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
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@ -20,10 +20,8 @@ normal driver parameters.
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See ``--ao=help`` for a list of compiled-in audio output drivers. The
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driver ``--ao=alsa`` is preferred. ``--ao=pulse`` is preferred on systems
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where PulseAudio is used. On Windows, ``--ao=wasapi`` is preferred,
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though it might cause trouble sometimes, in which case ``--ao=dsound``
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should be used. On BSD systems, ``--ao=oss`` or `--ao=sndio`` may work
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(the latter being experimental). On OS X systems, use ``--ao=coreaudio``.
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where PulseAudio is used. On BSD systems, ``--ao=oss`` or ``--ao=sndio``
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may work (the latter being experimental).
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.. admonition:: Examples
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@ -207,18 +205,6 @@ Available audio output drivers are:
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If you have stuttering video when using pulse, try to enable this
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option. (Or try to update PulseAudio.)
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``dsound`` (Windows only)
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DirectX DirectSound audio output driver
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.. note:: This driver is for compatibility with old systems.
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``device=<devicenum>``
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Sets the device number to use. Playing a file with ``-v`` will show a
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list of available devices.
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``buffersize=<ms>``
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DirectSound buffer size in milliseconds (default: 200).
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``sdl``
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SDL 1.2+ audio output driver. Should work on any platform supported by SDL
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1.2, but may require the ``SDL_AUDIODRIVER`` environment variable to be set
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@ -45,7 +45,6 @@ extern const struct ao_driver audio_out_jack;
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extern const struct ao_driver audio_out_openal;
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extern const struct ao_driver audio_out_null;
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extern const struct ao_driver audio_out_alsa;
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extern const struct ao_driver audio_out_dsound;
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extern const struct ao_driver audio_out_wasapi;
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extern const struct ao_driver audio_out_pcm;
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extern const struct ao_driver audio_out_lavc;
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@ -65,9 +64,6 @@ static const struct ao_driver * const audio_out_drivers[] = {
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#if HAVE_WASAPI
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&audio_out_wasapi,
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#endif
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#if HAVE_DSOUND
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&audio_out_dsound,
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#endif
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#if HAVE_OSS_AUDIO
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&audio_out_oss,
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#endif
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@ -1,707 +0,0 @@
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/*
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* Windows DirectSound interface
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*
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* Copyright (c) 2004 Gabor Szecsi <deje@miki.hu>
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*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* mpv is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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/**
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\todo verify/extend multichannel support
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <windows.h>
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#define DIRECTSOUND_VERSION 0x0600
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#include <dsound.h>
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#include <math.h>
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#include <libavutil/avutil.h>
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#include <libavutil/common.h>
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#include "config.h"
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#include "audio/format.h"
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#include "ao.h"
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#include "internal.h"
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#include "common/msg.h"
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#include "osdep/timer.h"
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#include "osdep/io.h"
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#include "options/m_option.h"
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/**
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\todo use the definitions from the win32 api headers when they define these
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*/
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#define WAVE_FORMAT_IEEE_FLOAT 0x0003
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#define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092
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#define WAVE_FORMAT_EXTENSIBLE 0xFFFE
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static const GUID KSDATAFORMAT_SUBTYPE_PCM = {
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0x1, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}
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};
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#if 0
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#define DSSPEAKER_HEADPHONE 0x00000001
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#define DSSPEAKER_MONO 0x00000002
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#define DSSPEAKER_QUAD 0x00000003
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#define DSSPEAKER_STEREO 0x00000004
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#define DSSPEAKER_SURROUND 0x00000005
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#define DSSPEAKER_5POINT1 0x00000006
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#endif
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#ifndef _WAVEFORMATEXTENSIBLE_
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typedef struct {
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WAVEFORMATEX Format;
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union {
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WORD wValidBitsPerSample; /* bits of precision */
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WORD wSamplesPerBlock; /* valid if wBitsPerSample==0 */
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WORD wReserved; /* If neither applies, set to zero. */
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} Samples;
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DWORD dwChannelMask; /* which channels are */
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/* present in stream */
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GUID SubFormat;
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} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
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#endif
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struct priv {
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HINSTANCE hdsound_dll; ///handle to the dll
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LPDIRECTSOUND hds; ///direct sound object
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LPDIRECTSOUNDBUFFER hdspribuf; ///primary direct sound buffer
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LPDIRECTSOUNDBUFFER hdsbuf; ///secondary direct sound buffer (stream buffer)
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int buffer_size; ///size in bytes of the direct sound buffer
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int write_offset; ///offset of the write cursor in the direct sound buffer
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int min_free_space; ///if the free space is below this value get_space() will return 0
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///there will always be at least this amout of free space to prevent
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///get_space() from returning wrong values when buffer is 100% full.
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///will be replaced with nBlockAlign in init()
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int underrun_check; ///0 or last reported free space (underrun detection)
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int device_num; ///wanted device number
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GUID device; ///guid of the device
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int audio_volume;
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int device_index;
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int outburst; ///play in multiple of chunks of this size
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int cfg_device;
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int cfg_buffersize;
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struct ao_device_list *listing; ///temporary during list_devs()
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};
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/***************************************************************************************/
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/**
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\brief output error message
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\param err error code
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\return string with the error message
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*/
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static char * dserr2str(int err)
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{
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switch (err) {
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case DS_OK: return "DS_OK";
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case DS_NO_VIRTUALIZATION: return "DS_NO_VIRTUALIZATION";
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case DSERR_ALLOCATED: return "DS_NO_VIRTUALIZATION";
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case DSERR_CONTROLUNAVAIL: return "DSERR_CONTROLUNAVAIL";
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case DSERR_INVALIDPARAM: return "DSERR_INVALIDPARAM";
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case DSERR_INVALIDCALL: return "DSERR_INVALIDCALL";
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case DSERR_GENERIC: return "DSERR_GENERIC";
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case DSERR_PRIOLEVELNEEDED: return "DSERR_PRIOLEVELNEEDED";
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case DSERR_OUTOFMEMORY: return "DSERR_OUTOFMEMORY";
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case DSERR_BADFORMAT: return "DSERR_BADFORMAT";
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case DSERR_UNSUPPORTED: return "DSERR_UNSUPPORTED";
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case DSERR_NODRIVER: return "DSERR_NODRIVER";
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case DSERR_ALREADYINITIALIZED: return "DSERR_ALREADYINITIALIZED";
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case DSERR_NOAGGREGATION: return "DSERR_NOAGGREGATION";
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case DSERR_BUFFERLOST: return "DSERR_BUFFERLOST";
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case DSERR_OTHERAPPHASPRIO: return "DSERR_OTHERAPPHASPRIO";
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case DSERR_UNINITIALIZED: return "DSERR_UNINITIALIZED";
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case DSERR_NOINTERFACE: return "DSERR_NOINTERFACE";
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case DSERR_ACCESSDENIED: return "DSERR_ACCESSDENIED";
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}
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return "unknown";
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}
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/**
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\brief uninitialize direct sound
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*/
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static void UninitDirectSound(struct ao *ao)
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{
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struct priv *p = ao->priv;
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// finally release the DirectSound object
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if (p->hds) {
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IDirectSound_Release(p->hds);
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p->hds = NULL;
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}
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// free DSOUND.DLL
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if (p->hdsound_dll) {
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FreeLibrary(p->hdsound_dll);
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p->hdsound_dll = NULL;
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}
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MP_VERBOSE(ao, "DirectSound uninitialized\n");
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}
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/**
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\brief enumerate direct sound devices
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\return TRUE to continue with the enumeration
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*/
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static BOOL CALLBACK DirectSoundEnum(LPGUID guid, LPCSTR desc, LPCSTR module,
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LPVOID context)
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{
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struct ao *ao = context;
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struct priv *p = ao->priv;
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MP_VERBOSE(ao, "%i %s ", p->device_index, desc);
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if (p->device_num == p->device_index) {
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MP_VERBOSE(ao, "<--");
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if (guid)
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memcpy(&p->device, guid, sizeof(GUID));
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}
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char *guidstr = talloc_strdup(NULL, "");
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if (guid) {
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wchar_t guidwstr[80] = {0};
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StringFromGUID2(guid, guidwstr, MP_ARRAY_SIZE(guidwstr));
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char *nstr = mp_to_utf8(NULL, guidwstr);
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if (nstr) {
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talloc_free(guidstr);
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guidstr = nstr;
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}
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}
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if (p->device_num < 0 && ao->device) {
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if (strcmp(ao->device, guidstr) == 0) {
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MP_VERBOSE(ao, "<--");
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p->device_num = p->device_index;
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if (guid)
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memcpy(&p->device, guid, sizeof(GUID));
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}
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}
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if (p->listing) {
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struct ao_device_desc e = {guidstr, desc};
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ao_device_list_add(p->listing, ao, &e);
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}
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talloc_free(guidstr);
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MP_VERBOSE(ao, "\n");
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p->device_index++;
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return TRUE;
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}
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static void EnumDevs(struct ao *ao)
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{
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struct priv *p = ao->priv;
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p->device_index = 0;
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p->device_num = p->cfg_device;
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HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
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OurDirectSoundEnumerate = (void *)GetProcAddress(p->hdsound_dll,
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"DirectSoundEnumerateA");
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if (OurDirectSoundEnumerate == NULL) {
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MP_ERR(ao, "GetProcAddress FAILED\n");
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return;
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}
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// Enumerate all directsound p->devices
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MP_VERBOSE(ao, "Output Devices:\n");
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OurDirectSoundEnumerate(DirectSoundEnum, ao);
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}
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static int LoadDirectSound(struct ao *ao)
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{
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struct priv *p = ao->priv;
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// initialize directsound
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p->hdsound_dll = LoadLibrary(L"DSOUND.DLL");
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if (p->hdsound_dll == NULL) {
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MP_ERR(ao, "cannot load DSOUND.DLL\n");
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return 0;
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}
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return 1;
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}
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static void list_devs(struct ao *ao, struct ao_device_list *list)
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{
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struct priv *p = ao->priv;
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bool need_init = !p->hdsound_dll;
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if (need_init && !LoadDirectSound(ao))
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return;
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p->listing = list;
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EnumDevs(ao);
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p->listing = NULL;
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if (need_init)
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UninitDirectSound(ao);
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}
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/**
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\brief initilize direct sound
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\return 0 if error, 1 if ok
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*/
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static int InitDirectSound(struct ao *ao)
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{
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struct priv *p = ao->priv;
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DSCAPS dscaps;
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if (!LoadDirectSound(ao))
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return 0;
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HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
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OurDirectSoundCreate =
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(void *)GetProcAddress(p->hdsound_dll, "DirectSoundCreate");
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if (OurDirectSoundCreate == NULL) {
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MP_ERR(ao, "GetProcAddress FAILED\n");
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FreeLibrary(p->hdsound_dll);
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return 0;
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}
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EnumDevs(ao);
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// Create the direct sound object
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if (FAILED(OurDirectSoundCreate((p->device_num > 0) ? &p->device : NULL,
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&p->hds, NULL)))
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{
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MP_ERR(ao, "cannot create a DirectSound device\n");
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FreeLibrary(p->hdsound_dll);
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return 0;
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}
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/* Set DirectSound Cooperative level, ie what control we want over Windows
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* sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the
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* settings of the primary buffer, but also that only the sound of our
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* application will be audible when it will have the focus.
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* !!! (this is not really working as intended yet because to set the
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* cooperative level you need the window handle of your application, and
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* I don't know of any easy way to get it. Especially since we might play
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* sound without any video, and so what window handle should we use ???
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* The hack for now is to use the Desktop window handle - it seems to be
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* working */
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if (IDirectSound_SetCooperativeLevel(p->hds, GetDesktopWindow(),
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DSSCL_EXCLUSIVE))
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{
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MP_ERR(ao, "cannot set direct sound cooperative level\n");
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IDirectSound_Release(p->hds);
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FreeLibrary(p->hdsound_dll);
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return 0;
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}
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MP_VERBOSE(ao, "DirectSound initialized\n");
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memset(&dscaps, 0, sizeof(DSCAPS));
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dscaps.dwSize = sizeof(DSCAPS);
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if (DS_OK == IDirectSound_GetCaps(p->hds, &dscaps)) {
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if (dscaps.dwFlags & DSCAPS_EMULDRIVER)
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MP_VERBOSE(ao, "DirectSound is emulated\n");
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} else {
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MP_VERBOSE(ao, "cannot get device capabilities\n");
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}
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return 1;
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}
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/**
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\brief destroy the direct sound buffer
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*/
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static void DestroyBuffer(struct ao *ao)
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{
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struct priv *p = ao->priv;
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if (p->hdsbuf) {
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IDirectSoundBuffer_Release(p->hdsbuf);
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p->hdsbuf = NULL;
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}
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if (p->hdspribuf) {
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IDirectSoundBuffer_Release(p->hdspribuf);
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p->hdspribuf = NULL;
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}
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}
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/**
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\brief fill sound buffer
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\param data pointer to the sound data to copy
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\param len length of the data to copy in bytes
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\return number of copyed bytes
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*/
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static int write_buffer(struct ao *ao, unsigned char *data, int len)
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{
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struct priv *p = ao->priv;
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HRESULT res;
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LPVOID lpvPtr1;
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DWORD dwBytes1;
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LPVOID lpvPtr2;
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DWORD dwBytes2;
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p->underrun_check = 0;
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// Lock the buffer
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res = IDirectSoundBuffer_Lock(p->hdsbuf, p->write_offset, len, &lpvPtr1,
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&dwBytes1, &lpvPtr2, &dwBytes2, 0);
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// If the buffer was lost, restore and retry lock.
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if (DSERR_BUFFERLOST == res) {
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IDirectSoundBuffer_Restore(p->hdsbuf);
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res = IDirectSoundBuffer_Lock(p->hdsbuf, p->write_offset, len, &lpvPtr1,
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&dwBytes1, &lpvPtr2, &dwBytes2, 0);
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}
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if (SUCCEEDED(res)) {
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memcpy(lpvPtr1, data, dwBytes1);
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if (NULL != lpvPtr2)
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memcpy(lpvPtr2, data + dwBytes1, dwBytes2);
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p->write_offset += dwBytes1 + dwBytes2;
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if (p->write_offset >= p->buffer_size)
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p->write_offset = dwBytes2;
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// Release the data back to DirectSound.
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res = IDirectSoundBuffer_Unlock(p->hdsbuf, lpvPtr1, dwBytes1, lpvPtr2,
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dwBytes2);
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if (SUCCEEDED(res)) {
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// Success.
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DWORD status;
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IDirectSoundBuffer_GetStatus(p->hdsbuf, &status);
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if (!(status & DSBSTATUS_PLAYING))
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res = IDirectSoundBuffer_Play(p->hdsbuf, 0, 0, DSBPLAY_LOOPING);
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return dwBytes1 + dwBytes2;
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}
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}
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// Lock, Unlock, or Restore failed.
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return 0;
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}
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/***************************************************************************************/
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/**
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\brief handle control commands
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\param cmd command
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\param arg argument
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\return CONTROL_OK or CONTROL_UNKNOWN in case the command is not supported
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*/
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static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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{
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struct priv *p = ao->priv;
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DWORD volume;
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switch (cmd) {
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case AOCONTROL_GET_VOLUME: {
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ao_control_vol_t *vol = (ao_control_vol_t *)arg;
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vol->left = vol->right = p->audio_volume;
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return CONTROL_OK;
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}
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case AOCONTROL_SET_VOLUME: {
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ao_control_vol_t *vol = (ao_control_vol_t *)arg;
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volume = p->audio_volume = vol->right;
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if (volume < 1)
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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(
|
||||
fragment=load_fragment('coreaudio.c'),
|
||||
framework_name=['CoreFoundation', 'CoreAudio', 'AudioUnit', 'AudioToolbox'])
|
||||
}, {
|
||||
'name': '--dsound',
|
||||
'desc': 'DirectSound audio output',
|
||||
'func': check_cc(header_name='dsound.h'),
|
||||
}, {
|
||||
'name': '--wasapi',
|
||||
'desc': 'WASAPI audio output',
|
||||
|
|
|
|||
|
|
@ -134,7 +134,6 @@ def build(ctx):
|
|||
( "audio/out/ao_coreaudio_exclusive.c", "coreaudio" ),
|
||||
( "audio/out/ao_coreaudio_properties.c", "coreaudio" ),
|
||||
( "audio/out/ao_coreaudio_utils.c", "coreaudio" ),
|
||||
( "audio/out/ao_dsound.c", "dsound" ),
|
||||
( "audio/out/ao_jack.c", "jack" ),
|
||||
( "audio/out/ao_lavc.c", "encoding" ),
|
||||
( "audio/out/ao_null.c" ),
|
||||
|
|
|
|||
Loading…
Reference in New Issue