mirror of
https://github.com/DrBeef/ioq3quest.git
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2326a060b9
When you start recording using SDL pulseaudio driver the client sends all audio captured while not recording. 240 milliseconds of audio is sent each frame until the capture buffer is empty. This is a problem for privacy and causes confusing to debug VoIP playback issues on other clients connected to server and when playing back demos.
454 lines
11 KiB
C
454 lines
11 KiB
C
/*
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===========================================================================
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Copyright (C) 1999-2005 Id Software, Inc.
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This file is part of Quake III Arena source code.
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Quake III Arena source code is free software; you can redistribute it
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and/or modify it under the terms of the GNU General Public License as
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published by the Free Software Foundation; either version 2 of the License,
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or (at your option) any later version.
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Quake III Arena source code is distributed in the hope that it will be
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useful, 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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You should have received a copy of the GNU General Public License
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along with Quake III Arena source code; if not, write to the Free Software
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Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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===========================================================================
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*/
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#include <stdlib.h>
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#include <stdio.h>
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#ifdef USE_LOCAL_HEADERS
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# include "SDL.h"
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#else
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# include <SDL.h>
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#endif
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#include "../qcommon/q_shared.h"
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#include "../client/snd_local.h"
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#include "../client/client.h"
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qboolean snd_inited = qfalse;
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cvar_t *s_sdlBits;
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cvar_t *s_sdlSpeed;
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cvar_t *s_sdlChannels;
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cvar_t *s_sdlDevSamps;
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cvar_t *s_sdlMixSamps;
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/* The audio callback. All the magic happens here. */
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static int dmapos = 0;
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static int dmasize = 0;
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static SDL_AudioDeviceID sdlPlaybackDevice;
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#if defined USE_VOIP && SDL_VERSION_ATLEAST( 2, 0, 5 )
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#define USE_SDL_AUDIO_CAPTURE
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static SDL_AudioDeviceID sdlCaptureDevice;
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static cvar_t *s_sdlCapture;
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static float sdlMasterGain = 1.0f;
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#endif
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/*
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===============
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SNDDMA_AudioCallback
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===============
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*/
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static void SNDDMA_AudioCallback(void *userdata, Uint8 *stream, int len)
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{
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int pos = (dmapos * (dma.samplebits/8));
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if (pos >= dmasize)
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dmapos = pos = 0;
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if (!snd_inited) /* shouldn't happen, but just in case... */
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{
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memset(stream, '\0', len);
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return;
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}
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else
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{
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int tobufend = dmasize - pos; /* bytes to buffer's end. */
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int len1 = len;
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int len2 = 0;
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if (len1 > tobufend)
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{
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len1 = tobufend;
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len2 = len - len1;
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}
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memcpy(stream, dma.buffer + pos, len1);
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if (len2 <= 0)
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dmapos += (len1 / (dma.samplebits/8));
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else /* wraparound? */
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{
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memcpy(stream+len1, dma.buffer, len2);
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dmapos = (len2 / (dma.samplebits/8));
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}
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}
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if (dmapos >= dmasize)
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dmapos = 0;
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#ifdef USE_SDL_AUDIO_CAPTURE
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if (sdlMasterGain != 1.0f)
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{
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int i;
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if (dma.isfloat && (dma.samplebits == 32))
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{
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float *ptr = (float *) stream;
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len /= sizeof (*ptr);
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for (i = 0; i < len; i++, ptr++)
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{
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*ptr *= sdlMasterGain;
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}
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}
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else if (dma.samplebits == 16)
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{
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Sint16 *ptr = (Sint16 *) stream;
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len /= sizeof (*ptr);
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for (i = 0; i < len; i++, ptr++)
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{
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*ptr = (Sint16) (((float) *ptr) * sdlMasterGain);
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}
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}
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else if (dma.samplebits == 8)
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{
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Uint8 *ptr = (Uint8 *) stream;
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len /= sizeof (*ptr);
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for (i = 0; i < len; i++, ptr++)
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{
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*ptr = (Uint8) (((float) *ptr) * sdlMasterGain);
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}
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}
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}
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#endif
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}
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static struct
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{
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Uint16 enumFormat;
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char *stringFormat;
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} formatToStringTable[ ] =
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{
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{ AUDIO_U8, "AUDIO_U8" },
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{ AUDIO_S8, "AUDIO_S8" },
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{ AUDIO_U16LSB, "AUDIO_U16LSB" },
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{ AUDIO_S16LSB, "AUDIO_S16LSB" },
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{ AUDIO_U16MSB, "AUDIO_U16MSB" },
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{ AUDIO_S16MSB, "AUDIO_S16MSB" },
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{ AUDIO_F32LSB, "AUDIO_F32LSB" },
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{ AUDIO_F32MSB, "AUDIO_F32MSB" }
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};
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static int formatToStringTableSize = ARRAY_LEN( formatToStringTable );
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/*
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===============
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SNDDMA_PrintAudiospec
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===============
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*/
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static void SNDDMA_PrintAudiospec(const char *str, const SDL_AudioSpec *spec)
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{
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int i;
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char *fmt = NULL;
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Com_Printf("%s:\n", str);
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for( i = 0; i < formatToStringTableSize; i++ ) {
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if( spec->format == formatToStringTable[ i ].enumFormat ) {
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fmt = formatToStringTable[ i ].stringFormat;
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}
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}
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if( fmt ) {
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Com_Printf( " Format: %s\n", fmt );
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} else {
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Com_Printf( " Format: " S_COLOR_RED "UNKNOWN\n");
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}
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Com_Printf( " Freq: %d\n", (int) spec->freq );
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Com_Printf( " Samples: %d\n", (int) spec->samples );
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Com_Printf( " Channels: %d\n", (int) spec->channels );
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}
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/*
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===============
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SNDDMA_Init
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===============
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*/
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qboolean SNDDMA_Init(void)
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{
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SDL_AudioSpec desired;
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SDL_AudioSpec obtained;
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int tmp;
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#ifdef USE_SDL_AUDIO_CAPTURE
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SDL_version sdlVersion;
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#endif
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if (snd_inited)
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return qtrue;
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if (!s_sdlBits) {
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s_sdlBits = Cvar_Get("s_sdlBits", "16", CVAR_ARCHIVE);
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s_sdlSpeed = Cvar_Get("s_sdlSpeed", "0", CVAR_ARCHIVE);
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s_sdlChannels = Cvar_Get("s_sdlChannels", "2", CVAR_ARCHIVE);
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s_sdlDevSamps = Cvar_Get("s_sdlDevSamps", "0", CVAR_ARCHIVE);
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s_sdlMixSamps = Cvar_Get("s_sdlMixSamps", "0", CVAR_ARCHIVE);
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}
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Com_Printf( "SDL_Init( SDL_INIT_AUDIO )... " );
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if (SDL_Init(SDL_INIT_AUDIO) != 0)
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{
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Com_Printf( "FAILED (%s)\n", SDL_GetError( ) );
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return qfalse;
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}
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Com_Printf( "OK\n" );
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Com_Printf( "SDL audio driver is \"%s\".\n", SDL_GetCurrentAudioDriver( ) );
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memset(&desired, '\0', sizeof (desired));
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memset(&obtained, '\0', sizeof (obtained));
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tmp = ((int) s_sdlBits->value);
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if ((tmp != 16) && (tmp != 8))
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tmp = 16;
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desired.freq = (int) s_sdlSpeed->value;
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if(!desired.freq) desired.freq = 22050;
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desired.format = ((tmp == 16) ? AUDIO_S16SYS : AUDIO_U8);
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// I dunno if this is the best idea, but I'll give it a try...
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// should probably check a cvar for this...
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if (s_sdlDevSamps->value)
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desired.samples = s_sdlDevSamps->value;
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else
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{
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// just pick a sane default.
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if (desired.freq <= 11025)
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desired.samples = 256;
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else if (desired.freq <= 22050)
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desired.samples = 512;
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else if (desired.freq <= 44100)
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desired.samples = 1024;
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else
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desired.samples = 2048; // (*shrug*)
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}
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desired.channels = (int) s_sdlChannels->value;
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desired.callback = SNDDMA_AudioCallback;
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sdlPlaybackDevice = SDL_OpenAudioDevice(NULL, SDL_FALSE, &desired, &obtained, SDL_AUDIO_ALLOW_ANY_CHANGE);
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if (sdlPlaybackDevice == 0)
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{
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Com_Printf("SDL_OpenAudioDevice() failed: %s\n", SDL_GetError());
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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return qfalse;
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}
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SNDDMA_PrintAudiospec("SDL_AudioSpec", &obtained);
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// dma.samples needs to be big, or id's mixer will just refuse to
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// work at all; we need to keep it significantly bigger than the
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// amount of SDL callback samples, and just copy a little each time
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// the callback runs.
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// 32768 is what the OSS driver filled in here on my system. I don't
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// know if it's a good value overall, but at least we know it's
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// reasonable...this is why I let the user override.
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tmp = s_sdlMixSamps->value;
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if (!tmp)
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tmp = (obtained.samples * obtained.channels) * 10;
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if (tmp & (tmp - 1)) // not a power of two? Seems to confuse something.
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{
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int val = 1;
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while (val < tmp)
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val <<= 1;
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tmp = val;
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}
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dmapos = 0;
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dma.samplebits = SDL_AUDIO_BITSIZE(obtained.format);
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dma.isfloat = SDL_AUDIO_ISFLOAT(obtained.format);
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dma.channels = obtained.channels;
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dma.samples = tmp;
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dma.submission_chunk = 1;
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dma.speed = obtained.freq;
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dmasize = (dma.samples * (dma.samplebits/8));
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dma.buffer = calloc(1, dmasize);
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#ifdef USE_SDL_AUDIO_CAPTURE
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// !!! FIXME: some of these SDL_OpenAudioDevice() values should be cvars.
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s_sdlCapture = Cvar_Get( "s_sdlCapture", "1", CVAR_ARCHIVE | CVAR_LATCH );
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// !!! FIXME: hopefully pulseaudio capture will be fixed in SDL 2.0.9... https://bugzilla.libsdl.org/show_bug.cgi?id=4087
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SDL_GetVersion(&sdlVersion);
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if (sdlVersion.major == 2 && sdlVersion.minor == 0 && sdlVersion.patch < 9 && Q_stricmp(SDL_GetCurrentAudioDriver(), "pulseaudio") == 0)
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{
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Com_Printf("SDL audio capture support disabled (pulseaudio capture does not work correctly with SDL %d.%d.%d)\n", sdlVersion.major, sdlVersion.minor, sdlVersion.patch);
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}
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else if (!s_sdlCapture->integer)
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{
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Com_Printf("SDL audio capture support disabled by user ('+set s_sdlCapture 1' to enable)\n");
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}
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#if USE_MUMBLE
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else if (cl_useMumble->integer)
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{
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Com_Printf("SDL audio capture support disabled for Mumble support\n");
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}
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#endif
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else
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{
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/* !!! FIXME: list available devices and let cvar specify one, like OpenAL does */
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SDL_AudioSpec spec;
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SDL_zero(spec);
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spec.freq = 48000;
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spec.format = AUDIO_S16SYS;
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spec.channels = 1;
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spec.samples = VOIP_MAX_PACKET_SAMPLES * 4;
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sdlCaptureDevice = SDL_OpenAudioDevice(NULL, SDL_TRUE, &spec, NULL, 0);
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Com_Printf( "SDL capture device %s.\n",
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(sdlCaptureDevice == 0) ? "failed to open" : "opened");
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}
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sdlMasterGain = 1.0f;
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#endif
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Com_Printf("Starting SDL audio callback...\n");
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SDL_PauseAudioDevice(sdlPlaybackDevice, 0); // start callback.
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// don't unpause the capture device; we'll do that in StartCapture.
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Com_Printf("SDL audio initialized.\n");
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snd_inited = qtrue;
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return qtrue;
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}
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/*
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===============
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SNDDMA_GetDMAPos
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===============
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*/
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int SNDDMA_GetDMAPos(void)
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{
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return dmapos;
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}
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/*
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===============
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SNDDMA_Shutdown
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===============
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*/
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void SNDDMA_Shutdown(void)
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{
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if (sdlPlaybackDevice != 0)
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{
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Com_Printf("Closing SDL audio playback device...\n");
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SDL_CloseAudioDevice(sdlPlaybackDevice);
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Com_Printf("SDL audio playback device closed.\n");
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sdlPlaybackDevice = 0;
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}
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#ifdef USE_SDL_AUDIO_CAPTURE
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if (sdlCaptureDevice)
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{
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Com_Printf("Closing SDL audio capture device...\n");
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SDL_CloseAudioDevice(sdlCaptureDevice);
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Com_Printf("SDL audio capture device closed.\n");
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sdlCaptureDevice = 0;
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}
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#endif
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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free(dma.buffer);
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dma.buffer = NULL;
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dmapos = dmasize = 0;
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snd_inited = qfalse;
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Com_Printf("SDL audio shut down.\n");
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}
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/*
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===============
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SNDDMA_Submit
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Send sound to device if buffer isn't really the dma buffer
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===============
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*/
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void SNDDMA_Submit(void)
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{
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SDL_UnlockAudioDevice(sdlPlaybackDevice);
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}
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/*
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===============
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SNDDMA_BeginPainting
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===============
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*/
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void SNDDMA_BeginPainting (void)
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{
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SDL_LockAudioDevice(sdlPlaybackDevice);
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}
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#ifdef USE_VOIP
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void SNDDMA_StartCapture(void)
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{
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#ifdef USE_SDL_AUDIO_CAPTURE
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if (sdlCaptureDevice)
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{
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SDL_ClearQueuedAudio(sdlCaptureDevice);
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SDL_PauseAudioDevice(sdlCaptureDevice, 0);
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}
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#endif
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}
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int SNDDMA_AvailableCaptureSamples(void)
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{
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#ifdef USE_SDL_AUDIO_CAPTURE
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// divided by 2 to convert from bytes to (mono16) samples.
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return sdlCaptureDevice ? (SDL_GetQueuedAudioSize(sdlCaptureDevice) / 2) : 0;
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#else
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return 0;
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#endif
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}
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void SNDDMA_Capture(int samples, byte *data)
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{
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#ifdef USE_SDL_AUDIO_CAPTURE
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// multiplied by 2 to convert from (mono16) samples to bytes.
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if (sdlCaptureDevice)
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{
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SDL_DequeueAudio(sdlCaptureDevice, data, samples * 2);
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}
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else
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#endif
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{
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SDL_memset(data, '\0', samples * 2);
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}
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}
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void SNDDMA_StopCapture(void)
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{
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#ifdef USE_SDL_AUDIO_CAPTURE
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if (sdlCaptureDevice)
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{
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SDL_PauseAudioDevice(sdlCaptureDevice, 1);
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}
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#endif
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}
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void SNDDMA_MasterGain( float val )
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{
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#ifdef USE_SDL_AUDIO_CAPTURE
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sdlMasterGain = val;
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#endif
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}
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#endif
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