Rename directories, move files (step 1 of 2) (does not compile)

Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.

Renames the following directories:
    libaf -> audio/filter
    libao2 -> audio/out
    libvo -> video/out
    libmpdemux -> demux

Split libmpcodecs:
    vf* -> video/filter
    vd*, dec_video.* -> video/decode
    mp_image*, img_format*, ... -> video/
    ad*, dec_audio.* -> audio/decode

libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.

Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.

sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).

Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.

1 files changed, 0 insertions, 648 deletions

diff --git a/libao2/ao_dsound.c b/libao2/ao_dsound.c
deleted file mode 100644
index f2f44dd401..0000000000
--- a/libao2/ao_dsound.c
+++ /dev/null
@@ -1,648 +0,0 @@
-/*
- * Windows DirectSound interface
- *
- * Copyright (c) 2004 Gabor Szecsi <deje@miki.hu>
- *
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-/**
-\todo verify/extend multichannel support
-*/
-
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <windows.h>
-#define DIRECTSOUND_VERSION 0x0600
-#include <dsound.h>
-#include <math.h>
-
-#include "config.h"
-#include "libaf/format.h"
-#include "audio_out.h"
-#include "audio_out_internal.h"
-#include "mp_msg.h"
-#include "osdep/timer.h"
-#include "subopt-helper.h"
-
-
-static const ao_info_t info =
-{
-	"Windows DirectSound audio output",
-	"dsound",
-	"Gabor Szecsi <deje@miki.hu>",
-	""
-};
-
-LIBAO_EXTERN(dsound)
-
-/**
-\todo use the definitions from the win32 api headers when they define these
-*/
-#define WAVE_FORMAT_IEEE_FLOAT 0x0003
-#define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092
-#define WAVE_FORMAT_EXTENSIBLE 0xFFFE
-
-static const GUID KSDATAFORMAT_SUBTYPE_PCM = {0x1,0x0000,0x0010, {0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}};
-
-#define SPEAKER_FRONT_LEFT             0x1
-#define SPEAKER_FRONT_RIGHT            0x2
-#define SPEAKER_FRONT_CENTER           0x4
-#define SPEAKER_LOW_FREQUENCY          0x8
-#define SPEAKER_BACK_LEFT              0x10
-#define SPEAKER_BACK_RIGHT             0x20
-#define SPEAKER_FRONT_LEFT_OF_CENTER   0x40
-#define SPEAKER_FRONT_RIGHT_OF_CENTER  0x80
-#define SPEAKER_BACK_CENTER            0x100
-#define SPEAKER_SIDE_LEFT              0x200
-#define SPEAKER_SIDE_RIGHT             0x400
-#define SPEAKER_TOP_CENTER             0x800
-#define SPEAKER_TOP_FRONT_LEFT         0x1000
-#define SPEAKER_TOP_FRONT_CENTER       0x2000
-#define SPEAKER_TOP_FRONT_RIGHT        0x4000
-#define SPEAKER_TOP_BACK_LEFT          0x8000
-#define SPEAKER_TOP_BACK_CENTER        0x10000
-#define SPEAKER_TOP_BACK_RIGHT         0x20000
-#define SPEAKER_RESERVED               0x80000000
-
-#if 0
-#define DSSPEAKER_HEADPHONE         0x00000001
-#define DSSPEAKER_MONO              0x00000002
-#define DSSPEAKER_QUAD              0x00000003
-#define DSSPEAKER_STEREO            0x00000004
-#define DSSPEAKER_SURROUND          0x00000005
-#define DSSPEAKER_5POINT1           0x00000006
-#endif
-
-#ifndef _WAVEFORMATEXTENSIBLE_
-typedef struct {
-    WAVEFORMATEX    Format;
-    union {
-        WORD wValidBitsPerSample;       /* bits of precision  */
-        WORD wSamplesPerBlock;          /* valid if wBitsPerSample==0 */
-        WORD wReserved;                 /* If neither applies, set to zero. */
-    } Samples;
-    DWORD           dwChannelMask;      /* which channels are */
-                                        /* present in stream  */
-    GUID            SubFormat;
-} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
-#endif
-
-static const int channel_mask[] = {
-  SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT  | SPEAKER_LOW_FREQUENCY,
-  SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT,
-  SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT   | SPEAKER_LOW_FREQUENCY,
-  SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT     | SPEAKER_LOW_FREQUENCY
-};
-
-static HINSTANCE hdsound_dll = NULL;      ///handle to the dll
-static LPDIRECTSOUND hds = NULL;          ///direct sound object
-static LPDIRECTSOUNDBUFFER hdspribuf = NULL; ///primary direct sound buffer
-static LPDIRECTSOUNDBUFFER hdsbuf = NULL; ///secondary direct sound buffer (stream buffer)
-static int buffer_size = 0;               ///size in bytes of the direct sound buffer
-static int write_offset = 0;              ///offset of the write cursor in the direct sound buffer
-static int min_free_space = 0;            ///if the free space is below this value get_space() will return 0
-                                          ///there will always be at least this amout of free space to prevent
-                                          ///get_space() from returning wrong values when buffer is 100% full.
-                                          ///will be replaced with nBlockAlign in init()
-static int underrun_check = 0;            ///0 or last reported free space (underrun detection)
-static int device_num = 0;                ///wanted device number
-static GUID device;                       ///guid of the device
-static int audio_volume;
-
-/***************************************************************************************/
-
-/**
-\brief output error message
-\param err error code
-\return string with the error message
-*/
-static char * dserr2str(int err)
-{
-	switch (err) {
-		case DS_OK: return "DS_OK";
-		case DS_NO_VIRTUALIZATION: return "DS_NO_VIRTUALIZATION";
-		case DSERR_ALLOCATED: return "DS_NO_VIRTUALIZATION";
-		case DSERR_CONTROLUNAVAIL: return "DSERR_CONTROLUNAVAIL";
-		case DSERR_INVALIDPARAM: return "DSERR_INVALIDPARAM";
-		case DSERR_INVALIDCALL: return "DSERR_INVALIDCALL";
-		case DSERR_GENERIC: return "DSERR_GENERIC";
-		case DSERR_PRIOLEVELNEEDED: return "DSERR_PRIOLEVELNEEDED";
-		case DSERR_OUTOFMEMORY: return "DSERR_OUTOFMEMORY";
-		case DSERR_BADFORMAT: return "DSERR_BADFORMAT";
-		case DSERR_UNSUPPORTED: return "DSERR_UNSUPPORTED";
-		case DSERR_NODRIVER: return "DSERR_NODRIVER";
-		case DSERR_ALREADYINITIALIZED: return "DSERR_ALREADYINITIALIZED";
-		case DSERR_NOAGGREGATION: return "DSERR_NOAGGREGATION";
-		case DSERR_BUFFERLOST: return "DSERR_BUFFERLOST";
-		case DSERR_OTHERAPPHASPRIO: return "DSERR_OTHERAPPHASPRIO";
-		case DSERR_UNINITIALIZED: return "DSERR_UNINITIALIZED";
-		case DSERR_NOINTERFACE: return "DSERR_NOINTERFACE";
-		case DSERR_ACCESSDENIED: return "DSERR_ACCESSDENIED";
-		default: return "unknown";
-	}
-}
-
-/**
-\brief uninitialize direct sound
-*/
-static void UninitDirectSound(void)
-{
-    // finally release the DirectSound object
-    if (hds) {
-    	IDirectSound_Release(hds);
-    	hds = NULL;
-    }
-    // free DSOUND.DLL
-    if (hdsound_dll) {
-    	FreeLibrary(hdsound_dll);
-    	hdsound_dll = NULL;
-    }
-	mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound uninitialized\n");
-}
-
-/**
-\brief print the commandline help
-*/
-static void print_help(void)
-{
-  mp_msg(MSGT_AO, MSGL_FATAL,
-           "\n-ao dsound commandline help:\n"
-           "Example: mpv -ao dsound:device=1\n"
-           "  sets 1st device\n"
-           "\nOptions:\n"
-           "  device=<device-number>\n"
-           "    Sets device number, use -v to get a list\n");
-}
-
-
-/**
-\brief enumerate direct sound devices
-\return TRUE to continue with the enumeration
-*/
-static BOOL CALLBACK DirectSoundEnum(LPGUID guid,LPCSTR desc,LPCSTR module,LPVOID context)
-{
-    int* device_index=context;
-    mp_msg(MSGT_AO, MSGL_V,"%i %s ",*device_index,desc);
-    if(device_num==*device_index){
-        mp_msg(MSGT_AO, MSGL_V,"<--");
-        if(guid){
-            memcpy(&device,guid,sizeof(GUID));
-        }
-    }
-    mp_msg(MSGT_AO, MSGL_V,"\n");
-    (*device_index)++;
-    return TRUE;
-}
-
-
-/**
-\brief initilize direct sound
-\return 0 if error, 1 if ok
-*/
-static int InitDirectSound(void)
-{
-	DSCAPS dscaps;
-
-	// initialize directsound
-    HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
-	HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
-	int device_index=0;
-	const opt_t subopts[] = {
-	  {"device", OPT_ARG_INT, &device_num,NULL},
-	  {NULL}
-	};
-	if (subopt_parse(ao_subdevice, subopts) != 0) {
-		print_help();
-		return 0;
-	}
-
-	hdsound_dll = LoadLibrary("DSOUND.DLL");
-	if (hdsound_dll == NULL) {
-		mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot load DSOUND.DLL\n");
-		return 0;
-	}
-	OurDirectSoundCreate = (void*)GetProcAddress(hdsound_dll, "DirectSoundCreate");
-	OurDirectSoundEnumerate = (void*)GetProcAddress(hdsound_dll, "DirectSoundEnumerateA");
-
-	if (OurDirectSoundCreate == NULL || OurDirectSoundEnumerate == NULL) {
-		mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: GetProcAddress FAILED\n");
-		FreeLibrary(hdsound_dll);
-		return 0;
-	}
-
-	// Enumerate all directsound devices
-	mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Output Devices:\n");
-	OurDirectSoundEnumerate(DirectSoundEnum,&device_index);
-
-	// Create the direct sound object
-	if FAILED(OurDirectSoundCreate((device_num)?&device:NULL, &hds, NULL )) {
-		mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create a DirectSound device\n");
-		FreeLibrary(hdsound_dll);
-		return 0;
-	}
-
-	/* Set DirectSound Cooperative level, ie what control we want over Windows
-	 * sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the
-	 * settings of the primary buffer, but also that only the sound of our
-	 * application will be hearable when it will have the focus.
-	 * !!! (this is not really working as intended yet because to set the
-	 * cooperative level you need the window handle of your application, and
-	 * I don't know of any easy way to get it. Especially since we might play
-	 * sound without any video, and so what window handle should we use ???
-	 * The hack for now is to use the Desktop window handle - it seems to be
-	 * working */
-	if (IDirectSound_SetCooperativeLevel(hds, GetDesktopWindow(), DSSCL_EXCLUSIVE)) {
-		mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot set direct sound cooperative level\n");
-		IDirectSound_Release(hds);
-		FreeLibrary(hdsound_dll);
-		return 0;
-	}
-	mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound initialized\n");
-
-	memset(&dscaps, 0, sizeof(DSCAPS));
-	dscaps.dwSize = sizeof(DSCAPS);
-	if (DS_OK == IDirectSound_GetCaps(hds, &dscaps)) {
-		if (dscaps.dwFlags & DSCAPS_EMULDRIVER) mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound is emulated, waveOut may give better performance\n");
-	} else {
-		mp_msg(MSGT_AO, MSGL_V, "ao_dsound: cannot get device capabilities\n");
-	}
-
-	return 1;
-}
-
-/**
-\brief destroy the direct sound buffer
-*/
-static void DestroyBuffer(void)
-{
-	if (hdsbuf) {
-		IDirectSoundBuffer_Release(hdsbuf);
-		hdsbuf = NULL;
-	}
-	if (hdspribuf) {
-		IDirectSoundBuffer_Release(hdspribuf);
-		hdspribuf = NULL;
-	}
-}
-
-/**
-\brief fill sound buffer
-\param data pointer to the sound data to copy
-\param len length of the data to copy in bytes
-\return number of copyed bytes
-*/
-static int write_buffer(unsigned char *data, int len)
-{
-  HRESULT res;
-  LPVOID lpvPtr1;
-  DWORD dwBytes1;
-  LPVOID lpvPtr2;
-  DWORD dwBytes2;
-
-  underrun_check = 0;
-
-  // Lock the buffer
-  res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
-  // If the buffer was lost, restore and retry lock.
-  if (DSERR_BUFFERLOST == res)
-  {
-    IDirectSoundBuffer_Restore(hdsbuf);
-	res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
-  }
-
-
-  if (SUCCEEDED(res))
-  {
-  	if( (ao_data.channels == 6) && !AF_FORMAT_IS_AC3(ao_data.format) ) {
-  	    // reorder channels while writing to pointers.
-  	    // it's this easy because buffer size and len are always
-  	    // aligned to multiples of channels*bytespersample
-  	    // there's probably some room for speed improvements here
-  	    const int chantable[6] = {0, 1, 4, 5, 2, 3}; // reorder "matrix"
-  	    int i, j;
-  	    int numsamp,sampsize;
-
-  	    sampsize = af_fmt2bits(ao_data.format)>>3; // bytes per sample
-  	    numsamp = dwBytes1 / (ao_data.channels * sampsize);  // number of samples for each channel in this buffer
-
-  	    for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) {
-                memcpy((char*)lpvPtr1+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize);
-  	    }
-
-  	    if (NULL != lpvPtr2 )
-  	    {
-  	        numsamp = dwBytes2 / (ao_data.channels * sampsize);
-  	        for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) {
-                    memcpy((char*)lpvPtr2+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+dwBytes1+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize);
-  	        }
-  	    }
-
-  	    write_offset+=dwBytes1+dwBytes2;
-  	    if(write_offset>=buffer_size)write_offset=dwBytes2;
-  	} else {
-  	    // Write to pointers without reordering.
-	memcpy(lpvPtr1,data,dwBytes1);
-    if (NULL != lpvPtr2 )memcpy(lpvPtr2,data+dwBytes1,dwBytes2);
-	write_offset+=dwBytes1+dwBytes2;
-    if(write_offset>=buffer_size)write_offset=dwBytes2;
-  	}
-
-   // Release the data back to DirectSound.
-    res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2);
-    if (SUCCEEDED(res))
-    {
-	  // Success.
-	  DWORD status;
-	  IDirectSoundBuffer_GetStatus(hdsbuf, &status);
-      if (!(status & DSBSTATUS_PLAYING)){
-	    res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
-	  }
-	  return dwBytes1+dwBytes2;
-    }
-  }
-  // Lock, Unlock, or Restore failed.
-  return 0;
-}
-
-/***************************************************************************************/
-
-/**
-\brief handle control commands
-\param cmd command
-\param arg argument
-\return CONTROL_OK or -1 in case the command can't be handled
-*/
-static int control(int cmd, void *arg)
-{
-	DWORD volume;
-	switch (cmd) {
-		case AOCONTROL_GET_VOLUME: {
-			ao_control_vol_t* vol = (ao_control_vol_t*)arg;
-			vol->left = vol->right = audio_volume;
-			return CONTROL_OK;
-		}
-		case AOCONTROL_SET_VOLUME: {
-			ao_control_vol_t* vol = (ao_control_vol_t*)arg;
-			volume = audio_volume = vol->right;
-			if (volume < 1)
-				volume = 1;
-			volume = (DWORD)(log10(volume) * 5000.0) - 10000;
-			IDirectSoundBuffer_SetVolume(hdsbuf, volume);
-			return CONTROL_OK;
-		}
-	}
-	return -1;
-}
-
-/**
-\brief setup sound device
-\param rate samplerate
-\param channels number of channels
-\param format format
-\param flags unused
-\return 1=success 0=fail
-*/
-static int init(int rate, int channels, int format, int flags)
-{
-    int res;
-	if (!InitDirectSound()) return 0;
-
-        global_ao->no_persistent_volume = true;
-	audio_volume = 100;
-
-	// ok, now create the buffers
-	WAVEFORMATEXTENSIBLE wformat;
-	DSBUFFERDESC dsbpridesc;
-	DSBUFFERDESC dsbdesc;
-
-	//check if the channel count and format is supported in general
-	if (channels > 6) {
-		UninitDirectSound();
-		mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: 8 channel audio not yet supported\n");
-		return 0;
-	}
-
-	if (AF_FORMAT_IS_AC3(format))
-		format = AF_FORMAT_AC3_NE;
-	switch(format){
-		case AF_FORMAT_AC3_NE:
-		case AF_FORMAT_S24_LE:
-		case AF_FORMAT_S16_LE:
-		case AF_FORMAT_U8:
-			break;
-		default:
-			mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
-			format=AF_FORMAT_S16_LE;
-	}
-	//fill global ao_data
-	ao_data.channels = channels;
-	ao_data.samplerate = rate;
-	ao_data.format = format;
-	ao_data.bps = channels * rate * (af_fmt2bits(format)>>3);
-	if(ao_data.buffersize==-1) ao_data.buffersize = ao_data.bps; // space for 1 sec
-	mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, channels, af_fmt2str_short(format));
-	mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Buffersize:%d bytes (%d msec)\n", ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000);
-
-	//fill waveformatex
-	ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE));
-	wformat.Format.cbSize          = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0;
-	wformat.Format.nChannels       = channels;
-	wformat.Format.nSamplesPerSec  = rate;
-	if (AF_FORMAT_IS_AC3(format)) {
-		wformat.Format.wFormatTag      = WAVE_FORMAT_DOLBY_AC3_SPDIF;
-		wformat.Format.wBitsPerSample  = 16;
-		wformat.Format.nBlockAlign     = 4;
-	} else {
-		wformat.Format.wFormatTag      = (channels > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
-		wformat.Format.wBitsPerSample  = af_fmt2bits(format);
-		wformat.Format.nBlockAlign     = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
-	}
-
-	// fill in primary sound buffer descriptor
-	memset(&dsbpridesc, 0, sizeof(DSBUFFERDESC));
-	dsbpridesc.dwSize = sizeof(DSBUFFERDESC);
-	dsbpridesc.dwFlags       = DSBCAPS_PRIMARYBUFFER;
-	dsbpridesc.dwBufferBytes = 0;
-	dsbpridesc.lpwfxFormat   = NULL;
-
-
-	// fill in the secondary sound buffer (=stream buffer) descriptor
-	memset(&dsbdesc, 0, sizeof(DSBUFFERDESC));
-	dsbdesc.dwSize = sizeof(DSBUFFERDESC);
-	dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 /** Better position accuracy */
-	                | DSBCAPS_GLOBALFOCUS         /** Allows background playing */
-	                | DSBCAPS_CTRLVOLUME;         /** volume control enabled */
-
-	if (channels > 2) {
-		wformat.dwChannelMask = channel_mask[channels - 3];
-		wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
-		wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample;
-		// Needed for 5.1 on emu101k - shit soundblaster
-		dsbdesc.dwFlags |= DSBCAPS_LOCHARDWARE;
-	}
-	wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
-
-	dsbdesc.dwBufferBytes = ao_data.buffersize;
-	dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat;
-	buffer_size = dsbdesc.dwBufferBytes;
-	write_offset = 0;
-	min_free_space = wformat.Format.nBlockAlign;
-	ao_data.outburst = wformat.Format.nBlockAlign * 512;
-
-	// create primary buffer and set its format
-
-	res = IDirectSound_CreateSoundBuffer( hds, &dsbpridesc, &hdspribuf, NULL );
-	if ( res != DS_OK ) {
-		UninitDirectSound();
-		mp_msg(MSGT_AO, MSGL_ERR,"ao_dsound: cannot create primary buffer (%s)\n", dserr2str(res));
-		return 0;
-	}
-	res = IDirectSoundBuffer_SetFormat( hdspribuf, (WAVEFORMATEX *)&wformat );
-	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));
-
-	mp_msg(MSGT_AO, MSGL_V, "ao_dsound: primary buffer created\n");
-
-	// now create the stream buffer
-
-	res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL);
-	if (res != DS_OK) {
-		if (dsbdesc.dwFlags & DSBCAPS_LOCHARDWARE) {
-			// Try without DSBCAPS_LOCHARDWARE
-			dsbdesc.dwFlags &= ~DSBCAPS_LOCHARDWARE;
-			res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL);
-		}
-		if (res != DS_OK) {
-			UninitDirectSound();
-			mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create secondary (stream)buffer (%s)\n", dserr2str(res));
-			return 0;
-		}
-	}
-	mp_msg(MSGT_AO, MSGL_V, "ao_dsound: secondary (stream)buffer created\n");
-	return 1;
-}
-
-
-
-/**
-\brief stop playing and empty buffers (for seeking/pause)
-*/
-static void reset(void)
-{
-	IDirectSoundBuffer_Stop(hdsbuf);
-	// reset directsound buffer
-	IDirectSoundBuffer_SetCurrentPosition(hdsbuf, 0);
-	write_offset=0;
-	underrun_check=0;
-}
-
-/**
-\brief stop playing, keep buffers (for pause)
-*/
-static void audio_pause(void)
-{
-	IDirectSoundBuffer_Stop(hdsbuf);
-}
-
-/**
-\brief resume playing, after audio_pause()
-*/
-static void audio_resume(void)
-{
-	IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
-}
-
-/**
-\brief close audio device
-\param immed stop playback immediately
-*/
-static void uninit(int immed)
-{
-	if (!immed)
-		usec_sleep(get_delay() * 1000000);
-	reset();
-
-	DestroyBuffer();
-	UninitDirectSound();
-}
-
-// return exact number of free (safe to write) bytes
-static int check_free_buffer_size(void)
-{
-	int space;
-	DWORD play_offset;
-	IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
-	space=buffer_size-(write_offset-play_offset);
-	//                |                                                      | <-- const --> |                |                 |
-	//                buffer start                                           play_cursor     write_cursor     write_offset      buffer end
-	// play_cursor is the actual postion of the play cursor
-	// write_cursor is the position after which it is assumed to be save to write data
-	// write_offset is the postion where we actually write the data to
-	if(space > buffer_size)space -= buffer_size; // write_offset < play_offset
-	// Check for buffer underruns. An underrun happens if DirectSound
-	// started to play old data beyond the current write_offset. Detect this
-	// by checking whether the free space shrinks, even though no data was
-	// written (i.e. no write_buffer). Doesn't always work, but the only
-	// reason we need this is to deal with the situation when playback ends,
-	// and the buffer is only half-filled.
-	if (space < underrun_check) {
-		// there's no useful data in the buffers
-		space = buffer_size;
-		reset();
-	}
-	underrun_check = space;
-	return space;
-}
-
-/**
-\brief find out how many bytes can be written into the audio buffer without
-\return free space in bytes, has to return 0 if the buffer is almost full
-*/
-static int get_space(void)
-{
-	int space = check_free_buffer_size();
-	if(space < min_free_space)return 0;
-	return space-min_free_space;
-}
-
-/**
-\brief play 'len' bytes of 'data'
-\param data pointer to the data to play
-\param len size in bytes of the data buffer, gets rounded down to outburst*n
-\param flags currently unused
-\return number of played bytes
-*/
-static int play(void* data, int len, int flags)
-{
-	int space = check_free_buffer_size();
-	if(space < len) len = space;
-
-	if (!(flags & AOPLAY_FINAL_CHUNK))
-	len = (len / ao_data.outburst) * ao_data.outburst;
-	return write_buffer(data, len);
-}
-
-/**
-\brief get the delay between the first and last sample in the buffer
-\return delay in seconds
-*/
-static float get_delay(void)
-{
-	int space = check_free_buffer_size();
-	return (float)(buffer_size - space) / (float)ao_data.bps;
-}