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fteqw/engine/client/snd_dma.c
Spoike 9e8bb446f4 implemented pm_stepdown. 
attempt to implement 'simple csqc' api.
handle qw+nq gunshot+blood+lightning differently - they do actually have different particle spawn patterns (qw is a single point, so spreads wider).
fix q3ui logo mesh thing. work around q3ui player meshes on d3d.
split video and renderer latching, so vid_reload delatches more stuff.
fix autosprite+autosprite2 in 6 different renderers...
added fog volumes to d3d9 renderer.
using matrix hacks instead of glDepthRange, this should give more consistent behaviour, especially now that we have r_viewmodel_fov.
small cleanup for gl shadowmaps to make the interface more consistent with other renderers.
added patchDef2 parsing to fte's .map loader, doesn't actually use it though.
some fixes for q3's shaders, including to try to get overbright working better.
updated customskin api to give more control.
first attempt at a packager system for fteqccgui. probably useless, but whatever.
menusys changes to try to support QSS's csqc.

git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@5200 fc73d0e0-1445-4013-8a0c-d673dee63da5
2018-01-22 19:18:04 +00:00

4183 lines
110 KiB
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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 CSQC_DAT
//for sounds following csqc ents
#include "pr_common.h"
extern world_t csqc_world;
#endif
static void S_Play(void);
static void S_PlayVol(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
{
int entnum;
vec3_t origin;
vec3_t velocity;
vec3_t forward;
vec3_t right;
vec3_t up;
} listener[MAX_SPLITS];
vec_t sound_nominal_clip_dist=1000.0;
#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 bgmvolume = CVARAFD( "musicvolume", "0.3", "bgmvolume", CVAR_ARCHIVE,
"Volume level for background music.");
cvar_t volume = CVARFD( "volume", "0.7", CVAR_ARCHIVE,
"Main volume level for all engine sound.");
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 precache = CVARAF( "s_precache", "1",
"precache", 0);
cvar_t loadas8bit = CVARAFD( "s_loadas8bit", "0",
"loadas8bit", CVAR_ARCHIVE,
"Downsample sounds on load as lower quality 8-bit sound.");
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 (ex. 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. "DISTRIBUTION" supports up to 6.");
cvar_t snd_buffersize = CVARAF( "s_buffersize", "0",
"snd_buffersize", 0);
cvar_t snd_samplebits = CVARAF( "s_bits", "16",
"snd_samplebits", CVAR_ARCHIVE);
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_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, "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, "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, "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 other 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.");
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.");
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.h>
#include <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.
};
#define VOIP_DEFAULT_CODEC (cls.protocol==CP_QUAKEWORLD?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 quakeworld.
static struct
{
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;
struct
{
qboolean inited;
qboolean loaded;
dllhandle_t *opuslib;
} opus;
unsigned char enccodec;
void *encoder;
unsigned int encframesize;
unsigned int encsamplerate;
void *decoder[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;
#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
#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
#ifdef AVAIL_OPENAL
extern snd_capture_driver_t OPENAL_Capture;
#endif
snd_capture_driver_t DSOUND_Capture;
snd_capture_driver_t OSS_Capture;
snd_capture_driver_t SDL_Capture;
snd_capture_driver_t *capturedrivers[] =
{
&DSOUND_Capture,
&SDL_Capture,
&OSS_Capture,
#ifdef AVAIL_OPENAL
&OPENAL_Capture,
#endif
NULL
};
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;
}
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;
}
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])
{
case VOIP_SPEEX_OLD:
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
qspeex_decoder_destroy(s_voip.decoder[sender]);
break;
case VOIP_RAW16:
break;
case VOIP_OPUS:
qopus_decoder_destroy(s_voip.decoder[sender]);
break;
}
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;
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;
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;
}
s_voip.deccodec[sender] = codec;
s_voip.decgen[sender] = gen;
s_voip.decseq[sender] = seq;
}
//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;
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;
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;
}
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;
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;
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;
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;
}
}
if (drops)
Con_DPrintf("%i dropped audio frames\n", drops);
if (decodesamps > 0)
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))
{
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_OLD:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
return S_Speex_Init();
case VOIP_PCMA:
case VOIP_PCMU:
return true;
case VOIP_OPUS:
return S_Opus_Init();
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)
{
if (s_voip.speexdsp.preproc)
qspeex_preprocess_run(s_voip.speexdsp.preproc, start);
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)
{
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_Printf("No microphone drivers supported\n");
else
Con_Printf("Microphone driver \"%s\" is not valid\n", driver);
}
else
Con_Printf("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
if (voipcodec == VOIP_SPEEX_OLD)
voipcodec = VOIP_SPEEX_WIDE;
}
voicevolumemod = s_voip.lastspoke_any > realtime?snd_voip_ducking.value:1;
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)
{
case VOIP_SPEEX_OLD:
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
break;
case VOIP_OPUS:
qopus_encoder_destroy(s_voip.encoder);
break;
}
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)
{
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;
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;
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;
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)
{
case VOIP_SPEEX_OLD:
case VOIP_SPEEX_NARROW:
case VOIP_SPEEX_WIDE:
case VOIP_SPEEX_ULTRAWIDE:
qspeex_bits_reset(&s_voip.speex.encbits);
break;
case VOIP_RAW16:
break;
case VOIP_OPUS:
qopus_encoder_ctl(s_voip.encoder, OPUS_RESET_STATE);
break;
}
}
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 (!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);
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;
}
switch(s_voip.enccodec)
{
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;
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;
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;
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)
{
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;
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;
case VOIP_OPUS:
NET_RTP_Transmit(initseq, inittimestamp, "opus@48000", outbuf, outpos);
break;
}
}
#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)
{
voipbutton = true;
}
static void S_Voip_Disable_f(void)
{
voipbutton = false;
}
static void S_Voip_f(void)
{
int i;
if (!strcmp(Cmd_Argv(1), "maxgain"))
{
i = atoi(Cmd_Argv(2));
if (s_voip.speexdsp.preproc)
qspeex_preprocess_ctl(s_voip.speexdsp.preproc, SPEEX_PREPROCESS_SET_AGC_MAX_GAIN, &i);
}
}
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.voiplevel > 100)
return 100;
if (!s_voip.cdriverctx || (!ignorevad && s_voip.dumps))
return -1;
return s_voip.voiplevel;
}
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");
Cvar_Register(&snd_voip_noisefilter, "Voice Chat");
Cvar_Register(&snd_voip_autogain, "Voice Chat");
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);
}
#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 SDL_Output;
#ifdef __linux__
sounddriver_t ALSA_Output;
#endif
sounddriver_t OSS_Output;
#ifdef AVAIL_OPENAL
extern sounddriver_t OPENAL_Output;
#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 MacOS_AudioOutput; //prefered on mac
#endif
#ifdef ANDROID
sounddriver_t Droid_AudioOutput; //prefered on android (java thread)
#endif
#if defined(__MORPHOS__)
sounddriver_t AHI_AudioOutput; //prefered on morphos
#endif
#ifdef NACL
extern sounddriver_t PPAPI_AudioOutput; //nacl
#endif
sounddriver_t 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
#ifdef __linux__
&ALSA_Output, //pure shite
#endif
&OSS_Output, //good, but not likely to work any more on linux (unlike every other unix system with a decent opengl driver)
#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
&Droid_AudioOutput, //prefered on android (java thread)
#endif
#if defined(__MORPHOS__)
&AHI_AudioOutput, //prefered on morphos
#endif
#ifdef NACL
&PPAPI_AudioOutput, //google's native client
#endif
&SNDIO_AudioOutput, //prefered on OpenBSD
#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, MAX_DYNAMIC_CHANNELS+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 && ( strchr(devicecode, ' ') ||
strchr(devicecode, '\"')))
fullintname = va("\"%s:%s\"", driver, devicecode); //it'll all get escaped anyway. but yeah, needs to be a single token or our multi-device stuff won't work properly. yes, this is a bit of a hack.
else 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 && ( strchr(devicecode, ' ') ||
strchr(devicecode, '\"')))
fullintname = va("\"%s:%s\"", driver, devicecode); //it'll all get escaped anyway. but yeah, needs to be a single token or our multi-device stuff won't work properly. yes, this is a bit of a hack.
else 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;
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 (!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 (!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][0])
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);
Cmd_AddCommand("play2", S_Play);
Cmd_AddCommand("playvol", S_PlayVol);
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(&volume, "Sound controls");
Cvar_Register(&precache, "Sound controls");
Cvar_Register(&loadas8bit, "Sound controls");
Cvar_Register(&bgmvolume, "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_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_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)
{
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_TouchSound
==================
*/
void S_TouchSound (char *name)
{
if (!sound_started)
return;
S_FindName (name, true, 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 (precache.ival && sndcardinfo)
S_LoadSound (sfx);
return sfx;
}
//=============================================================================
/*
=================
SND_PickChannel
=================
*/
channel_t *SND_PickChannel(soundcardinfo_t *sc, int entnum, int entchannel)
{
int ch_idx;
int oldestpos;
int oldest;
// Check for replacement sound, or find the best one to replace
oldest = -1;
oldestpos = -1;
for (ch_idx=DYNAMIC_FIRST; ch_idx < DYNAMIC_STOP ; 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;
}
// don't let monster sounds override player sounds
if (sc->seat != -1 && sc->channel[ch_idx].entnum == listener[sc->seat].entnum && entnum != listener[sc->seat].entnum && sc->channel[ch_idx].sfx)
continue;
if (!sc->channel[ch_idx].sfx)
{
oldestpos = 0x7fffffff;
oldest = ch_idx;
}
else if (sc->channel[ch_idx].pos > oldestpos)
{
oldestpos = sc->channel[ch_idx].pos;
oldest = ch_idx;
}
}
if (oldest == -1)
return NULL;
if (sc->channel[oldest].sfx)
sc->channel[oldest].sfx = NULL;
if (sc->total_chans <= oldest)
sc->total_chans = oldest+1;
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_ABSVOLUME)
volscale = 1;
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)
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(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_ABSVOLUME)
volscale = 1;
else
volscale = volume.value * voicevolumemod;
if (sc->seat == -1)
{
seat = 0;
VectorSubtract(ch->origin, listener[seat].origin, world_vec);
dist = DotProduct(world_vec,world_vec);
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;
}
}
}
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;
for (i = 0; i < sc->sn.numchannels; i++)
ch->vol[i] = bound(0, v, 255);
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)
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);
}
}
// =======================================================================
// 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?target_chan->pos:0;
extern cvar_t cl_demospeed;
int chanupdatetype = true;
if (!sfx)
sfx = target_chan->sfx;
if (fvol < 0 || !sfx)
{ //stopsound, apparently.
target_chan->sfx = NULL;
return;
}
if (ratemul <= 0)
ratemul = 1;
ratemul *= snd_playbackrate.value * (cls.state?cl.gamespeed:1) * (cls.demoplayback?cl_demospeed.value:1);
vol = fvol*255;
// spatialize
if (target_chan->sfx != sfx)
chanupdatetype = true;
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 / sound_nominal_clip_dist;
target_chan->master_vol = vol;
target_chan->entnum = entnum;
target_chan->entchannel = entchannel;
SND_Spatialize(sc, target_chan);
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)
{
target_chan->sfx = NULL;
return; // not audible at all
}
if (!S_LoadSound (sfx))
{
target_chan->sfx = NULL;
return; // couldn't load the sound's data
}
target_chan->sfx = sfx;
if (updateonly && sc->ChannelUpdate)
{
chanupdatetype = 2;
absstartpos = 0;
}
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 < DYNAMIC_STOP; 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;
}
}
}
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)
{
// 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)
{
result = (sc->channel[i].pos>>PITCHSHIFT) / (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;
}
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], true);
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], true);
}
}
sc->total_chans = MAX_DYNAMIC_CHANNELS + 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))
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 / sound_nominal_clip_dist;
ss->pos = 0;
ss->flags = CF_FORCELOOP;
SND_Spatialize (scard, ss);
if (scard->ChannelUpdate)
scard->ChannelUpdate(scard, ss, true);
}
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], true);
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++)
{
qboolean changed = false;
chan = &sc->channel[i];
if (!chan->sfx)
{
float time = 0;
sfx_t *newmusic = Media_NextTrack(i-MUSIC_FIRST, &time);
if (newmusic && newmusic->loadstate != SLS_FAILED)
{
chan->sfx = newmusic;
chan->rate = 1<<PITCHSHIFT;
chan->pos = (int)(time * sc->sn.speed) * chan->rate;
changed = true;
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_ABSVOLUME|CF_NOSPACIALISE|CF_NOREVERB; //bypasses volume cvar completely.
vol = 255*bgmvolume.value*voicevolumemod;
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, true);
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));
}
#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
return;
while(count --> 0)
{
sfx_t *sfx = sounds[count];
if (!sfx)
continue;
if (sfx->loadstate == SLS_NOTLOADED)
S_LoadSound(sfx);
if (sfx->loadstate != SLS_LOADED)
continue; //not ready yet
if (sc->ChannelUpdate)
{
for (c = NULL, j=DYNAMIC_FIRST; j < DYNAMIC_STOP ; j++)
{
if (sc->channel[j].entnum == entnums[count] && !sc->channel[j].entchannel && (sc->channel[j].flags & CF_AUTOSOUND))
{
c = &sc->channel[j];
break;
}
}
}
else
{
for (c = NULL, j=DYNAMIC_FIRST; j < DYNAMIC_STOP ; j++)
{
if (sc->channel[j].sfx == sfx && (sc->channel[j].flags & CF_AUTOSOUND))
{
c = &sc->channel[j];
break;
}
}
}
if (!c)
{
c = SND_PickChannel(sc, 0, 0);
if (!c)
continue;
c->flags = CF_AUTOSOUND|CF_FORCELOOP;
c->entnum = sc->ChannelUpdate?entnums[count]:0;
c->entchannel = 0;
c->dist_mult = 3 / sound_nominal_clip_dist;
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
SND_Spatialize(sc, c);
}
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, true);
}
}
}
}
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_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, true);
}
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, false);
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 (i >= DYNAMIC_STOP)
{
// 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)
{
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++;
}
Con_Printf ("----(%i+%i)----\n", active, mute);
}
#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;
S_LockMixer();
for (sc = sndcardinfo; sc; sc = sc->next)
S_UpdateCard(sc);
S_UnlockMixer();
}
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(void)
{ //plays a sound located around the player
int i;
char name[256];
sfx_t *sfx;
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));
sfx = S_PrecacheSound(name);
S_StartSound(0, -1, sfx, NULL, NULL, 1.0, 0.0, 0, 0, CF_NOSPACIALISE);
i++;
}
}
void S_PlayVol(void)
{
int i;
float vol;
char name[256];
sfx_t *sfx;
i = 1;
while (i<Cmd_Argc())
{
if (!Q_strrchr(Cmd_Argv(i), '.'))
{
Q_strncpy(name, Cmd_Argv(i), sizeof(name)-4);
Q_strcat(name, ".wav");
}
else
Q_strncpy(name, Cmd_Argv(i), sizeof(name));
sfx = S_PrecacheSound(name);
vol = Q_atof(Cmd_Argv(i+1));
S_StartSound(0, -1, sfx, NULL, NULL, vol, 0.0, 0, 0, CF_NOSPACIALISE);
i+=2;
}
}
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)*sc->width*(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",sc->width*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_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;
int width;
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->width = s->width;
}
return buf;
}
//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, int width, 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;
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_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;
}
s->sfx.decoder.buf = s;
s->sfx.decoder.decodedata = S_Raw_Locate;
s->sfx.loadstate = SLS_LOADED;
s->numchannels = channels;
s->width = width;
s->data = NULL;
s->length = 0;
s->id = sourceid;
s->inuse = true;
strcpy(s->sfx.name, "raw stream");
// Con_Printf("Added new raw stream\n");
}
S_LockMixer();
if (s->width != width || s->numchannels != channels)
{
s->width = width;
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) * s->width);
memcpy(newcache, (qbyte*)s->data + prepadl * (s->numchannels) * s->width, spare * (s->numchannels) * s->width);
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) * s->width);
SND_ResampleStream(data,
speed,
width,
channels,
samples,
outpos,
snd_speed,
s->width,
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], false);
break;
}
if (i == si->total_chans) //this one wasn't playing.
{
channel_t *c = SND_PickChannel(si, -1, 0);
if (c)
{
c->flags = CF_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, true);
}
}
}
S_UnlockMixer();
}
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