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https://github.com/xemu-project/xemu.git
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35f4b58c7a
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@5435 c046a42c-6fe2-441c-8c8c-71466251a162
1088 lines
26 KiB
C
1088 lines
26 KiB
C
/*
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* QEMU DirectSound audio driver
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*
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* Copyright (c) 2005 Vassili Karpov (malc)
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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/*
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* SEAL 1.07 by Carlos 'pel' Hasan was used as documentation
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*/
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#include "qemu-common.h"
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#include "audio.h"
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#define AUDIO_CAP "dsound"
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#include "audio_int.h"
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#define WIN32_LEAN_AND_MEAN
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#include <windows.h>
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#include <mmsystem.h>
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#include <objbase.h>
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#include <dsound.h>
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/* #define DEBUG_DSOUND */
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static struct {
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int lock_retries;
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int restore_retries;
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int getstatus_retries;
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int set_primary;
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int bufsize_in;
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int bufsize_out;
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audsettings_t settings;
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int latency_millis;
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} conf = {
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1,
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1,
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1,
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0,
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16384,
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16384,
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{
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44100,
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2,
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AUD_FMT_S16
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},
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10
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};
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typedef struct {
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LPDIRECTSOUND dsound;
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LPDIRECTSOUNDCAPTURE dsound_capture;
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LPDIRECTSOUNDBUFFER dsound_primary_buffer;
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audsettings_t settings;
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} dsound;
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static dsound glob_dsound;
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typedef struct {
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HWVoiceOut hw;
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LPDIRECTSOUNDBUFFER dsound_buffer;
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DWORD old_pos;
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int first_time;
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#ifdef DEBUG_DSOUND
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DWORD old_ppos;
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DWORD played;
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DWORD mixed;
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#endif
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} DSoundVoiceOut;
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typedef struct {
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HWVoiceIn hw;
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int first_time;
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LPDIRECTSOUNDCAPTUREBUFFER dsound_capture_buffer;
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} DSoundVoiceIn;
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static void dsound_log_hresult (HRESULT hr)
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{
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const char *str = "BUG";
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switch (hr) {
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case DS_OK:
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str = "The method succeeded";
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break;
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#ifdef DS_NO_VIRTUALIZATION
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case DS_NO_VIRTUALIZATION:
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str = "The buffer was created, but another 3D algorithm was substituted";
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break;
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#endif
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#ifdef DS_INCOMPLETE
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case DS_INCOMPLETE:
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str = "The method succeeded, but not all the optional effects were obtained";
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break;
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#endif
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#ifdef DSERR_ACCESSDENIED
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case DSERR_ACCESSDENIED:
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str = "The request failed because access was denied";
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break;
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#endif
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#ifdef DSERR_ALLOCATED
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case DSERR_ALLOCATED:
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str = "The request failed because resources, such as a priority level, were already in use by another caller";
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break;
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#endif
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#ifdef DSERR_ALREADYINITIALIZED
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case DSERR_ALREADYINITIALIZED:
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str = "The object is already initialized";
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break;
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#endif
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#ifdef DSERR_BADFORMAT
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case DSERR_BADFORMAT:
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str = "The specified wave format is not supported";
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break;
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#endif
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#ifdef DSERR_BADSENDBUFFERGUID
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case DSERR_BADSENDBUFFERGUID:
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str = "The GUID specified in an audiopath file does not match a valid mix-in buffer";
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break;
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#endif
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#ifdef DSERR_BUFFERLOST
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case DSERR_BUFFERLOST:
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str = "The buffer memory has been lost and must be restored";
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break;
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#endif
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#ifdef DSERR_BUFFERTOOSMALL
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case DSERR_BUFFERTOOSMALL:
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str = "The buffer size is not great enough to enable effects processing";
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break;
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#endif
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#ifdef DSERR_CONTROLUNAVAIL
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case DSERR_CONTROLUNAVAIL:
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str = "The buffer control (volume, pan, and so on) requested by the caller is not available. Controls must be specified when the buffer is created, using the dwFlags member of DSBUFFERDESC";
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break;
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#endif
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#ifdef DSERR_DS8_REQUIRED
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case DSERR_DS8_REQUIRED:
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str = "A DirectSound object of class CLSID_DirectSound8 or later is required for the requested functionality. For more information, see IDirectSound8 Interface";
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break;
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#endif
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#ifdef DSERR_FXUNAVAILABLE
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case DSERR_FXUNAVAILABLE:
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str = "The effects requested could not be found on the system, or they are in the wrong order or in the wrong location; for example, an effect expected in hardware was found in software";
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break;
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#endif
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#ifdef DSERR_GENERIC
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case DSERR_GENERIC :
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str = "An undetermined error occurred inside the DirectSound subsystem";
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break;
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#endif
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#ifdef DSERR_INVALIDCALL
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case DSERR_INVALIDCALL:
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str = "This function is not valid for the current state of this object";
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break;
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#endif
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#ifdef DSERR_INVALIDPARAM
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case DSERR_INVALIDPARAM:
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str = "An invalid parameter was passed to the returning function";
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break;
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#endif
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#ifdef DSERR_NOAGGREGATION
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case DSERR_NOAGGREGATION:
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str = "The object does not support aggregation";
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break;
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#endif
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#ifdef DSERR_NODRIVER
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case DSERR_NODRIVER:
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str = "No sound driver is available for use, or the given GUID is not a valid DirectSound device ID";
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break;
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#endif
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#ifdef DSERR_NOINTERFACE
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case DSERR_NOINTERFACE:
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str = "The requested COM interface is not available";
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break;
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#endif
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#ifdef DSERR_OBJECTNOTFOUND
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case DSERR_OBJECTNOTFOUND:
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str = "The requested object was not found";
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break;
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#endif
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#ifdef DSERR_OTHERAPPHASPRIO
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case DSERR_OTHERAPPHASPRIO:
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str = "Another application has a higher priority level, preventing this call from succeeding";
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break;
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#endif
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#ifdef DSERR_OUTOFMEMORY
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case DSERR_OUTOFMEMORY:
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str = "The DirectSound subsystem could not allocate sufficient memory to complete the caller's request";
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break;
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#endif
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#ifdef DSERR_PRIOLEVELNEEDED
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case DSERR_PRIOLEVELNEEDED:
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str = "A cooperative level of DSSCL_PRIORITY or higher is required";
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break;
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#endif
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#ifdef DSERR_SENDLOOP
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case DSERR_SENDLOOP:
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str = "A circular loop of send effects was detected";
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break;
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#endif
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#ifdef DSERR_UNINITIALIZED
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case DSERR_UNINITIALIZED:
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str = "The Initialize method has not been called or has not been called successfully before other methods were called";
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break;
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#endif
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#ifdef DSERR_UNSUPPORTED
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case DSERR_UNSUPPORTED:
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str = "The function called is not supported at this time";
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break;
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#endif
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default:
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AUD_log (AUDIO_CAP, "Reason: Unknown (HRESULT %#lx)\n", hr);
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return;
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}
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AUD_log (AUDIO_CAP, "Reason: %s\n", str);
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}
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static void GCC_FMT_ATTR (2, 3) dsound_logerr (
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HRESULT hr,
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const char *fmt,
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...
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)
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{
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va_list ap;
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va_start (ap, fmt);
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AUD_vlog (AUDIO_CAP, fmt, ap);
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va_end (ap);
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dsound_log_hresult (hr);
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}
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static void GCC_FMT_ATTR (3, 4) dsound_logerr2 (
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HRESULT hr,
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const char *typ,
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const char *fmt,
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...
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)
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{
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va_list ap;
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AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
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va_start (ap, fmt);
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AUD_vlog (AUDIO_CAP, fmt, ap);
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va_end (ap);
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dsound_log_hresult (hr);
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}
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static DWORD millis_to_bytes (struct audio_pcm_info *info, DWORD millis)
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{
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return (millis * info->bytes_per_second) / 1000;
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}
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#ifdef DEBUG_DSOUND
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static void print_wave_format (WAVEFORMATEX *wfx)
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{
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dolog ("tag = %d\n", wfx->wFormatTag);
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dolog ("nChannels = %d\n", wfx->nChannels);
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dolog ("nSamplesPerSec = %ld\n", wfx->nSamplesPerSec);
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dolog ("nAvgBytesPerSec = %ld\n", wfx->nAvgBytesPerSec);
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dolog ("nBlockAlign = %d\n", wfx->nBlockAlign);
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dolog ("wBitsPerSample = %d\n", wfx->wBitsPerSample);
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dolog ("cbSize = %d\n", wfx->cbSize);
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}
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#endif
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static int dsound_restore_out (LPDIRECTSOUNDBUFFER dsb)
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{
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HRESULT hr;
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int i;
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for (i = 0; i < conf.restore_retries; ++i) {
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hr = IDirectSoundBuffer_Restore (dsb);
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switch (hr) {
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case DS_OK:
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return 0;
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case DSERR_BUFFERLOST:
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continue;
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default:
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dsound_logerr (hr, "Could not restore playback buffer\n");
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return -1;
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}
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}
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dolog ("%d attempts to restore playback buffer failed\n", i);
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return -1;
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}
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static int waveformat_from_audio_settings (WAVEFORMATEX *wfx, audsettings_t *as)
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{
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memset (wfx, 0, sizeof (*wfx));
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wfx->wFormatTag = WAVE_FORMAT_PCM;
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wfx->nChannels = as->nchannels;
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wfx->nSamplesPerSec = as->freq;
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wfx->nAvgBytesPerSec = as->freq << (as->nchannels == 2);
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wfx->nBlockAlign = 1 << (as->nchannels == 2);
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wfx->cbSize = 0;
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switch (as->fmt) {
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case AUD_FMT_S8:
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case AUD_FMT_U8:
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wfx->wBitsPerSample = 8;
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break;
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case AUD_FMT_S16:
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case AUD_FMT_U16:
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wfx->wBitsPerSample = 16;
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wfx->nAvgBytesPerSec <<= 1;
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wfx->nBlockAlign <<= 1;
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break;
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case AUD_FMT_S32:
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case AUD_FMT_U32:
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wfx->wBitsPerSample = 32;
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wfx->nAvgBytesPerSec <<= 2;
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wfx->nBlockAlign <<= 2;
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break;
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default:
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dolog ("Internal logic error: Bad audio format %d\n", as->freq);
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return -1;
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}
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return 0;
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}
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static int waveformat_to_audio_settings (WAVEFORMATEX *wfx, audsettings_t *as)
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{
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if (wfx->wFormatTag != WAVE_FORMAT_PCM) {
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dolog ("Invalid wave format, tag is not PCM, but %d\n",
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wfx->wFormatTag);
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return -1;
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}
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if (!wfx->nSamplesPerSec) {
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dolog ("Invalid wave format, frequency is zero\n");
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return -1;
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}
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as->freq = wfx->nSamplesPerSec;
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switch (wfx->nChannels) {
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case 1:
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as->nchannels = 1;
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break;
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case 2:
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as->nchannels = 2;
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break;
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default:
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dolog (
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"Invalid wave format, number of channels is not 1 or 2, but %d\n",
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wfx->nChannels
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);
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return -1;
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}
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switch (wfx->wBitsPerSample) {
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case 8:
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as->fmt = AUD_FMT_U8;
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break;
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case 16:
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as->fmt = AUD_FMT_S16;
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break;
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case 32:
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as->fmt = AUD_FMT_S32;
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break;
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default:
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dolog ("Invalid wave format, bits per sample is not "
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"8, 16 or 32, but %d\n",
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wfx->wBitsPerSample);
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return -1;
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}
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return 0;
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}
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#include "dsound_template.h"
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#define DSBTYPE_IN
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#include "dsound_template.h"
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#undef DSBTYPE_IN
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static int dsound_get_status_out (LPDIRECTSOUNDBUFFER dsb, DWORD *statusp)
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{
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HRESULT hr;
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int i;
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for (i = 0; i < conf.getstatus_retries; ++i) {
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hr = IDirectSoundBuffer_GetStatus (dsb, statusp);
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if (FAILED (hr)) {
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dsound_logerr (hr, "Could not get playback buffer status\n");
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return -1;
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}
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|
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if (*statusp & DSERR_BUFFERLOST) {
|
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if (dsound_restore_out (dsb)) {
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return -1;
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}
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continue;
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}
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break;
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}
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return 0;
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}
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|
|
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static int dsound_get_status_in (LPDIRECTSOUNDCAPTUREBUFFER dscb,
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DWORD *statusp)
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{
|
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HRESULT hr;
|
|
|
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hr = IDirectSoundCaptureBuffer_GetStatus (dscb, statusp);
|
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if (FAILED (hr)) {
|
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dsound_logerr (hr, "Could not get capture buffer status\n");
|
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return -1;
|
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}
|
|
|
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return 0;
|
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}
|
|
|
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static void dsound_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
|
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{
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int src_len1 = dst_len;
|
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int src_len2 = 0;
|
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int pos = hw->rpos + dst_len;
|
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st_sample_t *src1 = hw->mix_buf + hw->rpos;
|
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st_sample_t *src2 = NULL;
|
|
|
|
if (pos > hw->samples) {
|
|
src_len1 = hw->samples - hw->rpos;
|
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src2 = hw->mix_buf;
|
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src_len2 = dst_len - src_len1;
|
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pos = src_len2;
|
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}
|
|
|
|
if (src_len1) {
|
|
hw->clip (dst, src1, src_len1);
|
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}
|
|
|
|
if (src_len2) {
|
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dst = advance (dst, src_len1 << hw->info.shift);
|
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hw->clip (dst, src2, src_len2);
|
|
}
|
|
|
|
hw->rpos = pos % hw->samples;
|
|
}
|
|
|
|
static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb)
|
|
{
|
|
int err;
|
|
LPVOID p1, p2;
|
|
DWORD blen1, blen2, len1, len2;
|
|
|
|
err = dsound_lock_out (
|
|
dsb,
|
|
&hw->info,
|
|
0,
|
|
hw->samples << hw->info.shift,
|
|
&p1, &p2,
|
|
&blen1, &blen2,
|
|
1
|
|
);
|
|
if (err) {
|
|
return;
|
|
}
|
|
|
|
len1 = blen1 >> hw->info.shift;
|
|
len2 = blen2 >> hw->info.shift;
|
|
|
|
#ifdef DEBUG_DSOUND
|
|
dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",
|
|
p1, blen1, len1,
|
|
p2, blen2, len2);
|
|
#endif
|
|
|
|
if (p1 && len1) {
|
|
audio_pcm_info_clear_buf (&hw->info, p1, len1);
|
|
}
|
|
|
|
if (p2 && len2) {
|
|
audio_pcm_info_clear_buf (&hw->info, p2, len2);
|
|
}
|
|
|
|
dsound_unlock_out (dsb, p1, p2, blen1, blen2);
|
|
}
|
|
|
|
static void dsound_close (dsound *s)
|
|
{
|
|
HRESULT hr;
|
|
|
|
if (s->dsound_primary_buffer) {
|
|
hr = IDirectSoundBuffer_Release (s->dsound_primary_buffer);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not release primary buffer\n");
|
|
}
|
|
s->dsound_primary_buffer = NULL;
|
|
}
|
|
}
|
|
|
|
static int dsound_open (dsound *s)
|
|
{
|
|
int err;
|
|
HRESULT hr;
|
|
WAVEFORMATEX wfx;
|
|
DSBUFFERDESC dsbd;
|
|
HWND hwnd;
|
|
|
|
hwnd = GetForegroundWindow ();
|
|
hr = IDirectSound_SetCooperativeLevel (
|
|
s->dsound,
|
|
hwnd,
|
|
DSSCL_PRIORITY
|
|
);
|
|
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not set cooperative level for window %p\n",
|
|
hwnd);
|
|
return -1;
|
|
}
|
|
|
|
if (!conf.set_primary) {
|
|
return 0;
|
|
}
|
|
|
|
err = waveformat_from_audio_settings (&wfx, &conf.settings);
|
|
if (err) {
|
|
return -1;
|
|
}
|
|
|
|
memset (&dsbd, 0, sizeof (dsbd));
|
|
dsbd.dwSize = sizeof (dsbd);
|
|
dsbd.dwFlags = DSBCAPS_PRIMARYBUFFER;
|
|
dsbd.dwBufferBytes = 0;
|
|
dsbd.lpwfxFormat = NULL;
|
|
|
|
hr = IDirectSound_CreateSoundBuffer (
|
|
s->dsound,
|
|
&dsbd,
|
|
&s->dsound_primary_buffer,
|
|
NULL
|
|
);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not create primary playback buffer\n");
|
|
return -1;
|
|
}
|
|
|
|
hr = IDirectSoundBuffer_SetFormat (s->dsound_primary_buffer, &wfx);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not set primary playback buffer format\n");
|
|
}
|
|
|
|
hr = IDirectSoundBuffer_GetFormat (
|
|
s->dsound_primary_buffer,
|
|
&wfx,
|
|
sizeof (wfx),
|
|
NULL
|
|
);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not get primary playback buffer format\n");
|
|
goto fail0;
|
|
}
|
|
|
|
#ifdef DEBUG_DSOUND
|
|
dolog ("Primary\n");
|
|
print_wave_format (&wfx);
|
|
#endif
|
|
|
|
err = waveformat_to_audio_settings (&wfx, &s->settings);
|
|
if (err) {
|
|
goto fail0;
|
|
}
|
|
|
|
return 0;
|
|
|
|
fail0:
|
|
dsound_close (s);
|
|
return -1;
|
|
}
|
|
|
|
static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
|
|
{
|
|
HRESULT hr;
|
|
DWORD status;
|
|
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
|
|
LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
|
|
|
|
if (!dsb) {
|
|
dolog ("Attempt to control voice without a buffer\n");
|
|
return 0;
|
|
}
|
|
|
|
switch (cmd) {
|
|
case VOICE_ENABLE:
|
|
if (dsound_get_status_out (dsb, &status)) {
|
|
return -1;
|
|
}
|
|
|
|
if (status & DSBSTATUS_PLAYING) {
|
|
dolog ("warning: Voice is already playing\n");
|
|
return 0;
|
|
}
|
|
|
|
dsound_clear_sample (hw, dsb);
|
|
|
|
hr = IDirectSoundBuffer_Play (dsb, 0, 0, DSBPLAY_LOOPING);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not start playing buffer\n");
|
|
return -1;
|
|
}
|
|
break;
|
|
|
|
case VOICE_DISABLE:
|
|
if (dsound_get_status_out (dsb, &status)) {
|
|
return -1;
|
|
}
|
|
|
|
if (status & DSBSTATUS_PLAYING) {
|
|
hr = IDirectSoundBuffer_Stop (dsb);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not stop playing buffer\n");
|
|
return -1;
|
|
}
|
|
}
|
|
else {
|
|
dolog ("warning: Voice is not playing\n");
|
|
}
|
|
break;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int dsound_write (SWVoiceOut *sw, void *buf, int len)
|
|
{
|
|
return audio_pcm_sw_write (sw, buf, len);
|
|
}
|
|
|
|
static int dsound_run_out (HWVoiceOut *hw)
|
|
{
|
|
int err;
|
|
HRESULT hr;
|
|
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
|
|
LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
|
|
int live, len, hwshift;
|
|
DWORD blen1, blen2;
|
|
DWORD len1, len2;
|
|
DWORD decr;
|
|
DWORD wpos, ppos, old_pos;
|
|
LPVOID p1, p2;
|
|
int bufsize;
|
|
|
|
if (!dsb) {
|
|
dolog ("Attempt to run empty with playback buffer\n");
|
|
return 0;
|
|
}
|
|
|
|
hwshift = hw->info.shift;
|
|
bufsize = hw->samples << hwshift;
|
|
|
|
live = audio_pcm_hw_get_live_out (hw);
|
|
|
|
hr = IDirectSoundBuffer_GetCurrentPosition (
|
|
dsb,
|
|
&ppos,
|
|
ds->first_time ? &wpos : NULL
|
|
);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not get playback buffer position\n");
|
|
return 0;
|
|
}
|
|
|
|
len = live << hwshift;
|
|
|
|
if (ds->first_time) {
|
|
if (conf.latency_millis) {
|
|
DWORD cur_blat;
|
|
|
|
cur_blat = audio_ring_dist (wpos, ppos, bufsize);
|
|
ds->first_time = 0;
|
|
old_pos = wpos;
|
|
old_pos +=
|
|
millis_to_bytes (&hw->info, conf.latency_millis) - cur_blat;
|
|
old_pos %= bufsize;
|
|
old_pos &= ~hw->info.align;
|
|
}
|
|
else {
|
|
old_pos = wpos;
|
|
}
|
|
#ifdef DEBUG_DSOUND
|
|
ds->played = 0;
|
|
ds->mixed = 0;
|
|
#endif
|
|
}
|
|
else {
|
|
if (ds->old_pos == ppos) {
|
|
#ifdef DEBUG_DSOUND
|
|
dolog ("old_pos == ppos\n");
|
|
#endif
|
|
return 0;
|
|
}
|
|
|
|
#ifdef DEBUG_DSOUND
|
|
ds->played += audio_ring_dist (ds->old_pos, ppos, hw->bufsize);
|
|
#endif
|
|
old_pos = ds->old_pos;
|
|
}
|
|
|
|
if ((old_pos < ppos) && ((old_pos + len) > ppos)) {
|
|
len = ppos - old_pos;
|
|
}
|
|
else {
|
|
if ((old_pos > ppos) && ((old_pos + len) > (ppos + bufsize))) {
|
|
len = bufsize - old_pos + ppos;
|
|
}
|
|
}
|
|
|
|
if (audio_bug (AUDIO_FUNC, len < 0 || len > bufsize)) {
|
|
dolog ("len=%d bufsize=%d old_pos=%ld ppos=%ld\n",
|
|
len, bufsize, old_pos, ppos);
|
|
return 0;
|
|
}
|
|
|
|
len &= ~hw->info.align;
|
|
if (!len) {
|
|
return 0;
|
|
}
|
|
|
|
#ifdef DEBUG_DSOUND
|
|
ds->old_ppos = ppos;
|
|
#endif
|
|
err = dsound_lock_out (
|
|
dsb,
|
|
&hw->info,
|
|
old_pos,
|
|
len,
|
|
&p1, &p2,
|
|
&blen1, &blen2,
|
|
0
|
|
);
|
|
if (err) {
|
|
return 0;
|
|
}
|
|
|
|
len1 = blen1 >> hwshift;
|
|
len2 = blen2 >> hwshift;
|
|
decr = len1 + len2;
|
|
|
|
if (p1 && len1) {
|
|
dsound_write_sample (hw, p1, len1);
|
|
}
|
|
|
|
if (p2 && len2) {
|
|
dsound_write_sample (hw, p2, len2);
|
|
}
|
|
|
|
dsound_unlock_out (dsb, p1, p2, blen1, blen2);
|
|
ds->old_pos = (old_pos + (decr << hwshift)) % bufsize;
|
|
|
|
#ifdef DEBUG_DSOUND
|
|
ds->mixed += decr << hwshift;
|
|
|
|
dolog ("played %lu mixed %lu diff %ld sec %f\n",
|
|
ds->played,
|
|
ds->mixed,
|
|
ds->mixed - ds->played,
|
|
abs (ds->mixed - ds->played) / (double) hw->info.bytes_per_second);
|
|
#endif
|
|
return decr;
|
|
}
|
|
|
|
static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
|
|
{
|
|
HRESULT hr;
|
|
DWORD status;
|
|
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
|
|
LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
|
|
|
|
if (!dscb) {
|
|
dolog ("Attempt to control capture voice without a buffer\n");
|
|
return -1;
|
|
}
|
|
|
|
switch (cmd) {
|
|
case VOICE_ENABLE:
|
|
if (dsound_get_status_in (dscb, &status)) {
|
|
return -1;
|
|
}
|
|
|
|
if (status & DSCBSTATUS_CAPTURING) {
|
|
dolog ("warning: Voice is already capturing\n");
|
|
return 0;
|
|
}
|
|
|
|
/* clear ?? */
|
|
|
|
hr = IDirectSoundCaptureBuffer_Start (dscb, DSCBSTART_LOOPING);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not start capturing\n");
|
|
return -1;
|
|
}
|
|
break;
|
|
|
|
case VOICE_DISABLE:
|
|
if (dsound_get_status_in (dscb, &status)) {
|
|
return -1;
|
|
}
|
|
|
|
if (status & DSCBSTATUS_CAPTURING) {
|
|
hr = IDirectSoundCaptureBuffer_Stop (dscb);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not stop capturing\n");
|
|
return -1;
|
|
}
|
|
}
|
|
else {
|
|
dolog ("warning: Voice is not capturing\n");
|
|
}
|
|
break;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int dsound_read (SWVoiceIn *sw, void *buf, int len)
|
|
{
|
|
return audio_pcm_sw_read (sw, buf, len);
|
|
}
|
|
|
|
static int dsound_run_in (HWVoiceIn *hw)
|
|
{
|
|
int err;
|
|
HRESULT hr;
|
|
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
|
|
LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
|
|
int live, len, dead;
|
|
DWORD blen1, blen2;
|
|
DWORD len1, len2;
|
|
DWORD decr;
|
|
DWORD cpos, rpos;
|
|
LPVOID p1, p2;
|
|
int hwshift;
|
|
|
|
if (!dscb) {
|
|
dolog ("Attempt to run without capture buffer\n");
|
|
return 0;
|
|
}
|
|
|
|
hwshift = hw->info.shift;
|
|
|
|
live = audio_pcm_hw_get_live_in (hw);
|
|
dead = hw->samples - live;
|
|
if (!dead) {
|
|
return 0;
|
|
}
|
|
|
|
hr = IDirectSoundCaptureBuffer_GetCurrentPosition (
|
|
dscb,
|
|
&cpos,
|
|
ds->first_time ? &rpos : NULL
|
|
);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not get capture buffer position\n");
|
|
return 0;
|
|
}
|
|
|
|
if (ds->first_time) {
|
|
ds->first_time = 0;
|
|
if (rpos & hw->info.align) {
|
|
ldebug ("warning: Misaligned capture read position %ld(%d)\n",
|
|
rpos, hw->info.align);
|
|
}
|
|
hw->wpos = rpos >> hwshift;
|
|
}
|
|
|
|
if (cpos & hw->info.align) {
|
|
ldebug ("warning: Misaligned capture position %ld(%d)\n",
|
|
cpos, hw->info.align);
|
|
}
|
|
cpos >>= hwshift;
|
|
|
|
len = audio_ring_dist (cpos, hw->wpos, hw->samples);
|
|
if (!len) {
|
|
return 0;
|
|
}
|
|
len = audio_MIN (len, dead);
|
|
|
|
err = dsound_lock_in (
|
|
dscb,
|
|
&hw->info,
|
|
hw->wpos << hwshift,
|
|
len << hwshift,
|
|
&p1,
|
|
&p2,
|
|
&blen1,
|
|
&blen2,
|
|
0
|
|
);
|
|
if (err) {
|
|
return 0;
|
|
}
|
|
|
|
len1 = blen1 >> hwshift;
|
|
len2 = blen2 >> hwshift;
|
|
decr = len1 + len2;
|
|
|
|
if (p1 && len1) {
|
|
hw->conv (hw->conv_buf + hw->wpos, p1, len1, &nominal_volume);
|
|
}
|
|
|
|
if (p2 && len2) {
|
|
hw->conv (hw->conv_buf, p2, len2, &nominal_volume);
|
|
}
|
|
|
|
dsound_unlock_in (dscb, p1, p2, blen1, blen2);
|
|
hw->wpos = (hw->wpos + decr) % hw->samples;
|
|
return decr;
|
|
}
|
|
|
|
static void dsound_audio_fini (void *opaque)
|
|
{
|
|
HRESULT hr;
|
|
dsound *s = opaque;
|
|
|
|
if (!s->dsound) {
|
|
return;
|
|
}
|
|
|
|
hr = IDirectSound_Release (s->dsound);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not release DirectSound\n");
|
|
}
|
|
s->dsound = NULL;
|
|
|
|
if (!s->dsound_capture) {
|
|
return;
|
|
}
|
|
|
|
hr = IDirectSoundCapture_Release (s->dsound_capture);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not release DirectSoundCapture\n");
|
|
}
|
|
s->dsound_capture = NULL;
|
|
}
|
|
|
|
static void *dsound_audio_init (void)
|
|
{
|
|
int err;
|
|
HRESULT hr;
|
|
dsound *s = &glob_dsound;
|
|
|
|
hr = CoInitialize (NULL);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not initialize COM\n");
|
|
return NULL;
|
|
}
|
|
|
|
hr = CoCreateInstance (
|
|
&CLSID_DirectSound,
|
|
NULL,
|
|
CLSCTX_ALL,
|
|
&IID_IDirectSound,
|
|
(void **) &s->dsound
|
|
);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not create DirectSound instance\n");
|
|
return NULL;
|
|
}
|
|
|
|
hr = IDirectSound_Initialize (s->dsound, NULL);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not initialize DirectSound\n");
|
|
|
|
hr = IDirectSound_Release (s->dsound);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not release DirectSound\n");
|
|
}
|
|
s->dsound = NULL;
|
|
return NULL;
|
|
}
|
|
|
|
hr = CoCreateInstance (
|
|
&CLSID_DirectSoundCapture,
|
|
NULL,
|
|
CLSCTX_ALL,
|
|
&IID_IDirectSoundCapture,
|
|
(void **) &s->dsound_capture
|
|
);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not create DirectSoundCapture instance\n");
|
|
}
|
|
else {
|
|
hr = IDirectSoundCapture_Initialize (s->dsound_capture, NULL);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not initialize DirectSoundCapture\n");
|
|
|
|
hr = IDirectSoundCapture_Release (s->dsound_capture);
|
|
if (FAILED (hr)) {
|
|
dsound_logerr (hr, "Could not release DirectSoundCapture\n");
|
|
}
|
|
s->dsound_capture = NULL;
|
|
}
|
|
}
|
|
|
|
err = dsound_open (s);
|
|
if (err) {
|
|
dsound_audio_fini (s);
|
|
return NULL;
|
|
}
|
|
|
|
return s;
|
|
}
|
|
|
|
static struct audio_option dsound_options[] = {
|
|
{"LOCK_RETRIES", AUD_OPT_INT, &conf.lock_retries,
|
|
"Number of times to attempt locking the buffer", NULL, 0},
|
|
{"RESTOURE_RETRIES", AUD_OPT_INT, &conf.restore_retries,
|
|
"Number of times to attempt restoring the buffer", NULL, 0},
|
|
{"GETSTATUS_RETRIES", AUD_OPT_INT, &conf.getstatus_retries,
|
|
"Number of times to attempt getting status of the buffer", NULL, 0},
|
|
{"SET_PRIMARY", AUD_OPT_BOOL, &conf.set_primary,
|
|
"Set the parameters of primary buffer", NULL, 0},
|
|
{"LATENCY_MILLIS", AUD_OPT_INT, &conf.latency_millis,
|
|
"(undocumented)", NULL, 0},
|
|
{"PRIMARY_FREQ", AUD_OPT_INT, &conf.settings.freq,
|
|
"Primary buffer frequency", NULL, 0},
|
|
{"PRIMARY_CHANNELS", AUD_OPT_INT, &conf.settings.nchannels,
|
|
"Primary buffer number of channels (1 - mono, 2 - stereo)", NULL, 0},
|
|
{"PRIMARY_FMT", AUD_OPT_FMT, &conf.settings.fmt,
|
|
"Primary buffer format", NULL, 0},
|
|
{"BUFSIZE_OUT", AUD_OPT_INT, &conf.bufsize_out,
|
|
"(undocumented)", NULL, 0},
|
|
{"BUFSIZE_IN", AUD_OPT_INT, &conf.bufsize_in,
|
|
"(undocumented)", NULL, 0},
|
|
{NULL, 0, NULL, NULL, NULL, 0}
|
|
};
|
|
|
|
static struct audio_pcm_ops dsound_pcm_ops = {
|
|
dsound_init_out,
|
|
dsound_fini_out,
|
|
dsound_run_out,
|
|
dsound_write,
|
|
dsound_ctl_out,
|
|
|
|
dsound_init_in,
|
|
dsound_fini_in,
|
|
dsound_run_in,
|
|
dsound_read,
|
|
dsound_ctl_in
|
|
};
|
|
|
|
struct audio_driver dsound_audio_driver = {
|
|
INIT_FIELD (name = ) "dsound",
|
|
INIT_FIELD (descr = )
|
|
"DirectSound http://wikipedia.org/wiki/DirectSound",
|
|
INIT_FIELD (options = ) dsound_options,
|
|
INIT_FIELD (init = ) dsound_audio_init,
|
|
INIT_FIELD (fini = ) dsound_audio_fini,
|
|
INIT_FIELD (pcm_ops = ) &dsound_pcm_ops,
|
|
INIT_FIELD (can_be_default = ) 1,
|
|
INIT_FIELD (max_voices_out = ) INT_MAX,
|
|
INIT_FIELD (max_voices_in = ) 1,
|
|
INIT_FIELD (voice_size_out = ) sizeof (DSoundVoiceOut),
|
|
INIT_FIELD (voice_size_in = ) sizeof (DSoundVoiceIn)
|
|
};
|