return "DSERR_GENERIC";
 145                 case DSERR_PRIOLEVELNEEDED: return "DSERR_PRIOLEVELNEEDED";
 146                 case DSERR_OUTOFMEMORY: return "DSERR_OUTOFMEMORY";
 147                 case DSERR_BADFORMAT: return "DSERR_BADFORMAT";
 148                 case DSERR_UNSUPPORTED: return "DSERR_UNSUPPORTED";
 149                 case DSERR_NODRIVER: return "DSERR_NODRIVER";
 150                 case DSERR_ALREADYINITIALIZED: return "DSERR_ALREADYINITIALIZED";
 151                 case DSERR_NOAGGREGATION: return "DSERR_NOAGGREGATION";
 152                 case DSERR_BUFFERLOST: return "DSERR_BUFFERLOST";
 153                 case DSERR_OTHERAPPHASPRIO: return "DSERR_OTHERAPPHASPRIO";
 154                 case DSERR_UNINITIALIZED: return "DSERR_UNINITIALIZED";
 155                 case DSERR_NOINTERFACE: return "DSERR_NOINTERFACE";
 156                 case DSERR_ACCESSDENIED: return "DSERR_ACCESSDENIED";
 157                 default: return "unknown";
 158         }
 159 }
 160
 161 /**
 162 \brief uninitialize direct sound
 163 */
 164 static void UninitDirectSound(void)
 165 {
 166     // finally release the DirectSound object
 167     if (hds) {
 168         IDirectSound_Release(hds);
 169         hds = NULL;
 170     }
 171     // free DSOUND.DLL
 172     if (hdsound_dll) {
 173         FreeLibrary(hdsound_dll);
 174         hdsound_dll = NULL;
 175     }
 176         mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound uninitialized\n");
 177 }
 178
 179 /**
 180 \brief print the commandline help
 181 */
 182 static void print_help(void)
 183 {
 184   mp_msg(MSGT_AO, MSGL_FATAL,
 185            "\n-ao dsound commandline help:\n"
 186            "Example: mplayer -ao dsound:device=1\n"
 187            "  sets 1st device\n"
 188            "\nOptions:\n"
 189            "  device=<device-number>\n"
 190            "    Sets device number, use -v to get a list\n");
 191 }
 192
 193
 194 /**
 195 \brief enumerate direct sound devices
 196 \return TRUE to continue with the enumeration
 197 */
 198 static BOOL CALLBACK DirectSoundEnum(LPGUID guid,LPCSTR desc,LPCSTR module,LPVOID context)
 199 {
 200     int* device_index=context;
 201     mp_msg(MSGT_AO, MSGL_V,"%i %s ",*device_index,desc);
 202     if(device_num==*device_index){
 203         mp_msg(MSGT_AO, MSGL_V,"<--");
 204         if(guid){
 205             memcpy(&device,guid,sizeof(GUID));
 206         }
 207     }
 208     mp_msg(MSGT_AO, MSGL_V,"\n");
 209     (*device_index)++;
 210     return TRUE;
 211 }
 212
 213
 214 /**
 215 \brief initilize direct sound
 216 \return 0 if error, 1 if ok
 217 */
 218 static int InitDirectSound(void)
 219 {
 220         DSCAPS dscaps;
 221
 222         // initialize directsound
 223     HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
 224         HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
 225         int device_index=0;
 226         const opt_t subopts[] = {
 227           {"device", OPT_ARG_INT, &device_num,NULL},
 228           {NULL}
 229         };
 230         if (subopt_parse(ao_subdevice, subopts) != 0) {
 231                 print_help();
 232                 return 0;
 233         }
 234
 235         hdsound_dll = LoadLibrary("DSOUND.DLL");
 236         if (hdsound_dll == NULL) {
 237                 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot load DSOUND.DLL\n");
 238                 return 0;
 239         }
 240         OurDirectSoundCreate = (void*)GetProcAddress(hdsound_dll, "DirectSoundCreate");
 241         OurDirectSoundEnumerate = (void*)GetProcAddress(hdsound_dll, "DirectSoundEnumerateA");
 242
 243         if (OurDirectSoundCreate == NULL || OurDirectSoundEnumerate == NULL) {
 244                 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: GetProcAddress FAILED\n");
 245                 FreeLibrary(hdsound_dll);
 246                 return 0;
 247         }
 248
 249         // Enumerate all directsound devices
 250         mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Output Devices:\n");
 251         OurDirectSoundEnumerate(DirectSoundEnum,&device_index);
 252
 253         // Create the direct sound object
 254         if FAILED(OurDirectSoundCreate((device_num)?&device:NULL, &hds, NULL )) {
 255                 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create a DirectSound device\n");
 256                 FreeLibrary(hdsound_dll);
 257                 return 0;
 258         }
 259
 260         /* Set DirectSound Cooperative level, ie what control we want over Windows
 261          * sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the
 262          * settings of the primary buffer, but also that only the sound of our
 263          * application will be hearable when it will have the focus.
 264          * !!! (this is not really working as intended yet because to set the
 265          * cooperative level you need the window handle of your application, and
 266          * I don't know of any easy way to get it. Especially since we might play
 267          * sound without any video, and so what window handle should we use ???
 268          * The hack for now is to use the Desktop window handle - it seems to be
 269          * working */
 270         if (IDirectSound_SetCooperativeLevel(hds, GetDesktopWindow(), DSSCL_EXCLUSIVE)) {
 271                 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot set direct sound cooperative level\n");
 272                 IDirectSound_Release(hds);
 273                 FreeLibrary(hdsound_dll);
 274                 return 0;
 275         }
 276         mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound initialized\n");
 277
 278         memset(&dscaps, 0, sizeof(DSCAPS));
 279         dscaps.dwSize = sizeof(DSCAPS);
 280         if (DS_OK == IDirectSound_GetCaps(hds, &dscaps)) {
 281                 if (dscaps.dwFlags & DSCAPS_EMULDRIVER) mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound is emulated, waveOut may give better performance\n");
 282         } else {
 283                 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: cannot get device capabilities\n");
 284         }
 285
 286         return 1;
 287 }
 288
 289 /**
 290 \brief destroy the direct sound buffer
 291 */
 292 static void DestroyBuffer(void)
 293 {
 294         if (hdsbuf) {
 295                 IDirectSoundBuffer_Release(hdsbuf);
 296                 hdsbuf = NULL;
 297         }
 298         if (hdspribuf) {
 299                 IDirectSoundBuffer_Release(hdspribuf);
 300                 hdspribuf = NULL;
 301         }
 302 }
 303
 304 /**
 305 \brief fill sound buffer
 306 \param data pointer to the sound data to copy
 307 \param len length of the data to copy in bytes
 308 \return number of copyed bytes
 309 */
 310 static int write_buffer(unsigned char *data, int len)
 311 {
 312   HRESULT res;
 313   LPVOID lpvPtr1;
 314   DWORD dwBytes1;
 315   LPVOID lpvPtr2;
 316   DWORD dwBytes2;
 317
 318   // Lock the buffer
 319   res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
 320   // If the buffer was lost, restore and retry lock.
 321   if (DSERR_BUFFERLOST == res)
 322   {
 323     IDirectSoundBuffer_Restore(hdsbuf);
 324         res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
 325   }
 326
 327
 328   if (SUCCEEDED(res))
 329   {
 330         if( (ao_data.channels == 6) && (ao_data.format!=AF_FORMAT_AC3) ) {
 331             // reorder channels while writing to pointers.
 332             // it's this easy because buffer size and len are always
 333             // aligned to multiples of channels*bytespersample
 334             // there's probably some room for speed improvements here
 335             const int chantable[6] = {0, 1, 4, 5, 2, 3}; // reorder "matrix"
 336             int i, j;
 337             int numsamp,sampsize;
 338
 339             sampsize = af_fmt2bits(ao_data.format)>>3; // bytes per sample
 340             numsamp = dwBytes1 / (ao_data.channels * sampsize);  // number of samples for each channel in this buffer
 341
 342             for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) {
 343                 memcpy(lpvPtr1+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize);
 344             }
 345
 346             if (NULL != lpvPtr2 )
 347             {
 348                 numsamp = dwBytes2 / (ao_data.channels * sampsize);
 349                 for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) {
 350                     memcpy(lpvPtr2+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+dwBytes1+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize);
 351                 }
 352             }
 353
 354             write_offset+=dwBytes1+dwBytes2;
 355             if(write_offset>=buffer_size)write_offset=dwBytes2;
 356         } else {
 357             // Write to pointers without reordering.
 358         fast_memcpy(lpvPtr1,data,dwBytes1);
 359     if (NULL != lpvPtr2 )fast_memcpy(lpvPtr2,data+dwBytes1,dwBytes2);
 360         write_offset+=dwBytes1+dwBytes2;
 361     if(write_offset>=buffer_size)write_offset=dwBytes2;
 362         }
 363
 364    // Release the data back to DirectSound.
 365     res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2);
 366     if (SUCCEEDED(res))
 367     {
 368           // Success.
 369           DWORD status;
 370           IDirectSoundBuffer_GetStatus(hdsbuf, &status);
 371       if (!(status & DSBSTATUS_PLAYING)){
 372             res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
 373           }
 374           return dwBytes1+dwBytes2;
 375     }
 376   }
 377   // Lock, Unlock, or Restore failed.
 378   return 0;
 379 }
 380
 381 /***************************************************************************************/
 382
 383 /**
 384 \brief handle control commands
 385 \param cmd command
 386 \param arg argument
 387 \return CONTROL_OK or -1 in case the command can't be handled
 388 */
 389 static int control(int cmd, void *arg)
 390 {
 391         DWORD volume;
 392         switch (cmd) {
 393                 case AOCONTROL_GET_VOLUME: {
 394                         ao_control_vol_t* vol = (ao_control_vol_t*)arg;
 395                         IDirectSoundBuffer_GetVolume(hdsbuf, &volume);
 396                         vol->left = vol->right = pow(10.0, (float)(volume+10000) / 5000.0);
 397                         //printf("ao_dsound: volume: %f\n",vol->left);
 398                         return CONTROL_OK;
 399                 }
 400                 case AOCONTROL_SET_VOLUME: {
 401                         ao_control_vol_t* vol = (ao_control_vol_t*)arg;
 402                         volume = (DWORD)(log10(vol->right) * 5000.0) - 10000;
 403                         IDirectSoundBuffer_SetVolume(hdsbuf, volume);
 404                         //printf("ao_dsound: volume: %f\n",vol->left);
 405                         return CONTROL_OK;
 406                 }
 407         }
 408         return -1;
 409 }
 410
 411 /**
 412 \brief setup sound device
 413 \param rate samplerate
 414 \param channels number of channels
 415 \param format format
 416 \param flags unused
 417 \return 1=success 0=fail
 418 */
 419 static int init(int rate, int channels, int format, int flags)
 420 {
 421     int res;
 422         if (!InitDirectSound()) return 0;
 423
 424         // ok, now create the buffers
 425         WAVEFORMATEXTENSIBLE wformat;
 426         DSBUFFERDESC dsbpridesc;
 427         DSBUFFERDESC dsbdesc;
 428
 429         //check if the channel count and format is supported in general
 430         if (channels > 6) {
 431                 UninitDirectSound();
 432                 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: 8 channel audio not yet supported\n");
 433                 return 0;
 434         }
 435         switch(format){
 436                 case AF_FORMAT_AC3:
 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 (format == AF_FORMAT_AC3) {
 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 }