From df22ebd757a4e555276889275d03b836b7042ac2 Mon Sep 17 00:00:00 2001 From: Spoike Date: Fri, 31 May 2013 01:16:07 +0000 Subject: [PATCH] Tweaks for voicechat: Added cl_voip_test cvar, so you can hear/test yourself. Added potential support for opus. audio ducking autogain switched off by default. its nice for pure speech but shite for push-to-talk, voice activation, or indeed anything else. git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@4375 fc73d0e0-1445-4013-8a0c-d673dee63da5 --- engine/client/snd_dma.c | 838 +++++++++++++++++++++++++++++----------- engine/client/snd_mix.c | 57 +-- engine/client/sound.h | 7 +- 3 files changed, 625 insertions(+), 277 deletions(-) diff --git a/engine/client/snd_dma.c b/engine/client/snd_dma.c index 3ec31362..b8a1017d 100644 --- a/engine/client/snd_dma.c +++ b/engine/client/snd_dma.c @@ -120,13 +120,18 @@ cvar_t snd_driver = CVARAF( "s_driver", "", #ifdef VOICECHAT static void S_Voip_Play_Callback(cvar_t *var, char *oldval); cvar_t cl_voip_send = CVARD("cl_voip_send", "0", "Sends voice-over-ip data to the server whenever it is set"); +cvar_t cl_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 cl_voip_vad_threshhold = CVARD("cl_voip_vad_threshhold", "15", "This is the threshhold for voice-activation-detection when sending voip data"); cvar_t cl_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 cl_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"); -cvar_t cl_voip_showmeter = CVARAFD("cl_voip_showmeter", "1", NULL, CVAR_ARCHIVE, "Shows your speech volume above the hud. 0=hide, 1=show when transmitting, 2=ignore voice-activation disable"); +cvar_t cl_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 cl_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 cl_voip_play = CVARAFDC("cl_voip_play", "1", NULL, CVAR_ARCHIVE, "Enables voip playback.", S_Voip_Play_Callback); +cvar_t cl_voip_play = CVARAFDC("cl_voip_play", "1", NULL, CVAR_ARCHIVE, "Enables voip playback. Value is a volume scaler.", S_Voip_Play_Callback); +cvar_t cl_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 cl_voip_micamp = CVARAFDC("cl_voip_micamp", "2", NULL, CVAR_ARCHIVE, "Amplifies your microphone when using voip.", 0); +cvar_t cl_voip_codec = CVARAFDC("cl_voip_codec", "0", NULL, CVAR_ARCHIVE, "0: speex. 1: raw. 2: opus.", 0); +cvar_t cl_voip_noisefilter = CVARAFDC("cl_voip_noisefilter", "1", NULL, CVAR_ARCHIVE, "Enable the use of the noise cancelation filter, which also normalises microphone volume levels.", 0); +cvar_t cl_voip_autogain = CVARAFDC("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.", 0); #endif extern vfsfile_t *rawwritefile; @@ -217,25 +222,60 @@ void S_SoundInfo_f(void) #ifdef VOICECHAT #include #include + +enum +{ + VOIP_SPEEX = 0, //original supported codec + VOIP_RAW = 1, //support is not recommended. + VOIP_OPUS = 2, //supposed to be better than speex. + + VOIP_INVALID = 16 //not currently generating audio. +}; static struct { - qboolean inited; - qboolean loaded; - dllhandle_t *speexlib; - dllhandle_t *speexdsplib; + struct + { + qboolean inited; + qboolean loaded; + dllhandle_t *speexlib; - SpeexBits encbits; + SpeexBits encbits; + SpeexBits decbits[MAX_CLIENTS]; + + const SpeexMode *mode; + } 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; - SpeexPreprocessState *preproc; - unsigned int framesize; - unsigned int samplerate; + unsigned int encframesize; + unsigned int encsamplerate; - SpeexBits decbits[MAX_CLIENTS]; 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*/ - float decamp[MAX_CLIENTS]; /*amplify them by this*/ + 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*/ + float lastspoke_any; unsigned char capturebuf[32768]; /*pending data*/ unsigned int capturepos;/*amount of pending data*/ @@ -246,9 +286,51 @@ static struct unsigned int dumps; /*trigger a new generation thing after a bit*/ unsigned int keeps; /*for vad_delay*/ - snd_capture_driver_t *driver;/*capture driver's functions*/ - void *driverctx; /*capture driver context*/ -} s_speex; + snd_capture_driver_t *cdriver;/*capture driver's functions*/ + void *cdriverctx; /*capture driver context*/ +} s_voip; + +#define OPUS_APPLICATION_VOIP 2048 +#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 @@ -274,6 +356,7 @@ 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); @@ -282,6 +365,7 @@ 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); @@ -297,6 +381,7 @@ static dllfunction_t qspeexfuncs[] = {(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"}, @@ -305,6 +390,7 @@ static dllfunction_t qspeexfuncs[] = 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"}, @@ -315,79 +401,253 @@ static dllfunction_t qspeexdspfuncs[] = snd_capture_driver_t DSOUND_Capture; snd_capture_driver_t OSS_Capture; -static qboolean S_Speex_Init(void) +static qboolean S_SpeexDSP_Init(void) { - int i; - const SpeexMode *mode; - #ifndef SPEEX_STATIC - if (s_speex.inited) - return s_speex.loaded; - s_speex.inited = true; + if (s_voip.speexdsp.inited) + return s_voip.speexdsp.loaded; + s_voip.speexdsp.inited = true; - s_speex.speexlib = Sys_LoadLibrary("libspeex", qspeexfuncs); - if (!s_speex.speexlib) + + s_voip.speexdsp.speexdsplib = Sys_LoadLibrary("libspeexdsp", qspeexdspfuncs); + if (!s_voip.speexdsp.speexdsplib) { - Con_Printf("libspeex not found. Voice chat is not available.\n"); - return false; - } - - s_speex.speexdsplib = Sys_LoadLibrary("libspeexdsp", qspeexdspfuncs); - if (!s_speex.speexdsplib) - { - Con_Printf("libspeexdsp not found. Voice chat is not available.\n"); + Con_Printf("libspeexdsp not found. Your mic may be noisy.\n"); return false; } #endif - mode = qspeex_lib_get_mode(SPEEX_MODEID_NB); + 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; - qspeex_bits_init(&s_speex.encbits); - qspeex_bits_reset(&s_speex.encbits); - - s_speex.encoder = qspeex_encoder_init(mode); - - qspeex_encoder_ctl(s_speex.encoder, SPEEX_GET_FRAME_SIZE, &s_speex.framesize); - qspeex_encoder_ctl(s_speex.encoder, SPEEX_GET_SAMPLING_RATE, &s_speex.samplerate); - s_speex.samplerate = 11025; - qspeex_encoder_ctl(s_speex.encoder, SPEEX_SET_SAMPLING_RATE, &s_speex.samplerate); - - s_speex.preproc = qspeex_preprocess_state_init(s_speex.framesize, s_speex.samplerate); - - i = 1; - qspeex_preprocess_ctl(s_speex.preproc, SPEEX_PREPROCESS_SET_DENOISE, &i); - - i = 1; - qspeex_preprocess_ctl(s_speex.preproc, SPEEX_PREPROCESS_SET_AGC, &i); - - for (i = 0; i < MAX_CLIENTS; i++) + s_voip.speex.speexlib = Sys_LoadLibrary("libspeex", qspeexfuncs); + if (!s_voip.speex.speexlib) { - qspeex_bits_init(&s_speex.decbits[i]); - qspeex_bits_reset(&s_speex.decbits[i]); - s_speex.decoder[i] = qspeex_decoder_init(mode); - s_speex.decamp[i] = 1; + Con_Printf("libspeex not found. Voice chat is not available.\n"); + return false; } - s_speex.loaded = true; - return s_speex.loaded; +#endif + + s_voip.speex.mode = qspeex_lib_get_mode(SPEEX_MODEID_NB); + + 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; +} + +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, newseq, drops; + int r; + + if (sender >= MAX_CLIENTS) + return; + + decodesamps = 0; + newseq = 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: + qspeex_decoder_destroy(s_voip.decoder[sender]); + 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_SPEEX: + if (!S_Speex_Init()) + return; //speex not usable. + + s_voip.decsamplerate[sender] = 11025; + s_voip.decframesize[sender] = 160; + 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(s_voip.speex.mode); + 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]) + { + s_voip.decframesize[sender] = (sizeof(decodebuf) / sizeof(decodebuf[0])) / 2; //this is the maximum size in a single frame. + //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; + } + 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; + } + + 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, cl_voip_play.value); + decodesamps = 0; + } + switch(codec) + { + default: + bytes = 0; + break; + case VOIP_SPEEX: + if (s_voip.decseq[sender] != seq) + { + //tell speex about missing packets + qspeex_decode_int(s_voip.decoder[sender], NULL, decodebuf + decodesamps); + drops++; + s_voip.decseq[sender]++; + } + else + { + //yay, we got data + bytes--; + len = *start++; + if (bytes < len) + break; + qspeex_bits_read_from(&s_voip.speex.decbits[sender], start, len); + bytes -= len; + start += len; + qspeex_decode_int(s_voip.decoder[sender], &s_voip.speex.decbits[sender], decodebuf + decodesamps); + newseq++; + } + decodesamps += s_voip.decframesize[sender]; + break; + case VOIP_OPUS: + bytes--; + len = *start++; + if (bytes < len) + break; + +// len = bytes; + + if (s_voip.decseq[sender] != seq) + { + r = qopus_decode(s_voip.decoder[sender], NULL, len, decodebuf + decodesamps, sizeof(decodebuf)/sizeof(decodebuf[0]) - decodesamps, false); + drops++; + s_voip.decseq[sender]++; + } + else + { + r = qopus_decode(s_voip.decoder[sender], start, len, decodebuf + decodesamps, sizeof(decodebuf)/sizeof(decodebuf[0]) - decodesamps, false); + newseq++; + } + if (r > 0) + decodesamps += r; + + bytes -= len; + start += len; + + s_voip.decseq[sender] = seq + 1; + break; + } + } + s_voip.decseq[sender] += newseq; + + 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, cl_voip_play.value); } void S_Voip_Parse(void) { unsigned int sender; unsigned int bytes; - unsigned char data[1024], *start; - short decodebuf[1024]; - unsigned int decodesamps, len, newseq, drops; + unsigned char data[1024]; unsigned char seq, gen; - float amp = 1; - unsigned int i; + unsigned char codec; sender = MSG_ReadByte(); gen = MSG_ReadByte(); + codec = gen>>4; + gen &= 0x0f; seq = MSG_ReadByte(); bytes = MSG_ReadShort(); - if (bytes > sizeof(data) || !cl_voip_play.ival || !S_Speex_Init() || (gen & 0xf0)) + if (bytes > sizeof(data) || cl_voip_play.value <= 0) { MSG_ReadSkip(bytes); return; @@ -396,64 +656,16 @@ void S_Voip_Parse(void) sender %= MAX_CLIENTS; - amp = s_speex.decamp[sender]; + //if testing, don't get confused if the server is echoing voice too! + if (cl_voip_test.ival) + if (sender == cl.playernum[0]) + return; - decodesamps = 0; - newseq = 0; - drops = 0; - start = data; - - s_speex.lastspoke[sender] = realtime + 0.5; - if (s_speex.decgen[sender] != gen) - { - qspeex_bits_reset(&s_speex.decbits[sender]); - s_speex.decgen[sender] = gen; - s_speex.decseq[sender] = seq; - } - - while (bytes > 0) - { - if (decodesamps + s_speex.framesize > sizeof(decodebuf)/sizeof(decodebuf[0])) - { - S_RawAudio(sender, (qbyte*)decodebuf, s_speex.samplerate, decodesamps, 1, 2); - decodesamps = 0; - } - - if (s_speex.decseq[sender] != seq) - { - qspeex_decode_int(s_speex.decoder[sender], NULL, decodebuf + decodesamps); - s_speex.decseq[sender]++; - drops++; - } - else - { - bytes--; - len = *start++; - qspeex_bits_read_from(&s_speex.decbits[sender], start, len); - bytes -= len; - start += len; - qspeex_decode_int(s_speex.decoder[sender], &s_speex.decbits[sender], decodebuf + decodesamps); - newseq++; - } - if (amp != 1) - { - for (i = decodesamps; i < decodesamps+s_speex.framesize; i++) - decodebuf[i] *= amp; - } - decodesamps += s_speex.framesize; - } - s_speex.decseq[sender] += newseq; - - if (drops) - Con_DPrintf("%i dropped audio frames\n", drops); - - if (decodesamps > 0) - S_RawAudio(sender, (qbyte*)decodebuf, s_speex.samplerate, decodesamps, 1, 2); + S_Voip_Decode(sender, codec, gen, seq, bytes, data); } - void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf) { - unsigned char outbuf[1024]; + unsigned char outbuf[8192]; unsigned int outpos;//in bytes unsigned int encpos;//in bytes short *start; @@ -461,162 +673,316 @@ void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf) unsigned int i; unsigned int samps; float level, f; + int len; float micamp = cl_voip_micamp.value; qboolean voipsendenable = true; + int voipcodec = cl_voip_codec.ival; - /*if you're sending sound, you should be prepared to accept others yelling at you to shut up*/ - if (!cl_voip_play.ival) - voipsendenable = false; - if (!(cls.fteprotocolextensions2 & PEXT2_VOICECHAT)) - voipsendenable = false; - - if (!voipsendenable) + if (buf) { - if (s_speex.driver) + /*if you're sending sound, you should be prepared to accept others yelling at you to shut up*/ + if (cl_voip_play.value <= 0) + voipsendenable = false; + if (!(cls.fteprotocolextensions2 & PEXT2_VOICECHAT)) + voipsendenable = false; + } + else + voipsendenable = cl_voip_test.ival; + + voicevolumemod = s_voip.lastspoke_any > realtime?cl_voip_ducking.value:1; + + if (!voipsendenable || (voipcodec != s_voip.enccodec && s_voip.cdriver)) + { + if (s_voip.cdriver) { - if (s_speex.wantsend) - s_speex.driver->Stop(s_speex.driverctx); - s_speex.driver->Shutdown(s_speex.driverctx); - s_speex.driverctx = NULL; - s_speex.driver = NULL; + 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; } - return; + switch(s_voip.enccodec) + { + case VOIP_SPEEX: + break; + case VOIP_OPUS: + qopus_encoder_destroy(s_voip.encoder); + break; + } + s_voip.encoder = NULL; + s_voip.enccodec = VOIP_INVALID; + + if (!voipsendenable) + return; } voipsendenable = cl_voip_send.ival>0; - if (!s_speex.driver) + if (!s_voip.cdriver) { - s_speex.voiplevel = -1; + s_voip.voiplevel = -1; /*only init the first time capturing is requested*/ if (!voipsendenable) return; /*Add new drivers in order of priority*/ - if (!s_speex.driver || !s_speex.driver->Init) - s_speex.driver = &DSOUND_Capture; - if (!s_speex.driver || !s_speex.driver->Init) - s_speex.driver = &OSS_Capture; + if (!s_voip.cdriver || !s_voip.cdriver->Init) + s_voip.cdriver = &DSOUND_Capture; + if (!s_voip.cdriver || !s_voip.cdriver->Init) + s_voip.cdriver = &OSS_Capture; /*no way to capture audio, give up*/ - if (!s_speex.driver || !s_speex.driver->Init) + if (!s_voip.cdriver || !s_voip.cdriver->Init) return; - /*see if we can init speex...*/ - if (!S_Speex_Init()) - return; + /*see if we can init our encoding codec...*/ + switch(voipcodec) + { + case VOIP_SPEEX: + if (!S_Speex_Init()) + { + Con_Printf("Unable to use speex codec - not installed\n"); + return; + } - s_speex.driverctx = s_speex.driver->Init(s_speex.samplerate); + qspeex_bits_init(&s_voip.speex.encbits); + qspeex_bits_reset(&s_voip.speex.encbits); + s_voip.encoder = qspeex_encoder_init(s_voip.speex.mode); + 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); + s_voip.encsamplerate = 11025; + qspeex_encoder_ctl(s_voip.encoder, SPEEX_SET_SAMPLING_RATE, &s_voip.encsamplerate); + 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; + +// opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate_bps)); +// opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND)); +// opus_encoder_ctl(enc, OPUS_SET_VBR(use_vbr)); +// opus_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(cvbr)); +// opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity)); +// opus_encoder_ctl(enc, OPUS_SET_INBAND_FEC(use_inbandfec)); +// opus_encoder_ctl(enc, OPUS_SET_FORCE_CHANNELS(forcechannels)); +// opus_encoder_ctl(enc, OPUS_SET_DTX(use_dtx)); +// opus_encoder_ctl(enc, OPUS_SET_PACKET_LOSS_PERC(packet_loss_perc)); + +// opus_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&skip)); +// opus_encoder_ctl(enc, 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.cdriverctx = s_voip.cdriver->Init(s_voip.encsamplerate); + + if (!s_voip.cdriverctx) + Con_Printf("No microphone detected\n"); } /*couldn't init a driver?*/ - if (!s_speex.driverctx) + if (!s_voip.cdriverctx) { return; } - if (!voipsendenable && s_speex.wantsend) + if (!voipsendenable && s_voip.wantsend) { - s_speex.wantsend = false; - s_speex.capturepos += s_speex.driver->Update(s_speex.driverctx, (unsigned char*)s_speex.capturebuf + s_speex.capturepos, 1, sizeof(s_speex.capturebuf) - s_speex.capturepos); - s_speex.driver->Stop(s_speex.driverctx); + 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_speex.wantsend) + else if (voipsendenable && !s_voip.wantsend) { - s_speex.wantsend = true; - if (!s_speex.capturepos) + 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_speex.dumps = 0; - s_speex.generation++; - s_speex.encsequence = 0; - qspeex_bits_reset(&s_speex.encbits); + 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: + qspeex_bits_reset(&s_voip.speex.encbits); + break; + case VOIP_OPUS: + qopus_encoder_ctl(s_voip.encoder, OPUS_RESET_STATE); + break; + } } else { - s_speex.capturepos += s_speex.driver->Update(s_speex.driverctx, (unsigned char*)s_speex.capturebuf + s_speex.capturepos, 1, sizeof(s_speex.capturebuf) - s_speex.capturepos); + 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_speex.driver->Start(s_speex.driverctx); - - voicevolumemod = cl_voip_capturingvol.value; + s_voip.cdriver->Start(s_voip.cdriverctx); } - s_speex.capturepos += s_speex.driver->Update(s_speex.driverctx, (unsigned char*)s_speex.capturebuf + s_speex.capturepos, s_speex.framesize*2, sizeof(s_speex.capturebuf) - s_speex.capturepos); + if (s_voip.wantsend) + voicevolumemod = min(voicevolumemod, cl_voip_capturingvol.value); - if (!s_speex.wantsend && s_speex.capturepos < s_speex.framesize*2) + 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_speex.voiplevel = -1; - s_speex.capturepos = 0; - voicevolumemod = 1; + s_voip.voiplevel = -1; + s_voip.capturepos = 0; return; } - initseq = s_speex.encsequence; + initseq = s_voip.encsequence; level = 0; samps=0; - for (encpos = 0, outpos = 0; s_speex.capturepos-encpos >= s_speex.framesize*2 && sizeof(outbuf)-outpos > 64; s_speex.encsequence++) + //*2 for 16bit audio input. + for (encpos = 0, outpos = 0; s_voip.capturepos-encpos >= s_voip.encframesize*2 && sizeof(outbuf)-outpos > 64; s_voip.encsequence++) { - start = (short*)(s_speex.capturebuf + encpos); + start = (short*)(s_voip.capturebuf + encpos); - qspeex_preprocess_run(s_speex.preproc, start); + if (cl_voip_noisefilter.ival || cl_voip_autogain.ival) + { + if (!s_voip.speexdsp.preproc || cl_voip_noisefilter.modified || cl_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 = cl_voip_noisefilter.ival; + qspeex_preprocess_ctl(s_voip.speexdsp.preproc, SPEEX_PREPROCESS_SET_DENOISE, &i); + i = cl_voip_autogain.ival; + qspeex_preprocess_ctl(s_voip.speexdsp.preproc, SPEEX_PREPROCESS_SET_AGC, &i); - for (i = 0; i < s_speex.framesize; i++) + s_voip.speexdsp.curframesize = s_voip.encframesize; + s_voip.speexdsp.cursamplerate = s_voip.encsamplerate; + } + } + if (s_voip.speexdsp.preproc) + qspeex_preprocess_run(s_voip.speexdsp.preproc, start); + } + + for (i = 0; i < s_voip.encframesize; i++) { f = start[i] * micamp; start[i] = f; f = fabs(start[i]); level += f*f; } - samps+=s_speex.framesize; - qspeex_bits_reset(&s_speex.encbits); - qspeex_encode_int(s_speex.encoder, start, &s_speex.encbits); - outbuf[outpos] = qspeex_bits_write(&s_speex.encbits, outbuf+outpos+1, sizeof(outbuf) - (outpos+1)); - outpos += 1+outbuf[outpos]; - encpos += s_speex.framesize*2; + switch(s_voip.enccodec) + { + case VOIP_SPEEX: + 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; + break; + case VOIP_OPUS: + len = qopus_encode(s_voip.encoder, start, s_voip.encframesize, outbuf+(outpos+1), max(255, sizeof(outbuf) - (outpos+1))); + if (len == 1) //packet does not need to be transmitted if it returns 1, supposedly. crazyness. + len = 0; + else if (len > 0) + { + outbuf[outpos] = len; + outpos += 1+len; + } + else + { + //error! + Con_Printf("Opus encoding error: %i\n", len); + } + break; + default: + outbuf[outpos] = 0; + break; + } + samps+=s_voip.encframesize; + encpos += s_voip.encframesize*2; } if (samps) { float nl; nl = (3000*level) / (32767.0f*32767*samps); - s_speex.voiplevel = (s_speex.voiplevel*7 + nl)/8; - if (s_speex.voiplevel < cl_voip_vad_threshhold.ival && !(cl_voip_send.ival & 2)) + s_voip.voiplevel = (s_voip.voiplevel*7 + nl)/8; + if (s_voip.voiplevel < cl_voip_vad_threshhold.ival && !(cl_voip_send.ival & 2)) { /*try and dump it, it was too quiet, and they're not pressing +voip*/ - if (s_speex.keeps > samps) + if (s_voip.keeps > samps) { /*but not instantly*/ - s_speex.keeps -= samps; + s_voip.keeps -= samps; } else { outpos = 0; - s_speex.dumps += samps; - s_speex.keeps = 0; + s_voip.dumps += samps; + s_voip.keeps = 0; } } else - s_speex.keeps = s_speex.samplerate * cl_voip_vad_delay.value; + s_voip.keeps = s_voip.encsamplerate * cl_voip_vad_delay.value; if (outpos) { - if (s_speex.dumps > s_speex.samplerate/4) - s_speex.generation++; - s_speex.dumps = 0; + if (s_voip.dumps > s_voip.encsamplerate/4) + s_voip.generation++; + s_voip.dumps = 0; } } - if (outpos && buf->maxsize - buf->cursize >= outpos+4) + if (outpos && (!buf || buf->maxsize - buf->cursize >= outpos+4)) { - MSG_WriteByte(buf, clc); - MSG_WriteByte(buf, (s_speex.generation & 0x0f)); /*gonna leave that nibble clear here... in this version, the client will ignore packets with those bits set. can use them for codec or something*/ - MSG_WriteByte(buf, initseq); - MSG_WriteShort(buf, outpos); - SZ_Write(buf, outbuf, outpos); + if (buf) + { + MSG_WriteByte(buf, clc); + MSG_WriteByte(buf, (s_voip.enccodec<<4) | (s_voip.generation & 0x0f)); /*gonna leave that nibble clear here... in this version, the client will ignore packets with those bits set. can use them for codec or something*/ + MSG_WriteByte(buf, initseq); + MSG_WriteShort(buf, outpos); + SZ_Write(buf, outbuf, outpos); + } + + if (cl_voip_test.ival) + S_Voip_Decode(cl.playernum[0], s_voip.enccodec, s_voip.generation & 0x0f, initseq, 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.playernum[0]] = realtime + 0.5; } /*remove sent data*/ - memmove(s_speex.capturebuf, s_speex.capturebuf + encpos, s_speex.capturepos-encpos); - s_speex.capturepos -= encpos; + 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) { @@ -636,14 +1002,15 @@ static void S_Voip_f(void) if (!strcmp(Cmd_Argv(1), "maxgain")) { i = atoi(Cmd_Argv(2)); - qspeex_preprocess_ctl(s_speex.preproc, SPEEX_PREPROCESS_SET_AGC_MAX_GAIN, &i); + if (s_voip.speexdsp.preproc) + qspeex_preprocess_ctl(s_voip.speexdsp.preproc, SPEEX_PREPROCESS_SET_AGC_MAX_GAIN, &i); } } static void S_Voip_Play_Callback(cvar_t *var, char *oldval) { if (cls.fteprotocolextensions2 & PEXT2_VOICECHAT) { - if (var->ival) + if (var->value > 0) CL_SendClientCommand(true, "unmuteall"); else CL_SendClientCommand(true, "muteall"); @@ -655,17 +1022,41 @@ void S_Voip_MapChange(void) } int S_Voip_Loudness(qboolean ignorevad) { - if (s_speex.voiplevel > 100) + if (s_voip.voiplevel > 100) return 100; - if (!s_speex.driverctx || (!ignorevad && s_speex.dumps)) + if (!s_voip.cdriverctx || (!ignorevad && s_voip.dumps)) return -1; - return s_speex.voiplevel; + return s_voip.voiplevel; } qboolean S_Voip_Speaking(unsigned int plno) { if (plno >= MAX_CLIENTS) return false; - return s_speex.lastspoke[plno] > realtime; + 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(&cl_voip_send, "Voice Chat"); + Cvar_Register(&cl_voip_vad_threshhold, "Voice Chat"); + Cvar_Register(&cl_voip_vad_delay, "Voice Chat"); + Cvar_Register(&cl_voip_capturingvol, "Voice Chat"); + Cvar_Register(&cl_voip_showmeter, "Voice Chat"); + Cvar_Register(&cl_voip_play, "Voice Chat"); + Cvar_Register(&cl_voip_test, "Voice Chat"); + Cvar_Register(&cl_voip_ducking, "Voice Chat"); + Cvar_Register(&cl_voip_micamp, "Voice Chat"); + Cvar_Register(&cl_voip_codec, "Voice Chat"); + Cvar_Register(&cl_voip_noisefilter, "Voice Chat"); + Cvar_Register(&cl_voip_autogain, "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) @@ -1133,16 +1524,7 @@ void S_Init (void) Cvar_Register(&snd_playbackrate, "Sound controls"); #ifdef VOICECHAT - Cvar_Register(&cl_voip_send, "Voice Chat"); - Cvar_Register(&cl_voip_vad_threshhold, "Voice Chat"); - Cvar_Register(&cl_voip_vad_delay, "Voice Chat"); - Cvar_Register(&cl_voip_capturingvol, "Voice Chat"); - Cvar_Register(&cl_voip_showmeter, "Voice Chat"); - Cvar_Register(&cl_voip_play, "Voice Chat"); - Cvar_Register(&cl_voip_micamp, "Voice Chat"); - Cmd_AddCommand("+voip", S_Voip_Enable_f); - Cmd_AddCommand("-voip", S_Voip_Disable_f); - Cmd_AddCommand("voip", S_Voip_f); + S_Voip_Init(); #endif Cvar_Register(&snd_inactive, "Sound controls"); @@ -1443,14 +1825,25 @@ void SND_Spatialize(soundcardinfo_t *sc, channel_t *ch) vec_t dist; vec_t scale; vec3_t world_vec; - int i; + int i, v; // anything coming from the view entity will always be full volume - if (ch->entnum == -1 || ch->entnum == cl.playernum[0]+1) + if (ch->flags & CF_ABSVOLUME) { + v = ch->master_vol; for (i = 0; i < sc->sn.numchannels; i++) { - ch->vol[i] = ch->master_vol * (ruleset_allow_localvolume.value ? snd_playersoundvolume.value : 1); + ch->vol[i] = v; + } + return; + } + if (ch->entnum == -1 || ch->entnum == cl.playernum[0]+1) + { + v = ch->master_vol * (ruleset_allow_localvolume.value ? snd_playersoundvolume.value : 1) * volume.value * voicevolumemod; + v = bound(0, v, 255); + for (i = 0; i < sc->sn.numchannels; i++) + { + ch->vol[i] = v; } return; } @@ -1472,9 +1865,8 @@ void SND_Spatialize(soundcardinfo_t *sc, channel_t *ch) { scale = 1 + DotProduct(listener_vec, sc->speakerdir[i]); scale = (1.0 - dist) * scale * sc->dist[i]; - ch->vol[i] = (int) (ch->master_vol * scale); - if (ch->vol[i] < 0) - ch->vol[i] = 0; + v = ch->master_vol * scale * volume.value * voicevolumemod; + ch->vol[i] = bound(0, v, 255); } } @@ -1520,6 +1912,7 @@ static void S_StartSoundCard(soundcardinfo_t *sc, int entnum, int entchannel, sf { VectorCopy(origin, target_chan->origin); } + target_chan->flags = 0; target_chan->dist_mult = attenuation / sound_nominal_clip_dist; target_chan->master_vol = vol; target_chan->entnum = entnum; @@ -1820,12 +2213,14 @@ void S_UpdateAmbientSounds (soundcardinfo_t *sc) chan->sfx = newmusic; chan->rate = 1<pos = 0; - chan->vol[0] = chan->vol[1] = chan->vol[2] = chan->vol[3] = chan->vol[4] = chan->vol[5] = chan->master_vol = 100; } } } if (chan->sfx) - chan->vol[0] = chan->vol[1] = chan->vol[2] = chan->vol[3] = chan->vol[4] = chan->vol[5] = chan->master_vol = (255/volume.value)*bgmvolume.value; + { + chan->master_vol = 255; //bypasses volume cvar completely. + chan->vol[0] = chan->vol[1] = chan->vol[2] = chan->vol[3] = chan->vol[4] = chan->vol[5] = bound(0, chan->master_vol*bgmvolume.value*voicevolumemod, 255); + } } @@ -1878,7 +2273,7 @@ void S_UpdateAmbientSounds (soundcardinfo_t *sc) } chan->master_vol = level[ambient_channel]; - chan->vol[0] = chan->vol[1] = chan->vol[2] = chan->vol[3] = chan->vol[4] = chan->vol[5] = chan->master_vol; + 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) ^ (level[ambient_channel] == 0)); @@ -2318,7 +2713,7 @@ sfxcache_t *S_Raw_Locate(sfx_t *sfx, sfxcache_t *buf, int start, int length) } //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) +void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels, int width, float volume) { soundcardinfo_t *si; int i; @@ -2469,11 +2864,12 @@ void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels, if (i == si->total_chans) //this one wasn't playing. { channel_t *c = SND_PickChannel(si, -1, 0); + c->flags = CF_ABSVOLUME; c->entnum = -1; c->entchannel = 0; c->dist_mult = 0; c->looping = false; - c->master_vol = 255; + c->master_vol = 255 * volume; c->pos = 0; c->rate = 1<sfx = &s->sfx; diff --git a/engine/client/snd_mix.c b/engine/client/snd_mix.c index 7e688107..aa9c8dcd 100644 --- a/engine/client/snd_mix.c +++ b/engine/client/snd_mix.c @@ -36,7 +36,7 @@ void S_TransferPaintBuffer(soundcardinfo_t *sc, int endtime) unsigned int outlimit; int *p; int val; - int snd_vol; +// int snd_vol; short *pbuf; int i, numc; @@ -44,7 +44,7 @@ void S_TransferPaintBuffer(soundcardinfo_t *sc, int endtime) count = (endtime - sc->paintedtime) * sc->sn.numchannels; outlimit = sc->sn.samples; out_idx = (sc->paintedtime * sc->sn.numchannels) % outlimit; - snd_vol = (volume.value*voicevolumemod)*256; +// snd_vol = (volume.value*voicevolumemod)*256; numc = sc->sn.numchannels; pbuf = sc->Lock(sc, &out_idx); @@ -58,7 +58,7 @@ void S_TransferPaintBuffer(soundcardinfo_t *sc, int endtime) { for (i = 0; i < numc; i++) { - val = (*p * snd_vol) >> 8; + val = *p;// * snd_vol) >> 8; p++; if (val > 0x7fff) val = 0x7fff; @@ -78,7 +78,7 @@ void S_TransferPaintBuffer(soundcardinfo_t *sc, int endtime) { for (i = 0; i < numc; i++) { - val = (*p * snd_vol) >> 8; + val = *p;// * snd_vol) >> 8; p++; if (val > 0x7fff) val = 0x7fff; @@ -289,11 +289,6 @@ void SND_PaintChannelFrom8 (channel_t *ch, sfxcache_t *sc, int count) int i; unsigned int pos = ch->pos-(sc->soundoffset<vol[0] > 255) - ch->vol[0] = 255; - if (ch->vol[1] > 255) - ch->vol[1] = 255; - if (ch->rate != (1<data; @@ -324,11 +319,6 @@ void SND_PaintChannelFrom8Stereo (channel_t *ch, sfxcache_t *sc, int count) int i; unsigned int pos = ch->pos-(sc->soundoffset<vol[0] > 255) - ch->vol[0] = 255; - if (ch->vol[1] > 255) - ch->vol[1] = 255; - if (ch->rate != (1<data; @@ -356,15 +346,6 @@ void SND_PaintChannelFrom8_4Speaker (channel_t *ch, sfxcache_t *sc, int count) int i; unsigned int pos = ch->pos-(sc->soundoffset<vol[0] > 255) - ch->vol[0] = 255; - if (ch->vol[1] > 255) - ch->vol[1] = 255; - if (ch->vol[2] > 255) - ch->vol[2] = 255; - if (ch->vol[3] > 255) - ch->vol[3] = 255; - if (ch->rate != (1<pos-(sc->soundoffset<vol[0] > 255) - ch->vol[0] = 255; - if (ch->vol[1] > 255) - ch->vol[1] = 255; - if (ch->vol[2] > 255) - ch->vol[2] = 255; - if (ch->vol[3] > 255) - ch->vol[3] = 255; - if (ch->vol[4] > 255) - ch->vol[4] = 255; - if (ch->vol[5] > 255) - ch->vol[5] = 255; - if (ch->rate != (1<pos-(sc->soundoffset<vol[0] > 255) - ch->vol[0] = 255; - if (ch->vol[1] > 255) - ch->vol[1] = 255; - if (ch->vol[2] > 255) - ch->vol[2] = 255; - if (ch->vol[3] > 255) - ch->vol[3] = 255; - if (ch->vol[4] > 255) - ch->vol[4] = 255; - if (ch->vol[5] > 255) - ch->vol[5] = 255; - if (ch->vol[6] > 255) - ch->vol[6] = 255; - if (ch->vol[7] > 255) - ch->vol[7] = 255; - if (ch->rate != (1<>PITCHSHIFT)/KHZ*/ +#define CF_ABSVOLUME 1 // ignores volume cvar. typedef struct { sfx_t *sfx; // sfx number - int vol[MAXSOUNDCHANNELS]; // 0-255 volume -// int delay[MAXSOUNDCHANNELS]; + int vol[MAXSOUNDCHANNELS]; // volume, .8 fixed point. int start; // start time in global paintsamples int pos; // sample position in sfx, <0 means delay sound start (shifted up by 8) int rate; // 24.8 fixed point rate scaling + int flags; // cf_ flags int looping; // where to loop, -1 = no looping int entnum; // to allow overriding a specific sound int entchannel; //int audio_fd @@ -180,7 +181,7 @@ void S_SetUnderWater(qboolean underwater); void S_Restart_f (void); //plays streaming audio -void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels, int width); +void S_RawAudio(int sourceid, qbyte *data, int speed, int samples, int channels, int width, float volume); void CLVC_Poll (void);