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https://github.com/UberGames/lilium-voyager.git
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92935df37b
The version check is required for supporting macOS PPC with SDL 2.0.1 and Travis-CI (Ubuntu Trusty) with SDL 2.0.2. The client now requires SDL 2.0.5 runtime if compiled against SDL 2.0.5 or newer.
445 lines
10 KiB
C
445 lines
10 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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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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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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