mirror of https://github.com/mpv-player/mpv
ao_dsound: remove this audio output
It existed for XP-compatibility only. There was also a time where ao_wasapi caused issues, but we're relatively confident that ao_wasapi works better or at least as good as ao_dsound on Windows Vista and later.
This commit is contained in:
wm4
parent
27ccad541a
commit
3e90a5fe81
|
|
@ -20,10 +20,8 @@ normal driver parameters.
|
||||||
|
|
||||||
See ``--ao=help`` for a list of compiled-in audio output drivers. The
|
See ``--ao=help`` for a list of compiled-in audio output drivers. The
|
||||||
driver ``--ao=alsa`` is preferred. ``--ao=pulse`` is preferred on systems
|
driver ``--ao=alsa`` is preferred. ``--ao=pulse`` is preferred on systems
|
||||||
where PulseAudio is used. On Windows, ``--ao=wasapi`` is preferred,
|
where PulseAudio is used. On BSD systems, ``--ao=oss`` or ``--ao=sndio``
|
||||||
though it might cause trouble sometimes, in which case ``--ao=dsound``
|
may work (the latter being experimental).
|
||||||
should be used. On BSD systems, ``--ao=oss`` or `--ao=sndio`` may work
|
|
||||||
(the latter being experimental). On OS X systems, use ``--ao=coreaudio``.
|
|
||||||
|
|
||||||
.. admonition:: Examples
|
.. admonition:: Examples
|
||||||
|
|
||||||
|
|
@ -207,18 +205,6 @@ Available audio output drivers are:
|
||||||
If you have stuttering video when using pulse, try to enable this
|
If you have stuttering video when using pulse, try to enable this
|
||||||
option. (Or try to update PulseAudio.)
|
option. (Or try to update PulseAudio.)
|
||||||
|
|
||||||
``dsound`` (Windows only)
|
|
||||||
DirectX DirectSound audio output driver
|
|
||||||
|
|
||||||
.. note:: This driver is for compatibility with old systems.
|
|
||||||
|
|
||||||
``device=<devicenum>``
|
|
||||||
Sets the device number to use. Playing a file with ``-v`` will show a
|
|
||||||
list of available devices.
|
|
||||||
|
|
||||||
``buffersize=<ms>``
|
|
||||||
DirectSound buffer size in milliseconds (default: 200).
|
|
||||||
|
|
||||||
``sdl``
|
``sdl``
|
||||||
SDL 1.2+ audio output driver. Should work on any platform supported by SDL
|
SDL 1.2+ audio output driver. Should work on any platform supported by SDL
|
||||||
1.2, but may require the ``SDL_AUDIODRIVER`` environment variable to be set
|
1.2, but may require the ``SDL_AUDIODRIVER`` environment variable to be set
|
||||||
|
|
|
||||||
|
|
@ -45,7 +45,6 @@ extern const struct ao_driver audio_out_jack;
|
||||||
extern const struct ao_driver audio_out_openal;
|
extern const struct ao_driver audio_out_openal;
|
||||||
extern const struct ao_driver audio_out_null;
|
extern const struct ao_driver audio_out_null;
|
||||||
extern const struct ao_driver audio_out_alsa;
|
extern const struct ao_driver audio_out_alsa;
|
||||||
extern const struct ao_driver audio_out_dsound;
|
|
||||||
extern const struct ao_driver audio_out_wasapi;
|
extern const struct ao_driver audio_out_wasapi;
|
||||||
extern const struct ao_driver audio_out_pcm;
|
extern const struct ao_driver audio_out_pcm;
|
||||||
extern const struct ao_driver audio_out_lavc;
|
extern const struct ao_driver audio_out_lavc;
|
||||||
|
|
@ -65,9 +64,6 @@ static const struct ao_driver * const audio_out_drivers[] = {
|
||||||
#if HAVE_WASAPI
|
#if HAVE_WASAPI
|
||||||
&audio_out_wasapi,
|
&audio_out_wasapi,
|
||||||
#endif
|
#endif
|
||||||
#if HAVE_DSOUND
|
|
||||||
&audio_out_dsound,
|
|
||||||
#endif
|
|
||||||
#if HAVE_OSS_AUDIO
|
#if HAVE_OSS_AUDIO
|
||||||
&audio_out_oss,
|
&audio_out_oss,
|
||||||
#endif
|
#endif
|
||||||
|
|
|
||||||
|
|
@ -1,707 +0,0 @@
|
||||||
/*
|
|
||||||
* Windows DirectSound interface
|
|
||||||
*
|
|
||||||
* Copyright (c) 2004 Gabor Szecsi <deje@miki.hu>
|
|
||||||
*
|
|
||||||
* This file is part of mpv.
|
|
||||||
*
|
|
||||||
* mpv 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.
|
|
||||||
*
|
|
||||||
* mpv 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 mpv. If not, see <http://www.gnu.org/licenses/>.
|
|
||||||
*/
|
|
||||||
|
|
||||||
/**
|
|
||||||
\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 <libavutil/avutil.h>
|
|
||||||
#include <libavutil/common.h>
|
|
||||||
|
|
||||||
#include "config.h"
|
|
||||||
#include "audio/format.h"
|
|
||||||
#include "ao.h"
|
|
||||||
#include "internal.h"
|
|
||||||
#include "common/msg.h"
|
|
||||||
#include "osdep/timer.h"
|
|
||||||
#include "osdep/io.h"
|
|
||||||
#include "options/m_option.h"
|
|
||||||
|
|
||||||
/**
|
|
||||||
\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}
|
|
||||||
};
|
|
||||||
|
|
||||||
#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
|
|
||||||
|
|
||||||
struct priv {
|
|
||||||
HINSTANCE hdsound_dll; ///handle to the dll
|
|
||||||
LPDIRECTSOUND hds; ///direct sound object
|
|
||||||
LPDIRECTSOUNDBUFFER hdspribuf; ///primary direct sound buffer
|
|
||||||
LPDIRECTSOUNDBUFFER hdsbuf; ///secondary direct sound buffer (stream buffer)
|
|
||||||
int buffer_size; ///size in bytes of the direct sound buffer
|
|
||||||
int write_offset; ///offset of the write cursor in the direct sound buffer
|
|
||||||
int min_free_space; ///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()
|
|
||||||
int underrun_check; ///0 or last reported free space (underrun detection)
|
|
||||||
int device_num; ///wanted device number
|
|
||||||
GUID device; ///guid of the device
|
|
||||||
int audio_volume;
|
|
||||||
|
|
||||||
int device_index;
|
|
||||||
|
|
||||||
int outburst; ///play in multiple of chunks of this size
|
|
||||||
|
|
||||||
int cfg_device;
|
|
||||||
int cfg_buffersize;
|
|
||||||
|
|
||||||
struct ao_device_list *listing; ///temporary during list_devs()
|
|
||||||
};
|
|
||||||
|
|
||||||
/***************************************************************************************/
|
|
||||||
|
|
||||||
/**
|
|
||||||
\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";
|
|
||||||
}
|
|
||||||
return "unknown";
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
\brief uninitialize direct sound
|
|
||||||
*/
|
|
||||||
static void UninitDirectSound(struct ao *ao)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
|
|
||||||
// finally release the DirectSound object
|
|
||||||
if (p->hds) {
|
|
||||||
IDirectSound_Release(p->hds);
|
|
||||||
p->hds = NULL;
|
|
||||||
}
|
|
||||||
// free DSOUND.DLL
|
|
||||||
if (p->hdsound_dll) {
|
|
||||||
FreeLibrary(p->hdsound_dll);
|
|
||||||
p->hdsound_dll = NULL;
|
|
||||||
}
|
|
||||||
MP_VERBOSE(ao, "DirectSound uninitialized\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)
|
|
||||||
{
|
|
||||||
struct ao *ao = context;
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
|
|
||||||
MP_VERBOSE(ao, "%i %s ", p->device_index, desc);
|
|
||||||
if (p->device_num == p->device_index) {
|
|
||||||
MP_VERBOSE(ao, "<--");
|
|
||||||
if (guid)
|
|
||||||
memcpy(&p->device, guid, sizeof(GUID));
|
|
||||||
}
|
|
||||||
char *guidstr = talloc_strdup(NULL, "");
|
|
||||||
if (guid) {
|
|
||||||
wchar_t guidwstr[80] = {0};
|
|
||||||
StringFromGUID2(guid, guidwstr, MP_ARRAY_SIZE(guidwstr));
|
|
||||||
char *nstr = mp_to_utf8(NULL, guidwstr);
|
|
||||||
if (nstr) {
|
|
||||||
talloc_free(guidstr);
|
|
||||||
guidstr = nstr;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
if (p->device_num < 0 && ao->device) {
|
|
||||||
if (strcmp(ao->device, guidstr) == 0) {
|
|
||||||
MP_VERBOSE(ao, "<--");
|
|
||||||
p->device_num = p->device_index;
|
|
||||||
if (guid)
|
|
||||||
memcpy(&p->device, guid, sizeof(GUID));
|
|
||||||
}
|
|
||||||
}
|
|
||||||
if (p->listing) {
|
|
||||||
struct ao_device_desc e = {guidstr, desc};
|
|
||||||
ao_device_list_add(p->listing, ao, &e);
|
|
||||||
}
|
|
||||||
talloc_free(guidstr);
|
|
||||||
|
|
||||||
MP_VERBOSE(ao, "\n");
|
|
||||||
p->device_index++;
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
static void EnumDevs(struct ao *ao)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
|
|
||||||
p->device_index = 0;
|
|
||||||
p->device_num = p->cfg_device;
|
|
||||||
|
|
||||||
HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
|
|
||||||
OurDirectSoundEnumerate = (void *)GetProcAddress(p->hdsound_dll,
|
|
||||||
"DirectSoundEnumerateA");
|
|
||||||
|
|
||||||
if (OurDirectSoundEnumerate == NULL) {
|
|
||||||
MP_ERR(ao, "GetProcAddress FAILED\n");
|
|
||||||
return;
|
|
||||||
}
|
|
||||||
|
|
||||||
// Enumerate all directsound p->devices
|
|
||||||
MP_VERBOSE(ao, "Output Devices:\n");
|
|
||||||
OurDirectSoundEnumerate(DirectSoundEnum, ao);
|
|
||||||
}
|
|
||||||
|
|
||||||
static int LoadDirectSound(struct ao *ao)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
|
|
||||||
// initialize directsound
|
|
||||||
p->hdsound_dll = LoadLibrary(L"DSOUND.DLL");
|
|
||||||
if (p->hdsound_dll == NULL) {
|
|
||||||
MP_ERR(ao, "cannot load DSOUND.DLL\n");
|
|
||||||
return 0;
|
|
||||||
}
|
|
||||||
|
|
||||||
return 1;
|
|
||||||
}
|
|
||||||
|
|
||||||
static void list_devs(struct ao *ao, struct ao_device_list *list)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
bool need_init = !p->hdsound_dll;
|
|
||||||
if (need_init && !LoadDirectSound(ao))
|
|
||||||
return;
|
|
||||||
|
|
||||||
p->listing = list;
|
|
||||||
EnumDevs(ao);
|
|
||||||
p->listing = NULL;
|
|
||||||
|
|
||||||
if (need_init)
|
|
||||||
UninitDirectSound(ao);
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
\brief initilize direct sound
|
|
||||||
\return 0 if error, 1 if ok
|
|
||||||
*/
|
|
||||||
static int InitDirectSound(struct ao *ao)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
|
|
||||||
DSCAPS dscaps;
|
|
||||||
|
|
||||||
if (!LoadDirectSound(ao))
|
|
||||||
return 0;
|
|
||||||
|
|
||||||
HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
|
|
||||||
OurDirectSoundCreate =
|
|
||||||
(void *)GetProcAddress(p->hdsound_dll, "DirectSoundCreate");
|
|
||||||
|
|
||||||
if (OurDirectSoundCreate == NULL) {
|
|
||||||
MP_ERR(ao, "GetProcAddress FAILED\n");
|
|
||||||
FreeLibrary(p->hdsound_dll);
|
|
||||||
return 0;
|
|
||||||
}
|
|
||||||
|
|
||||||
EnumDevs(ao);
|
|
||||||
|
|
||||||
// Create the direct sound object
|
|
||||||
if (FAILED(OurDirectSoundCreate((p->device_num > 0) ? &p->device : NULL,
|
|
||||||
&p->hds, NULL)))
|
|
||||||
{
|
|
||||||
MP_ERR(ao, "cannot create a DirectSound device\n");
|
|
||||||
FreeLibrary(p->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 audible 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(p->hds, GetDesktopWindow(),
|
|
||||||
DSSCL_EXCLUSIVE))
|
|
||||||
{
|
|
||||||
MP_ERR(ao, "cannot set direct sound cooperative level\n");
|
|
||||||
IDirectSound_Release(p->hds);
|
|
||||||
FreeLibrary(p->hdsound_dll);
|
|
||||||
return 0;
|
|
||||||
}
|
|
||||||
MP_VERBOSE(ao, "DirectSound initialized\n");
|
|
||||||
|
|
||||||
memset(&dscaps, 0, sizeof(DSCAPS));
|
|
||||||
dscaps.dwSize = sizeof(DSCAPS);
|
|
||||||
if (DS_OK == IDirectSound_GetCaps(p->hds, &dscaps)) {
|
|
||||||
if (dscaps.dwFlags & DSCAPS_EMULDRIVER)
|
|
||||||
MP_VERBOSE(ao, "DirectSound is emulated\n");
|
|
||||||
} else {
|
|
||||||
MP_VERBOSE(ao, "cannot get device capabilities\n");
|
|
||||||
}
|
|
||||||
|
|
||||||
return 1;
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
\brief destroy the direct sound buffer
|
|
||||||
*/
|
|
||||||
static void DestroyBuffer(struct ao *ao)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
|
|
||||||
if (p->hdsbuf) {
|
|
||||||
IDirectSoundBuffer_Release(p->hdsbuf);
|
|
||||||
p->hdsbuf = NULL;
|
|
||||||
}
|
|
||||||
if (p->hdspribuf) {
|
|
||||||
IDirectSoundBuffer_Release(p->hdspribuf);
|
|
||||||
p->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(struct ao *ao, unsigned char *data, int len)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
HRESULT res;
|
|
||||||
LPVOID lpvPtr1;
|
|
||||||
DWORD dwBytes1;
|
|
||||||
LPVOID lpvPtr2;
|
|
||||||
DWORD dwBytes2;
|
|
||||||
|
|
||||||
p->underrun_check = 0;
|
|
||||||
|
|
||||||
// Lock the buffer
|
|
||||||
res = IDirectSoundBuffer_Lock(p->hdsbuf, p->write_offset, len, &lpvPtr1,
|
|
||||||
&dwBytes1, &lpvPtr2, &dwBytes2, 0);
|
|
||||||
// If the buffer was lost, restore and retry lock.
|
|
||||||
if (DSERR_BUFFERLOST == res) {
|
|
||||||
IDirectSoundBuffer_Restore(p->hdsbuf);
|
|
||||||
res = IDirectSoundBuffer_Lock(p->hdsbuf, p->write_offset, len, &lpvPtr1,
|
|
||||||
&dwBytes1, &lpvPtr2, &dwBytes2, 0);
|
|
||||||
}
|
|
||||||
|
|
||||||
|
|
||||||
if (SUCCEEDED(res)) {
|
|
||||||
memcpy(lpvPtr1, data, dwBytes1);
|
|
||||||
if (NULL != lpvPtr2)
|
|
||||||
memcpy(lpvPtr2, data + dwBytes1, dwBytes2);
|
|
||||||
p->write_offset += dwBytes1 + dwBytes2;
|
|
||||||
if (p->write_offset >= p->buffer_size)
|
|
||||||
p->write_offset = dwBytes2;
|
|
||||||
|
|
||||||
// Release the data back to DirectSound.
|
|
||||||
res = IDirectSoundBuffer_Unlock(p->hdsbuf, lpvPtr1, dwBytes1, lpvPtr2,
|
|
||||||
dwBytes2);
|
|
||||||
if (SUCCEEDED(res)) {
|
|
||||||
// Success.
|
|
||||||
DWORD status;
|
|
||||||
IDirectSoundBuffer_GetStatus(p->hdsbuf, &status);
|
|
||||||
if (!(status & DSBSTATUS_PLAYING))
|
|
||||||
res = IDirectSoundBuffer_Play(p->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 CONTROL_UNKNOWN in case the command is not supported
|
|
||||||
*/
|
|
||||||
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
DWORD volume;
|
|
||||||
switch (cmd) {
|
|
||||||
case AOCONTROL_GET_VOLUME: {
|
|
||||||
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
|
|
||||||
vol->left = vol->right = p->audio_volume;
|
|
||||||
return CONTROL_OK;
|
|
||||||
}
|
|
||||||
case AOCONTROL_SET_VOLUME: {
|
|
||||||
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
|
|
||||||
volume = p->audio_volume = vol->right;
|
|
||||||
if (volume < 1)
|
|
||||||
volume = 1;
|
|
||||||
volume = (DWORD)(log10(volume) * 5000.0) - 10000;
|
|
||||||
IDirectSoundBuffer_SetVolume(p->hdsbuf, volume);
|
|
||||||
return CONTROL_OK;
|
|
||||||
}
|
|
||||||
case AOCONTROL_HAS_SOFT_VOLUME:
|
|
||||||
return CONTROL_TRUE;
|
|
||||||
}
|
|
||||||
return CONTROL_UNKNOWN;
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
\brief setup sound device
|
|
||||||
\param rate samplerate
|
|
||||||
\param channels number of channels
|
|
||||||
\param format format
|
|
||||||
\param flags unused
|
|
||||||
\return 0=success -1=fail
|
|
||||||
*/
|
|
||||||
static int init(struct ao *ao)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
int res;
|
|
||||||
|
|
||||||
if (!InitDirectSound(ao))
|
|
||||||
return -1;
|
|
||||||
|
|
||||||
p->audio_volume = 100;
|
|
||||||
|
|
||||||
// ok, now create the buffers
|
|
||||||
WAVEFORMATEXTENSIBLE wformat;
|
|
||||||
DSBUFFERDESC dsbpridesc;
|
|
||||||
DSBUFFERDESC dsbdesc;
|
|
||||||
int format = af_fmt_from_planar(ao->format);
|
|
||||||
int rate = ao->samplerate;
|
|
||||||
|
|
||||||
if (!af_fmt_is_spdif(format)) {
|
|
||||||
struct mp_chmap_sel sel = {0};
|
|
||||||
mp_chmap_sel_add_waveext(&sel);
|
|
||||||
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
|
|
||||||
return -1;
|
|
||||||
}
|
|
||||||
switch (format) {
|
|
||||||
case AF_FORMAT_S24:
|
|
||||||
case AF_FORMAT_S16:
|
|
||||||
case AF_FORMAT_U8:
|
|
||||||
break;
|
|
||||||
default:
|
|
||||||
if (af_fmt_is_spdif(format))
|
|
||||||
break;
|
|
||||||
MP_VERBOSE(ao, "format %s not supported defaulting to Signed 16-bit Little-Endian\n",
|
|
||||||
af_fmt_to_str(format));
|
|
||||||
format = AF_FORMAT_S16;
|
|
||||||
}
|
|
||||||
//set our audio parameters
|
|
||||||
ao->samplerate = rate;
|
|
||||||
ao->format = format;
|
|
||||||
ao->bps = ao->channels.num * rate * af_fmt_to_bytes(format);
|
|
||||||
int buffersize = ao->bps * p->cfg_buffersize / 1000;
|
|
||||||
MP_VERBOSE(ao, "Samplerate:%iHz Channels:%i Format:%s\n", rate,
|
|
||||||
ao->channels.num, af_fmt_to_str(format));
|
|
||||||
MP_VERBOSE(ao, "Buffersize:%d bytes (%f msec)\n",
|
|
||||||
buffersize, buffersize * 1000.0 / ao->bps);
|
|
||||||
|
|
||||||
//fill waveformatex
|
|
||||||
ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE));
|
|
||||||
wformat.Format.cbSize = (ao->channels.num > 2)
|
|
||||||
? sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX) : 0;
|
|
||||||
wformat.Format.nChannels = ao->channels.num;
|
|
||||||
wformat.Format.nSamplesPerSec = rate;
|
|
||||||
if (af_fmt_is_spdif(format)) {
|
|
||||||
// Whether it also works with e.g. DTS is unknown, but probably does.
|
|
||||||
wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
|
|
||||||
wformat.Format.wBitsPerSample = 16;
|
|
||||||
wformat.Format.nBlockAlign = 4;
|
|
||||||
} else {
|
|
||||||
wformat.Format.wFormatTag = (ao->channels.num > 2)
|
|
||||||
? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
|
|
||||||
int bps = af_fmt_to_bytes(format);
|
|
||||||
wformat.Format.wBitsPerSample = bps * 8;
|
|
||||||
wformat.Format.nBlockAlign = wformat.Format.nChannels * bps;
|
|
||||||
}
|
|
||||||
|
|
||||||
// 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 (ao->channels.num > 2) {
|
|
||||||
wformat.dwChannelMask = mp_chmap_to_waveext(&ao->channels);
|
|
||||||
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 = buffersize;
|
|
||||||
dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat;
|
|
||||||
p->buffer_size = dsbdesc.dwBufferBytes;
|
|
||||||
p->write_offset = 0;
|
|
||||||
p->min_free_space = wformat.Format.nBlockAlign;
|
|
||||||
p->outburst = wformat.Format.nBlockAlign * 512;
|
|
||||||
|
|
||||||
// create primary buffer and set its format
|
|
||||||
|
|
||||||
res = IDirectSound_CreateSoundBuffer(p->hds, &dsbpridesc, &p->hdspribuf, NULL);
|
|
||||||
if (res != DS_OK) {
|
|
||||||
UninitDirectSound(ao);
|
|
||||||
MP_ERR(ao, "cannot create primary buffer (%s)\n", dserr2str(res));
|
|
||||||
return -1;
|
|
||||||
}
|
|
||||||
res = IDirectSoundBuffer_SetFormat(p->hdspribuf, (WAVEFORMATEX *)&wformat);
|
|
||||||
if (res != DS_OK) {
|
|
||||||
MP_WARN(ao, "cannot set primary buffer format (%s), using "
|
|
||||||
"standard setting (bad quality)", dserr2str(res));
|
|
||||||
}
|
|
||||||
|
|
||||||
MP_VERBOSE(ao, "primary buffer created\n");
|
|
||||||
|
|
||||||
// now create the stream buffer
|
|
||||||
|
|
||||||
res = IDirectSound_CreateSoundBuffer(p->hds, &dsbdesc, &p->hdsbuf, NULL);
|
|
||||||
if (res != DS_OK) {
|
|
||||||
if (dsbdesc.dwFlags & DSBCAPS_LOCHARDWARE) {
|
|
||||||
// Try without DSBCAPS_LOCHARDWARE
|
|
||||||
dsbdesc.dwFlags &= ~DSBCAPS_LOCHARDWARE;
|
|
||||||
res = IDirectSound_CreateSoundBuffer(p->hds, &dsbdesc, &p->hdsbuf, NULL);
|
|
||||||
}
|
|
||||||
if (res != DS_OK) {
|
|
||||||
UninitDirectSound(ao);
|
|
||||||
MP_ERR(ao, "cannot create secondary (stream)buffer (%s)\n",
|
|
||||||
dserr2str(res));
|
|
||||||
return -1;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
MP_VERBOSE(ao, "secondary (stream)buffer created\n");
|
|
||||||
return 0;
|
|
||||||
}
|
|
||||||
|
|
||||||
|
|
||||||
|
|
||||||
/**
|
|
||||||
\brief stop playing and empty buffers (for seeking/pause)
|
|
||||||
*/
|
|
||||||
static void reset(struct ao *ao)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
|
|
||||||
IDirectSoundBuffer_Stop(p->hdsbuf);
|
|
||||||
// reset directsound buffer
|
|
||||||
IDirectSoundBuffer_SetCurrentPosition(p->hdsbuf, 0);
|
|
||||||
p->write_offset = 0;
|
|
||||||
p->underrun_check = 0;
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
\brief stop playing, keep buffers (for pause)
|
|
||||||
*/
|
|
||||||
static void audio_pause(struct ao *ao)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
|
|
||||||
IDirectSoundBuffer_Stop(p->hdsbuf);
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
\brief resume playing, after audio_pause()
|
|
||||||
*/
|
|
||||||
static void audio_resume(struct ao *ao)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
|
|
||||||
IDirectSoundBuffer_Play(p->hdsbuf, 0, 0, DSBPLAY_LOOPING);
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
\brief close audio device
|
|
||||||
\param immed stop playback immediately
|
|
||||||
*/
|
|
||||||
static void uninit(struct ao *ao)
|
|
||||||
{
|
|
||||||
reset(ao);
|
|
||||||
|
|
||||||
DestroyBuffer(ao);
|
|
||||||
UninitDirectSound(ao);
|
|
||||||
}
|
|
||||||
|
|
||||||
// return exact number of free (safe to write) bytes
|
|
||||||
static int check_free_buffer_size(struct ao *ao)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
int space;
|
|
||||||
DWORD play_offset;
|
|
||||||
IDirectSoundBuffer_GetCurrentPosition(p->hdsbuf, &play_offset, NULL);
|
|
||||||
space = p->buffer_size - (p->write_offset - play_offset);
|
|
||||||
// | | <-- const --> | | |
|
|
||||||
// buffer start play_cursor write_cursor p->write_offset buffer end
|
|
||||||
// play_cursor is the actual position of the play cursor
|
|
||||||
// write_cursor is the position after which it is assumed to be save to write data
|
|
||||||
// p->write_offset is the position where we actually write the data to
|
|
||||||
if (space > p->buffer_size)
|
|
||||||
space -= p->buffer_size; // p->write_offset < play_offset
|
|
||||||
// Check for buffer underruns. An underrun happens if DirectSound
|
|
||||||
// started to play old data beyond the current p->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 < p->underrun_check) {
|
|
||||||
// there's no useful data in the buffers
|
|
||||||
space = p->buffer_size;
|
|
||||||
reset(ao);
|
|
||||||
}
|
|
||||||
p->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(struct ao *ao)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
|
|
||||||
int space = check_free_buffer_size(ao);
|
|
||||||
if (space < p->min_free_space)
|
|
||||||
return 0;
|
|
||||||
return (space - p->min_free_space) / p->outburst * p->outburst / ao->sstride;
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
\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(struct ao *ao, void **data, int samples, int flags)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
int len = samples * ao->sstride;
|
|
||||||
|
|
||||||
int space = check_free_buffer_size(ao);
|
|
||||||
if (space < len)
|
|
||||||
len = space;
|
|
||||||
|
|
||||||
if (!(flags & AOPLAY_FINAL_CHUNK))
|
|
||||||
len = (len / p->outburst) * p->outburst;
|
|
||||||
return write_buffer(ao, data[0], len) / ao->sstride;
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
\brief get the delay between the first and last sample in the buffer
|
|
||||||
\return delay in seconds
|
|
||||||
*/
|
|
||||||
static double get_delay(struct ao *ao)
|
|
||||||
{
|
|
||||||
struct priv *p = ao->priv;
|
|
||||||
|
|
||||||
int space = check_free_buffer_size(ao);
|
|
||||||
return (p->buffer_size - space) / (double)ao->bps;
|
|
||||||
}
|
|
||||||
|
|
||||||
#define OPT_BASE_STRUCT struct priv
|
|
||||||
|
|
||||||
const struct ao_driver audio_out_dsound = {
|
|
||||||
.description = "Windows DirectSound audio output",
|
|
||||||
.name = "dsound",
|
|
||||||
.init = init,
|
|
||||||
.uninit = uninit,
|
|
||||||
.control = control,
|
|
||||||
.get_space = get_space,
|
|
||||||
.play = play,
|
|
||||||
.get_delay = get_delay,
|
|
||||||
.pause = audio_pause,
|
|
||||||
.resume = audio_resume,
|
|
||||||
.reset = reset,
|
|
||||||
.list_devs = list_devs,
|
|
||||||
.priv_size = sizeof(struct priv),
|
|
||||||
.options = (const struct m_option[]) {
|
|
||||||
OPT_INT("device", cfg_device, 0, OPTDEF_INT(-1)),
|
|
||||||
OPT_INTRANGE("buffersize", cfg_buffersize, 0, 1, 10000, OPTDEF_INT(200)),
|
|
||||||
{0}
|
|
||||||
},
|
|
||||||
};
|
|
||||||
4
wscript
4
wscript
|
|
@ -559,10 +559,6 @@ audio_output_features = [
|
||||||
'func': check_cc(
|
'func': check_cc(
|
||||||
fragment=load_fragment('coreaudio.c'),
|
fragment=load_fragment('coreaudio.c'),
|
||||||
framework_name=['CoreFoundation', 'CoreAudio', 'AudioUnit', 'AudioToolbox'])
|
framework_name=['CoreFoundation', 'CoreAudio', 'AudioUnit', 'AudioToolbox'])
|
||||||
}, {
|
|
||||||
'name': '--dsound',
|
|
||||||
'desc': 'DirectSound audio output',
|
|
||||||
'func': check_cc(header_name='dsound.h'),
|
|
||||||
}, {
|
}, {
|
||||||
'name': '--wasapi',
|
'name': '--wasapi',
|
||||||
'desc': 'WASAPI audio output',
|
'desc': 'WASAPI audio output',
|
||||||
|
|
|
||||||
|
|
@ -134,7 +134,6 @@ def build(ctx):
|
||||||
( "audio/out/ao_coreaudio_exclusive.c", "coreaudio" ),
|
( "audio/out/ao_coreaudio_exclusive.c", "coreaudio" ),
|
||||||
( "audio/out/ao_coreaudio_properties.c", "coreaudio" ),
|
( "audio/out/ao_coreaudio_properties.c", "coreaudio" ),
|
||||||
( "audio/out/ao_coreaudio_utils.c", "coreaudio" ),
|
( "audio/out/ao_coreaudio_utils.c", "coreaudio" ),
|
||||||
( "audio/out/ao_dsound.c", "dsound" ),
|
|
||||||
( "audio/out/ao_jack.c", "jack" ),
|
( "audio/out/ao_jack.c", "jack" ),
|
||||||
( "audio/out/ao_lavc.c", "encoding" ),
|
( "audio/out/ao_lavc.c", "encoding" ),
|
||||||
( "audio/out/ao_null.c" ),
|
( "audio/out/ao_null.c" ),
|
||||||
|
|
|
||||||
Loading…
Reference in New Issue