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mpv/libao2/ao_dsound.c

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/******************************************************************************
* ao_dsound.c: Windows DirectSound interface for MPlayer
* Copyright (c) 2004 Gabor Szecsi <deje@miki.hu>
*
* This program 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.
*
* This program 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 this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
*
*****************************************************************************/
/**
\todo verify/extend multichannel support
*/
#include <stdio.h>
#include <stdlib.h>
#include <windows.h>
#define DIRECTSOUND_VERSION 0x0600
#include <dsound.h>
#include "libaf/af_format.h"
#include "audio_out.h"
#include "audio_out_internal.h"
#include "mp_msg.h"
#include "libvo/fastmemcpy.h"
#include "osdep/timer.h"
static 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
*/
#if 1
#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
#define DSSPEAKER_HEADPHONE 0x00000001
#define DSSPEAKER_MONO 0x00000002
#define DSSPEAKER_QUAD 0x00000003
#define DSSPEAKER_STEREO 0x00000004
#define DSSPEAKER_SURROUND 0x00000005
#define DSSPEAKER_5POINT1 0x00000006
#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
#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 = 4096; ///if the free space is below this value get_space() will return 0
/***************************************************************************************/
/**
\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 initilize direct sound
\return 0 if error, 1 if ok
*/
static int InitDirectSound(void)
{
DSCAPS dscaps;
// initialize directsound
HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
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");
if (OurDirectSoundCreate == NULL) {
mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: GetProcAddress FAILED\n");
FreeLibrary(hdsound_dll);
return 0;
}
// Create the direct sound object
if FAILED(OurDirectSoundCreate(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;
// 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) && (ao_data.format!=AF_FORMAT_AC3) ) {
// 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(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(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;
IDirectSoundBuffer_GetVolume(hdsbuf, &volume);
vol->left = vol->right = (float)(volume+10000) / 100.0;
//printf("ao_dsound: volume: %f\n",vol->left);
return CONTROL_OK;
}
case AOCONTROL_SET_VOLUME: {
ao_control_vol_t* vol = (ao_control_vol_t*)arg;
volume = (vol->right * 100.0)-10000;
IDirectSoundBuffer_SetVolume(hdsbuf, volume);
//printf("ao_dsound: volume: %f\n",vol->left);
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;
// ok, now create the buffers
WAVEFORMATEXTENSIBLE wformat;
DSBUFFERDESC dsbpridesc;
DSBUFFERDESC dsbdesc;
//check if the format is supported in general
switch(format){
case AF_FORMAT_AC3:
case AF_FORMAT_S24_LE:
case AF_FORMAT_S16_LE:
case AF_FORMAT_S8:
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 (format == AF_FORMAT_AC3) {
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;
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()
{
IDirectSoundBuffer_Stop(hdsbuf);
// reset directsound buffer
IDirectSoundBuffer_SetCurrentPosition(hdsbuf, 0);
write_offset=0;
}
/**
\brief stop playing, keep buffers (for pause)
*/
static void audio_pause()
{
IDirectSoundBuffer_Stop(hdsbuf);
}
/**
\brief resume playing, after audio_pause()
*/
static void audio_resume()
{
IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
}
/**
\brief close audio device
\param immed stop playback immediately, currently not supported
*/
static void uninit(int immed)
{
reset();
DestroyBuffer();
UninitDirectSound();
}
/**
\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()
{
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
if(space < min_free_space)return 0;
return 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)
{
DWORD play_offset;
int space;
// make sure we have enough space to write data
IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
space=buffer_size-(write_offset-play_offset);
if(space > buffer_size)space -= buffer_size; // write_offset < play_offset
if(space < len) len = space;
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()
{
DWORD play_offset;
int space;
IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
space=play_offset-write_offset;
if(space <= 0)space += buffer_size;
return (float)(buffer_size - space) / (float)ao_data.bps;
}
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