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fteqw/engine/client/snd_dma.c
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2024-07-14 19:58:25 +01:00

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/*
Copyright (C) 1996-1997 Id Software, Inc.
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
See the GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
// snd_dma.c -- main control for any streaming sound output devices
#include "quakedef.h"
#ifdef __GNUC__
#define fte_weakstruct __attribute__((weak))
#else
//msvc's uninitialised symbols are always weak, so this is fine.
#define fte_weakstruct
#endif
#ifdef CSQC_DAT
//for sounds following csqc ents
#include "pr_common.h"
extern world_t csqc_world;
#endif
static void S_Play_f(void);
static void S_SoundList_f(void);
#ifdef HAVE_MIXER
static void S_Update_(soundcardinfo_t *sc);
#endif
void S_StopAllSounds(qboolean clear);
static void S_StopAllSounds_f (void);
static void S_UpdateCard(soundcardinfo_t *sc);
static void S_ClearBuffer (soundcardinfo_t *sc);
// =======================================================================
// Internal sound data & structures
// =======================================================================
soundcardinfo_t *sndcardinfo; //the master card.
int snd_blocked = 0;
static qboolean snd_ambient = 1;
qboolean snd_initialized = false;
int snd_speed;
float voicevolumemod = 1;
static struct listener_s
{
int entnum;
vec3_t origin;
vec3_t velocity;
vec3_t forward;
vec3_t right;
vec3_t up;
} listener[MAX_SPLITS];
cvar_t snd_nominaldistance = CVARAFD("s_nominaldistance", "1000", "snd_soundradius", CVAR_CHEAT, "This cvar defines how far an attenuation=1 sound can be heard.");
#define MAX_SFX 8192
sfx_t *known_sfx; // hunk allocated [MAX_SFX]
int num_sfx;
sfx_t *ambient_sfx[NUM_AMBIENTS];
//int desired_speed = 44100;
int desired_bits = 16;
int sound_started=0;
cvar_t mastervolume = CVARFD( "mastervolume", "1", CVAR_ARCHIVE, "Additional multiplier for all other sounds.");
cvar_t bgmvolume = CVARAFD( "musicvolume", "0.3", "bgmvolume", CVAR_ARCHIVE,
"Volume level for background music.");
cvar_t volume = CVARAFD( "volume", "0.7", /*q3*/"s_volume",CVAR_ARCHIVE,
"Volume level for game sounds (does not affect music, voice, or cinematics).");
cvar_t nosound = CVARFD( "nosound", "0", CVAR_ARCHIVE,
"Disable all sound from the engine. Cannot be overriden by configs or anything if set via the -nosound commandline argument.");
cvar_t snd_precache = CVARAF( "s_precache", "1",
"precache", 0);
cvar_t snd_loadas8bit = CVARAFD( "s_loadas8bit", "0",
"loadas8bit", CVAR_ARCHIVE,
"Downsample sounds on load as lower quality 8-bit sound, to save memory.");
cvar_t snd_loadasstereo = CVARD( "snd_loadasstereo", "0",
"Force mono sounds to load as if stereo ones, to waste memory. Not normally useful.");
cvar_t ambient_level = CVARAFD( "s_ambientlevel", "0.3",
"ambient_level", CVAR_ARCHIVE,
"This controls the volume levels of automatic area-based sounds (like water or sky), and is quite annoying. If you're playing deathmatch you'll definitely want this OFF.");
cvar_t ambient_fade = CVARAF( "s_ambientfade", "100",
"ambient_fade", CVAR_ARCHIVE);
cvar_t snd_noextraupdate = CVARAF( "s_noextraupdate", "0",
"snd_noextraupdate", 0);
cvar_t snd_show = CVARAF( "s_show", "0",
"snd_show", 0);
#ifdef __DJGPP__
#define DEFAULT_SND_KHZ "11"
#else
//fixme: are android devices more likely to use 44.1khz?
#define DEFAULT_SND_KHZ "48" //most modern systems should go with 48khz audio (dvd quality). various hardware codecs support nothing else.
#endif
cvar_t snd_khz = CVARAFD( "s_khz", DEFAULT_SND_KHZ,
"snd_khz", CVAR_ARCHIVE, "Sound speed, in kilohertz. Common values are 11, 22, 44, 48. Values above 1000 are explicitly in hertz.");
cvar_t snd_inactive = CVARAFD( "s_inactive", "1",
"snd_inactive", CVAR_ARCHIVE,
"Play sound while application is inactive (ie: tabbed out). Needs a snd_restart if changed."
); //set if you want sound even when tabbed out.
cvar_t _snd_mixahead = CVARAFD( "s_mixahead", "0.1",
"_snd_mixahead", CVAR_ARCHIVE, "Specifies how many seconds to prebuffer audio. Lower values give less latency, but might result in crackling. Different hardware/drivers have different tolerances, and this setting may be ignored completely where drivers are expected to provide their own tolerances.");
cvar_t snd_leftisright = CVARAF( "s_swapstereo", "0",
"snd_leftisright", CVAR_ARCHIVE);
cvar_t snd_eax = CVARAF( "s_eax", "0",
"snd_eax", 0);
cvar_t snd_speakers = CVARAFD( "s_numspeakers", "2",
"snd_numspeakers", CVAR_ARCHIVE, "Number of hardware audio channels to use. "FULLENGINENAME" supports up to 6.");
cvar_t snd_buffersize = CVARAF( "s_buffersize", "0",
"snd_buffersize", 0);
#if defined(MIXER_F32) && defined(FTE_TARGET_WEB) //let emscripten have float audio
cvar_t snd_samplebits = CVARAF( "s_bits", "32",
"snd_samplebits", CVAR_ARCHIVE);
#else
cvar_t snd_samplebits = CVARAF( "s_bits", "16",
"snd_samplebits", CVAR_ARCHIVE);
#endif
cvar_t snd_playersoundvolume = CVARAFD( "s_localvolume", "1",
"snd_localvolume", CVAR_ARCHIVE,
"Sound level for sounds local or originating from the player such as firing and pain sounds."); //sugested by crunch
cvar_t snd_doppler = CVARAFD( "s_doppler", "0",
"snd_doppler", CVAR_ARCHIVE,
"Enables doppler, with a multiplier for the scale.");
cvar_t snd_doppler_min = CVARAFD( "s_doppler_min", "0.5",
"snd_doppler_min", CVAR_ARCHIVE,
"Slowest allowed doppler scale.");
cvar_t snd_doppler_max = CVARAFD( "s_doppler_max", "2",
"snd_doppler_max", CVAR_ARCHIVE,
"Highest allowed doppler scale, to avoid things getting too weird.");
cvar_t snd_playbackrate = CVARFD( "snd_playbackrate", "1", CVAR_CHEAT, "Debugging cvar that changes the playback rate of all new sounds.");
cvar_t snd_ignoregamespeed = CVARFD( "snd_ignoregamespeed", "0", 0, "When set, allows sounds to desynchronise with game time or demo speeds.");
cvar_t snd_ignorecueloops = CVARD( "snd_ignorecueloops", "0", "Ignores cue commands in wav files, for q3 compat.");
cvar_t snd_linearresample = CVARAF( "s_linearresample", "1",
"snd_linearresample", 0);
cvar_t snd_linearresample_stream = CVARAF( "s_linearresample_stream", "0",
"snd_linearresample_stream", 0);
cvar_t snd_mixerthread = CVARAD( "s_mixerthread", "1",
"snd_mixerthread", "When enabled sound mixing will be run on a separate thread. Currently supported only by directsound. Other drivers may unconditionally thread audio. Set to 0 only if you have issues.");
cvar_t snd_device = CVARAFD( "s_device", "",
"snd_device", CVAR_ARCHIVE, "This is the sound device(s) to use, in the form of driver:device.\nIf desired, multiple devices can be listed in space-seperated (quoted) tokens. _s_device_opts contains any enumerated options.\nIn all seriousness, use the menu to set this if you wish to keep your hair.");
cvar_t snd_device_opts = CVARFD( "_s_device_opts", "", CVAR_NOSET|CVAR_NOSAVE, "The possible audio output devices, in \"value\" \"description\" pairs, for gamecode to read.");
#ifdef VOICECHAT
static void QDECL S_Voip_Play_Callback(cvar_t *var, char *oldval);
cvar_t snd_voip_capturedevice = CVARF("cl_voip_capturedevice", "", CVAR_ARCHIVE);
cvar_t snd_voip_capturedevice_opts = CVARFD("_cl_voip_capturedevice_opts", "", CVAR_NOSET|CVAR_NOSAVE, "The possible audio capture devices, in \"value\" \"description\" pairs, for gamecode to read.");
int voipbutton; //+voip, no longer part of cl_voip_send to avoid it getting saved
cvar_t snd_voip_send = CVARFD("cl_voip_send", "0", CVAR_ARCHIVE|CVAR_NOTFROMSERVER, "Sends voice-over-ip data to the server whenever it is set.\n0: only send voice if +voip is pressed.\n1: voice activation.\n2: constantly send.\n+4: Do not send to game, only to rtp sessions.");
cvar_t snd_voip_test = CVARD("cl_voip_test", "0", "If 1, enables you to hear your own voice directly, bypassing the server and thus without networking latency, but is fine for checking audio levels. Note that sv_voip_echo can be set if you want to include latency and packetloss considerations, but setting that cvar requires server admin access and is thus much harder to use.");
cvar_t snd_voip_vad_threshhold = CVARFD("cl_voip_vad_threshhold", "15", CVAR_ARCHIVE, "This is the threshhold for voice-activation-detection when sending voip data");
cvar_t snd_voip_vad_delay = CVARD("cl_voip_vad_delay", "0.3", "Keeps sending voice data for this many seconds after voice activation would normally stop");
cvar_t snd_voip_capturingvol = CVARAFD("cl_voip_capturingvol", "0.5", NULL, CVAR_ARCHIVE, "Volume multiplier applied while capturing, to avoid your audio from being heard by others. Does not affect game volume when others speak (minimum of cl_voip_capturingvol and cl_voip_ducking is used).");
cvar_t snd_voip_showmeter = CVARAFD("cl_voip_showmeter", "1", NULL, CVAR_ARCHIVE, "Shows your speech volume above the standard hud. 0=hide, 1=show when transmitting, 2=ignore voice-activation disable");
cvar_t snd_voip_play = CVARAFCD("cl_voip_play", "1", NULL, CVAR_ARCHIVE, S_Voip_Play_Callback, "Enables voip playback. Value is a volume scaler.");
cvar_t snd_voip_ducking = CVARAFD("cl_voip_ducking", "0.5", NULL, CVAR_ARCHIVE, "Scales game audio by this much when someone is talking to you. Does not affect your speaker volume when you speak (minimum of cl_voip_capturingvol and cl_voip_ducking is used).");
cvar_t snd_voip_micamp = CVARAFD("cl_voip_micamp", "2", NULL, CVAR_ARCHIVE, "Amplifies your microphone when using voip.");
cvar_t snd_voip_codec = CVARAFD("cl_voip_codec", "", NULL, CVAR_ARCHIVE, "0: speex(@11khz). 1: raw. 2: opus. 3: speex(@8khz). 4: speex(@16). 5:speex(@32). 6: pcma. 7: pcmu.");
#ifdef HAVE_SPEEX
cvar_t snd_voip_noisefilter = CVARAFD("cl_voip_noisefilter", "1", NULL, CVAR_ARCHIVE, "Enable the use of the noise cancelation filter.");
cvar_t snd_voip_autogain = CVARAFD("cl_voip_autogain", "0", NULL, CVAR_ARCHIVE, "Attempts to normalize your voice levels to a standard level. Useful for lazy people, but interferes with voice activation levels.");
#endif
cvar_t snd_voip_bitrate = CVARAFD("cl_voip_bitrate", "3000", NULL, CVAR_ARCHIVE, "For codecs with non-specific bitrates, this specifies the target bitrate to use.");
#endif
extern vfsfile_t *rawwritefile;
#ifdef MULTITHREAD
void *mixermutex;
void S_LockMixer(void)
{
Sys_LockMutex(mixermutex);
}
void S_UnlockMixer(void)
{
Sys_UnlockMutex(mixermutex);
}
#else
void S_LockMixer(void)
{
}
void S_UnlockMixer(void)
{
}
#endif
void S_AmbientOff (void)
{
snd_ambient = false;
}
void S_AmbientOn (void)
{
snd_ambient = true;
}
qboolean S_HaveOutput(void)
{
return sound_started && sndcardinfo;
}
void S_SoundInfo_f(void)
{
int i, j;
int active, known;
soundcardinfo_t *sc;
if (!sound_started)
{
Con_Printf ("sound system not started\n");
return;
}
if (!sndcardinfo)
{
Con_Printf ("No sound cards\n");
return;
}
for (sc = sndcardinfo; sc; sc = sc->next)
{
Con_Printf("Audio Device: %s\n", sc->name);
Con_Printf(" %d channels, %gkhz, %d bit audio%s\n", sc->sn.numchannels, sc->sn.speed/1000.0, sc->sn.samplebytes*8, sc->selfpainting?", threaded":"");
Con_Printf(" %d samples in buffer\n", sc->sn.samples);
for (i = 0, active = 0, known = 0; i < sc->total_chans; i++)
{
if (sc->channel[i].sfx)
{
known++;
for (j = 0; j < MAXSOUNDCHANNELS; j++)
{
if (sc->channel[i].vol[j])
{
active++;
break;
}
}
if (j<MAXSOUNDCHANNELS)
Con_Printf(" %s (%i %i, %g %g %g, active)\n", sc->channel[i].sfx->name, sc->channel[i].entnum, sc->channel[i].entchannel, sc->channel[i].origin[0], sc->channel[i].origin[1], sc->channel[i].origin[2]);
else
Con_DPrintf(" %s (%i %i, %g %g %g, inactive)\n", sc->channel[i].sfx->name, sc->channel[i].entnum, sc->channel[i].entchannel, sc->channel[i].origin[0], sc->channel[i].origin[1], sc->channel[i].origin[2]);
}
}
Con_Printf(" %d/%d/%"PRIiSIZE"/%"PRIiSIZE" active/known/highest/max\n", active, known, sc->total_chans, sc->max_chans);
for (i = 0; i < sc->sn.numchannels; i++)
{
Con_Printf(" chan %i: fwd:%-4g rt:%-4g up:%-4g dist:%-4g\n", i, sc->speakerdir[i][0], sc->speakerdir[i][1], sc->speakerdir[i][2], sc->dist[i]);
}
}
}
#ifdef VOICECHAT
#ifdef SPEEX_STATIC
#include <speex/speex.h>
#include <speex/speex_preprocess.h>
#else
typedef struct {int stuff[15];} SpeexBits;
typedef struct SpeexMode SpeexMode;
typedef struct SpeexPreprocessState SpeexPreprocessState;
typedef qint16_t spx_int16_t;
#define SPEEX_MODEID_NB 0
#define SPEEX_MODEID_WB 1
#define SPEEX_MODEID_UWB 2
#define SPEEX_GET_FRAME_SIZE 3
#define SPEEX_SET_SAMPLING_RATE 24
#define SPEEX_GET_SAMPLING_RATE 25
#define SPEEX_PREPROCESS_SET_DENOISE 0
#define SPEEX_PREPROCESS_SET_AGC 2
#define SPEEX_PREPROCESS_SET_AGC_MAX_GAIN 30
#endif
enum
{
VOIP_SPEEX_OLD = 0, //original supported codec (with needless padding and at the wrong rate to keep quake implementations easy)
VOIP_RAW16 = 1, //support is not recommended.
VOIP_OPUS = 2, //supposed to be better than speex.
VOIP_SPEEX_NARROW = 3, //narrowband speex. packed data.
VOIP_SPEEX_WIDE = 4, //wideband speex. packed data.
VOIP_SPEEX_ULTRAWIDE = 5,//wideband speex. packed data.
VOIP_PCMA = 6, //G711 is kinda shit, encoding audio at 8khz with funny truncation for 13bit to 8bit
VOIP_PCMU = 7, //ulaw version of g711 (instead of alaw)
VOIP_INVALID = 16 //not currently generating audio.
};
#if defined(HAVE_LEGACY) && defined(HAVE_OPUS) && defined(HAVE_SPEEX)
#define VOIP_DEFAULT_CODEC ((cls.protocol==CP_QUAKEWORLD && !(cls.fteprotocolextensions2&PEXT2_REPLACEMENTDELTAS))?VOIP_SPEEX_OLD:VOIP_OPUS) //opus is preferred, but ezquake is still common and only supports my first attempt at voice compression so favour that for mvdsv servers.
#elif defined(HAVE_OPUS)
#define VOIP_DEFAULT_CODEC VOIP_OPUS
#elif defined(HAVE_SPEEX)
#define VOIP_DEFAULT_CODEC VOIP_SPEEX_OLD
#else
#define VOIP_DEFAULT_CODEC VOIP_PCMA
#endif
static struct
{
#ifdef HAVE_SPEEX
struct
{
qboolean inited;
qboolean loaded;
dllhandle_t *speexlib;
SpeexBits encbits;
SpeexBits decbits[MAX_CLIENTS];
const SpeexMode *modenb;
const SpeexMode *modewb;
const SpeexMode *modeuwb;
} speex;
struct
{
qboolean inited;
qboolean loaded;
dllhandle_t *speexdsplib;
SpeexPreprocessState *preproc; //filter out noise
int curframesize;
int cursamplerate;
} speexdsp;
#endif
#ifdef HAVE_OPUS
struct
{
qboolean inited;
qboolean loaded;
dllhandle_t *opuslib;
} opus;
#endif
unsigned char enccodec;
void *encoder;
unsigned int encframesize;
unsigned int encsamplerate;
void *decoder[MAX_CLIENTS];
float declevel[MAX_CLIENTS];
unsigned char deccodec[MAX_CLIENTS];
unsigned char decseq[MAX_CLIENTS]; /*sender's sequence, to detect+cover minor packetloss*/
unsigned char decgen[MAX_CLIENTS]; /*last generation. if it changes, we flush speex to reset packet loss*/
unsigned int decsamplerate[MAX_CLIENTS];
unsigned int decframesize[MAX_CLIENTS];
float lastspoke[MAX_CLIENTS]; /*time when they're no longer considered talking. if future, they're talking (timeout avoids flickering, and harder to troll with fake-tourettes when noone is looking)*/
float lastspoke_any;
unsigned char capturebuf[32768]; /*pending data*/
unsigned int capturepos;/*amount of pending data*/
unsigned int encsequence;/*the outgoing sequence count*/
unsigned int enctimestamp;/*for rtp streaming*/
unsigned int generation;/*incremented whenever capture is restarted*/
qboolean wantsend; /*set if we're capturing data to send*/
float voiplevel; /*your own voice level*/
unsigned int dumps; /*trigger a new generation thing after a bit*/
unsigned int keeps; /*for vad_delay*/
int curbitrate;
snd_capture_driver_t *cdriver;/*capture driver's functions*/
void *cdriverctx; /*capture driver context*/
} s_voip;
#ifdef HAVE_OPUS
#define OPUS_APPLICATION_VOIP 2048
#define OPUS_SET_BITRATE_REQUEST 4002
#define OPUS_RESET_STATE 4028
#ifdef OPUS_STATIC
#include "opus.h"
#define qopus_encoder_create opus_encoder_create
#define qopus_encoder_destroy opus_encoder_destroy
#define qopus_encoder_ctl opus_encoder_ctl
#define qopus_encode opus_encode
#define qopus_decoder_create opus_decoder_create
#define qopus_decoder_destroy opus_decoder_destroy
#define qopus_decoder_ctl opus_decoder_ctl
#define qopus_decode opus_decode
#else
#define opus_int32 int
#define opus_int16 short
#define OpusEncoder void
#define OpusDecoder void
static OpusEncoder *(VARGS *qopus_encoder_create)(opus_int32 Fs, int channels, int application, int *error);
static void (VARGS *qopus_encoder_destroy)(OpusEncoder *st);
static int (VARGS *qopus_encoder_ctl)(OpusEncoder *st, int request, ...);
static opus_int32 (VARGS *qopus_encode)(OpusEncoder *st, const opus_int16 *pcm, int frame_size, unsigned char *data, opus_int32 max_data_bytes);
static OpusDecoder *(VARGS *qopus_decoder_create)(opus_int32 Fs, int channels, int *error);
static void (VARGS *qopus_decoder_destroy)(OpusDecoder *st);
static int (VARGS *qopus_decoder_ctl)(OpusDecoder *st, int request, ...);
static int (VARGS *qopus_decode)(OpusDecoder *st, const unsigned char *data, opus_int32 len, opus_int16 *pcm, int frame_size, int decode_fec);
static dllfunction_t qopusfuncs[] =
{
{(void*)&qopus_encoder_create, "opus_encoder_create"},
{(void*)&qopus_encoder_destroy, "opus_encoder_destroy"},
{(void*)&qopus_encoder_ctl, "opus_encoder_ctl"},
{(void*)&qopus_encode, "opus_encode"},
{(void*)&qopus_decoder_create, "opus_decoder_create"},
{(void*)&qopus_decoder_destroy, "opus_decoder_destroy"},
{(void*)&qopus_decoder_ctl, "opus_decoder_ctl"},
{(void*)&qopus_decode, "opus_decode"},
{NULL}
};
#endif
static qboolean S_Opus_Init(void)
{
#ifndef OPUS_STATIC
#ifdef _WIN32
char *modulename = "libopus-0" ARCH_DL_POSTFIX;
#else
char *modulename = "libopus"ARCH_DL_POSTFIX".0";
#endif
if (s_voip.opus.inited)
return s_voip.opus.loaded;
s_voip.opus.inited = true;
s_voip.opus.opuslib = Sys_LoadLibrary(modulename, qopusfuncs);
if (!s_voip.opus.opuslib)
{
Con_Printf("%s not found. Voice chat is not available.\n", modulename);
return false;
}
#endif
s_voip.opus.loaded = true;
return s_voip.opus.loaded;
}
#endif
#ifdef HAVE_SPEEX
#ifdef SPEEX_STATIC
#define qspeex_lib_get_mode speex_lib_get_mode
#define qspeex_bits_init speex_bits_init
#define qspeex_bits_reset speex_bits_reset
#define qspeex_bits_write speex_bits_write
#define qspeex_preprocess_state_init speex_preprocess_state_init
#define qspeex_preprocess_state_destroy speex_preprocess_state_destroy
#define qspeex_preprocess_ctl speex_preprocess_ctl
#define qspeex_preprocess_run speex_preprocess_run
#define qspeex_encoder_init speex_encoder_init
#define qspeex_encoder_destroy speex_encoder_destroy
#define qspeex_encoder_ctl speex_encoder_ctl
#define qspeex_encode_int speex_encode_int
#define qspeex_decoder_init speex_decoder_init
#define qspeex_decoder_destroy speex_decoder_destroy
#define qspeex_decode_int speex_decode_int
#define qspeex_bits_read_from speex_bits_read_from
#else
static const SpeexMode *(VARGS *qspeex_lib_get_mode)(int mode);
static void (VARGS *qspeex_bits_init)(SpeexBits *bits);
static void (VARGS *qspeex_bits_reset)(SpeexBits *bits);
static int (VARGS *qspeex_bits_write)(SpeexBits *bits, char *bytes, int max_len);
static SpeexPreprocessState *(VARGS *qspeex_preprocess_state_init)(int frame_size, int sampling_rate);
static void (VARGS *qspeex_preprocess_state_destroy)(SpeexPreprocessState *st);
static int (VARGS *qspeex_preprocess_ctl)(SpeexPreprocessState *st, int request, void *ptr);
static int (VARGS *qspeex_preprocess_run)(SpeexPreprocessState *st, spx_int16_t *x);
static void * (VARGS *qspeex_encoder_init)(const SpeexMode *mode);
static int (VARGS *qspeex_encoder_ctl)(void *state, int request, void *ptr);
static int (VARGS *qspeex_encode_int)(void *state, spx_int16_t *in, SpeexBits *bits);
static void *(VARGS *qspeex_decoder_init)(const SpeexMode *mode);
static void (VARGS *qspeex_decoder_destroy)(void *state);
static int (VARGS *qspeex_decode_int)(void *state, SpeexBits *bits, spx_int16_t *out);
static void (VARGS *qspeex_bits_read_from)(SpeexBits *bits, char *bytes, int len);
static dllfunction_t qspeexfuncs[] =
{
{(void*)&qspeex_lib_get_mode, "speex_lib_get_mode"},
{(void*)&qspeex_bits_init, "speex_bits_init"},
{(void*)&qspeex_bits_reset, "speex_bits_reset"},
{(void*)&qspeex_bits_write, "speex_bits_write"},
{(void*)&qspeex_encoder_init, "speex_encoder_init"},
{(void*)&qspeex_encoder_ctl, "speex_encoder_ctl"},
{(void*)&qspeex_encode_int, "speex_encode_int"},
{(void*)&qspeex_decoder_init, "speex_decoder_init"},
{(void*)&qspeex_decoder_destroy, "speex_decoder_destroy"},
{(void*)&qspeex_decode_int, "speex_decode_int"},
{(void*)&qspeex_bits_read_from, "speex_bits_read_from"},
{NULL}
};
static dllfunction_t qspeexdspfuncs[] =
{
{(void*)&qspeex_preprocess_state_init, "speex_preprocess_state_init"},
{(void*)&qspeex_preprocess_state_destroy, "speex_preprocess_state_destroy"},
{(void*)&qspeex_preprocess_ctl, "speex_preprocess_ctl"},
{(void*)&qspeex_preprocess_run, "speex_preprocess_run"},
{NULL}
};
#endif
static qboolean S_SpeexDSP_Init(void)
{
#ifndef SPEEX_STATIC
if (s_voip.speexdsp.inited)
return s_voip.speexdsp.loaded;
s_voip.speexdsp.inited = true;
s_voip.speexdsp.speexdsplib = Sys_LoadLibrary("libspeexdsp", qspeexdspfuncs);
if (!s_voip.speexdsp.speexdsplib)
{
Con_Printf("libspeexdsp not found. Your mic may be noisy.\n");
return false;
}
#endif
s_voip.speexdsp.loaded = true;
return s_voip.speexdsp.loaded;
}
static qboolean S_Speex_Init(void)
{
#ifndef SPEEX_STATIC
if (s_voip.speex.inited)
return s_voip.speex.loaded;
s_voip.speex.inited = true;
s_voip.speex.speexlib = Sys_LoadLibrary("libspeex", qspeexfuncs);
if (!s_voip.speex.speexlib)
{
Con_Printf("libspeex not found. Voice chat is not available.\n");
return false;
}
#endif
s_voip.speex.modenb = qspeex_lib_get_mode(SPEEX_MODEID_NB);
s_voip.speex.modewb = qspeex_lib_get_mode(SPEEX_MODEID_WB);
s_voip.speex.modeuwb = qspeex_lib_get_mode(SPEEX_MODEID_UWB);
s_voip.speex.loaded = true;
return s_voip.speex.loaded;
}
#endif
#ifdef AVAIL_OPENAL
extern snd_capture_driver_t OPENAL_Capture;
#endif
#ifdef _WIN32
snd_capture_driver_t fte_weakstruct DSOUND_Capture;
#endif
snd_capture_driver_t fte_weakstruct OSS_Capture;
snd_capture_driver_t fte_weakstruct SDL_Capture;
snd_capture_driver_t *capturedrivers[] =
{
#ifdef _WIN32
&DSOUND_Capture,
#endif
&SDL_Capture,
&OSS_Capture,
#ifdef AVAIL_OPENAL
&OPENAL_Capture,
#endif
NULL
};
size_t PCMA_Decode(short *out, unsigned char *in, size_t samples)
{
size_t i = 0;
for (i = 0; i < samples; i++)
{
unsigned char inv = in[i]^0x55; //g711 alaw inverts every other bit
int m = inv&0xf;
int e = (inv&0x70)>>4;
if (e)
m = (((m)<<1)|0x21) << (e-1);
else
m = (((m)<<1)|1);
if (inv & 0x80)
out[i] = -m;
else
out[i] = m;
}
return i;
}
size_t PCMA_Encode(unsigned char *out, size_t outsize, short *in, size_t samples)
{
size_t i = 0;
for (i = 0; i < samples; i++)
{
int o = in[i];
unsigned char b;
if (o < 0)
{
o = -o;
b = 0x80;
}
else
b = 0;
if (o >= 0x0800)
b |= ((o>>7)&0xf) | 0x70;
else if (o >= 0x0400)
b |= ((o>>6)&0xf) | 0x60;
else if (o >= 0x0200)
b |= ((o>>5)&0xf) | 0x50;
else if (o >= 0x0100)
b |= ((o>>4)&0xf) | 0x40;
else if (o >= 0x0080)
b |= ((o>>3)&0xf) | 0x30;
else if (o >= 0x0040)
b |= ((o>>2)&0xf) | 0x20;
else if (o >= 0x0020)
b |= ((o>>1)&0xf) | 0x10;
else
b |= ((o>>1)&0xf) | 0x00;
out[i] = b^0x55; //invert every-other bit.
}
return samples;
}
size_t PCMU_Decode(short *out, unsigned char *in, size_t samples)
{
size_t i = 0;
for (i = 0; i < samples; i++)
{
unsigned char inv = in[i]^0xff;
int m = (((inv&0xf)<<1)|0x21) << ((inv&0x70)>>4);
m -= 33;
if (inv & 0x80)
out[i] = -m;
else
out[i] = m;
}
return i;
}
size_t PCMU_Encode(unsigned char *out, size_t outsize, short *in, size_t samples)
{
size_t i = 0;
for (i = 0; i < samples; i++)
{
int o = in[i];
unsigned char b;
if (o < 0)
{
o = ~o;
b = 0x80;
}
else
b = 0;
o+=33;
if (o >= 0x1000)
b |= ((o>>8)&0xf) | 0x70;
else if (o >= 0x0800)
b |= ((o>>7)&0xf) | 0x60;
else if (o >= 0x0400)
b |= ((o>>6)&0xf) | 0x50;
else if (o >= 0x0200)
b |= ((o>>5)&0xf) | 0x40;
else if (o >= 0x0100)
b |= ((o>>4)&0xf) | 0x30;
else if (o >= 0x0080)
b |= ((o>>3)&0xf) | 0x20;
else if (o >= 0x0040)
b |= ((o>>2)&0xf) | 0x10;
else
b |= ((o>>1)&0xf) | 0x00;
out[i] = b^0xff;
}
return samples;
}
void S_Voip_Decode(unsigned int sender, unsigned int codec, unsigned int gen, unsigned char seq, unsigned int bytes, unsigned char *data)
{
unsigned char *start;
short decodebuf[8192];
unsigned int decodesamps, len, drops;
int r;
if (sender >= MAX_CLIENTS)
return;
decodesamps = 0;
drops = 0;
start = data;
s_voip.lastspoke[sender] = realtime + 0.5;
if (s_voip.lastspoke[sender] > s_voip.lastspoke_any)
s_voip.lastspoke_any = s_voip.lastspoke[sender];
//if they re-started speaking, flush any old state to avoid things getting weirdly delayed and reset the codec properly.
if (s_voip.decgen[sender] != gen || s_voip.deccodec[sender] != codec)
{
S_RawAudio(sender, NULL, s_voip.decsamplerate[sender], 0, 1, 2, 0);
if (s_voip.deccodec[sender] != codec)
{
//make sure old state is closed properly.
switch(s_voip.deccodec[sender])
{
#ifdef HAVE_SPEEX
case VOIP_SPEEX_OLD:
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
qspeex_decoder_destroy(s_voip.decoder[sender]);
break;
#endif
case VOIP_RAW16:
break;
#ifdef HAVE_OPUS
case VOIP_OPUS:
qopus_decoder_destroy(s_voip.decoder[sender]);
break;
#endif
}
s_voip.decoder[sender] = NULL;
s_voip.deccodec[sender] = VOIP_INVALID;
}
switch(codec)
{
default: //codec not supported.
return;
case VOIP_RAW16:
s_voip.decsamplerate[sender] = 11025;
break;
case VOIP_PCMA:
case VOIP_PCMU:
s_voip.decsamplerate[sender] = 8000;
s_voip.decframesize[sender] = 8000/20;
break;
#ifdef HAVE_SPEEX
case VOIP_SPEEX_OLD:
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
{
const SpeexMode *smode;
if (!S_Speex_Init())
return; //speex not usable.
if (codec == VOIP_SPEEX_NARROW)
{
s_voip.decsamplerate[sender] = 8000;
s_voip.decframesize[sender] = 160;
smode = s_voip.speex.modenb;
}
else if (codec == VOIP_SPEEX_WIDE)
{
s_voip.decsamplerate[sender] = 16000;
s_voip.decframesize[sender] = 320;
smode = s_voip.speex.modewb;
}
else if (codec == VOIP_SPEEX_ULTRAWIDE)
{
s_voip.decsamplerate[sender] = 32000;
s_voip.decframesize[sender] = 640;
smode = s_voip.speex.modeuwb;
}
else
{
s_voip.decsamplerate[sender] = 11025;
s_voip.decframesize[sender] = 160;
smode = s_voip.speex.modenb;
}
if (!s_voip.decoder[sender])
{
qspeex_bits_init(&s_voip.speex.decbits[sender]);
qspeex_bits_reset(&s_voip.speex.decbits[sender]);
s_voip.decoder[sender] = qspeex_decoder_init(smode);
if (!s_voip.decoder[sender])
return;
}
else
qspeex_bits_reset(&s_voip.speex.decbits[sender]);
}
break;
#endif
#ifdef HAVE_OPUS
case VOIP_OPUS:
if (!S_Opus_Init())
return;
//the lazy way to reset the codec!
if (!s_voip.decoder[sender])
{
//opus outputs to 8, 12, 16, 24, or 48khz. pick whichever has least excess samples and resample to fit it.
if (snd_speed <= 8000)
s_voip.decsamplerate[sender] = 8000;
else if (snd_speed <= 12000)
s_voip.decsamplerate[sender] = 12000;
else if (snd_speed <= 16000)
s_voip.decsamplerate[sender] = 16000;
else if (snd_speed <= 24000)
s_voip.decsamplerate[sender] = 24000;
else
s_voip.decsamplerate[sender] = 48000;
s_voip.decoder[sender] = qopus_decoder_create(s_voip.decsamplerate[sender], 1/*FIXME: support stereo where possible*/, NULL);
if (!s_voip.decoder[sender])
return;
s_voip.decframesize[sender] = s_voip.decsamplerate[sender]/400; //this is the maximum size in a single frame.
}
else
qopus_decoder_ctl(s_voip.decoder[sender], OPUS_RESET_STATE);
break;
#endif
}
s_voip.deccodec[sender] = codec;
s_voip.decgen[sender] = gen;
s_voip.decseq[sender] = seq;
s_voip.declevel[sender] = 0;
}
//if there's packetloss, tell the decoder about the missing parts.
//no infinite loops please.
if ((unsigned)(seq - s_voip.decseq[sender]) > 10)
s_voip.decseq[sender] = seq - 10;
while(s_voip.decseq[sender] != seq)
{
if (decodesamps + s_voip.decframesize[sender] > sizeof(decodebuf)/sizeof(decodebuf[0]))
{
S_RawAudio(sender, (qbyte*)decodebuf, s_voip.decsamplerate[sender], decodesamps, 1, 2, snd_voip_play.value);
decodesamps = 0;
}
switch(codec)
{
case VOIP_RAW16:
case VOIP_PCMA:
case VOIP_PCMU:
break;
#ifdef HAVE_SPEEX
case VOIP_SPEEX_OLD:
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
qspeex_decode_int(s_voip.decoder[sender], NULL, decodebuf + decodesamps);
decodesamps += s_voip.decframesize[sender];
break;
#endif
#ifdef HAVE_OPUS
case VOIP_OPUS:
r = qopus_decode(s_voip.decoder[sender], NULL, 0, decodebuf + decodesamps, min(s_voip.decframesize[sender], sizeof(decodebuf)/sizeof(decodebuf[0]) - decodesamps), false);
if (r > 0)
decodesamps += r;
break;
#endif
}
s_voip.decseq[sender]++;
}
while (bytes > 0)
{
if (decodesamps + s_voip.decframesize[sender] >= sizeof(decodebuf)/sizeof(decodebuf[0]))
{
S_RawAudio(sender, (qbyte*)decodebuf, s_voip.decsamplerate[sender], decodesamps, 1, 2, snd_voip_play.value);
decodesamps = 0;
}
switch(codec)
{
default:
bytes = 0;
break;
#ifdef HAVE_SPEEX
case VOIP_SPEEX_OLD:
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
if (codec == VOIP_SPEEX_OLD)
{ //older versions support only this, and require this extra bit.
bytes--;
len = *start++;
if (bytes < len)
break;
}
else
len = bytes;
qspeex_bits_read_from(&s_voip.speex.decbits[sender], start, len);
bytes -= len;
start += len;
while (qspeex_decode_int(s_voip.decoder[sender], &s_voip.speex.decbits[sender], decodebuf + decodesamps) == 0)
{
decodesamps += s_voip.decframesize[sender];
s_voip.decseq[sender]++;
seq++;
if (decodesamps + s_voip.decframesize[sender] >= sizeof(decodebuf)/sizeof(decodebuf[0]))
{
S_RawAudio(sender, (qbyte*)decodebuf, s_voip.decsamplerate[sender], decodesamps, 1, 2, snd_voip_play.value);
decodesamps = 0;
}
}
break;
#endif
case VOIP_RAW16:
len = min(bytes, sizeof(decodebuf)-(sizeof(decodebuf[0])*decodesamps));
memcpy(decodebuf+decodesamps, start, len);
decodesamps += len / sizeof(decodebuf[0]);
s_voip.decseq[sender]++;
bytes -= len;
start += len;
break;
case VOIP_PCMA:
case VOIP_PCMU:
len = min(bytes, sizeof(decodebuf)-(sizeof(decodebuf[0])*decodesamps));
if (len > s_voip.decframesize[sender]*2)
len = s_voip.decframesize[sender]*2;
if (codec == VOIP_PCMA)
decodesamps += PCMA_Decode(decodebuf+decodesamps, start, len);
else
decodesamps += PCMU_Decode(decodebuf+decodesamps, start, len);
s_voip.decseq[sender]++;
bytes -= len;
start += len;
break;
#ifdef HAVE_OPUS
case VOIP_OPUS:
len = bytes;
if (decodesamps > 0)
{
S_RawAudio(sender, (qbyte*)decodebuf, s_voip.decsamplerate[sender], decodesamps, 1, 2, snd_voip_play.value);
decodesamps = 0;
}
r = qopus_decode(s_voip.decoder[sender], start, len, decodebuf + decodesamps, sizeof(decodebuf)/sizeof(decodebuf[0]) - decodesamps, false);
// Con_Printf("Decoded %i frames from %i bytes\n", r, len);
if (r > 0)
{
int frames = r / s_voip.decframesize[sender];
decodesamps += r;
s_voip.decseq[sender] = (s_voip.decseq[sender] + frames) & 0xff;
seq = (seq+frames)&0xff;
}
else if (r < 0)
Con_Printf("Opus decoding error %i\n", r);
bytes -= len;
start += len;
break;
#endif
}
}
if (drops)
Con_DPrintf("%i dropped audio frames\n", drops);
if (decodesamps > 0)
{ //calculate levels of other people. eukara demanded this.
float level = 0;
float f;
for (len = 0; len < decodesamps; len++)
{
f = decodebuf[len];
level += f*f;
}
level = (3000*level) / (32767.0f*32767*decodesamps);
s_voip.declevel[sender] = (s_voip.declevel[sender]*7 + level)/8;
S_RawAudio(sender, (qbyte*)decodebuf, s_voip.decsamplerate[sender], decodesamps, 1, 2, snd_voip_play.value);
}
}
#ifdef SUPPORT_ICE
static int S_Voip_NameToId(const char *codec)
{
if (!Q_strcasecmp(codec, "speex@8000"))
return VOIP_SPEEX_NARROW;
else if (!Q_strcasecmp(codec, "speex@11025"))
return VOIP_SPEEX_OLD;
else if (!Q_strcasecmp(codec, "speex@16000"))
return VOIP_SPEEX_WIDE;
else if (!Q_strcasecmp(codec, "speex@32000"))
return VOIP_SPEEX_ULTRAWIDE;
else if (!Q_strcasecmp(codec, "opus") || !strcmp(codec, "opus@48000"))
return VOIP_OPUS;
else if (!Q_strcasecmp(codec, "pcma@8000"))
return VOIP_PCMA;
else if (!Q_strcasecmp(codec, "pcmu@8000"))
return VOIP_PCMU;
else
return VOIP_INVALID;
}
qboolean S_Voip_RTP_CodecOkay(const char *codec)
{
switch(S_Voip_NameToId(codec))
{
#ifdef HAVE_SPEEX
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_OLD:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
return S_Speex_Init();
#endif
case VOIP_PCMA:
case VOIP_PCMU:
return true;
#ifdef HAVE_OPUS
case VOIP_OPUS:
return S_Opus_Init();
#endif
default:
return false;
}
}
void S_Voip_RTP_Parse(unsigned short sequence, char *codec, unsigned char *data, unsigned int datalen)
{
S_Voip_Decode(MAX_CLIENTS-1, S_Voip_NameToId(codec), 0, sequence&0xff, datalen, data);
}
qboolean NET_RTP_Transmit(unsigned int sequence, unsigned int timestamp, const char *codec, char *cdata, int clength);
qboolean NET_RTP_Active(void);
#else
#define NET_RTP_Active() false
#endif
void S_Voip_Parse(void)
{
unsigned int sender;
unsigned int bytes;
unsigned char data[1024];
unsigned char seq, gen;
unsigned char codec;
sender = MSG_ReadByte();
gen = MSG_ReadByte();
codec = gen>>4;
gen &= 0x0f;
seq = MSG_ReadByte();
bytes = MSG_ReadShort();
if (bytes > sizeof(data) || snd_voip_play.value <= 0)
{
MSG_ReadSkip(bytes);
return;
}
MSG_ReadData(data, bytes);
sender %= MAX_CLIENTS;
//if testing, don't get confused if the server is echoing voice too!
if (snd_voip_test.ival)
if (sender == cl.playerview[0].playernum)
return;
S_Voip_Decode(sender, codec, gen, seq, bytes, data);
}
static float S_Voip_Preprocess(short *start, unsigned int samples, float micamp)
{
int i;
float level = 0, f;
int framesize = s_voip.encframesize;
while(samples >= framesize)
{
#ifdef HAVE_SPEEX
if (s_voip.speexdsp.preproc)
qspeex_preprocess_run(s_voip.speexdsp.preproc, start);
#endif
for (i = 0; i < framesize; i++)
{
f = start[i] * micamp;
start[i] = bound(-32768, f, 32767); //clamp it carefully, so it doesn't go to crap when given far too high a mic amp
level += f*f;
}
start += framesize;
samples -= framesize;
}
return level;
}
static void S_Voip_TryInitCaptureContext(char *driver, char *device, int rate)
{
static float throttle;
int i;
s_voip.cdriver = NULL;
/*Add new drivers in order of priority*/
for (i = 0; capturedrivers[i]; i++)
{
if (capturedrivers[i]->Init && (!driver || !strcmp(capturedrivers[i]->drivername, driver)))
{
s_voip.cdriver = capturedrivers[i];
s_voip.cdriverctx = s_voip.cdriver->Init(s_voip.encsamplerate, device);
if (s_voip.cdriverctx)
{
//success!
return;
}
}
}
if (!s_voip.cdriver)
{
if (!driver)
Con_ThrottlePrintf(&throttle, 0, CON_ERROR"No microphone drivers supported\n");
else
Con_ThrottlePrintf(&throttle, 0, CON_ERROR"Microphone driver \"%s\" is not valid\n", driver);
}
else
Con_ThrottlePrintf(&throttle, 0, CON_ERROR"No microphone detected\n");
s_voip.cdriver = NULL;
}
static void S_Voip_InitCaptureContext(int rate)
{
char *s;
s_voip.cdriver = NULL;
s_voip.cdriverctx = NULL;
for (s = snd_voip_capturedevice.string; ; )
{
char *sep;
s = COM_Parse(s);
if (!*com_token)
break;
sep = strchr(com_token, ':');
if (sep)
*sep++ = 0;
S_Voip_TryInitCaptureContext(com_token, sep, rate);
}
if (!s_voip.cdriver)
S_Voip_TryInitCaptureContext(NULL, NULL, rate);
}
void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf)
{
unsigned char outbuf[8192];
unsigned int outpos;//in bytes
unsigned int encpos;//in bytes
short *start;
unsigned int initseq;//in frames
#ifdef SUPPORT_ICE
unsigned int inittimestamp;//in samples
#endif
unsigned int samps;
float level;
int len;
float micamp = snd_voip_micamp.value;
qboolean voipsendenable = true;
int voipcodec = *snd_voip_codec.string?snd_voip_codec.ival:VOIP_DEFAULT_CODEC;
qboolean rtpstream = NET_RTP_Active();
if (buf)
{
/*if you're sending sound, you should be prepared to accept others yelling at you to shut up*/
if (snd_voip_play.value <= 0)
voipsendenable = false;
/*don't send sound if its not supported. that'll break stuff*/
if (!(cls.fteprotocolextensions2 & PEXT2_VOICECHAT))
voipsendenable = false;
}
else
{
/*we're not sending it to a server. the above considerations don't matter*/
voipsendenable = snd_voip_test.ival;
}
/*don't send sound if mic volume won't send anything anyway*/
if (micamp <= 0)
voipsendenable = false;
if (rtpstream)
{
voipsendenable = true;
//if rtp streaming is enabled, hack the codec to something better supported
#ifdef HAVE_SPEEX
if (voipcodec == VOIP_SPEEX_OLD)
voipcodec = VOIP_SPEEX_WIDE;
#endif
}
voicevolumemod = s_voip.lastspoke_any > realtime?snd_voip_ducking.value:1;
voicevolumemod *= mastervolume.value;
if (!voipsendenable || (voipcodec != s_voip.enccodec && s_voip.cdriver))
{
if (s_voip.cdriver)
{
if (s_voip.cdriverctx)
{
if (s_voip.wantsend)
{
s_voip.cdriver->Stop(s_voip.cdriverctx);
s_voip.wantsend = false;
}
s_voip.cdriver->Shutdown(s_voip.cdriverctx);
s_voip.cdriverctx = NULL;
}
s_voip.cdriver = NULL;
}
switch(s_voip.enccodec)
{
#ifdef HAVE_SPEEX
case VOIP_SPEEX_OLD:
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
break;
#endif
#ifdef HAVE_OPUS
case VOIP_OPUS:
qopus_encoder_destroy(s_voip.encoder);
break;
#endif
}
s_voip.encoder = NULL;
s_voip.enccodec = VOIP_INVALID;
if (!voipsendenable)
return;
}
voipsendenable = voipbutton || (snd_voip_send.ival>0);
if (!s_voip.cdriver)
{
s_voip.voiplevel = -1;
/*only init the first time capturing is requested*/
if (!voipsendenable)
return;
/*see if we can init our encoding codec...*/
switch(voipcodec)
{
#ifdef HAVE_SPEEX
case VOIP_SPEEX_OLD:
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
{
const SpeexMode *smode;
if (!S_Speex_Init())
{
Con_Printf("Unable to use speex codec - not installed\n");
return;
}
if (voipcodec == VOIP_SPEEX_ULTRAWIDE)
smode = s_voip.speex.modeuwb;
else if (voipcodec == VOIP_SPEEX_WIDE)
smode = s_voip.speex.modewb;
else
smode = s_voip.speex.modenb;
qspeex_bits_init(&s_voip.speex.encbits);
qspeex_bits_reset(&s_voip.speex.encbits);
s_voip.encoder = qspeex_encoder_init(smode);
if (!s_voip.encoder)
return;
qspeex_encoder_ctl(s_voip.encoder, SPEEX_GET_FRAME_SIZE, &s_voip.encframesize);
qspeex_encoder_ctl(s_voip.encoder, SPEEX_GET_SAMPLING_RATE, &s_voip.encsamplerate);
if (voipcodec == VOIP_SPEEX_NARROW)
s_voip.encsamplerate = 8000;
else if (voipcodec == VOIP_SPEEX_WIDE)
s_voip.encsamplerate = 16000;
else if (voipcodec == VOIP_SPEEX_ULTRAWIDE)
s_voip.encsamplerate = 32000;
else
s_voip.encsamplerate = 11025;
qspeex_encoder_ctl(s_voip.encoder, SPEEX_SET_SAMPLING_RATE, &s_voip.encsamplerate);
}
break;
#endif
case VOIP_PCMA:
case VOIP_PCMU:
s_voip.encsamplerate = 8000;
s_voip.encframesize = 8000/20;
break;
case VOIP_RAW16:
s_voip.encsamplerate = 11025;
s_voip.encframesize = 256;
break;
#ifdef HAVE_OPUS
case VOIP_OPUS:
if (!S_Opus_Init())
{
Con_Printf("Unable to use opus codec - not installed\n");
return;
}
//use whatever is convienient.
s_voip.encsamplerate = 48000;
s_voip.encframesize = s_voip.encsamplerate / 400; //2.5ms frames, at a minimum.
s_voip.encoder = qopus_encoder_create(s_voip.encsamplerate, 1, OPUS_APPLICATION_VOIP, NULL);
if (!s_voip.encoder)
return;
s_voip.curbitrate = 0;
// opus_encoder_ctl(s_voip.encoder, OPUS_SET_BITRATE(bitrate_bps));
// opus_encoder_ctl(s_voip.encoder, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND));
// opus_encoder_ctl(s_voip.encoder, OPUS_SET_VBR(use_vbr));
// opus_encoder_ctl(s_voip.encoder, OPUS_SET_VBR_CONSTRAINT(cvbr));
// opus_encoder_ctl(s_voip.encoder, OPUS_SET_COMPLEXITY(complexity));
// opus_encoder_ctl(s_voip.encoder, OPUS_SET_INBAND_FEC(use_inbandfec));
// opus_encoder_ctl(s_voip.encoder, OPUS_SET_FORCE_CHANNELS(forcechannels));
// opus_encoder_ctl(s_voip.encoder, OPUS_SET_DTX(use_dtx));
// opus_encoder_ctl(s_voip.encoder, OPUS_SET_PACKET_LOSS_PERC(packet_loss_perc));
// opus_encoder_ctl(s_voip.encoder, OPUS_GET_LOOKAHEAD(&skip));
// opus_encoder_ctl(s_voip.encoder, OPUS_SET_LSB_DEPTH(16));
break;
#endif
default:
Con_Printf("Unable to use that codec - not implemented\n");
//can't start up other coedcs, cos we're too lame.
return;
}
s_voip.enccodec = voipcodec;
S_Voip_InitCaptureContext(s_voip.encsamplerate); //sets cdriver+cdriverctx
}
/*couldn't init a driver?*/
if (!s_voip.cdriverctx || !s_voip.cdriver)
{
return;
}
if (!voipsendenable && s_voip.wantsend)
{
s_voip.wantsend = false;
s_voip.capturepos += s_voip.cdriver->Update(s_voip.cdriverctx, (unsigned char*)s_voip.capturebuf + s_voip.capturepos, 1, sizeof(s_voip.capturebuf) - s_voip.capturepos);
s_voip.cdriver->Stop(s_voip.cdriverctx);
/*note: we still grab audio to flush everything that was captured while it was active*/
}
else if (voipsendenable && !s_voip.wantsend)
{
s_voip.wantsend = true;
if (!s_voip.capturepos)
{ /*if we were actually still sending, it was probably only off for a single frame, in which case don't reset it*/
s_voip.dumps = 0;
s_voip.generation++;
s_voip.encsequence = 0;
//reset codecs so they start with a clean slate when new audio blocks are generated.
switch(s_voip.enccodec)
{
#ifdef HAVE_SPEEX
case VOIP_SPEEX_OLD:
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
qspeex_bits_reset(&s_voip.speex.encbits);
break;
#endif
case VOIP_RAW16:
break;
#ifdef HAVE_OPUS
case VOIP_OPUS:
qopus_encoder_ctl(s_voip.encoder, OPUS_RESET_STATE);
break;
#endif
}
}
else
{
s_voip.capturepos += s_voip.cdriver->Update(s_voip.cdriverctx, (unsigned char*)s_voip.capturebuf + s_voip.capturepos, 1, sizeof(s_voip.capturebuf) - s_voip.capturepos);
}
s_voip.cdriver->Start(s_voip.cdriverctx);
}
if (s_voip.wantsend)
voicevolumemod = min(voicevolumemod, snd_voip_capturingvol.value);
s_voip.capturepos += s_voip.cdriver->Update(s_voip.cdriverctx, (unsigned char*)s_voip.capturebuf + s_voip.capturepos, s_voip.encframesize*2, sizeof(s_voip.capturebuf) - s_voip.capturepos);
if (!voipsendenable || (!s_voip.wantsend && s_voip.capturepos < s_voip.encframesize*2))
{
s_voip.voiplevel = -1;
s_voip.capturepos = 0;
return;
}
initseq = s_voip.encsequence;
#ifdef SUPPORT_ICE
inittimestamp = s_voip.enctimestamp;
#endif
level = 0;
samps=0;
//*2 for 16bit audio input.
for (encpos = 0, outpos = 0; encpos+s_voip.encframesize*2 <= s_voip.capturepos && outpos+256 < sizeof(outbuf); )
{
start = (short*)(s_voip.capturebuf + encpos);
#ifdef HAVE_SPEEX
if (snd_voip_noisefilter.ival || snd_voip_autogain.ival)
{
if (!s_voip.speexdsp.preproc || snd_voip_noisefilter.modified || snd_voip_noisefilter.modified || s_voip.speexdsp.curframesize != s_voip.encframesize || s_voip.speexdsp.cursamplerate != s_voip.encsamplerate)
{
if (s_voip.speexdsp.preproc)
qspeex_preprocess_state_destroy(s_voip.speexdsp.preproc);
s_voip.speexdsp.preproc = NULL;
if (S_SpeexDSP_Init())
{
int i;
s_voip.speexdsp.preproc = qspeex_preprocess_state_init(s_voip.encframesize, s_voip.encsamplerate);
i = snd_voip_noisefilter.ival;
qspeex_preprocess_ctl(s_voip.speexdsp.preproc, SPEEX_PREPROCESS_SET_DENOISE, &i);
i = snd_voip_autogain.ival;
qspeex_preprocess_ctl(s_voip.speexdsp.preproc, SPEEX_PREPROCESS_SET_AGC, &i);
s_voip.speexdsp.curframesize = s_voip.encframesize;
s_voip.speexdsp.cursamplerate = s_voip.encsamplerate;
}
}
}
else if (s_voip.speexdsp.preproc)
{
qspeex_preprocess_state_destroy(s_voip.speexdsp.preproc);
s_voip.speexdsp.preproc = NULL;
}
#endif
switch(s_voip.enccodec)
{
#ifdef HAVE_SPEEX
case VOIP_SPEEX_OLD:
//this is from before I understood speex properly.
level += S_Voip_Preprocess(start, s_voip.encframesize, micamp);
qspeex_bits_reset(&s_voip.speex.encbits);
qspeex_encode_int(s_voip.encoder, start, &s_voip.speex.encbits);
len = qspeex_bits_write(&s_voip.speex.encbits, outbuf+(outpos+1), sizeof(outbuf) - (outpos+1));
if (len < 0 || len > 255)
len = 0;
outbuf[outpos] = len;
outpos += 1+len;
s_voip.encsequence++;
samps+=s_voip.encframesize;
encpos += s_voip.encframesize*2;
break;
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
//write multiple speex frames into a single merged frame
qspeex_bits_reset(&s_voip.speex.encbits);
for (; encpos+s_voip.encframesize*2 <= s_voip.capturepos; )
{
start = (short*)(s_voip.capturebuf + encpos);
level += S_Voip_Preprocess(start, s_voip.encframesize, micamp);
qspeex_encode_int(s_voip.encoder, start, &s_voip.speex.encbits);
s_voip.encsequence++;
samps+=s_voip.encframesize;
encpos += s_voip.encframesize*2;
if (rtpstream) //FIXME: why?
break;
}
len = qspeex_bits_write(&s_voip.speex.encbits, outbuf+outpos, sizeof(outbuf) - outpos);
outpos += len;
break;
#endif
case VOIP_RAW16:
len = s_voip.capturepos-encpos; //amount of data to be eaten in this frame
len = min(len, sizeof(outbuf)-outpos);
len &= ~((s_voip.encframesize*2)-1);
level += S_Voip_Preprocess(start, len/2, micamp);
memcpy(outbuf+outpos, start, len); //'encode'
outpos += len; //bytes written to output
encpos += len; //number of bytes consumed
s_voip.encsequence++; //increment number of packets, for packetloss detection.
samps+=len / 2; //number of samplepairs eaten in this packet. for stats.
break;
case VOIP_PCMA:
case VOIP_PCMU:
//FIXME: what's with this /2? these are just 8-bit mono (logarithmic) pcm...
len = s_voip.capturepos-encpos; //amount of data to be eaten in this frame
len = min(len, sizeof(outbuf)-outpos);
len = min(len, s_voip.encframesize*2);
level += S_Voip_Preprocess(start, len/2, micamp);
if (s_voip.enccodec == VOIP_PCMA)
outpos += PCMA_Encode(outbuf+outpos, sizeof(outbuf)-outpos, start, len/2);
else
outpos += PCMU_Encode(outbuf+outpos, sizeof(outbuf)-outpos, start, len/2);
encpos += len; //number of bytes consumed
s_voip.encsequence++; //increment number of packets, for packetloss detection.
samps+=len / 2; //number of samplepairs eaten in this packet. for stats.
break;
#ifdef HAVE_OPUS
case VOIP_OPUS:
{
//opus rtp only supports/allows a single chunk in each packet.
int frames;
int nrate;
//densely pack the frames.
start = (short*)(s_voip.capturebuf + encpos);
frames = (s_voip.capturepos-encpos)/2;
nrate = snd_voip_bitrate.value;
if (nrate != s_voip.curbitrate)
{
s_voip.curbitrate = nrate;
if (nrate == 0)
nrate = -1000;
qopus_encoder_ctl(s_voip.encoder, OPUS_SET_BITRATE_REQUEST, (int)nrate);
nrate = 10000;
}
if (frames >= 2880)
frames = 2880;
else if (frames >= 1920 && nrate > 100)
frames = 1920;
else if (frames >= 960 && nrate > 500)
frames = 960;
else if (frames >= 480 && nrate > 1000)
frames = 480;
else if (snd_voip_send.ival & 4)
break; //don't send small rtp packets, its abusive.
else if (frames >= 240 && nrate > 2000)
frames = 240;
else if (frames >= 120 && nrate > 4000)
frames = 120;
else
break; //invalid size, wait for more.
level += S_Voip_Preprocess(start, frames, micamp);
len = qopus_encode(s_voip.encoder, start, frames, outbuf+outpos, sizeof(outbuf) - outpos);
if (len >= 0)
{
s_voip.encsequence += frames / s_voip.encframesize;
outpos += len;
samps+=frames;
encpos += frames*2;
}
else
{
Con_Printf("Opus encoding error: %i\n", len);
encpos = s_voip.capturepos;
}
}
break;
#endif
default:
outbuf[outpos] = 0;
break;
}
//opus has no way to detect the end properly.
//standard rtp favours many small packets.
if (rtpstream || s_voip.enccodec == VOIP_OPUS)
break;
}
if (samps)
{
float nl;
s_voip.enctimestamp += samps;
nl = (3000*level) / (32767.0f*32767*samps);
s_voip.voiplevel = (s_voip.voiplevel*7 + nl)/8;
if (s_voip.voiplevel < snd_voip_vad_threshhold.ival && !voipbutton && !(snd_voip_send.ival & 6))
{
/*try and dump it, it was too quiet, and they're not pressing +voip*/
if (s_voip.keeps > samps)
{
/*but not instantly*/
s_voip.keeps -= samps;
}
else
{
outpos = 0;
s_voip.dumps += samps;
s_voip.keeps = 0;
}
}
else
s_voip.keeps = s_voip.encsamplerate * snd_voip_vad_delay.value;
if (outpos)
{
if (s_voip.dumps > s_voip.encsamplerate/4)
s_voip.generation++;
s_voip.dumps = 0;
}
}
if (outpos)
{
if (buf && !(snd_voip_send.ival & 4))
{
if (buf->maxsize - buf->cursize >= 5+outpos)
{
qbyte cgen = ((s_voip.enccodec&0x7)<<4) | (s_voip.generation & 0x0f);
if (s_voip.enccodec >= 8 || 0)
cgen |= 0x80;
MSG_WriteByte(buf, clc);
MSG_WriteByte(buf, cgen);
MSG_WriteByte(buf, initseq&0xff);
/*if (cgen & 0x80)
{
MSG_WriteShort(buf, 1+outpos);
MSG_WriteByte(buf, s_voip.enccodec>>3);
}
else*/
MSG_WriteShort(buf, outpos); //even with codecs where the size is easy to determine, this is still useful for servers (which are unaware of the actual codec)
SZ_Write(buf, outbuf, outpos);
}
else
Con_Printf("Audio frame too small %i vs %i\n", outpos+4, buf->maxsize - buf->cursize);
}
#ifdef SUPPORT_ICE
if (rtpstream)
{
switch(s_voip.enccodec)
{
#ifdef HAVE_SPEEX
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
case VOIP_SPEEX_OLD:
NET_RTP_Transmit(initseq, inittimestamp, va("speex@%i", s_voip.encsamplerate), outbuf, outpos);
break;
#endif
case VOIP_PCMA:
NET_RTP_Transmit(initseq, inittimestamp, "pcma@8000", outbuf, outpos);
break;
case VOIP_PCMU:
NET_RTP_Transmit(initseq, inittimestamp, "pcmu@8000", outbuf, outpos);
break;
#ifdef HAVE_OPUS
case VOIP_OPUS:
NET_RTP_Transmit(initseq, inittimestamp, "opus@48000", outbuf, outpos);
break;
#endif
}
}
#endif
if (snd_voip_test.ival)
S_Voip_Decode(cl.playerview[0].playernum, s_voip.enccodec, s_voip.generation & 0x0f, initseq&0xff, outpos, outbuf);
//update our own lastspoke, so queries shows that we're speaking when we're speaking in a generic way, even if we can't hear ourselves.
//but don't update general lastspoke, so ducking applies only when others speak. use capturingvol for yourself. they're more explicit that way.
s_voip.lastspoke[cl.playerview[0].playernum] = realtime + 0.5;
}
/*remove sent data*/
if (encpos)
{
memmove(s_voip.capturebuf, s_voip.capturebuf + encpos, s_voip.capturepos-encpos);
s_voip.capturepos -= encpos;
}
}
void S_Voip_Ignore(unsigned int slot, qboolean ignore)
{
CL_SendClientCommand(true, "vignore %i %i", slot, ignore);
}
static void S_Voip_Enable_f(void)
{
if (Cmd_IsInsecure())
return;
voipbutton = true;
}
static void S_Voip_Disable_f(void)
{
voipbutton = false;
}
static void S_Voip_f(void)
{
#ifdef HAVE_SPEEX
if (!strcmp(Cmd_Argv(1), "maxgain"))
{
int i = atoi(Cmd_Argv(2));
if (s_voip.speexdsp.preproc)
qspeex_preprocess_ctl(s_voip.speexdsp.preproc, SPEEX_PREPROCESS_SET_AGC_MAX_GAIN, &i);
}
else
#endif
{
Con_Printf("unrecognised parameter \"%s\"\n", Cmd_Argv(1));
}
}
static void QDECL S_Voip_Play_Callback(cvar_t *var, char *oldval)
{
if (cls.fteprotocolextensions2 & PEXT2_VOICECHAT)
{
if (var->value > 0)
CL_SendClientCommand(true, "unmuteall");
else
CL_SendClientCommand(true, "muteall");
}
}
void S_Voip_MapChange(void)
{
voipbutton = false;
Cvar_ForceCallback(&snd_voip_play);
}
int S_Voip_Loudness(qboolean ignorevad)
{
if (!s_voip.cdriverctx || (!ignorevad && s_voip.dumps))
return -1;
if (s_voip.voiplevel > 100)
return 100;
return s_voip.voiplevel;
}
int S_Voip_ClientLoudness(unsigned int plno)
{
if (plno >= MAX_CLIENTS)
return 0;
if (s_voip.lastspoke[plno] > realtime)
return s_voip.declevel[plno];
return -1;
}
qboolean S_Voip_Speaking(unsigned int plno)
{
if (plno >= MAX_CLIENTS)
return false;
return s_voip.lastspoke[plno] > realtime;
}
void S_Voip_Init(void)
{
int i;
for (i = 0; i < MAX_CLIENTS; i++)
s_voip.deccodec[i] = VOIP_INVALID;
s_voip.enccodec = VOIP_INVALID;
Cvar_Register(&snd_voip_capturedevice, "Voice Chat");
Cvar_Register(&snd_voip_capturedevice_opts, "Voice Chat");
Cvar_Register(&snd_voip_send, "Voice Chat");
Cvar_Register(&snd_voip_vad_threshhold, "Voice Chat");
Cvar_Register(&snd_voip_vad_delay, "Voice Chat");
Cvar_Register(&snd_voip_capturingvol, "Voice Chat");
Cvar_Register(&snd_voip_showmeter, "Voice Chat");
Cvar_Register(&snd_voip_play, "Voice Chat");
Cvar_Register(&snd_voip_test, "Voice Chat");
Cvar_Register(&snd_voip_ducking, "Voice Chat");
Cvar_Register(&snd_voip_micamp, "Voice Chat");
Cvar_Register(&snd_voip_codec, "Voice Chat");
#ifdef HAVE_SPEEX
Cvar_Register(&snd_voip_noisefilter, "Voice Chat");
Cvar_Register(&snd_voip_autogain, "Voice Chat");
#endif
Cvar_Register(&snd_voip_bitrate, "Voice Chat");
Cmd_AddCommand("+voip", S_Voip_Enable_f);
Cmd_AddCommand("-voip", S_Voip_Disable_f);
Cmd_AddCommand("voip", S_Voip_f);
}
#else
void S_Voip_Parse(void)
{
unsigned int bytes;
MSG_ReadByte();
MSG_ReadByte();
MSG_ReadByte();
bytes = MSG_ReadShort();
MSG_ReadSkip(bytes);
}
int S_Voip_ClientLoudness(unsigned int plno)
{
return -1;
}
#endif
void S_DefaultSpeakerConfiguration(soundcardinfo_t *sc)
{
sc->dist[0] = 1;
sc->dist[1] = 1;
sc->dist[2] = 1;
sc->dist[3] = 1;
sc->dist[4] = 1;
sc->dist[5] = 1;
switch (sc->sn.numchannels)
{
case 1:
VectorSet(sc->speakerdir[0], 0, 0, 0);
break;
case 2:
case 3:
VectorSet(sc->speakerdir[0], 0, -1, 0);
VectorSet(sc->speakerdir[1], 0, 1, 0);
VectorSet(sc->speakerdir[2], 0, 0, 0);
break;
case 4: // quad
case 5:
VectorSet(sc->speakerdir[0], 0.7, -0.7, 0);
VectorSet(sc->speakerdir[1], 0.7, 0.7, 0);
VectorSet(sc->speakerdir[2], -0.7, -0.7, 0);
VectorSet(sc->speakerdir[3], -0.7, 0.7, 0);
VectorSet(sc->speakerdir[4], 0, 0, 0);
break;
case 6: // 5.1
case 7:
VectorSet(sc->speakerdir[0], 0.7, -0.7, 0); //front-left
VectorSet(sc->speakerdir[1], 0.7, 0.7, 0); //front-right
VectorSet(sc->speakerdir[2], 1, 0, 0); //center
VectorSet(sc->speakerdir[3], 0, 0, 0); //bass
VectorSet(sc->speakerdir[4], -0.7, -0.7, 0);//back-left
VectorSet(sc->speakerdir[5], -0.7, 0.7, 0); //back-right
VectorSet(sc->speakerdir[6], 0, 0, 0);
break;
case 8: // 7.1
default:
VectorSet(sc->speakerdir[0], 0.7, -0.7, 0);
VectorSet(sc->speakerdir[1], 0.7, 0.7, 0);
VectorSet(sc->speakerdir[2], 1, 0, 0);
VectorSet(sc->speakerdir[3], 0, 0, 0);
VectorSet(sc->speakerdir[4], -0.7, -0.7, 0);
VectorSet(sc->speakerdir[5], -0.7, 0.7, 0);
VectorSet(sc->speakerdir[6], 0, -1, 0);
VectorSet(sc->speakerdir[7], 0, 1, 0);
break;
}
}
#ifdef AVAIL_WASAPI
extern sounddriver_t WASAPI_Output;
#endif
#ifdef AVAIL_XAUDIO2
extern sounddriver_t XAUDIO2_Output;
#endif
#ifdef AVAIL_DSOUND
extern sounddriver_t DSOUND_Output;
#endif
sounddriver_t fte_weakstruct SDL_Output;
#ifdef __linux__
extern sounddriver_t ALSA_Output;
extern sounddriver_t Pulse_Output;
#endif
sounddriver_t fte_weakstruct OSS_Output;
#ifdef AVAIL_OPENAL
extern sounddriver_t OPENAL_Output;
extern sounddriver_t OPENAL_Output_Lame;
#endif
#ifdef __DJGPP__
extern sounddriver_t SBLASTER_Output;
#endif
#if defined(_WIN32) && !defined(WINRT) && !defined(FTE_SDL)
extern sounddriver_t WaveOut_Output;
#endif
#ifdef MACOSX
sounddriver_t fte_weakstruct MacOS_AudioOutput; //prefered on mac
#endif
#ifdef ANDROID
sounddriver_t fte_weakstruct OSL_Output; //general audio library, but android has all kinds of quirks.
sounddriver_t fte_weakstruct Droid_AudioOutput;
#endif
#if defined(__MORPHOS__)
sounddriver_t fte_weakstruct AHI_AudioOutput; //prefered on morphos
#endif
sounddriver_t fte_weakstruct SNDIO_AudioOutput; //bsd
//in order of preference
static sounddriver_t *outputdrivers[] =
{
#ifdef AVAIL_OPENAL
&OPENAL_Output, //refuses to run as the default device, at least until its perfected.
#endif
#ifdef HAVE_MIXER
#ifdef AVAIL_DSOUND
&DSOUND_Output,
#endif
#ifdef AVAIL_XAUDIO2
&XAUDIO2_Output,
#endif
#ifdef AVAIL_WASAPI
&WASAPI_Output, //this is last, so that we can default to exclusive. woot.
#endif
&SDL_Output, //prefered on linux. distros can ensure that its configured correctly.
#ifdef AUDIO_PULSE
&Pulse_Output, //wasteful, and availability generally means Alsa is broken/defective.
#endif
#ifdef AUDIO_ALSA
&ALSA_Output, //pure shite, and availability generally means OSS is broken/defective.
#endif
#ifdef AUDIO_OSS
&OSS_Output, //good for low latency audio, but not likely to work any more on linux (unlike every other unix system with a decent opengl driver)
#endif
#ifdef __DJGPP__
&SBLASTER_Output, //zomgwtfdos?
#endif
#if defined(_WIN32) && !defined(WINRT) && !defined(FTE_SDL)
&WaveOut_Output, //doesn't work properly in vista, etc.
#endif
#ifdef MACOSX
&MacOS_AudioOutput, //prefered on mac
#endif
#ifdef ANDROID
&OSL_Output, //opensl(es)
#endif
#if defined(__MORPHOS__)
&AHI_AudioOutput, //prefered on morphos
#endif
&SNDIO_AudioOutput, //prefered on OpenBSD
#ifdef AVAIL_OPENAL
&OPENAL_Output_Lame,//streaming quake's audio via openal instead of using openal properly. used in our browser port to work around issues with webaudio (at least in chromium).
#endif
#endif
NULL
};
static soundcardinfo_t *SNDDMA_Init(char *driver, char *device, int seat)
{
soundcardinfo_t *sc = Z_Malloc(sizeof(soundcardinfo_t));
sounddriver_t *sd;
int i;
int st;
memset(sc, 0, sizeof(*sc));
sc->seat = seat;
sc->next = sndcardinfo;
sndcardinfo = sc;
// set requested rate
if (snd_khz.ival >= 1000)
sc->sn.speed = snd_khz.ival;
else if (snd_khz.ival <= 0)
sc->sn.speed = 22050;
/* else if (snd_khz.ival >= 195)
sc->sn.speed = 200000;
else if (snd_khz.ival >= 180)
sc->sn.speed = 192000;
else if (snd_khz.ival >= 90)
sc->sn.speed = 96000; */
else if (snd_khz.ival >= 45)
sc->sn.speed = 48000;
else if (snd_khz.ival >= 30)
sc->sn.speed = 44100;
else if (snd_khz.ival >= 20)
sc->sn.speed = 22050;
else if (snd_khz.ival >= 10)
sc->sn.speed = 11025;
else
sc->sn.speed = 8000;
// set requested speaker count
if (snd_speakers.ival > MAXSOUNDCHANNELS)
sc->sn.numchannels = MAXSOUNDCHANNELS;
else if (snd_speakers.ival > 1)
sc->sn.numchannels = (int)snd_speakers.ival;
else
sc->sn.numchannels = 1;
// set requested sample bits
if (snd_samplebits.ival >= 32)
sc->sn.samplebytes = 4;
else if (snd_samplebits.ival >= 16)
sc->sn.samplebytes = 2;
else
sc->sn.samplebytes = 1;
// set requested buffer size
if (snd_buffersize.ival > 0)
sc->sn.samples = snd_buffersize.ival * sc->sn.numchannels;
else
sc->sn.samples = 0;
for (i = 0; outputdrivers[i]; i++)
{
sd = outputdrivers[i];
if (sd && sd->name && (!driver || !Q_strcasecmp(sd->name, driver)))
{
//skip drivers which are not present.
if (!sd->InitCard)
continue;
st = (**sd->InitCard)(sc, device);
if (st)
{
if (!sc->sn.sampleformat)
{
Con_TPrintf("S_Startup: Ignoring soundcard %s due to unspecified sample format.\n", sc->name);
S_ShutdownCard(sc);
continue;
}
S_DefaultSpeakerConfiguration(sc);
if (snd_speed)
{ //if the sample speeds of multiple soundcards do not match, it'll fail.
if (snd_speed != sc->sn.speed)
{
Con_TPrintf("S_Startup: Ignoring soundcard %s due to mismatched sample speeds.\n", sc->name);
S_ShutdownCard(sc);
return NULL;
}
}
else
snd_speed = sc->sn.speed;
if (sc->seat == -1 && sc->ListenerUpdate)
sc->seat = 0; //hardware rendering won't cope with seat=-1
Z_ReallocElements((void**)&sc->channel, &sc->max_chans, NUM_AMBIENTS+NUM_MUSICS, sizeof(*sc->channel));
return sc;
}
}
}
S_ShutdownCard(sc);
if (!driver)
Con_TPrintf("Could not start audio device \"%s\"\n", device?device:"default");
else
Con_TPrintf("Could not start \"%s\" device \"%s\"\n", driver, device?device:"default");
return NULL;
}
soundcardinfo_t *S_SetupDeviceSeat(char *driver, char *device, int seat)
{
return SNDDMA_Init(driver, device, seat);
/*
soundcardinfo_t *sc;
for (sc = sndcardinfo; sc; sc = sc->next)
{
sc->seat = seat;
}*/
}
static void QDECL S_EnumeratedOutDevice(const char *driver, const char *devicecode, const char *readabledevice)
{
const char *fullintname;
char opts[8192];
char nbuf[1024];
char dbuf[1024];
if (devicecode)
fullintname = va("%s:%s", driver, devicecode);
else
fullintname = driver;
Q_snprintfz(opts, sizeof(opts), "%s%s%s %s", snd_device_opts.string, *snd_device_opts.string?" ":"", COM_QuotedString(fullintname, nbuf, sizeof(nbuf), false), COM_QuotedString(readabledevice, dbuf, sizeof(dbuf), false));
Cvar_ForceSet(&snd_device_opts, opts);
}
#ifdef VOICECHAT
static void QDECL S_Voip_EnumeratedCaptureDevice(const char *driver, const char *devicecode, const char *readabledevice)
{
const char *fullintname;
char opts[8192];
char nbuf[1024];
char dbuf[1024];
if (devicecode)
fullintname = va("%s:%s", driver, devicecode);
else
fullintname = driver;
Q_snprintfz(opts, sizeof(opts), "%s%s%s %s", snd_voip_capturedevice_opts.string, *snd_voip_capturedevice_opts.string?" ":"", COM_QuotedString(fullintname, nbuf, sizeof(nbuf), false), COM_QuotedString(readabledevice, dbuf, sizeof(dbuf), false));
Cvar_ForceSet(&snd_voip_capturedevice_opts, opts);
}
#endif
void S_EnumerateDevices(void)
{
int i;
sounddriver_t *sd;
qboolean safe = COM_CheckParm("-noenumerate") || COM_CheckParm("-safe");
Cvar_ForceSet(&snd_device_opts, "");
S_EnumeratedOutDevice("", NULL, "Default");
S_EnumeratedOutDevice("none", NULL, "None");
for (i = 0; outputdrivers[i]; i++)
{
sd = outputdrivers[i];
if (sd && sd->name)
{
if (safe || !sd->Enumerate || !sd->Enumerate(S_EnumeratedOutDevice))
S_EnumeratedOutDevice(sd->name, "", va("Default %s", sd->name));
}
}
#ifdef VOICECHAT
Cvar_ForceSet(&snd_voip_capturedevice_opts, "");
S_Voip_EnumeratedCaptureDevice("", NULL, "Default");
for (i = 0; capturedrivers[i]; i++)
{
if (!capturedrivers[i]->Init)
continue;
if (safe || !capturedrivers[i]->Enumerate || !capturedrivers[i]->Enumerate(S_Voip_EnumeratedCaptureDevice))
S_Voip_EnumeratedCaptureDevice(capturedrivers[i]->drivername, NULL, va("Default %s", capturedrivers[i]->drivername));
}
#endif
}
/*
================
S_Startup
================
*/
void S_ClearRaw(void);
void S_Startup (void)
{
qboolean nodefault = false;
char *s;
if (!snd_initialized)
return;
if (sound_started)
S_Shutdown(false);
snd_blocked = 0;
snd_speed = 0;
S_UpdateReverb(0, NULL, 0);
{ //we can actually use underwater hints automatically easily enough. q3 also does this.
//its other things that are more awkward.
struct reverbproperties_s underwater = REVERB_PRESET_UNDERWATER;
S_UpdateReverb(1, &underwater, sizeof(underwater));
}
for (s = snd_device.string; ; )
{
char *sep;
s = COM_Parse(s);
if (!*com_token)
break;
if (!Q_strcasecmp(com_token, "none"))
nodefault = true;
else
{
sep = strchr(com_token, ':');
if (sep)
*sep++ = 0;
SNDDMA_Init(com_token, sep, -1);
}
}
if (!sndcardinfo && !nodefault)
{
#if defined(_WIN32) && !defined(FTE_SDL)
INS_SetupControllerAudioDevices(true);
#endif
if (!sndcardinfo)
SNDDMA_Init(NULL, NULL, -1);
}
sound_started = true;
S_ClearRaw();
if (!known_sfx)
known_sfx = Z_Malloc(MAX_SFX*sizeof(sfx_t));
num_sfx = 0;
CL_InitTEntSounds();
ambient_sfx[AMBIENT_WATER] = S_PrecacheSound ("ambience/water1.wav");
ambient_sfx[AMBIENT_SKY] = S_PrecacheSound ("ambience/wind2.wav");
}
//why isn't this part of S_Restart_f anymore?
//so that the video code can call it directly without flushing the models it's just loaded.
void S_DoRestart (qboolean onlyifneeded)
{
int i;
if (onlyifneeded && sound_started)
return; //don't need to if its already running.
S_StopAllSounds (true);
S_Shutdown(false);
if (nosound.ival)
return;
S_Startup();
S_StopAllSounds (true);
for (i=1 ; i<MAX_PRECACHE_SOUNDS ; i++)
{
if (!cl.sound_name[i])
break;
cl.sound_precache[i] = S_FindName (cl.sound_name[i], true, false);
}
}
void S_Restart_f (void)
{
S_EnumerateDevices();
S_DoRestart(false);
}
void S_Control_f (void)
{
int i;
char *command;
command = Cmd_Argv (1);
if (!Q_strcasecmp(command, "off"))
{
Cache_Flush();//forget the old sounds.
S_StopAllSounds (true);
S_Shutdown(false);
sound_started = 0;
}
if (!Q_strcasecmp(command, "rate") || !Q_strcasecmp(command, "speed"))
{
Cvar_SetValue(&snd_khz, atof(Cmd_Argv (2))/1000);
S_Restart_f();
return;
}
//individual device control
if (!Q_strncasecmp(command, "card", 4))
{
int card;
soundcardinfo_t *sc;
card = atoi(command+4);
for (i = 0, sc = sndcardinfo; i < card && sc; i++,sc=sc->next)
;
if (!sc)
{
Con_Printf("Sound card %i is invalid (try resetting first)\n", card);
return;
}
if (Cmd_Argc() < 3)
{
Con_Printf("Scard %i is %s\n", card, sc->name);
return;
}
command = Cmd_Argv (2);
if (!Q_strcasecmp(command, "mono"))
{
for (i = 0; i < MAXSOUNDCHANNELS; i++)
{
VectorSet(sc->speakerdir[i], 0, 0, 0);
sc->dist[i] = 1;
}
}
else if (!Q_strcasecmp(command, "standard") || !Q_strcasecmp(command, "stereo"))
{
for (i = 0; i < MAXSOUNDCHANNELS; i++)
{
VectorSet(sc->speakerdir[i], 0, (i&1)?1:-1, 0);
sc->dist[i] = 1;
}
}
else if (!Q_strcasecmp(command, "swap"))
{
for (i = 0; i < MAXSOUNDCHANNELS; i++)
{
sc->speakerdir[i][1] *= -1;
}
}
else if (!Q_strcasecmp(command, "front"))
{
for (i = 0; i < MAXSOUNDCHANNELS; i++)
{
VectorSet(sc->speakerdir[i], 0.7, (i&1)?-0.7:0.7, 0);
sc->dist[i] = 1;
}
}
else if (!Q_strcasecmp(command, "back"))
{
for (i = 0; i < MAXSOUNDCHANNELS; i++)
{
VectorSet(sc->speakerdir[i], -0.7, (i&1)?-0.7:0.7, 0);
sc->dist[i] = 1;
}
}
return;
}
else
Con_Printf("valid commands are: off, single, multi, cardX mono, cardX stereo, cardX front, cardX back\n");
}
/*
================
S_Init
================
*/
void S_Init (void)
{
int p, i;
Con_DPrintf("\nSound Initialization\n");
Cmd_AddCommand("play", S_Play_f); //sound that doesn't follow the player
Cmd_AddCommand("play2", S_Play_f); //sound that DOES follow the player
Cmd_AddCommand("playvol", S_Play_f);
Cmd_AddCommand("stopsound", S_StopAllSounds_f);
Cmd_AddCommand("soundlist", S_SoundList_f);
Cmd_AddCommand("soundinfo", S_SoundInfo_f);
Cmd_AddCommand("snd_restart", S_Restart_f);
Cmd_AddCommand("soundcontrol", S_Control_f);
Cvar_Register(&nosound, "Sound controls");
Cvar_Register(&mastervolume, "Sound controls");
Cvar_Register(&volume, "Sound controls");
Cvar_Register(&snd_precache, "Sound controls");
Cvar_Register(&snd_loadas8bit, "Sound controls");
Cvar_Register(&snd_loadasstereo, "Sound controls");
Cvar_Register(&bgmvolume, "Sound controls");
Cvar_Register(&snd_nominaldistance, "Sound controls");
Cvar_Register(&ambient_level, "Sound controls");
Cvar_Register(&ambient_fade, "Sound controls");
Cvar_Register(&snd_noextraupdate, "Sound controls");
Cvar_Register(&snd_show, "Sound controls");
Cvar_Register(&_snd_mixahead, "Sound controls");
Cvar_Register(&snd_khz, "Sound controls");
Cvar_Register(&snd_leftisright, "Sound controls");
Cvar_Register(&snd_eax, "Sound controls");
Cvar_Register(&snd_speakers, "Sound controls");
Cvar_Register(&snd_buffersize, "Sound controls");
Cvar_Register(&snd_samplebits, "Sound controls");
Cvar_Register(&snd_playbackrate, "Sound controls");
Cvar_Register(&snd_ignoregamespeed, "Sound controls");
Cvar_Register(&snd_doppler, "Sound controls");
Cvar_Register(&snd_doppler_min, "Sound controls");
Cvar_Register(&snd_doppler_max, "Sound controls");
Cvar_Register(&snd_inactive, "Sound controls");
#ifdef MULTITHREAD
Cvar_Register(&snd_mixerthread, "Sound controls");
#endif
Cvar_Register(&snd_playersoundvolume, "Sound controls");
Cvar_Register(&snd_device, "Sound controls");
Cvar_Register(&snd_device_opts, "Sound controls");
Cvar_Register(&snd_ignorecueloops, "Sound controls");
Cvar_Register(&snd_linearresample, "Sound controls");
Cvar_Register(&snd_linearresample_stream, "Sound controls");
#ifdef VOICECHAT
S_Voip_Init();
#endif
#ifdef MULTITHREAD
mixermutex = Sys_CreateMutex();
#endif
for (i = 0; outputdrivers[i]; i++)
{
sounddriver_t *sd = outputdrivers[i];
if (sd && sd->name && sd->RegisterCvars)
sd->RegisterCvars();
}
if (COM_CheckParm("-nosound"))
{
Cvar_ForceSet(&nosound, "1");
nosound.flags |= CVAR_NOSET;
return;
}
S_EnumerateDevices();
p = COM_CheckParm ("-soundspeed");
if (!p)
p = COM_CheckParm ("-sspeed");
if (!p)
p = COM_CheckParm ("-sndspeed");
if (p)
{
if (p < com_argc-1)
Cvar_SetValue(&snd_khz, atof(com_argv[p+1]));
else
Sys_Error ("S_Init: you must specify a speed in KB after -soundspeed");
}
snd_initialized = true;
known_sfx = Z_Malloc(MAX_SFX*sizeof(sfx_t));
num_sfx = 0;
}
// =======================================================================
// Shutdown sound engine
// =======================================================================
void S_ShutdownCard(soundcardinfo_t *sc)
{
soundcardinfo_t **link;
for (link = &sndcardinfo; *link; link = &(*link)->next)
{
if (*link == sc)
{
*link = sc->next;
if (sc->Shutdown)
sc->Shutdown(sc);
Z_Free(sc->channel);
Z_Free(sc);
break;
}
}
}
void S_Shutdown(qboolean final)
{
soundcardinfo_t *sc, *next;
#if defined(_WIN32) && !defined(FTE_SDL)
INS_SetupControllerAudioDevices(false);
#endif
for (sc = sndcardinfo; sc; sc=next)
{
next = sc->next;
sc->Shutdown(sc);
Z_Free(sc->channel);
Z_Free(sc);
sndcardinfo = next;
}
sound_started = 0;
S_Purge(false);
Z_Free(known_sfx);
known_sfx = NULL;
num_sfx = 0;
if (final)
{
Z_Free(reverbproperties);
reverbproperties = NULL;
numreverbproperties = 0;
}
#ifdef MULTITHREAD
if (final && mixermutex)
{
Sys_DestroyMutex(mixermutex);
mixermutex = NULL;
}
#endif
}
// =======================================================================
// Load a sound
// =======================================================================
/*
==================
S_FindName
also touches it
==================
*/
sfx_t *S_FindName (const char *name, qboolean create, qboolean syspath)
{
int i;
sfx_t *sfx;
if (!name)
Sys_Error ("S_FindName: NULL\n");
if (Q_strlen(name) >= MAX_OSPATH)
Sys_Error ("Sound name too long: %s", name);
// see if already loaded
for (i=0 ; i < num_sfx ; i++)
if (!Q_strcmp(known_sfx[i].name, name) && known_sfx[i].syspath == syspath)
{
known_sfx[i].touched = true;
return &known_sfx[i];
}
if (num_sfx == MAX_SFX)
Sys_Error ("S_FindName: out of sfx_t");
if (create && known_sfx)
{
sfx = &known_sfx[i];
strcpy (sfx->name, name);
sfx->syspath = syspath;
sfx->touched = true;
num_sfx++;
}
else
sfx = NULL;
return sfx;
}
void S_Purge(qboolean retaintouched)
{
sfx_t *sfx;
int i;
//make sure ambients are kept. silly ambients.
if (retaintouched)
{
ambient_sfx[AMBIENT_WATER] = S_PrecacheSound ("ambience/water1.wav");
ambient_sfx[AMBIENT_SKY] = S_PrecacheSound ("ambience/wind2.wav");
}
if (!num_sfx)
return;
S_LockMixer();
for (i=0 ; i < num_sfx ; i++)
{
sfx = &known_sfx[i];
/*don't hurt sounds if they're being processed by a worker thread*/
if (sfx->loadstate == SLS_LOADING)
{
if (retaintouched)
continue; //don't bother waiting
//trying to shut down or something.
//make sure there's no worker about to write to sfx after the memory is freed
COM_WorkerPartialSync(sfx, &sfx->loadstate, SLS_LOADING);
}
/*don't purge the file if its still relevent*/
if (retaintouched && sfx->touched)
continue;
if (S_IsPlayingSomewhere(sfx))
continue; //eep?!?
sfx->loadstate = SLS_NOTLOADED;
/*nothing to do if there's no data within*/
if (!sfx->decoder.buf)
continue;
/*stop the decoder first*/
if (sfx->decoder.purge)
sfx->decoder.purge(sfx);
else if (sfx->decoder.ended)
sfx->decoder.ended(sfx);
/*if there's any data associated still, kill it. if present, it should be a single sfxcache_t (with data in same alloc)*/
if (sfx->decoder.buf)
BZ_Free(sfx->decoder.buf);
memset(&sfx->decoder, 0, sizeof(sfx->decoder));
}
S_UnlockMixer();
}
void S_ResetFailedLoad(void)
{
int i;
for (i=0 ; i < num_sfx ; i++)
{
if (known_sfx[i].loadstate == SLS_FAILED)
known_sfx[i].loadstate = SLS_NOTLOADED;
}
}
void S_UntouchAll(void)
{
int i;
for (i=0 ; i < num_sfx ; i++)
known_sfx[i].touched = false;
}
/*
==================
S_PrecacheSound
==================
*/
sfx_t *S_PrecacheSound2 (const char *name, qboolean syspath)
{
sfx_t *sfx;
if (nosound.ival || !known_sfx || !*name)
return NULL;
sfx = S_FindName (name, true, syspath);
// cache it in
if (snd_precache.ival && snd_precache.ival != 2 && sndcardinfo)
S_LoadSound (sfx, true);
return sfx;
}
//=============================================================================
/*
=================
SND_PickChannel
=================
*/
channel_t *SND_PickChannel(soundcardinfo_t *sc, int entnum, int entchannel)
{
int ch_idx;
int oldest;
// Check for replacement sound, or an idle channel
oldest = -1;
for (ch_idx=DYNAMIC_FIRST; ch_idx < sc->max_chans ; ch_idx++)
{
if (entchannel != 0 // channel 0 never overrides
&& sc->channel[ch_idx].entnum == entnum
&& sc->channel[ch_idx].entchannel == entchannel)
{ // always override sound from same entity
oldest = ch_idx;
break;
}
if (!sc->channel[ch_idx].sfx)
oldest = ch_idx;
}
if (oldest == -1)
{
oldest = sc->max_chans;
Z_ReallocElements((void**)&sc->channel, &sc->max_chans, oldest+1, sizeof(*sc->channel));
}
sc->channel[oldest].sfx = NULL;
if (sc->total_chans <= oldest)
sc->total_chans = oldest+1;
#ifdef Q3CLIENT //presumably we should be using this instead of pos for oldest, but blurgh.
sc->channel[oldest].starttime = Sys_Milliseconds();
#endif
return &sc->channel[oldest];
}
static void SND_AccumulateSpacialization(soundcardinfo_t *sc, channel_t *ch, vec3_t origin)
{
vec3_t listener_vec;
vec_t dist;
vec_t scale;
vec3_t world_vec;
int i, v;
float volscale;
int seat;
if (ch->flags & CF_CL_ABSVOLUME)
volscale = mastervolume.value;
else
volscale = volume.value * voicevolumemod;
if (sc->seat == -1)
{
seat = 0;
VectorSubtract(origin, listener[seat].origin, world_vec);
dist = DotProduct(world_vec,world_vec);
for (i = 1; i < cl.splitclients; i++)
{
VectorSubtract(origin, listener[i].origin, world_vec);
scale = DotProduct(world_vec,world_vec);
if (scale < dist)
{
dist = scale;
seat = i;
}
}
}
else
{
seat = sc->seat;
}
// anything coming from the view entity will always be full volume
if (ch->entnum == listener[seat].entnum)
{
v = ch->master_vol * (ruleset_allow_localvolume.value ? snd_playersoundvolume.value : 1) * volscale;
v = bound(0, v, 255);
for (i = 0; i < sc->sn.numchannels; i++)
ch->vol[i] = v;
return;
}
// calculate stereo seperation and distance attenuation
VectorSubtract(origin, listener[seat].origin, world_vec);
dist = VectorNormalize(world_vec) * ch->dist_mult;
if ((ch->flags & CF_NOSPACIALISE) || !ch->dist_mult)
{
scale = 1;
scale = (1.0 - dist) * scale;
v = ch->master_vol * scale * volscale;
for (i = 0; i < sc->sn.numchannels; i++)
ch->vol[i] += bound(0, v, 255);
return;
}
//rotate the world_vec into listener space, so that the audio direction stored in the speakerdir array can be used directly.
listener_vec[0] = DotProduct(listener[seat].forward, world_vec);
listener_vec[1] = DotProduct(listener[seat].right, world_vec);
listener_vec[2] = DotProduct(listener[seat].up, world_vec);
if (snd_leftisright.ival^r_xflip.ival)
listener_vec[1] = -listener_vec[1];
for (i = 0; i < sc->sn.numchannels; i++)
{
scale = 1 + DotProduct(listener_vec, sc->speakerdir[i]);
scale = (1.0 - dist) * scale * sc->dist[i];
v = ch->master_vol * scale * volscale;
ch->vol[i] += bound(0, v, 255);
}
}
/*
=================
SND_Spatialize
=================
*/
static void SND_Spatialize(soundcardinfo_t *sc, channel_t *ch)
{
vec3_t listener_vec, sound_vel;
vec_t dist;
vec_t scale;
vec3_t world_vec;
int i, v;
float volscale;
int seat;
if (ch->flags & CF_FOLLOW)
{
//sounds following ents should update their position to match that ent's position.
//its important that they do not snap back to where they were if the entity vanishes, so we just overwrite the channel origin for that. its simpler.
#ifdef CSQC_DAT
if (ch->entnum < 0 && -ch->entnum < csqc_world.num_edicts)
{
wedict_t *ed = WEDICT_NUM_PB(csqc_world.progs, -ch->entnum);
if (ed->ereftype == ER_ENTITY)
{
VectorCopy(ed->v->origin, ch->origin);
VectorCopy(ed->v->velocity, ch->velocity);
if (ed->v->solid == SOLID_BSP)
{
VectorMA(ch->origin, 0.5, ed->v->absmin, ch->origin);
VectorMA(ch->origin, 0.5, ed->v->absmax, ch->origin);
}
}
}
else
#endif
if (ch->entnum > 0 && ch->entnum < cl.maxlerpents && cl.lerpents[ch->entnum].sequence == cl.lerpentssequence)
{
lerpents_t *le = cl.lerpents+ch->entnum;
int midx = le->entstate->modelindex;
VectorCopy(le->origin, ch->origin);
//VectorCopy(le->velocity, ch->velocity); //fixme: bmodels should use their center rather than their origin. check le->state->solid?
//bmodels should report the center of the entity rather than the origin (which is frequently at 0 0 0 or merely used as a pivot)
if (le->entstate->solidsize == ES_SOLID_BSP && midx > 0 && midx < countof(cl.model_precache))
{
if (cl.model_precache[midx] && cl.model_precache[midx]->loadstate == MLS_LOADED && cl.model_precache[midx]->type == mod_brush)
{
//fixme: should probably deal with rotations.
VectorMA(ch->origin, 0.5, cl.model_precache[midx]->mins, ch->origin);
VectorMA(ch->origin, 0.5, cl.model_precache[midx]->maxs, ch->origin);
}
}
}
//FIXME: update rate to provide doppler
}
//sounds with absvolume ignore all volume etc cvars+settings
if (ch->flags & CF_CL_ABSVOLUME)
volscale = mastervolume.value;
else
volscale = volume.value * voicevolumemod;
if (!vid.activeapp && !snd_inactive.ival && !(ch->flags & CF_CLI_INACTIVE))
volscale = 0;
if (sc->seat == -1)
{
seat = 0;
VectorSubtract(ch->origin, listener[seat].origin, world_vec);
dist = DotProduct(world_vec,world_vec);
#if MAX_SPLITS > 1
for (i = 1; i < cl.splitclients; i++)
{
VectorSubtract(ch->origin, listener[i].origin, world_vec);
scale = DotProduct(world_vec,world_vec);
if (scale < dist)
{
dist = scale;
seat = i;
}
}
#endif
}
else
{
seat = sc->seat;
}
// anything coming from the view entity will always be full volume
// (no, I don't like this hack)
if (ch->entnum == listener[seat].entnum && ch->entnum)
{
v = ch->master_vol * (ruleset_allow_localvolume.value ? snd_playersoundvolume.value : 1) * volscale;
v = bound(0, v, 255);
for (i = 0; i < sc->sn.numchannels; i++)
ch->vol[i] = v;
return;
}
// calculate stereo seperation and distance attenuation
VectorSubtract(ch->origin, listener[seat].origin, world_vec);
dist = VectorNormalize(world_vec) * ch->dist_mult;
if ((ch->flags & CF_NOSPACIALISE) || !ch->dist_mult)
{
scale = 1;
scale = (1.0 - dist) * scale;
v = ch->master_vol * scale * volscale;
v = bound(0, v, 255);
for (i = 0; i < sc->sn.numchannels; i++)
ch->vol[i] = v;
return;
}
//an attempt at doppler.
if (snd_doppler.value)
{
//according to feh, the speed of sound is about 9000 qu/s.
VectorAdd(listener[seat].velocity, ch->velocity, sound_vel);
scale = 1 + snd_doppler.value * DotProduct(world_vec, sound_vel) / (9000.0);
if (scale > snd_doppler_max.value)
scale = snd_doppler_max.value;
if (scale < snd_doppler_min.value)
scale = snd_doppler_min.value;
ch->rate = (1<<PITCHSHIFT) * scale + 0.5;
if (ch->rate < 1) //too small values result in crashes.
ch->rate = 1;
}
//rotate the world_vec into listener space, so that the audio direction stored in the speakerdir array can be used directly.
listener_vec[0] = DotProduct(listener[seat].forward, world_vec);
listener_vec[1] = DotProduct(listener[seat].right, world_vec);
listener_vec[2] = DotProduct(listener[seat].up, world_vec);
if (snd_leftisright.ival^r_xflip.ival)
listener_vec[1] = -listener_vec[1];
for (i = 0; i < sc->sn.numchannels; i++)
{
scale = 1 + DotProduct(listener_vec, sc->speakerdir[i]);
scale = (1.0 - dist) * scale * sc->dist[i];
v = ch->master_vol * scale * volscale;
v = bound(0, v, 255);
ch->vol[i] = v;
}
}
// =======================================================================
// Start a sound effect
// =======================================================================
static void S_UpdateSoundCard(soundcardinfo_t *sc, qboolean updateonly, channel_t *target_chan, int entnum, int entchannel, sfx_t *sfx, vec3_t origin, vec3_t velocity, float fvol, float attenuation, float timeoffset, float ratemul, unsigned int flags)
{
channel_t *check;
int vol;
int ch_idx;
int skip;
int absstartpos = updateonly?sc->GetChannelPos?sc->GetChannelPos(sc, target_chan)<<PITCHSHIFT:target_chan->pos:0;
extern cvar_t cl_demospeed;
chanupdatereason_t chanupdatetype = updateonly?CUR_UPDATE:CUR_EVERYTHING;
if (!sfx)
sfx = target_chan->sfx;
if (fvol < 0 || !sfx)
{ //stopsound, apparently.
target_chan->sfx = NULL;
return;
}
if (timeoffset != 0.0)
chanupdatetype |= CUR_OFFSET;
if (!ratemul) //rate of 0
ratemul = 1;
ratemul *= snd_playbackrate.value;
if (!snd_ignoregamespeed.ival)
ratemul *= (cls.state?cl.gamespeed:1) * (cls.demoplayback?cl_demospeed.value:1);
if (ratemul <= 0) //in case the user set the cvars weirdly
ratemul = 1;
vol = fvol*255;
// spatialize
if (target_chan->sfx != sfx)
chanupdatetype |= CUR_SOUNDCHANGE;
memset (target_chan, 0, sizeof(*target_chan));
if (!origin)
{
if (sc->seat == -1)
{
VectorClear(target_chan->origin);
attenuation = 0;
flags |= CF_NOSPACIALISE;
}
else
VectorCopy(listener[sc->seat].origin, target_chan->origin);
}
else
{
VectorCopy(origin, target_chan->origin);
}
if (velocity)
VectorCopy(velocity, target_chan->velocity);
else
VectorClear(target_chan->velocity);
target_chan->flags = flags;
target_chan->dist_mult = attenuation / snd_nominaldistance.value;
target_chan->master_vol = vol;
target_chan->entnum = entnum;
target_chan->entchannel = entchannel;
SND_Spatialize(sc, target_chan);
if (!S_LoadSound (sfx, false))
{
target_chan->sfx = NULL;
return; // couldn't load the sound's data
}
//FIXME: why does this only filter for openal devices? its weird.
if (!updateonly && !target_chan->vol[0] && !target_chan->vol[1] && !target_chan->vol[2] && !target_chan->vol[3] && !target_chan->vol[4] && !target_chan->vol[5] && sc->ChannelUpdate)
if (sfx->loopstart == -1 && !(flags&CF_FORCELOOP)) //only skip if its not looping.
{
target_chan->sfx = NULL;
goto updatechannel;
}
target_chan->sfx = sfx;
target_chan->rate = ((1<<PITCHSHIFT) * ratemul); //*sfx->rate/sc->sn.speed;
if (target_chan->rate < 1) /*make sure the rate won't crash us*/
target_chan->rate = 1;
target_chan->pos = absstartpos + (int)(timeoffset*sc->sn.speed*target_chan->rate);
if (!updateonly)
{
// if an identical sound has also been started this frame, offset the pos
// a bit to keep it from just making the first one louder
check = &sc->channel[DYNAMIC_FIRST];
for (ch_idx=DYNAMIC_FIRST; ch_idx < sc->total_chans; ch_idx++, check++)
{
if (check == target_chan)
continue;
if (check->sfx == sfx && !check->pos)
{
skip = rand () % (int)(0.1*sc->sn.speed);
target_chan->pos -= skip*target_chan->rate;
break;
}
}
}
updatechannel:
if (sc->ChannelUpdate)
sc->ChannelUpdate(sc, target_chan, chanupdatetype);
}
float S_UpdateSound(int entnum, int entchannel, sfx_t *sfx, vec3_t origin, vec3_t velocity, float fvol, float attenuation, float timeofs, float pitchadj, unsigned int flags)
{
int i;
int result = 0;
int cards = 0;
soundcardinfo_t *sc;
channel_t *chan;
if (cls.demoseeking)
return result;
S_LockMixer();
for (sc = sndcardinfo; sc; sc = sc->next)
{
cards++;
for (i = 0; i < sc->total_chans; i++)
{
if (sc->channel[i].entnum == entnum && sc->channel[i].entchannel == entchannel && sc->channel[i].sfx)
{
S_UpdateSoundCard(sc, true, &sc->channel[i], entnum, entchannel, sfx, origin, velocity, fvol, attenuation, timeofs, pitchadj, flags);
result++;
break;
}
}
//start it if we couldn't find it.
if (i == sc->total_chans && sfx)
{
chan = SND_PickChannel(sc, entnum, entchannel);
if (chan)
S_UpdateSoundCard(sc, false, chan, entnum, entchannel, sfx, origin, velocity, fvol, attenuation, timeofs, pitchadj, flags);
}
}
S_UnlockMixer();
if (!cards)
cards=1;
return result / (float)cards;
}
void S_StartSound(int entnum, int entchannel, sfx_t *sfx, vec3_t origin, vec3_t velocity, float fvol, float attenuation, float timeofs, float pitchadj, unsigned int flags)
{
soundcardinfo_t *sc;
channel_t *target_chan;
if (!sfx || !*sfx->name) //no named sounds would need specific starting.
return;
if (cls.demoseeking)
return;
if (!sound_started)
return;
if (nosound.ival)
return;
S_LockMixer();
for (sc = sndcardinfo; sc; sc = sc->next)
{
if (flags & CF_NOREPLACE)
{
int i;
for (i = 0; i < sc->total_chans; i++)
if (sc->channel[i].entnum == entnum && sc->channel[i].entchannel == entchannel)
break;
if (i < sc->total_chans)
continue;
}
#ifdef Q3CLIENT
if (flags & CF_CLI_NODUPES)
{ //don't start too many simultaneous sounds. q3 sucks or something.
int active = 0, i;
unsigned int time = Sys_Milliseconds();
for (i = 0; i < sc->total_chans; i++)
{ //as per q3, channel isn't important.
if (sc->channel[i].entnum == entnum && sc->channel[i].sfx == sfx)
{
//never allow a new sound within 50ms of the previous one
if (time - sc->channel[i].starttime < 50)
break;
active++;
}
}
if (active >= 4 || i < sc->total_chans)
{
Con_DPrintf("CF_CLI_NODUPES strikes again!\n");
break;
}
}
#endif
// pick a channel to play on
target_chan = SND_PickChannel(sc, entnum, entchannel);
if (!target_chan)
break;
S_UpdateSoundCard(sc, false, target_chan, entnum, entchannel, sfx, origin, velocity, fvol, attenuation, timeofs, pitchadj, flags);
}
S_UnlockMixer();
}
qboolean S_GetMusicInfo(int musicchannel, float *time, float *duration, char *title, size_t titlesize)
{
qboolean result = false;
soundcardinfo_t *sc;
sfx_t *sfx;
*time = 0;
*duration = 0;
if (titlesize)
*title = 0;
musicchannel += MUSIC_FIRST;
S_LockMixer();
for (sc = sndcardinfo; sc; sc = sc->next)
{
sfx = sc->channel[musicchannel].sfx;
if (sfx)
{
Q_strncpyz(title, COM_SkipPath(sfx->name), titlesize);
if (sfx->loadstate == SLS_LOADED)
{
if (sfx->decoder.querydata)
*duration = sfx->decoder.querydata(sfx, NULL, title, titlesize);
else if (sfx->decoder.buf)
{
sfxcache_t *c = sfx->decoder.buf;
*duration = (float)c->length / c->speed;
}
else
*duration = 0;
//FIXME: openal doesn't report the actual time.
*time = (sc->channel[musicchannel].pos>>PITCHSHIFT) / (float)snd_speed; //the time into the sound, ignoring play rate.
result = true;
}
}
}
S_UnlockMixer();
return result;
}
float S_GetSoundTime(int entnum, int entchannel)
{
int i;
float result = -1; //if we didn't find one
soundcardinfo_t *sc;
S_LockMixer();
for (sc = sndcardinfo; sc && result == -1; sc = sc->next)
{
for (i = 0; i < sc->total_chans; i++)
{
if (sc->channel[i].entnum == entnum && sc->channel[i].entchannel == entchannel && sc->channel[i].sfx)
{
ssamplepos_t spos = sc->GetChannelPos?sc->GetChannelPos(sc, &sc->channel[i]):(sc->channel[i].pos>>PITCHSHIFT);
result = spos / (float)snd_speed; //the time into the sound, ignoring play rate.
break;
}
}
//we found one on this sound device card, ignore others.
if (result != -1)
break;
}
S_UnlockMixer();
return result;
}
float S_GetChannelLevel(int entnum, int entchannel)
{
int i, j;
float result = -1; //if we didn't find one
soundcardinfo_t *sc;
sfxcache_t scachebuf, *scache;
S_LockMixer();
for (sc = sndcardinfo; sc && result == -1; sc = sc->next)
{
for (i = 0; i < sc->total_chans; i++)
{
if (sc->channel[i].entnum == entnum && sc->channel[i].entchannel == entchannel && sc->channel[i].sfx)
{
ssamplepos_t spos = sc->GetChannelPos?sc->GetChannelPos(sc, &sc->channel[i]):(sc->channel[i].pos>>PITCHSHIFT);
if (sc->channel[i].sfx->decoder.decodedata)
scache = sc->channel[i].sfx->decoder.decodedata(sc->channel[i].sfx, &scachebuf, spos, 1);
else
scache = NULL;
if (!scache)
scache = sc->channel[i].sfx->decoder.buf;
if (scache && spos >= scache->soundoffset && spos < scache->soundoffset+scache->length)
{
spos -= scache->soundoffset;
spos *= scache->numchannels;
switch(scache->format)
{
#ifdef FTE_TARGET_WEB
case QAF_BLOB:
result = 0; //sorry. you're going to have to use .wav :(
break;
#endif
case QAF_S8:
for (j = 0; j < scache->numchannels; j++) //average the channels
result += abs(((signed char*)scache->data)[spos+j]);
result /= scache->numchannels*127.0;
break;
case QAF_S16:
for (j = 0; j < scache->numchannels; j++) //average the channels
result += abs(((signed short*)scache->data)[spos+j]);
result /= scache->numchannels*32767.0;
break;
#ifdef MIXER_F32
case QAF_F32:
for (j = 0; j < scache->numchannels; j++) //average the channels
result += fabs(((float*)scache->data)[spos+j]);
result /= scache->numchannels;
break;
#endif
}
}
else
result = 0;
break;
}
}
//we found one on this sound device card, ignore others.
if (result != -1)
break;
}
S_UnlockMixer();
return result;
}
qboolean S_IsPlayingSomewhere(sfx_t *s)
{
soundcardinfo_t *si;
int i;
for (si = sndcardinfo; si; si=si->next)
{
for (i = 0; i < si->total_chans; i++)
if (si->channel[i].sfx == s)
return true;
}
return false;
}
static void S_StopSoundCard(soundcardinfo_t *sc, int entnum, int entchannel)
{
int i;
for (i=0 ; i<sc->total_chans ; i++)
{
if (sc->channel[i].entnum == entnum
&& (!entchannel || sc->channel[i].entchannel == entchannel))
{
sc->channel[i].sfx = NULL;
if (sc->ChannelUpdate)
sc->ChannelUpdate(sc, &sc->channel[i], CUR_EVERYTHING);
if (entchannel)
break;
}
}
}
void S_StopSound(int entnum, int entchannel)
{
soundcardinfo_t *sc;
S_LockMixer();
for (sc = sndcardinfo; sc; sc = sc->next)
S_StopSoundCard(sc, entnum, entchannel);
S_UnlockMixer();
}
void S_StopAllSounds(qboolean clear)
{
int i;
sfx_t *s;
channel_t musics[NUM_MUSICS];
soundcardinfo_t *sc;
if (!sound_started)
return;
S_LockMixer();
for (sc = sndcardinfo; sc; sc = sc->next)
{
for (i=sc->total_chans ; i --> 0 ; )
{
if (i >= MUSIC_FIRST && i < MUSIC_FIRST+NUM_MUSICS && sc->selfpainting)
continue; //don't reset music if is safe to continue playing it without stuttering
s = sc->channel[i].sfx;
if (s)
{
sc->channel[i].sfx = NULL;
if (s->loadstate == SLS_LOADED && s->decoder.ended)
if (!S_IsPlayingSomewhere(s)) //if we aint playing it elsewhere, free it compleatly.
{
if (s->decoder.ended)
s->decoder.ended(s);
}
if (sc->ChannelUpdate)
sc->ChannelUpdate(sc, &sc->channel[i], CUR_EVERYTHING);
}
}
sc->total_chans = NUM_AMBIENTS + NUM_MUSICS; // no statics
Z_ReallocElements((void**)&sc->channel, &sc->max_chans, sc->total_chans, sizeof(*sc->channel));
memcpy(musics, &sc->channel[MUSIC_FIRST], sizeof(musics));
Q_memset(sc->channel, 0, sc->max_chans * sizeof(channel_t));
memcpy(&sc->channel[MUSIC_FIRST], musics, sizeof(musics));
if (clear && !sc->selfpainting) //if its self-painting, then the mixer will continue painting anyway (which is important if its still painting music, but otherwise don't stutter at all when loading)
S_ClearBuffer (sc);
}
S_UnlockMixer();
}
static void S_StopAllSounds_f (void)
{
S_StopAllSounds (true);
}
static void S_ClearBuffer (soundcardinfo_t *sc)
{
void *buffer;
unsigned int dummy;
int clear;
if (!sound_started || !sc->sn.buffer)
return;
if (sc->sn.sampleformat == QSF_U8)
clear = 0x80;
else
clear = 0;
dummy = 0;
buffer = sc->Lock(sc, &dummy);
if (buffer)
{
Q_memset(buffer, clear, sc->sn.samples * sc->sn.samplebytes);
sc->Unlock(sc, buffer);
}
}
/*
=================
S_StaticSound
=================
*/
void S_StaticSound (sfx_t *sfx, vec3_t origin, float vol, float attenuation)
{
channel_t *ss;
soundcardinfo_t *scard;
if (!sfx)
return;
S_LockMixer();
for (scard = sndcardinfo; scard; scard = scard->next)
{
if (scard->total_chans == scard->max_chans)
{
if (!ZF_ReallocElements((void**)&scard->channel, &scard->max_chans, scard->max_chans+64, sizeof(*scard->channel)))
{
Con_Printf ("total_channels == MAX_CHANNELS\n");
continue;
}
}
if (!S_LoadSound (sfx, true))
break;
ss = &scard->channel[scard->total_chans];
scard->total_chans++;
ss->entnum = 0;
ss->sfx = sfx;
ss->rate = 1<<PITCHSHIFT;
VectorCopy (origin, ss->origin);
ss->master_vol = vol*255;
ss->dist_mult = attenuation / snd_nominaldistance.value;
ss->pos = 0;
ss->flags = CF_FORCELOOP|CF_CLI_STATIC;
SND_Spatialize (scard, ss);
if (scard->ChannelUpdate)
scard->ChannelUpdate(scard, ss, CUR_EVERYTHING);
}
S_UnlockMixer();
}
//=============================================================================
void S_Music_Clear(sfx_t *onlyifsample)
{
//stops the current BGM music
//calling this will trigger Media_NextTrack later
sfx_t *s;
soundcardinfo_t *sc;
int i;
for (i = MUSIC_FIRST; i < MUSIC_STOP; i++)
{
for (sc = sndcardinfo; sc; sc=sc->next)
{
s = sc->channel[i].sfx;
if (!s)
continue;
if (onlyifsample && s != onlyifsample)
continue;
sc->channel[i].pos = 0;
sc->channel[i].sfx = NULL;
if (sc->ChannelUpdate)
sc->ChannelUpdate(sc, &sc->channel[i], CUR_EVERYTHING);
if (s && s->decoder.ended && !S_IsPlayingSomewhere(s)) //if we aint playing it elsewhere, free it compleatly.
s->decoder.ended(s);
}
}
}
void S_Music_Seek(float time)
{
soundcardinfo_t *sc;
int i;
for (i = MUSIC_FIRST; i < MUSIC_STOP; i++)
{
for (sc = sndcardinfo; sc; sc=sc->next)
{
sc->channel[i].pos += sc->sn.speed*time * sc->channel[i].rate;
if (sc->channel[i].pos < 0)
{ //clamp to the start of the track
sc->channel[i].pos=0;
}
//if we seek over the end, ignore it. The sound playing code will spot that.
}
}
}
//mixer must be locked
qboolean S_Music_Playing(int musicchannel)
{
soundcardinfo_t *sc;
musicchannel += MUSIC_FIRST;
for (sc = sndcardinfo; sc; sc=sc->next)
{
if (sc->channel[musicchannel].sfx)
return true;
}
return false;
}
/*
===================
S_UpdateAmbientSounds
===================
*/
mleaf_t *Q1BSP_LeafForPoint (model_t *model, vec3_t p);
void S_UpdateAmbientSounds (soundcardinfo_t *sc)
{
float vol;
channel_t *chan;
int i;
#ifdef Q1BSPS
mleaf_t *l;
float oldvol;
int ambientlevel[NUM_AMBIENTS];
#endif
if (!snd_ambient)
return;
for (i = MUSIC_FIRST; i < MUSIC_STOP; i++)
{
chanupdatereason_t changed = CUR_SPACIALISEONLY;
chan = &sc->channel[i];
if (!chan->sfx)
{
float time = 0;
sfx_t *newmusic;
if (!S_Music_Playing(i-MUSIC_FIRST))
{
newmusic = Media_NextTrack(i-MUSIC_FIRST, &time);
if (newmusic && newmusic->loadstate != SLS_FAILED)
{ //okay, now we know which track we're meant to be playing, all devices can play it at once.
soundcardinfo_t *sc2;
for (sc2 = sndcardinfo; sc2; sc2=sc2->next)
{
channel_t *chan = &sc2->channel[i];
chan->sfx = newmusic;
chan->rate = 1<<PITCHSHIFT;
chan->pos = (int)(time * sc->sn.speed) * chan->rate;
changed = CUR_EVERYTHING;
chan->master_vol = bound(0, 1, 255);
chan->vol[0] = chan->vol[1] = chan->vol[2] = chan->vol[3] = chan->vol[4] = chan->vol[5] = chan->master_vol;
if (sc->ChannelUpdate)
sc->ChannelUpdate(sc, chan, changed);
}
}
}
}
if (chan->sfx)
{
chan->flags = /*CF_CL_INACTIVE|*/CF_CL_ABSVOLUME|CF_NOSPACIALISE|CF_NOREVERB; //bypasses volume cvar completely.
vol = 255*bgmvolume.value*voicevolumemod;
if (!vid.activeapp && !snd_inactive.ival && !(chan->flags & CF_CLI_INACTIVE))
vol = 0;
vol = bound(0, vol, 255);
vol = Media_CrossFade(i-MUSIC_FIRST, vol, (chan->pos>>PITCHSHIFT) / (float)snd_speed);
if (vol < 0)
{ //cross fading wants to KILL this track now, apparently.
sfx_t *s = chan->sfx;
if (s->loadstate != SLS_LOADING)
{
chan->pos = 0;
chan->sfx = NULL;
if (sc->ChannelUpdate)
sc->ChannelUpdate(sc, chan, CUR_EVERYTHING);
if (s && s->decoder.ended && !S_IsPlayingSomewhere(s)) //if we aint playing it elsewhere, free it compleatly.
s->decoder.ended(s);
}
}
else
{
chan->master_vol = bound(0, vol, 255);
chan->vol[0] = chan->vol[1] = chan->vol[2] = chan->vol[3] = chan->vol[4] = chan->vol[5] = chan->master_vol;
if (sc->ChannelUpdate)
sc->ChannelUpdate(sc, chan, changed);
}
}
}
#ifdef Q1BSPS
// calc ambient sound levels
for (i = 0; i < NUM_AMBIENTS; i++)
ambientlevel[i] = 0;
if (cl.worldmodel && cl.worldmodel->type == mod_brush && cl.worldmodel->fromgame == fg_quake && cl.worldmodel->loadstate == MLS_LOADED)
{
if (ambient_level.value)
{
if (sc->seat < 0)
{
int seat = max(1,cl.splitclients);
while(seat --> 0)
{
l = Q1BSP_LeafForPoint(cl.worldmodel, listener[seat].origin);
if (!l)
continue;
for (i = 0; i < NUM_AMBIENTS; i++)
ambientlevel[i] = max(ambientlevel[i], l->ambient_sound_level[i]);
}
}
else
{
l = Q1BSP_LeafForPoint(cl.worldmodel, listener[sc->seat].origin);
if (l)
for (i = 0; i < NUM_AMBIENTS; i++)
ambientlevel[i] = l->ambient_sound_level[i];
}
}
}
for (i = 0 ; i< NUM_AMBIENTS ; i++)
{
chan = &sc->channel[AMBIENT_FIRST+i];
chan->sfx = ambient_sfx[AMBIENT_FIRST+i];
chan->entnum = 0;
chan->flags = CF_FORCELOOP | CF_NOSPACIALISE;
chan->rate = 1<<PITCHSHIFT;
VectorClear(chan->origin);
vol = ambient_level.value * ambientlevel[i];
if (vol < 8)
vol = 0;
oldvol = sc->ambientlevels[i];
// don't adjust volume too fast
if (sc->ambientlevels[i] < vol)
{
sc->ambientlevels[i] += host_frametime * ambient_fade.value;
if (sc->ambientlevels[i] > vol)
sc->ambientlevels[i] = vol;
}
else if (chan->master_vol > vol)
{
sc->ambientlevels[i] -= host_frametime * ambient_fade.value;
if (sc->ambientlevels[i] < vol)
sc->ambientlevels[i] = vol;
}
chan->master_vol = sc->ambientlevels[i];
chan->vol[0] = chan->vol[1] = chan->vol[2] = chan->vol[3] = chan->vol[4] = chan->vol[5] = bound(0, chan->master_vol * (volume.value*voicevolumemod), 255);
if (sc->ChannelUpdate)
sc->ChannelUpdate(sc, chan, ((oldvol == 0) ^ (sc->ambientlevels[i] == 0))?CUR_EVERYTHING:CUR_SPACIALISEONLY);
}
#endif
}
struct sndreverbproperties_s *reverbproperties;
size_t numreverbproperties;
qboolean S_UpdateReverb(size_t slot, void *reverb, size_t reverbsize)
{
struct reverbproperties_s newprops;
if (slot >= 1024)
return false;
if (slot >= numreverbproperties)
{
int slots = slot+1;
void *n = BZ_Realloc(reverbproperties, sizeof(*reverbproperties)*slots);
if (!n)
return false;
reverbproperties = n;
memset(reverbproperties+numreverbproperties, 0, sizeof(*reverbproperties) * (slots-numreverbproperties));
numreverbproperties = slots;
}
memset(&newprops, 0, sizeof(newprops));
if (reverb)
{
//clamp the size for possible future extensibility
if (reverbsize > sizeof(newprops))
reverbsize = sizeof(newprops);
memcpy(&newprops, reverb, reverbsize);
}
if (memcmp(&newprops, &reverbproperties[slot].props, sizeof(newprops)))
{
reverbproperties[slot].props = newprops;
reverbproperties[slot].modificationcount++;
}
return true;
}
/*
============
S_Update
Called once each time through the main loop
============
*/
void S_UpdateListener(int seat, int entnum, vec3_t origin, vec3_t forward, vec3_t right, vec3_t up, size_t reverbtype, vec3_t velocity)
{
soundcardinfo_t *sc;
listener[seat].entnum = entnum;
VectorCopy(origin, listener[seat].origin);
VectorCopy(forward, listener[seat].forward);
VectorCopy(right, listener[seat].right);
VectorCopy(up, listener[seat].up);
VectorCopy(velocity, listener[seat].velocity);
for (sc = sndcardinfo; sc; sc=sc->next)
if (sc->SetEnvironmentReverb && (sc->seat == seat || (sc->seat == -1 && seat == 0)))
sc->SetEnvironmentReverb(sc, reverbtype);
}
void S_GetListenerInfo(int seat, float *origin, float *forward, float *right, float *up)
{
VectorCopy(listener[seat].origin, origin);
VectorCopy(listener[seat].forward, forward);
VectorCopy(listener[seat].right, right);
VectorCopy(listener[seat].up, up);
}
static void S_Q2_AddEntitySounds(soundcardinfo_t *sc)
{
vec3_t positions[2048];
int entnums[countof(positions)];
sfx_t *sounds[countof(positions)];
unsigned int count;
unsigned int j;
channel_t *c;
#ifdef Q2CLIENT
if (cls.protocol == CP_QUAKE2)
count = CLQ2_GatherSounds(positions, entnums, sounds, countof(sounds));
else
#endif
#ifdef VM_CG
if (cls.protocol == CP_QUAKE3 && q3)
count = q3->cg.GatherLoopingSounds(positions, entnums, sounds, countof(sounds));
else
#endif
return;
while(count --> 0)
{
sfx_t *sfx = sounds[count];
if (!sfx)
continue;
if (sfx->loadstate == SLS_NOTLOADED)
S_LoadSound(sfx, true);
if (sfx->loadstate != SLS_LOADED)
continue; //not ready yet
if (sc->ChannelUpdate)
{
for (c = NULL, j=DYNAMIC_FIRST; j < sc->total_chans ; j++)
{
if (sc->channel[j].entnum == entnums[count] && !sc->channel[j].entchannel && (sc->channel[j].flags & CF_CLI_AUTOSOUND))
{
c = &sc->channel[j];
break;
}
}
}
else
{
for (c = NULL, j=DYNAMIC_FIRST; j < sc->total_chans ; j++)
{
if (sc->channel[j].sfx == sfx && (sc->channel[j].flags & CF_CLI_AUTOSOUND))
{
c = &sc->channel[j];
break;
}
}
}
if (!c)
{
c = SND_PickChannel(sc, 0, 0);
if (!c)
continue;
c->flags = CF_CLI_AUTOSOUND|CF_FORCELOOP;
c->entnum = sc->ChannelUpdate?entnums[count]:0;
c->entchannel = 0;
c->dist_mult = 3 / snd_nominaldistance.value;
c->master_vol = 255 * 1;
c->pos = 0<<PITCHSHIFT; //q2 does weird stuff with the pos. we just forceloop and detect when it became irrelevant. this is required for stream decoding or openal
c->rate = 1<<PITCHSHIFT;
for (j = 0; j < countof(c->vol); j++)
c->vol[j] = 0;
c->sfx = NULL;
}
if (sc->ChannelUpdate)
{ //hardware mixing doesn't support merging
VectorCopy(positions[count], c->origin);
SND_Spatialize(sc, c);
if (c->sfx)
sc->ChannelUpdate(sc, c, CUR_SPACIALISEONLY);
}
else
{ //merge with any other ents, if we can
for (j = 0; j <= count; j++)
{
if (sounds[j] == sfx)
{
sounds[j] = NULL;
SND_AccumulateSpacialization(sc, c, positions[j]);
}
}
}
if (!c->sfx)
{
for (j = 0; j < countof(c->vol); j++)
if (c->vol[j])
break;
if (j == countof(c->vol))
c->sfx = NULL; //err, never mind
else
{
c->sfx = sfx;
if (sc->ChannelUpdate)
sc->ChannelUpdate(sc, c, CUR_EVERYTHING);
}
}
}
}
static void S_UpdateCard(soundcardinfo_t *sc)
{
int i, j;
channel_t *ch;
channel_t *combine;
if (!sound_started)
return;
if ((snd_blocked > 0))
{
if (!sc->inactive_sound)
return;
}
#ifdef AVAIL_OPENAL
if (sc->ListenerUpdate)
{
sc->ListenerUpdate(sc, listener[sc->seat].entnum, listener[sc->seat].origin, listener[sc->seat].forward, listener[sc->seat].right, listener[sc->seat].up, listener[sc->seat].velocity);
}
#endif
// update general area ambient sound sources
S_UpdateAmbientSounds (sc);
combine = NULL;
// update spatialization for static and dynamic sounds
ch = sc->channel+DYNAMIC_FIRST;
for (i=DYNAMIC_FIRST ; i<sc->total_chans; i++, ch++)
{
if (!ch->sfx)
continue;
if (ch->flags & CF_CLI_AUTOSOUND)
{
if (!ch->vol[0] && !ch->vol[1] && !ch->vol[2] && !ch->vol[3] && !ch->vol[4] && !ch->vol[5])
{
ch->sfx = NULL;
if (sc->ChannelUpdate)
sc->ChannelUpdate(sc, ch, CUR_EVERYTHING);
}
ch->vol[0] = ch->vol[1] = ch->vol[2] = ch->vol[3] = ch->vol[4] = ch->vol[5] = 0;
continue;
}
if (sc->ChannelUpdate)
{
if (ch->flags & CF_FOLLOW)
SND_Spatialize(sc, ch); //update it a little
sc->ChannelUpdate(sc, ch, CUR_SPACIALISEONLY);
continue;
}
SND_Spatialize(sc, ch); // respatialize channel
if (!ch->vol[0] && !ch->vol[1] && !ch->vol[2] && !ch->vol[3] && !ch->vol[4] && !ch->vol[5])
continue;
// try to combine static sounds with a previous channel of the same
// sound effect so we don't mix five torches every frame
if (ch->flags & CF_CLI_STATIC)
{
// see if it can just use the last one
if (combine && combine->sfx == ch->sfx)
{
combine->vol[0] += ch->vol[0];
combine->vol[1] += ch->vol[1];
combine->vol[2] += ch->vol[2];
combine->vol[3] += ch->vol[3];
combine->vol[4] += ch->vol[4];
combine->vol[5] += ch->vol[5];
ch->vol[0] = ch->vol[1] = ch->vol[2] = ch->vol[3] = ch->vol[4] = ch->vol[5] = 0;
continue;
}
// search for one
combine = sc->channel+DYNAMIC_FIRST;
for (j=DYNAMIC_FIRST ; j<i; j++, combine++)
if (combine->sfx == ch->sfx)
break;
if (j == sc->total_chans)
{
combine = NULL;
}
else
{
if (combine != ch)
{
combine->vol[0] += ch->vol[0];
combine->vol[1] += ch->vol[1];
combine->vol[2] += ch->vol[2];
combine->vol[3] += ch->vol[3];
combine->vol[4] += ch->vol[4];
combine->vol[5] += ch->vol[5];
ch->vol[0] = ch->vol[1] = ch->vol[2] = ch->vol[3] = ch->vol[4] = ch->vol[5] = 0;
}
continue;
}
}
}
S_Q2_AddEntitySounds(sc);
//
// debugging output
//
if (snd_show.ival)
{
struct listener_s *l;
int active, mute;
active = 0;
mute = 0;
ch = sc->channel;
for (i=0 ; i<sc->total_chans; i++, ch++)
{
if (ch->sfx && (ch->vol[0] || ch->vol[1]) )
{
if (snd_show.ival > 1)
Con_Printf ("%i, %i/%i/%i/%i/%i/%i %s\n", i, ch->vol[0], ch->vol[1], ch->vol[2], ch->vol[3], ch->vol[4], ch->vol[5], ch->sfx->name);
active++;
}
else if (ch->sfx)
mute++;
}
if (sc->seat < 0)
l = &listener[0];
else
l = &listener[sc->seat];
Con_Printf ("----(%i+%i %s %i %.1f %.1f %.1f)----\n", active, mute, sc->name, l->entnum, l->origin[0], l->origin[1], l->origin[2]);
}
#ifdef HAVE_MIXER
// mix some sound
if (sc->selfpainting)
return;
if (snd_blocked > 0)
{
if (!sc->inactive_sound)
return;
}
S_Update_(sc);
#endif
}
#ifdef HAVE_MIXER
int S_GetMixerTime(soundcardinfo_t *sc)
{
int samplepos;
int fullsamples;
fullsamples = sc->sn.samples / sc->sn.numchannels;
// it is possible to miscount buffers if it has wrapped twice between
// calls to S_Update. Oh well.
samplepos = sc->GetDMAPos(sc);
if (sc->samplequeue > 0)
samplepos -= sc->samplequeue;
if (samplepos < 0)
{
samplepos = 0;
}
if (samplepos < sc->oldsamplepos)
{
int bias;
sc->buffers++; // buffer wrapped
if (sc->paintedtime > 0x40000000)
{
//when things get too large, we push everything back to prevent overflows
bias = sc->paintedtime;
bias -= bias % fullsamples;
sc->paintedtime -= bias;
sc->buffers -= bias / fullsamples;
}
}
sc->oldsamplepos = samplepos;
return sc->buffers*fullsamples + samplepos/sc->sn.numchannels;
}
#endif
void S_Update (void)
{
soundcardinfo_t *sc;
RSpeedMark();
S_LockMixer();
for (sc = sndcardinfo; sc; sc = sc->next)
S_UpdateCard(sc);
S_UnlockMixer();
RSpeedEnd(RSPEED_AUDIO);
}
void S_ExtraUpdate (void)
{
#ifdef HAVE_MIXER
soundcardinfo_t *sc;
#endif
if (!sound_started)
return;
#if defined(_WIN32) && !defined(WINRT)
INS_Accumulate ();
#endif
#ifdef HAVE_MIXER
if (snd_noextraupdate.ival)
return; // don't pollute timings
for (sc = sndcardinfo; sc; sc = sc->next)
{
if (sc->selfpainting)
continue;
if (snd_blocked > 0)
{
if (!sc->inactive_sound)
continue;
}
S_LockMixer();
S_Update_(sc);
S_UnlockMixer();
}
#endif
}
#ifdef HAVE_MIXER
static void S_Update_(soundcardinfo_t *sc)
{
int soundtime; /*in pairs*/
unsigned endtime;
int samps;
// Updates DMA time
soundtime = S_GetMixerTime(sc);
if (sc->samplequeue > 0)
{
/*device uses a write-once queue*/
endtime = soundtime + sc->samplequeue/sc->sn.numchannels;
soundtime = sc->paintedtime;
samps = sc->samplequeue / sc->sn.numchannels;
}
else if (sc->samplequeue < 0)
{ /*device is telling us the exact point that we should be mixing to*/
endtime = soundtime;
soundtime = sc->paintedtime;
samps = sc->sn.samples / sc->sn.numchannels;
}
else
{
/*device uses memory-mapped output*/
// check to make sure that we haven't overshot
if (sc->paintedtime < soundtime)
{
//Con_Printf ("S_Update_ : overflow\n");
sc->paintedtime = soundtime;
}
// mix ahead of current position
endtime = soundtime + (int)(_snd_mixahead.value * sc->sn.speed);
samps = sc->sn.samples / sc->sn.numchannels;
}
if (endtime - soundtime > samps)
{
endtime = soundtime + samps;
}
/*DirectSound may have killed us to give priority to another app, ask to restore it*/
if (sc->Restore)
sc->Restore(sc);
S_PaintChannels (sc, endtime);
sc->Submit(sc, soundtime, endtime);
}
/*
called periodically by dedicated mixer threads.
do any blocking calls AFTER this returns. note that this means you can't use the Submit/unlock method to submit blocking audio.
*/
void S_MixerThread(soundcardinfo_t *sc)
{
S_LockMixer();
S_Update_(sc);
S_UnlockMixer();
}
#endif
/*
===============================================================================
console functions
===============================================================================
*/
void S_Play_f(void)
{ //plays a sound located around the player
int i;
char name[256];
sfx_t *sfx;
const char *cmdname = Cmd_Argv(0);
float vol, attenuation = 0;
unsigned int flags = CF_NOSPACIALISE;
int entnum = 0;
float *origin = NULL;
/* //Vanilla compat (breaks modern QW mods):
if (!strcmp(cmdname, "play"))
{
flags = 0;
attenuation = 1;
origin = listener[0].origin;
entnum = listener[0].entnum;
}
*/
i = 1;
while (i<Cmd_Argc())
{
if (!Q_strrchr(Cmd_Argv(i), '.'))
{
Q_strncpyz(name, Cmd_Argv(i), sizeof(name)-4);
Q_strcat(name, ".wav");
}
else
Q_strncpyz(name, Cmd_Argv(i), sizeof(name));
i++;
sfx = S_PrecacheSound(name);
if (!strcmp(cmdname, "playvol"))
vol = Q_atof(Cmd_Argv(i++));
else
vol = 1.0;
S_StartSound(entnum, 0, sfx, origin, NULL, vol, attenuation, 0, 0, flags);
}
}
void S_SoundList_f(void)
{
int i;
sfx_t *sfx;
sfxcache_t *sc;
sfxcache_t scachebuf;
int size, total;
int duration;
S_LockMixer();
total = 0;
for (sfx=known_sfx, i=0 ; i<num_sfx ; i++, sfx++)
{
if (sfx->loadstate != SLS_LOADED)
sc = NULL;
else if (sfx->decoder.decodedata)
{
if (sfx->decoder.querydata)
sc = (sfx->decoder.querydata(sfx, &scachebuf, NULL, 0) < 0)?NULL:&scachebuf;
else
sc = NULL; //don't bother trying to actually decode anything here.
if (!sc)
{
Con_Printf("S( ) : %s\n", sfx->name);
continue;
}
}
else
sc = sfx->decoder.buf;
if (!sc)
{
Con_Printf("?( ) : %s\n", sfx->name);
continue;
}
size = (sc->soundoffset+sc->length)*QAF_BYTES(sc->format)*(sc->numchannels);
duration = (sc->soundoffset+sc->length) / sc->speed;
total += size;
if (sfx->loopstart >= 0)
Con_Printf ("L");
else
Con_Printf (" ");
Con_Printf("(%2db%2ic) %6i %2is : %s\n",QAF_BYTES(sc->format)*8, sc->numchannels, size, duration, sfx->name);
}
Con_Printf ("Total resident: %i\n", total);
S_UnlockMixer();
}
void S_LocalSound2 (const char *sound, int channel, float volume)
{
sfx_t *sfx;
if (nosound.ival)
return;
if (!sound_started)
return;
sfx = S_PrecacheSound (sound);
if (!sfx)
{
Con_Printf ("S_LocalSound: can't cache %s\n", sound);
return;
}
S_StartSound (0, channel, sfx, NULL, NULL, volume, 0, 0, 0, CF_CLI_INACTIVE|CF_NOSPACIALISE|CF_NOREVERB);
}
void S_LocalSound (const char *sound)
{
S_LocalSound2(sound, 256, 1);
}
typedef struct {
sfxdecode_t decoder;
qboolean inuse;
int id;
sfx_t *sfx;
int numchannels;
qaudiofmt_t format;
int length;
void *data;
} streaming_t;
#define MAX_RAW_SOURCES (MAX_CLIENTS+1)
streaming_t s_streamers[MAX_RAW_SOURCES];
void S_ClearRaw(void)
{
memset(s_streamers, 0, sizeof(s_streamers));
}
//returns an sfxcache_t stating where the data is
sfxcache_t *QDECL S_Raw_Locate(sfx_t *sfx, sfxcache_t *buf, ssamplepos_t start, int length)
{
streaming_t *s = sfx->decoder.buf;
if (buf)
{
buf->data = s->data;
buf->length = s->length;
buf->numchannels = s->numchannels;
buf->soundoffset = 0;
buf->speed = snd_speed;
buf->format = s->format;
}
if (start >= s->length)
return NULL; //eof...
return buf;
}
void QDECL S_Raw_Ended(sfx_t *sfx)
{ //no longer playing anywhere...
streaming_t *s = sfx->decoder.buf;
s->inuse = false; //let it get reused now.
}
void QDECL S_Raw_Purge(sfx_t *sfx)
{ //flush all caches, will be re-read from disk (or not, because this is streamed)
streaming_t *s = sfx->decoder.buf;
s->length = 0;
s->numchannels = 0;
BZ_Free(s->data);
s->data = NULL;
s->inuse = false;
memset(&sfx->decoder, 0, sizeof(sfx->decoder));
}
//streaming audio. //this is useful when there is one source, and the sound is to be played with no attenuation
void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels, qaudiofmt_t format, float volume)
{
soundcardinfo_t *si;
int i;
int prepadl; //this is the amount of data that was previously available, and will be removed from the buffer.
int spare; //the amount of existing data that is still left to be played
int outsamples; //the amount of data we're going to add (at the output rate)
double speedfactor;
qbyte *newcache;
streaming_t *s, *free=NULL;
if (!sound_started)
return;
for (s = s_streamers, i = 0; i < MAX_RAW_SOURCES; i++, s++)
{
if (!s->inuse)
{
if (!free)
free = s;
continue;
}
if (s->id == sourceid)
break;
}
if (!data)
{
if (i == MAX_RAW_SOURCES)
return; //wierd, it wasn't even playing.
s->inuse = false;
S_LockMixer();
for (si = sndcardinfo; si; si=si->next)
for (i = 0; i < si->total_chans; i++)
if (si->channel[i].sfx == s->sfx)
{
si->channel[i].sfx = NULL;
break;
}
BZ_Free(s->data);
s->data = NULL;
S_UnlockMixer();
return;
}
if (i == MAX_RAW_SOURCES || !s->inuse) //whoops.
{
if (i == MAX_RAW_SOURCES)
{
if (!free)
{
Con_Printf("No free audio streams\n");
return;
}
s = free;
}
if (!s->sfx)
s->sfx = S_FindName(va("***stream_%i***", i), true, false);
s->sfx->decoder.buf = s;
s->sfx->decoder.decodedata = S_Raw_Locate;
s->sfx->decoder.ended = S_Raw_Ended;
s->sfx->decoder.purge = S_Raw_Purge;
s->sfx->loopstart = -1; //non-looping...
s->sfx->loadstate = SLS_LOADED;
s->numchannels = channels;
s->format = format;
s->data = NULL;
s->length = 0;
s->id = sourceid;
s->inuse = true;
// Con_Printf("Added new raw stream\n");
}
S_LockMixer();
if (s->format != format || s->numchannels != channels)
{
s->format = format;
s->numchannels = channels;
s->length = 0;
Con_Printf("Restarting raw stream\n");
}
speedfactor = (double)speed/snd_speed;
outsamples = samples/speedfactor;
prepadl = 0x7fffffff;
for (si = sndcardinfo; si; si=si->next) //make sure all cards are playing, and that we still get a prepad if just one is.
{
for (i = 0; i < si->total_chans; i++)
if (si->channel[i].sfx == s->sfx)
{
if (prepadl > (si->channel[i].pos>>PITCHSHIFT))
prepadl = (si->channel[i].pos>>PITCHSHIFT);
break;
}
}
if (prepadl == 0x7fffffff)
{
if (snd_show.ival)
Con_Printf("Wasn't playing\n");
prepadl = 0;
spare = 0;
if (spare > snd_speed)
{
Con_DPrintf("Sacrificed raw sound stream\n");
spare = 0; //too far out. sacrifice it all
}
}
else
{
if (prepadl < 0)
prepadl = 0;
spare = s->length - prepadl;
if (spare < 0) //remaining samples since last time
spare = 0;
if (spare > snd_speed*2) // more than 2 seconds of sound. don't buffer more than 2 seconds. 1: its probably buggy if we need to. 2: takes too much memory, and we use malloc+copies.
{
Con_DPrintf("Sacrificed raw sound stream\n");
spare = 0; //too far out. sacrifice it all
}
}
newcache = BZ_Malloc((spare+outsamples) * (s->numchannels) * QAF_BYTES(s->format));
memcpy(newcache, (qbyte*)s->data + prepadl * (s->numchannels) * QAF_BYTES(s->format), spare * (s->numchannels) * QAF_BYTES(s->format));
BZ_Free(s->data);
s->data = newcache;
s->length = spare + outsamples;
{
extern cvar_t snd_linearresample_stream;
short *outpos = (short *)((char*)s->data + spare * (s->numchannels) * QAF_BYTES(s->format));
SND_ResampleStream(data,
speed,
format,
channels,
samples,
outpos,
snd_speed,
s->format,
s->numchannels,
snd_linearresample_stream.ival);
}
for (si = sndcardinfo; si; si=si->next)
{
for (i = 0; i < si->total_chans; i++)
if (si->channel[i].sfx == s->sfx)
{
si->channel[i].pos -= prepadl*si->channel[i].rate;
if (si->channel[i].pos < 0)
si->channel[i].pos = 0;
si->channel[i].master_vol = 255 * volume;
if (si->ChannelUpdate)
si->ChannelUpdate(si, &si->channel[i], CUR_SPACIALISEONLY);
break;
}
if (i == si->total_chans) //this one wasn't playing.
{
channel_t *c = SND_PickChannel(si, -1, 0);
if (c)
{
c->flags = (sourceid>=0?CF_CLI_INACTIVE:0)|CF_CL_ABSVOLUME|CF_NOSPACIALISE;
c->entnum = 0;
c->entchannel = 0;
c->dist_mult = 0;
c->master_vol = 255 * volume;
c->pos = 0;
c->rate = 1<<PITCHSHIFT;
c->sfx = s->sfx;
SND_Spatialize(si, c);
if (si->ChannelUpdate)
si->ChannelUpdate(si, c, CUR_EVERYTHING);
}
}
}
S_UnlockMixer();
}
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