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virtualx-engine/servers/audio/audio_mixer_sw.cpp

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GODOT IS OPEN SOURCE
2014-02-10 02:10:30 +01:00
/*************************************************************************/
/* audio_mixer_sw.cpp */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* http://www.godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2014 Juan Linietsky, Ariel Manzur. */
/* */
/* Permission is hereby granted, free of charge, to any person obtaining */
/* a copy of this software and associated documentation files (the */
/* "Software"), to deal in the Software without restriction, including */
/* without limitation the rights to use, copy, modify, merge, publish, */
/* distribute, sublicense, and/or sell copies of the Software, and to */
/* permit persons to whom the Software is furnished to do so, subject to */
/* the following conditions: */
/* */
/* The above copyright notice and this permission notice shall be */
/* included in all copies or substantial portions of the Software. */
/* */
/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
/*************************************************************************/
#include "audio_mixer_sw.h"
#include "print_string.h"
#include "os/os.h"
//TODO implement FAST_AUDIO macro
#ifdef FAST_AUDIO
#define NO_REVERB
#endif
template<class Depth,bool is_stereo,bool use_filter,bool use_fx,AudioMixerSW::InterpolationType type,AudioMixerSW::MixChannels mix_mode>
void AudioMixerSW::do_resample(const Depth* p_src, int32_t *p_dst, ResamplerState *p_state) {
// this function will be compiled branchless by any decent compiler
int32_t final,final_r,next,next_r;
int32_t *reverb_dst = p_state->reverb_buffer;
while (p_state->amount--) {
int32_t pos=p_state->pos >> MIX_FRAC_BITS;
if (is_stereo)
pos<<=1;
final=p_src[pos];
if (is_stereo)
final_r=p_src[pos+1];
if (sizeof(Depth)==1) { /* conditions will not exist anymore when compiled! */
final<<=8;
if (is_stereo)
final_r<<=8;
}
if (type==INTERPOLATION_LINEAR) {
if (is_stereo) {
next=p_src[pos+2];
next_r=p_src[pos+3];
} else {
next=p_src[pos+1];
}
if (sizeof(Depth)==1) {
next<<=8;
if (is_stereo)
next_r<<=8;
}
int32_t frac=int32_t(p_state->pos&MIX_FRAC_MASK);
final=final+((next-final)*frac >> MIX_FRAC_BITS);
if (is_stereo)
final_r=final_r+((next_r-final_r)*frac >> MIX_FRAC_BITS);
}
if (use_filter) {
Channel::Mix::Filter *f = p_state->filter_l;
float finalf=final;
float pre = finalf;
finalf = ((finalf*p_state->coefs.b0) + (f->hb[0]*p_state->coefs.b1) + (f->hb[1]*p_state->coefs.b2) + (f->ha[0]*p_state->coefs.a1) + (f->ha[1]*p_state->coefs.a2)
);
f->ha[1]=f->ha[0];
f->hb[1]=f->hb[0];
f->hb[0]=pre;
f->ha[0]=finalf;
final=Math::fast_ftoi(finalf);
if (is_stereo) {
f = p_state->filter_r;
finalf=final_r;
pre = finalf;
finalf = ((finalf*p_state->coefs.b0) + (f->hb[0]*p_state->coefs.b1) + (f->hb[1]*p_state->coefs.b2) + (f->ha[0]*p_state->coefs.a1) + (f->ha[1]*p_state->coefs.a2)
);
f->ha[1]=f->ha[0];
f->hb[1]=f->hb[0];
f->hb[0]=pre;
f->ha[0]=finalf;
final_r=Math::fast_ftoi(finalf);
}
p_state->coefs.b0+=p_state->coefs_inc.b0;
p_state->coefs.b1+=p_state->coefs_inc.b1;
p_state->coefs.b2+=p_state->coefs_inc.b2;
p_state->coefs.a1+=p_state->coefs_inc.a1;
p_state->coefs.a2+=p_state->coefs_inc.a2;
}
if (!is_stereo) {
final_r=final; //copy to right channel if stereo
}
//convert back to 24 bits and mix to buffers
if (mix_mode==MIX_STEREO) {
*p_dst++ +=(final*(p_state->vol[0]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
*p_dst++ +=(final_r*(p_state->vol[1]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
p_state->vol[0]+=p_state->vol_inc[0];
p_state->vol[1]+=p_state->vol_inc[1];
if (use_fx) {
*reverb_dst++ +=(final*(p_state->reverb_vol[0]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
*reverb_dst++ +=(final_r*(p_state->reverb_vol[1]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
p_state->reverb_vol[0]+=p_state->reverb_vol_inc[0];
p_state->reverb_vol[1]+=p_state->reverb_vol_inc[1];
}
} else if (mix_mode==MIX_QUAD) {
*p_dst++ +=(final*(p_state->vol[0]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
*p_dst++ +=(final_r*(p_state->vol[1]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
*p_dst++ +=(final*(p_state->vol[2]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
*p_dst++ +=(final_r*(p_state->vol[3]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
p_state->vol[0]+=p_state->vol_inc[0];
p_state->vol[1]+=p_state->vol_inc[1];
p_state->vol[2]+=p_state->vol_inc[2];
p_state->vol[3]+=p_state->vol_inc[3];
if (use_fx) {
*reverb_dst++ +=(final*(p_state->reverb_vol[0]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
*reverb_dst++ +=(final_r*(p_state->reverb_vol[1]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
*reverb_dst++ +=(final*(p_state->reverb_vol[2]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
*reverb_dst++ +=(final_r*(p_state->reverb_vol[3]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
p_state->reverb_vol[0]+=p_state->reverb_vol_inc[0];
p_state->reverb_vol[1]+=p_state->reverb_vol_inc[1];
p_state->reverb_vol[2]+=p_state->reverb_vol_inc[2];
p_state->reverb_vol[3]+=p_state->reverb_vol_inc[3];
}
}
p_state->pos+=p_state->increment;
}
}
void AudioMixerSW::mix_channel(Channel& c) {
if (!sample_manager->is_sample(c.sample)) {
// sample is gone!
c.active=false;
return;
}
/* some 64-bit fixed point precaches */
int64_t loop_begin_fp=((int64_t)sample_manager->sample_get_loop_begin(c.sample) << MIX_FRAC_BITS);
int64_t loop_end_fp=((int64_t)sample_manager->sample_get_loop_end(c.sample) << MIX_FRAC_BITS);
int64_t length_fp=((int64_t)sample_manager->sample_get_length(c.sample) << MIX_FRAC_BITS);
int64_t begin_limit=(sample_manager->sample_get_loop_format(c.sample)!=AS::SAMPLE_LOOP_NONE)?loop_begin_fp:0;
int64_t end_limit=(sample_manager->sample_get_loop_format(c.sample)!=AS::SAMPLE_LOOP_NONE)?loop_end_fp:length_fp;
bool is_stereo=sample_manager->sample_is_stereo(c.sample);
int32_t todo=mix_chunk_size;
// int mixed=0;
bool use_filter=false;
ResamplerState rstate;
/* compute voume ramps, increment, etc */
for(int i=0;i<mix_channels;i++) {
c.mix.old_vol[i]=c.mix.vol[i];
c.mix.old_reverb_vol[i]=c.mix.reverb_vol[i];
c.mix.old_chorus_vol[i]=c.mix.chorus_vol[i];
}
float vol = c.vol*channel_nrg;
float reverb_vol = c.reverb_send*channel_nrg;
float chorus_vol = c.chorus_send*channel_nrg;
if (mix_channels==2) {
//stereo pan
float pan = c.pan * 0.5 + 0.5;
float panv[2]={
(1.0 - pan)*(1<<MIX_VOL_FRAC_BITS),
(pan)*(1<<MIX_VOL_FRAC_BITS)
};
for(int i=0;i<2;i++) {
c.mix.vol[i]=Math::fast_ftoi(vol*panv[i]);
c.mix.reverb_vol[i]=Math::fast_ftoi(reverb_vol*panv[i]);
c.mix.chorus_vol[i]=Math::fast_ftoi(chorus_vol*panv[i]);
}
} else {
//qudra pan
float panx = c.pan * 0.5 + 0.5;
float pany = c.depth * 0.5 + 0.5;
// with this model every speaker plays at 0.25 energy at the center.. i'm not sure if it's correct but it seems to be balanced
float panv[4]={
(1.0-pany)*(1.0-panx)*(1<<MIX_VOL_FRAC_BITS),
(1.0-pany)*( panx)*(1<<MIX_VOL_FRAC_BITS),
( pany)*(1.0-panx)*(1<<MIX_VOL_FRAC_BITS),
( pany)*( panx)*(1<<MIX_VOL_FRAC_BITS)
};
for(int i=0;i<4;i++) {
c.mix.vol[i]=Math::fast_ftoi(vol*panv[i]);
c.mix.reverb_vol[i]=Math::fast_ftoi(reverb_vol*panv[i]);
c.mix.chorus_vol[i]=Math::fast_ftoi(chorus_vol*panv[i]);
}
}
if (c.first_mix) { // avoid ramp up
for(int i=0;i<mix_channels;i++) {
c.mix.old_vol[i]=c.mix.vol[i];
c.mix.old_reverb_vol[i]=c.mix.reverb_vol[i];
c.mix.old_chorus_vol[i]=c.mix.chorus_vol[i];
}
c.first_mix=false;
}
Channel::Filter::Coefs filter_coefs;
Channel::Filter::Coefs filter_inc;
if (c.filter.type!=AudioMixer::FILTER_NONE) {
filter_coefs=c.filter.old_coefs;
filter_inc.b0=(c.filter.coefs.b0-filter_coefs.b0)/(1<<mix_chunk_bits);
filter_inc.b1=(c.filter.coefs.b1-filter_coefs.b1)/(1<<mix_chunk_bits);
filter_inc.b2=(c.filter.coefs.b2-filter_coefs.b2)/(1<<mix_chunk_bits);
filter_inc.a1=(c.filter.coefs.a1-filter_coefs.a1)/(1<<mix_chunk_bits);
filter_inc.a2=(c.filter.coefs.a2-filter_coefs.a2)/(1<<mix_chunk_bits);
use_filter=true;
}
if (c.mix.increment>0)
c.mix.increment=((int64_t)c.speed<<MIX_FRAC_BITS)/mix_rate;
else
c.mix.increment=-((int64_t)c.speed<<MIX_FRAC_BITS)/mix_rate;
//volume ramp
for(int i=0;i<mix_channels;i++) {
rstate.vol_inc[i]=((c.mix.vol[i]-c.mix.old_vol[i])<<MIX_VOLRAMP_FRAC_BITS)>>mix_chunk_bits;
rstate.vol[i]=c.mix.old_vol[i]<<MIX_VOLRAMP_FRAC_BITS;
rstate.reverb_vol_inc[i]=((c.mix.reverb_vol[i]-c.mix.old_reverb_vol[i])<<MIX_VOLRAMP_FRAC_BITS)>>mix_chunk_bits;
rstate.reverb_vol[i]=c.mix.old_reverb_vol[i]<<MIX_VOLRAMP_FRAC_BITS;
rstate.chorus_vol_inc[i]=((c.mix.chorus_vol[i]-c.mix.old_chorus_vol[i])<<MIX_VOLRAMP_FRAC_BITS)>>mix_chunk_bits;
rstate.chorus_vol[i]=c.mix.old_chorus_vol[i]<<MIX_VOLRAMP_FRAC_BITS;
}
//looping
AS::SampleLoopFormat loop_format=sample_manager->sample_get_loop_format(c.sample);
AS::SampleFormat format=sample_manager->sample_get_format(c.sample);
bool use_fx=fx_enabled && (c.mix.old_reverb_vol || c.mix.reverb_vol || c.mix.old_chorus_vol || c.mix.chorus_vol );
/* audio data */
const void *data=sample_manager->sample_get_data_ptr(c.sample);
int32_t *dst_buff=mix_buffer;
#ifndef NO_REVERB
rstate.reverb_buffer=reverb_state[c.reverb_room].buffer;
#endif
/* @TODO validar loops al registrar? */
rstate.coefs=filter_coefs;
rstate.coefs_inc=filter_inc;
rstate.filter_l=&c.mix.filter_l;
rstate.filter_r=&c.mix.filter_r;
while (todo>0) {
int64_t limit=0;
int32_t target=0,aux=0;
/** LOOP CHECKING **/
if ( c.mix.increment < 0 ) {
/* going backwards */
if ( loop_format!=AS::SAMPLE_LOOP_NONE && c.mix.offset < loop_begin_fp ) {
/* loopstart reached */
if ( loop_format==AS::SAMPLE_LOOP_PING_PONG ) {
/* bounce ping pong */
c.mix.offset= loop_begin_fp + ( loop_begin_fp-c.mix.offset );
c.mix.increment=-c.mix.increment;
} else {
/* go to loop-end */
c.mix.offset=loop_end_fp-(loop_begin_fp-c.mix.offset);
}
} else {
/* check for sample not reaching begining */
if(c.mix.offset < 0) {
c.active=false;
break;
}
}
} else {
/* going forward */
if( loop_format!=AS::SAMPLE_LOOP_NONE && c.mix.offset >= loop_end_fp ) {
/* loopend reached */
if ( loop_format==AS::SAMPLE_LOOP_PING_PONG ) {
/* bounce ping pong */
c.mix.offset=loop_end_fp-(c.mix.offset-loop_end_fp);
c.mix.increment=-c.mix.increment;
} else {
/* go to loop-begin */
c.mix.offset=loop_begin_fp+(c.mix.offset-loop_end_fp);
}
} else {
/* no loop, check for end of sample */
if(c.mix.offset >= length_fp) {
c.active=false;
break;
}
}
}
/** MIXCOUNT COMPUTING **/
/* next possible limit (looppoints or sample begin/end */
limit=(c.mix.increment < 0) ?begin_limit:end_limit;
/* compute what is shorter, the todo or the limit? */
aux=(limit-c.mix.offset)/c.mix.increment+1;
target=(aux<todo)?aux:todo; /* mix target is the shorter buffer */
/* check just in case */
if ( target<=0 ) {
c.active=false;
break;
}
todo-=target;
int32_t offset=c.mix.offset&mix_chunk_mask; /* strip integer */
c.mix.offset-=offset;
rstate.increment=c.mix.increment;
rstate.amount=target;
rstate.pos=offset;
/* Macros to call the resample function for all possibilities, creating a dedicated-non branchy function call for each thanks to template magic*/
#define CALL_RESAMPLE_FUNC( m_depth, m_stereo, m_use_filter, m_use_fx, m_interp, m_mode)\
do_resample<m_depth,m_stereo,m_use_filter,m_use_fx,m_interp, m_mode>(\
src_ptr,\
dst_buff,&rstate);
#define CALL_RESAMPLE_INTERP( m_depth, m_stereo, m_use_filter, m_use_fx, m_interp, m_mode)\
if(m_interp==INTERPOLATION_RAW) {\
CALL_RESAMPLE_FUNC(m_depth,m_stereo,m_use_filter,m_use_fx,INTERPOLATION_RAW,m_mode);\
} else if(m_interp==INTERPOLATION_LINEAR) {\
CALL_RESAMPLE_FUNC(m_depth,m_stereo,m_use_filter,m_use_fx,INTERPOLATION_LINEAR,m_mode);\
} else if(m_interp==INTERPOLATION_CUBIC) {\
CALL_RESAMPLE_FUNC(m_depth,m_stereo,m_use_filter,m_use_fx,INTERPOLATION_CUBIC,m_mode);\
}\
#define CALL_RESAMPLE_FX( m_depth, m_stereo, m_use_filter, m_use_fx, m_interp, m_mode)\
if(m_use_fx) {\
CALL_RESAMPLE_INTERP(m_depth,m_stereo,m_use_filter,true,m_interp, m_mode);\
} else {\
CALL_RESAMPLE_INTERP(m_depth,m_stereo,m_use_filter,false,m_interp, m_mode);\
}\
#define CALL_RESAMPLE_FILTER( m_depth, m_stereo, m_use_filter, m_use_fx, m_interp, m_mode)\
if(m_use_filter) {\
CALL_RESAMPLE_FX(m_depth,m_stereo,true,m_use_fx,m_interp, m_mode);\
} else {\
CALL_RESAMPLE_FX(m_depth,m_stereo,false,m_use_fx,m_interp, m_mode);\
}\
#define CALL_RESAMPLE_STEREO( m_depth, m_stereo, m_use_filter, m_use_fx, m_interp, m_mode)\
if(m_stereo) {\
CALL_RESAMPLE_FILTER(m_depth,true,m_use_filter,m_use_fx,m_interp, m_mode);\
} else {\
CALL_RESAMPLE_FILTER(m_depth,false,m_use_filter,m_use_fx,m_interp, m_mode);\
}\
#define CALL_RESAMPLE_MODE( m_depth, m_stereo, m_use_filter, m_use_fx, m_interp, m_mode)\
if(m_mode==MIX_STEREO) {\
CALL_RESAMPLE_STEREO(m_depth,m_stereo,m_use_filter,m_use_fx,m_interp, MIX_STEREO);\
} else {\
CALL_RESAMPLE_STEREO(m_depth,m_stereo,m_use_filter,m_use_fx,m_interp, MIX_QUAD);\
}\
if (format==AS::SAMPLE_FORMAT_PCM8) {
int8_t *src_ptr = &((int8_t*)data)[(c.mix.offset >> MIX_FRAC_BITS)<<(is_stereo?1:0) ];
CALL_RESAMPLE_MODE(int8_t,is_stereo,use_filter,use_fx,interpolation_type,mix_channels);
} else if (format==AS::SAMPLE_FORMAT_PCM16) {
int16_t *src_ptr = &((int16_t*)data)[(c.mix.offset >> MIX_FRAC_BITS)<<(is_stereo?1:0) ];
CALL_RESAMPLE_MODE(int16_t,is_stereo,use_filter,use_fx,interpolation_type,mix_channels);
}
c.mix.offset+=rstate.pos;
dst_buff+=target*2;
}
c.filter.old_coefs=c.filter.coefs;
}
void AudioMixerSW::mix_chunk() {
ERR_FAIL_COND(mix_chunk_left);
inside_mix=true;
// emit tick in usecs
for (int i=0;i<mix_chunk_size*mix_channels;i++) {
mix_buffer[i]=0;
}
#ifndef NO_REVERB
for(int i=0;i<max_reverbs;i++)
reverb_state[i].used_in_chunk=false;
#endif
audio_mixer_chunk_call(mix_chunk_size);
int ac=0;
for (int i=0;i<MAX_CHANNELS;i++) {
if (!channels[i].active)
continue;
ac++;
/* process volume */
Channel&c=channels[i];
#ifndef NO_REVERB
bool has_reverb = c.reverb_send>CMP_EPSILON && fx_enabled;
if (has_reverb || c.had_prev_reverb) {
if (!reverb_state[c.reverb_room].used_in_chunk) {
//zero the room
int32_t *buff = reverb_state[c.reverb_room].buffer;
int len = mix_chunk_size*mix_channels;
for (int j=0;j<len;j++) {
buff[j]=0; // buffer in use, clear it for appending
}
reverb_state[c.reverb_room].used_in_chunk=true;
}
}
#else
bool has_reverb = false;
#endif
bool has_chorus = c.chorus_send>CMP_EPSILON && fx_enabled;
mix_channel(c);
c.had_prev_reverb=has_reverb;
c.had_prev_chorus=has_chorus;
}
//process reverb
#ifndef NO_REVERB
if (fx_enabled) {
for(int i=0;i<max_reverbs;i++) {
if (!reverb_state[i].enabled && !reverb_state[i].used_in_chunk)
continue; //this reverb is not in use
int32_t *src=NULL;
if (reverb_state[i].used_in_chunk)
src=reverb_state[i].buffer;
else
src=zero_buffer;
bool in_use=false;
int passes=mix_channels/2;
for(int j=0;j<passes;j++) {
if (reverb_state[i].reverb[j].process((int*)&src[j*2],(int*)&mix_buffer[j*2],mix_chunk_size,passes))
in_use=true;
}
if (in_use) {
reverb_state[i].enabled=true;
reverb_state[i].frames_idle=0;
//copy data over
} else {
reverb_state[i].frames_idle+=mix_chunk_size;
if (false) { // go idle because too many frames passed
//disable this reverb, as nothing important happened on it
reverb_state[i].enabled=false;
reverb_state[i].frames_idle=0;
}
}
}
}
#endif
mix_chunk_left=mix_chunk_size;
inside_mix=false;
}
int AudioMixerSW::mix(int32_t *p_buffer,int p_frames) {
int todo=p_frames;
int mixes=0;
while(todo) {
if (!mix_chunk_left) {
if (step_callback)
step_callback(step_udata);
mix_chunk();
mixes++;
}
int to_mix=MIN(mix_chunk_left,todo);
int from=mix_chunk_size-mix_chunk_left;
for (int i=0;i<to_mix*2;i++) {
(*p_buffer++)=mix_buffer[from*2+i];
}
mix_chunk_left-=to_mix;
todo-=to_mix;
}
return mixes;
}
uint64_t AudioMixerSW::get_step_usecs() const {
double mct = (1<<mix_chunk_bits)/double(mix_rate);
return mct*1000000.0;
}
int AudioMixerSW::_get_channel(ChannelID p_channel) const {
if (p_channel<0) {
return -1;
}
int idx=p_channel%MAX_CHANNELS;
int check=p_channel/MAX_CHANNELS;
ERR_FAIL_INDEX_V(idx,MAX_CHANNELS,-1);
if (channels[idx].check!=check) {
return -1;
}
if (!channels[idx].active) {
return -1;
}
return idx;
}
AudioMixer::ChannelID AudioMixerSW::channel_alloc(RID p_sample) {
ERR_FAIL_COND_V( !sample_manager->is_sample(p_sample), INVALID_CHANNEL );
int index=-1;
for (int i=0;i<MAX_CHANNELS;i++) {
if (!channels[i].active) {
index=i;
break;
}
}
if (index==-1)
return INVALID_CHANNEL;
Channel &c=channels[index];
// init variables
c.sample=p_sample;
c.vol=1;
c.pan=0;
c.depth=0;
c.height=0;
c.chorus_send=0;
c.reverb_send=0;
c.reverb_room=REVERB_HALL;
c.positional=false;
c.filter.type=FILTER_NONE;
c.speed=sample_manager->sample_get_mix_rate(p_sample);
c.active=true;
c.check=channel_id_count++;
c.first_mix=true;
// init mix variables
c.mix.offset=0;
c.mix.increment=1;
//zero everything when this errors
for(int i=0;i<4;i++) {
c.mix.old_vol[i]=0;
c.mix.old_reverb_vol[i]=0;
c.mix.old_chorus_vol[i]=0;
}
c.had_prev_chorus=false;
c.had_prev_reverb=false;
c.had_prev_vol=false;
ChannelID ret_id = index+c.check*MAX_CHANNELS;
return ret_id;
}
void AudioMixerSW::channel_set_volume(ChannelID p_channel, float p_gain) {
if (p_gain>3) // avoid gain going too high
p_gain=3;
if (p_gain<0)
p_gain=0;
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return;
Channel &c = channels[chan];
//Math::exp( p_db * 0.11512925464970228420089957273422 );
c.vol=p_gain;
}
void AudioMixerSW::channel_set_pan(ChannelID p_channel, float p_pan, float p_depth,float p_height) {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return;
Channel &c = channels[chan];
c.pan=p_pan;
c.depth=p_depth;
c.height=p_height;
}
void AudioMixerSW::channel_set_filter(ChannelID p_channel, FilterType p_type, float p_cutoff, float p_resonance, float p_gain) {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return;
Channel &c = channels[chan];
if (c.filter.type==p_type && c.filter.cutoff==p_cutoff && c.filter.resonance==p_resonance && c.filter.gain==p_gain)
return; //bye
bool type_changed = p_type!=c.filter.type;
c.filter.type=p_type;
c.filter.cutoff=p_cutoff;
c.filter.resonance=p_resonance;
c.filter.gain=p_gain;
AudioFilterSW filter;
switch(p_type) {
case FILTER_NONE: {
return; //do nothing else
} break;
case FILTER_LOWPASS: {
filter.set_mode(AudioFilterSW::LOWPASS);
} break;
case FILTER_BANDPASS: {
filter.set_mode(AudioFilterSW::BANDPASS);
} break;
case FILTER_HIPASS: {
filter.set_mode(AudioFilterSW::HIGHPASS);
} break;
case FILTER_NOTCH: {
filter.set_mode(AudioFilterSW::NOTCH);
} break;
case FILTER_PEAK: {
filter.set_mode(AudioFilterSW::PEAK);
} break;
case FILTER_BANDLIMIT: {
filter.set_mode(AudioFilterSW::BANDLIMIT);
} break;
case FILTER_LOW_SHELF: {
filter.set_mode(AudioFilterSW::LOWSHELF);
} break;
case FILTER_HIGH_SHELF: {
filter.set_mode(AudioFilterSW::HIGHSHELF);
} break;
}
filter.set_cutoff(p_cutoff);
filter.set_resonance(p_resonance);
filter.set_gain(p_gain);
filter.set_sampling_rate(mix_rate);
filter.set_stages(1);
AudioFilterSW::Coeffs coefs;
filter.prepare_coefficients(&coefs);
if (!type_changed)
c.filter.old_coefs=c.filter.coefs;
c.filter.coefs.b0=coefs.b0;
c.filter.coefs.b1=coefs.b1;
c.filter.coefs.b2=coefs.b2;
c.filter.coefs.a1=coefs.a1;
c.filter.coefs.a2=coefs.a2;
if (type_changed) {
//type changed reset filter
c.filter.old_coefs=c.filter.coefs;
c.mix.filter_l.ha[0]=0;
c.mix.filter_l.ha[1]=0;
c.mix.filter_l.hb[0]=0;
c.mix.filter_l.hb[1]=0;
c.mix.filter_r.ha[0]=0;
c.mix.filter_r.ha[1]=0;
c.mix.filter_r.hb[0]=0;
c.mix.filter_r.hb[1]=0;
}
}
void AudioMixerSW::channel_set_chorus(ChannelID p_channel, float p_chorus ) {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return;
Channel &c = channels[chan];
c.chorus_send=p_chorus;
}
void AudioMixerSW::channel_set_reverb(ChannelID p_channel, ReverbRoomType p_room_type, float p_reverb) {
ERR_FAIL_INDEX(p_room_type,MAX_REVERBS);
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return;
Channel &c = channels[chan];
c.reverb_room=p_room_type;
c.reverb_send=p_reverb;
}
void AudioMixerSW::channel_set_mix_rate(ChannelID p_channel, int p_mix_rate) {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return;
Channel &c = channels[chan];
c.speed=p_mix_rate;
}
void AudioMixerSW::channel_set_positional(ChannelID p_channel, bool p_positional) {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return;
Channel &c = channels[chan];
c.positional=p_positional;
}
float AudioMixerSW::channel_get_volume(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return 0;
const Channel &c = channels[chan];
//Math::log( c.vol ) * 8.6858896380650365530225783783321;
return c.vol;
}
float AudioMixerSW::channel_get_pan(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return 0;
const Channel &c = channels[chan];
return c.pan;
}
float AudioMixerSW::channel_get_pan_depth(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return 0;
const Channel &c = channels[chan];
return c.depth;
}
float AudioMixerSW::channel_get_pan_height(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return 0;
const Channel &c = channels[chan];
return c.height;
}
AudioMixer::FilterType AudioMixerSW::channel_get_filter_type(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return FILTER_NONE;
const Channel &c = channels[chan];
return c.filter.type;
}
float AudioMixerSW::channel_get_filter_cutoff(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return 0;
const Channel &c = channels[chan];
return c.filter.cutoff;
}
float AudioMixerSW::channel_get_filter_resonance(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return 0;
const Channel &c = channels[chan];
return c.filter.resonance;
}
float AudioMixerSW::channel_get_filter_gain(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return 0;
const Channel &c = channels[chan];
return c.filter.gain;
}
float AudioMixerSW::channel_get_chorus(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return 0;
const Channel &c = channels[chan];
return c.chorus_send;
}
AudioMixer::ReverbRoomType AudioMixerSW::channel_get_reverb_type(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return REVERB_HALL;
const Channel &c = channels[chan];
return c.reverb_room;
}
float AudioMixerSW::channel_get_reverb(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return 0;
const Channel &c = channels[chan];
return c.reverb_send;
}
int AudioMixerSW::channel_get_mix_rate(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return 0;
const Channel &c = channels[chan];
return c.speed;
}
bool AudioMixerSW::channel_is_positional(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return false;
const Channel &c = channels[chan];
return c.positional;
}
bool AudioMixerSW::channel_is_valid(ChannelID p_channel) const {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return false;
return channels[chan].active;
}
void AudioMixerSW::channel_free(ChannelID p_channel) {
int chan = _get_channel(p_channel);
if (chan<0 || chan >=MAX_CHANNELS)
return;
Channel &c=channels[chan];
if (!c.active)
return;
bool has_vol=false;
for(int i=0;i<mix_channels;i++) {
if (c.mix.vol[i])
has_vol=true;
if (c.mix.reverb_vol[i])
has_vol=true;
if (c.mix.chorus_vol[i])
has_vol=true;
}
if (c.active && has_vol && inside_mix) {
// drive voice to zero, and run a chunk, the VRAMP will fade it good
c.vol=0;
c.reverb_send=0;
c.chorus_send=0;
mix_channel(c);
}
/* @TODO RAMP DOWN ON STOP */
c.active=false;
}
AudioMixerSW::AudioMixerSW(SampleManagerSW *p_sample_manager,int p_desired_latency_ms,int p_mix_rate,MixChannels p_mix_channels,bool p_use_fx,InterpolationType p_interp,MixStepCallback p_step_callback,void *p_step_udata) {
if (OS::get_singleton()->is_stdout_verbose()) {
print_line("AudioServerSW Params: ");
print_line(" -mix chans: "+itos(p_mix_channels));
print_line(" -mix rate: "+itos(p_mix_rate));
print_line(" -latency: "+itos(p_desired_latency_ms));
print_line(" -fx: "+itos(p_use_fx));
print_line(" -interp: "+itos(p_interp));
}
sample_manager=p_sample_manager;
mix_channels=p_mix_channels;
mix_rate=p_mix_rate;
step_callback=p_step_callback;
step_udata=p_step_udata;
mix_chunk_bits=nearest_shift( p_desired_latency_ms * p_mix_rate / 1000 );
mix_chunk_size=(1<<mix_chunk_bits);
mix_chunk_mask=mix_chunk_size-1;
mix_buffer = memnew_arr(int32_t,mix_chunk_size*mix_channels);
#ifndef NO_REVERB
zero_buffer = memnew_arr(int32_t,mix_chunk_size*mix_channels);
for(int i=0;i<mix_chunk_size*mix_channels;i++)
zero_buffer[i]=0; //zero buffer is zero...
max_reverbs=MAX_REVERBS;
int reverberators=mix_channels/2;
reverb_state = memnew_arr(ReverbState,max_reverbs);
for(int i=0;i<max_reverbs;i++) {
reverb_state[i].enabled=false;
reverb_state[i].reverb = memnew_arr(ReverbSW,reverberators);
reverb_state[i].buffer = memnew_arr(int32_t,mix_chunk_size*mix_channels);
reverb_state[i].frames_idle=0;
for(int j=0;j<reverberators;j++) {
static ReverbSW::ReverbMode modes[MAX_REVERBS]={ReverbSW::REVERB_MODE_STUDIO_SMALL,ReverbSW::REVERB_MODE_STUDIO_MEDIUM,ReverbSW::REVERB_MODE_STUDIO_LARGE,ReverbSW::REVERB_MODE_HALL};
reverb_state[i].reverb[j].set_mix_rate(p_mix_rate);
reverb_state[i].reverb[j].set_mode(modes[i]);
}
}
fx_enabled=p_use_fx;
#else
fx_enabled=false;
#endif
mix_chunk_left=0;
interpolation_type=p_interp;
channel_id_count=1;
inside_mix=false;
channel_nrg=1.0;
}
void AudioMixerSW::set_mixer_volume(float p_volume) {
channel_nrg=p_volume;
}
AudioMixerSW::~AudioMixerSW() {
memdelete_arr(mix_buffer);
#ifndef NO_REVERB
memdelete_arr(zero_buffer);
for(int i=0;i<max_reverbs;i++) {
memdelete_arr(reverb_state[i].reverb);
memdelete_arr(reverb_state[i].buffer);
}
memdelete_arr(reverb_state);
#endif
}
Reference in a new issue Copy permalink