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godot/servers/audio/audio_server_sw.cpp

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GODOT IS OPEN SOURCE
2014-02-10 01:10:30 +00:00
/*************************************************************************/
/* audio_server_sw.cpp */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* http://www.godotengine.org */
/*************************************************************************/
Welcome in 2017, dear changelog reader! That year should bring the long-awaited OpenGL ES 3.0 compatible renderer with state-of-the-art rendering techniques tuned to work as low as middle end handheld devices - without compromising with the possibilities given for higher end desktop games of course. Great times ahead for the Godot community and the gamers that will play our games!
2017-01-01 21:01:57 +00:00
/* Copyright (c) 2007-2017 Juan Linietsky, Ariel Manzur. */
GODOT IS OPEN SOURCE
2014-02-10 01:10:30 +00:00
/* */
/* 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_server_sw.h"
#include "globals.h"
#include "os/os.h"
struct _AudioDriverLock {
_AudioDriverLock() { if (AudioDriverSW::get_singleton()) AudioDriverSW::get_singleton()->lock(); }
~_AudioDriverLock() { if (AudioDriverSW::get_singleton()) AudioDriverSW::get_singleton()->unlock(); }
};
#define AUDIO_LOCK _AudioDriverLock _adlock;
AudioMixer *AudioServerSW::get_mixer() {
return mixer;
}
/* CALLBACKS */
void AudioServerSW::audio_mixer_chunk_callback(int p_frames) {
/*
for(List<Stream*>::Element *E=event_streams.front();E;E=E->next()) {
if (E->get()->active)
E->get()->audio_stream->mix(NULL,p_frames);
}
*/
}
void AudioServerSW::_mixer_callback(void *p_udata) {
AudioServerSW *self = (AudioServerSW*)p_udata;
for(List<Stream*>::Element *E=self->active_audio_streams.front();E;E=E->next()) {
if (!E->get()->active)
continue;
EventStream *es=E->get()->event_stream;
if (!es)
continue;
es->update(self->mixer_step_usecs);
}
}
void AudioServerSW::driver_process_chunk(int p_frames,int32_t *p_buffer) {
int samples=p_frames*internal_buffer_channels;
for(int i=0;i<samples;i++) {
internal_buffer[i]=0;
}
while(voice_rb.commands_left()) {
VoiceRBSW::Command cmd = voice_rb.pop_command();
if (cmd.type==VoiceRBSW::Command::CMD_CHANGE_ALL_FX_VOLUMES) {
SelfList<Voice>*al = active_list.first();
while(al) {
Voice *v=al->self();
if (v->channel!=AudioMixer::INVALID_CHANNEL) {
mixer->channel_set_volume(v->channel,v->volume*fx_volume_scale);
}
al=al->next();
}
continue;
}
if (!voice_owner.owns(cmd.voice))
continue;
Voice *v = voice_owner.get(cmd.voice);
switch(cmd.type) {
case VoiceRBSW::Command::CMD_NONE: {
} break;
case VoiceRBSW::Command::CMD_PLAY: {
if (v->channel!=AudioMixer::INVALID_CHANNEL)
mixer->channel_free(v->channel);
RID sample = cmd.play.sample;
if (!sample_manager->is_sample(sample))
continue;
v->channel=mixer->channel_alloc(sample);
v->volume=1.0;
mixer->channel_set_volume(v->channel,fx_volume_scale);
if (v->channel==AudioMixer::INVALID_CHANNEL) {
#ifdef AUDIO_DEBUG
WARN_PRINT("AUDIO: all channels used, failed to allocate voice");
#endif
v->active=false;
break; // no voices left?
}
v->active=true; // this kind of ensures it works
if (!v->active_item.in_list())
active_list.add(&v->active_item);
} break;
case VoiceRBSW::Command::CMD_STOP: {
if (v->channel!=AudioMixer::INVALID_CHANNEL) {
mixer->channel_free(v->channel);
if (v->active_item.in_list()) {
active_list.remove(&v->active_item);
}
}
v->active=false;
} break;
case VoiceRBSW::Command::CMD_SET_VOLUME: {
if (v->channel!=AudioMixer::INVALID_CHANNEL) {
v->volume=cmd.volume.volume;
mixer->channel_set_volume(v->channel,cmd.volume.volume*fx_volume_scale);
}
} break;
case VoiceRBSW::Command::CMD_SET_PAN: {
if (v->channel!=AudioMixer::INVALID_CHANNEL)
mixer->channel_set_pan(v->channel,cmd.pan.pan,cmd.pan.depth,cmd.pan.height);
} break;
case VoiceRBSW::Command::CMD_SET_FILTER: {
if (v->channel!=AudioMixer::INVALID_CHANNEL)
mixer->channel_set_filter(v->channel,(AudioMixer::FilterType)cmd.filter.type,cmd.filter.cutoff,cmd.filter.resonance,cmd.filter.gain);
} break;
case VoiceRBSW::Command::CMD_SET_CHORUS: {
if (v->channel!=AudioMixer::INVALID_CHANNEL)
mixer->channel_set_chorus(v->channel,cmd.chorus.send);
} break;
case VoiceRBSW::Command::CMD_SET_REVERB: {
if (v->channel!=AudioMixer::INVALID_CHANNEL)
mixer->channel_set_reverb(v->channel,(AudioMixer::ReverbRoomType)cmd.reverb.room,cmd.reverb.send);
} break;
case VoiceRBSW::Command::CMD_SET_MIX_RATE: {
if (v->channel!=AudioMixer::INVALID_CHANNEL)
mixer->channel_set_mix_rate(v->channel,cmd.mix_rate.mix_rate);
} break;
case VoiceRBSW::Command::CMD_SET_POSITIONAL: {
if (v->channel!=AudioMixer::INVALID_CHANNEL)
mixer->channel_set_positional(v->channel,cmd.positional.positional);
} break;
default: {}
}
}
mixer->mix(internal_buffer,p_frames);
//uint64_t stepsize=mixer->get_step_usecs();
for(List<Stream*>::Element *E=active_audio_streams.front();E;E=E->next()) {
ERR_CONTINUE(!E->get()->active); // bug?
AudioStream *as=E->get()->audio_stream;
if (!as)
continue;
int channels=as->get_channel_count();
if (channels==0)
continue; // does not want mix
if (!as->mix(stream_buffer,p_frames))
continue; //nothing was mixed!!
int32_t stream_vol_scale=(stream_volume*stream_volume_scale*E->get()->volume_scale)*(1<<STREAM_SCALE_BITS);
#define STRSCALE(m_val) (((m_val>>STREAM_SCALE_BITS)*stream_vol_scale)>>8)
switch(internal_buffer_channels) {
case 2: {
switch(channels) {
case 1: {
for(int i=0;i<p_frames;i++) {
internal_buffer[(i<<1)+0]+=STRSCALE(stream_buffer[i]);
internal_buffer[(i<<1)+1]+=STRSCALE(stream_buffer[i]);
}
} break;
case 2: {
for(int i=0;i<p_frames*2;i++) {
internal_buffer[i]+=STRSCALE(stream_buffer[i]);
}
} break;
case 4: {
for(int i=0;i<p_frames;i++) {
internal_buffer[(i<<2)+0]+=STRSCALE((stream_buffer[(i<<2)+0]+stream_buffer[(i<<2)+2])>>1);
internal_buffer[(i<<2)+1]+=STRSCALE((stream_buffer[(i<<2)+1]+stream_buffer[(i<<2)+3])>>1);
}
} break;
} break;
} break;
case 4: {
switch(channels) {
case 1: {
for(int i=0;i<p_frames;i++) {
internal_buffer[(i<<2)+0]+=STRSCALE(stream_buffer[i]);
internal_buffer[(i<<2)+1]+=STRSCALE(stream_buffer[i]);
internal_buffer[(i<<2)+2]+=STRSCALE(stream_buffer[i]);
internal_buffer[(i<<2)+3]+=STRSCALE(stream_buffer[i]);
}
} break;
case 2: {
for(int i=0;i<p_frames*2;i++) {
internal_buffer[(i<<2)+0]+=STRSCALE(stream_buffer[(i<<1)+0]);
internal_buffer[(i<<2)+1]+=STRSCALE(stream_buffer[(i<<1)+1]);
internal_buffer[(i<<2)+2]+=STRSCALE(stream_buffer[(i<<1)+0]);
internal_buffer[(i<<2)+3]+=STRSCALE(stream_buffer[(i<<1)+1]);
}
} break;
case 4: {
for(int i=0;i<p_frames*4;i++) {
internal_buffer[i]+=STRSCALE(stream_buffer[i]);
}
} break;
} break;
} break;
case 6: {
} break;
}
#undef STRSCALE
}
SelfList<Voice> *activeE=active_list.first();
while(activeE) {
SelfList<Voice> *activeN=activeE->next();
if (activeE->self()->channel==AudioMixer::INVALID_CHANNEL || !mixer->channel_is_valid(activeE->self()->channel)) {
active_list.remove(activeE);
activeE->self()->active=false;
}
activeE=activeN;
}
uint32_t peak=0;
for(int i=0;i<samples;i++) {
//clamp to (1<<24) using branchless code
int32_t in = internal_buffer[i];
#ifdef DEBUG_ENABLED
{
int mask = (in >> (32 - 1));
uint32_t p = (in + mask) ^ mask;
if (p>peak)
peak=p;
}
#endif
int32_t lo = -0x800000, hi=0x7FFFFF;
lo-=in;
hi-=in;
in += (lo & ((lo < 0) - 1)) + (hi & ((hi > 0) - 1));
p_buffer[i]=in<<8;
}
if (peak>max_peak)
max_peak=peak;
}
void AudioServerSW::driver_process(int p_frames,int32_t *p_buffer) {
3D Import Import & UDP -=-=-=-=-=-=-=-=-=-=- -Animation Import filter support -Animation Clip import support -Animation Optimizer Fixes, Improvements and Visibile Options -Extremely Experimental UDP support.
2014-11-12 14:23:23 +00:00
_output_delay=p_frames/double(AudioDriverSW::get_singleton()->get_mix_rate());
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//process in chunks to make sure to never process more than INTERNAL_BUFFER_SIZE
int todo=p_frames;
while(todo) {
int tomix=MIN(todo,INTERNAL_BUFFER_SIZE);
driver_process_chunk(tomix,p_buffer);
p_buffer+=tomix;
todo-=tomix;
}
}
/* SAMPLE API */
RID AudioServerSW::sample_create(SampleFormat p_format, bool p_stereo, int p_length) {
AUDIO_LOCK
return sample_manager->sample_create(p_format,p_stereo,p_length);
}
void AudioServerSW::sample_set_description(RID p_sample, const String& p_description) {
AUDIO_LOCK
sample_manager->sample_set_description(p_sample,p_description);
}
Remove bogus argument in sample_get_description Copy-paste mistake from the setter most likely.
2015-12-28 00:18:37 +00:00
String AudioServerSW::sample_get_description(RID p_sample) const {
GODOT IS OPEN SOURCE
2014-02-10 01:10:30 +00:00
AUDIO_LOCK
return sample_manager->sample_get_description(p_sample);
}
AS::SampleFormat AudioServerSW::sample_get_format(RID p_sample) const {
//AUDIO_LOCK
return sample_manager->sample_get_format(p_sample);
}
bool AudioServerSW::sample_is_stereo(RID p_sample) const {
//AUDIO_LOCK
return sample_manager->sample_is_stereo(p_sample);
}
int AudioServerSW::sample_get_length(RID p_sample) const {
///AUDIO_LOCK
return sample_manager->sample_get_length(p_sample);
}
const void* AudioServerSW::sample_get_data_ptr(RID p_sample) const {
///AUDIO_LOCK
return sample_manager->sample_get_data_ptr(p_sample);
}
Memory pool vectors (DVector) have been enormously simplified in code, and renamed to PoolVector
2017-01-07 21:25:37 +00:00
void AudioServerSW::sample_set_data(RID p_sample, const PoolVector<uint8_t>& p_buffer) {
GODOT IS OPEN SOURCE
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AUDIO_LOCK
sample_manager->sample_set_data(p_sample,p_buffer);
}
Memory pool vectors (DVector) have been enormously simplified in code, and renamed to PoolVector
2017-01-07 21:25:37 +00:00
PoolVector<uint8_t> AudioServerSW::sample_get_data(RID p_sample) const {
GODOT IS OPEN SOURCE
2014-02-10 01:10:30 +00:00
AUDIO_LOCK
return sample_manager->sample_get_data(p_sample);
}
void AudioServerSW::sample_set_mix_rate(RID p_sample,int p_rate) {
AUDIO_LOCK
sample_manager->sample_set_mix_rate(p_sample,p_rate);
}
int AudioServerSW::sample_get_mix_rate(RID p_sample) const {
AUDIO_LOCK
return sample_manager->sample_get_mix_rate(p_sample);
}
void AudioServerSW::sample_set_loop_format(RID p_sample,SampleLoopFormat p_format) {
AUDIO_LOCK
sample_manager->sample_set_loop_format(p_sample,p_format);
}
AS::SampleLoopFormat AudioServerSW::sample_get_loop_format(RID p_sample) const {
AUDIO_LOCK
return sample_manager->sample_get_loop_format(p_sample);
}
void AudioServerSW::sample_set_loop_begin(RID p_sample,int p_pos) {
AUDIO_LOCK
sample_manager->sample_set_loop_begin(p_sample,p_pos);
}
int AudioServerSW::sample_get_loop_begin(RID p_sample) const {
AUDIO_LOCK
return sample_manager->sample_get_loop_begin(p_sample);
}
void AudioServerSW::sample_set_loop_end(RID p_sample,int p_pos) {
AUDIO_LOCK
sample_manager->sample_set_loop_end(p_sample,p_pos);
}
int AudioServerSW::sample_get_loop_end(RID p_sample) const {
AUDIO_LOCK
return sample_manager->sample_get_loop_end(p_sample);
}
/* VOICE API */
RID AudioServerSW::voice_create() {
Voice * v = memnew( Voice );
v->channel=AudioMixer::INVALID_CHANNEL;
AUDIO_LOCK
return voice_owner.make_rid(v);
}
void AudioServerSW::voice_play(RID p_voice, RID p_sample) {
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND(!v);
v->active=true; // force actvive (will be disabled later i gues..)
//stop old, start new
VoiceRBSW::Command cmd;
cmd.type=VoiceRBSW::Command::CMD_PLAY;
cmd.voice=p_voice;
cmd.play.sample=p_sample;
voice_rb.push_command(cmd);
}
Rename voice_set_volume argument to p_volume to avoid confusion This argument is indeed expected to be a volume on a linear scale (not dB), typically from 0.0 to 1.0, though it can go higher than 1.0.
2015-09-26 20:42:43 +00:00
void AudioServerSW::voice_set_volume(RID p_voice, float p_volume) {
GODOT IS OPEN SOURCE
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VoiceRBSW::Command cmd;
cmd.type=VoiceRBSW::Command::CMD_SET_VOLUME;
cmd.voice=p_voice;
Rename voice_set_volume argument to p_volume to avoid confusion This argument is indeed expected to be a volume on a linear scale (not dB), typically from 0.0 to 1.0, though it can go higher than 1.0.
2015-09-26 20:42:43 +00:00
cmd.volume.volume=p_volume;
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voice_rb.push_command(cmd);
}
void AudioServerSW::voice_set_pan(RID p_voice, float p_pan, float p_depth,float p_height) {
VoiceRBSW::Command cmd;
cmd.type=VoiceRBSW::Command::CMD_SET_PAN;
cmd.voice=p_voice;
cmd.pan.pan=p_pan;
cmd.pan.depth=p_depth;
cmd.pan.height=p_height;
voice_rb.push_command(cmd);
}
void AudioServerSW::voice_set_filter(RID p_voice, FilterType p_type, float p_cutoff, float p_resonance,float p_gain) {
VoiceRBSW::Command cmd;
cmd.type=VoiceRBSW::Command::CMD_SET_FILTER;
cmd.voice=p_voice;
cmd.filter.type=p_type;
cmd.filter.cutoff=p_cutoff;
cmd.filter.resonance=p_resonance;
cmd.filter.gain=p_gain;
voice_rb.push_command(cmd);
}
void AudioServerSW::voice_set_chorus(RID p_voice, float p_chorus ) {
VoiceRBSW::Command cmd;
cmd.type=VoiceRBSW::Command::CMD_SET_CHORUS;
cmd.voice=p_voice;
cmd.chorus.send=p_chorus;
voice_rb.push_command(cmd);
}
void AudioServerSW::voice_set_reverb(RID p_voice, ReverbRoomType p_room_type, float p_reverb) {
VoiceRBSW::Command cmd;
cmd.type=VoiceRBSW::Command::CMD_SET_REVERB;
cmd.voice=p_voice;
cmd.reverb.room=p_room_type;
cmd.reverb.send=p_reverb;
More Bug Fixes -=-=-=-=-=-=- -Fixed a few bugs in Mixer, now playback of chiptunes works great :) -Changed how visibility AABB generation from skeletons work, it's fully automatic and real-time now, generated from current skeleton pose for the frame. -Fixed camera in 3D kinematic character demo.
2014-09-17 23:03:10 +00:00
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voice_rb.push_command(cmd);
}
void AudioServerSW::voice_set_mix_rate(RID p_voice, int p_mix_rate) {
VoiceRBSW::Command cmd;
cmd.type=VoiceRBSW::Command::CMD_SET_MIX_RATE;
cmd.voice=p_voice;
cmd.mix_rate.mix_rate=p_mix_rate;
voice_rb.push_command(cmd);
}
void AudioServerSW::voice_set_positional(RID p_voice, bool p_positional) {
VoiceRBSW::Command cmd;
cmd.type=VoiceRBSW::Command::CMD_SET_POSITIONAL;
cmd.voice=p_voice;
cmd.positional.positional=p_positional;
voice_rb.push_command(cmd);
}
float AudioServerSW::voice_get_volume(RID p_voice) const {
AUDIO_LOCK
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND_V(!v, 0);
return mixer->channel_get_volume( v->channel );
}
float AudioServerSW::voice_get_pan(RID p_voice) const {
AUDIO_LOCK
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND_V(!v, 0);
return mixer->channel_get_pan( v->channel );
}
float AudioServerSW::voice_get_pan_depth(RID p_voice) const {
AUDIO_LOCK
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND_V(!v, 0);
return mixer->channel_get_pan_depth( v->channel );
}
float AudioServerSW::voice_get_pan_height(RID p_voice) const {
AUDIO_LOCK
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND_V(!v, 0);
return mixer->channel_get_pan_height( v->channel );
}
AS::FilterType AudioServerSW::voice_get_filter_type(RID p_voice) const {
AUDIO_LOCK
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND_V(!v, AS::FILTER_NONE);
return (AS::FilterType)mixer->channel_get_filter_type(v->channel);
}
float AudioServerSW::voice_get_filter_cutoff(RID p_voice) const {
AUDIO_LOCK
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND_V(!v, 0);
return mixer->channel_get_filter_cutoff( v->channel );
}
float AudioServerSW::voice_get_filter_resonance(RID p_voice) const {
AUDIO_LOCK
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND_V(!v, 0);
return mixer->channel_get_filter_resonance( v->channel );
}
float AudioServerSW::voice_get_chorus(RID p_voice) const {
AUDIO_LOCK
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND_V(!v, 0);
return mixer->channel_get_chorus( v->channel );
}
AS::ReverbRoomType AudioServerSW::voice_get_reverb_type(RID p_voice) const {
AUDIO_LOCK
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND_V(!v, REVERB_SMALL);
return (AS::ReverbRoomType)mixer->channel_get_reverb_type( v->channel );
}
float AudioServerSW::voice_get_reverb(RID p_voice) const {
AUDIO_LOCK
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND_V(!v, 0);
return mixer->channel_get_reverb( v->channel );
}
int AudioServerSW::voice_get_mix_rate(RID p_voice) const {
AUDIO_LOCK
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND_V(!v, 0);
return mixer->channel_get_mix_rate( v->channel );
}
bool AudioServerSW::voice_is_positional(RID p_voice) const {
AUDIO_LOCK
Voice *v = voice_owner.get( p_voice );
ERR_FAIL_COND_V(!v, 0);
return mixer->channel_is_positional( v->channel );
}
void AudioServerSW::voice_stop(RID p_voice) {
VoiceRBSW::Command cmd;
cmd.type=VoiceRBSW::Command::CMD_STOP;
cmd.voice=p_voice;
voice_rb.push_command(cmd);
//return mixer->channel_free( v->channel );
}
bool AudioServerSW::voice_is_active(RID p_voice) const {
Voice *v = voice_owner.get(p_voice);
ERR_FAIL_COND_V(!v,false);
return v->active;
}
/* STREAM API */
RID AudioServerSW::audio_stream_create(AudioStream *p_stream) {
-another approach to solving the deadlock problem :|
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AUDIO_LOCK
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Stream *s = memnew(Stream);
s->audio_stream=p_stream;
s->event_stream=NULL;
s->active=false;
s->E=NULL;
s->volume_scale=1.0;
p_stream->set_mix_rate(AudioDriverSW::get_singleton()->get_mix_rate());
return stream_owner.make_rid(s);
}
RID AudioServerSW::event_stream_create(EventStream *p_stream) {
AUDIO_LOCK
Stream *s = memnew(Stream);
s->audio_stream=NULL;
s->event_stream=p_stream;
s->active=false;
s->E=NULL;
s->volume_scale=1.0;
//p_stream->set_mix_rate(AudioDriverSW::get_singleton()->get_mix_rate());
return stream_owner.make_rid(s);
}
void AudioServerSW::stream_set_active(RID p_stream, bool p_active) {
Stream *s = stream_owner.get(p_stream);
ERR_FAIL_COND(!s);
-different attempt to avod deadlock problem
2015-12-21 13:25:46 +00:00
_THREAD_SAFE_METHOD_
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if (s->active==p_active)
return;
-another approach to solving the deadlock problem :|
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AUDIO_LOCK;
s->active=p_active;
if (p_active)
s->E=active_audio_streams.push_back(s);
else {
active_audio_streams.erase(s->E);
s->E=NULL;
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}
-different attempt to avod deadlock problem
2015-12-21 13:25:46 +00:00
-another approach to solving the deadlock problem :|
2015-12-21 13:51:27 +00:00
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}
bool AudioServerSW::stream_is_active(RID p_stream) const {
Stream *s = stream_owner.get(p_stream);
ERR_FAIL_COND_V(!s,false);
return s->active;
}
void AudioServerSW::stream_set_volume_scale(RID p_stream, float p_scale) {
Stream *s = stream_owner.get(p_stream);
ERR_FAIL_COND(!s);
s->volume_scale=p_scale;
}
float AudioServerSW::stream_set_volume_scale(RID p_stream) const {
Stream *s = stream_owner.get(p_stream);
ERR_FAIL_COND_V(!s,0);
return s->volume_scale;
}
void AudioServerSW::free(RID p_id) {
if(voice_owner.owns(p_id)) {
Voice *v = voice_owner.get(p_id);
AUDIO_LOCK
mixer->channel_free( v->channel );
voice_owner.free(p_id);
memdelete(v);
} else if (stream_owner.owns(p_id)) {
Stream *s=stream_owner.get(p_id);
if (s->active) {
stream_set_active(p_id,false);
}
memdelete(s);
stream_owner.free(p_id);
} else if (sample_manager->is_sample(p_id)) {
AUDIO_LOCK
sample_manager->free(p_id);
}
}
void AudioServerSW::_thread_func(void *self) {
thread renaming
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Thread::set_name("AudioServerSW");
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AudioServerSW *as=(AudioServerSW *)self;
while (!as->exit_update_thread) {
as->_update_streams(true);
OS::get_singleton()->delay_usec(5000);
}
}
void AudioServerSW::init() {
int latency = GLOBAL_DEF("audio/mixer_latency",10);
internal_buffer_channels=2; // read from driver
internal_buffer = memnew_arr(int32_t,INTERNAL_BUFFER_SIZE*internal_buffer_channels);
stream_buffer = memnew_arr(int32_t,INTERNAL_BUFFER_SIZE*4); //max 4 channels
AudioMixerSW::MixChannels mix_chans = AudioMixerSW::MIX_STEREO;
switch(AudioDriverSW::get_singleton()->get_output_format()) {
case AudioDriverSW::OUTPUT_MONO:
case AudioDriverSW::OUTPUT_STEREO:
mix_chans=AudioMixerSW::MIX_STEREO;
break;
case AudioDriverSW::OUTPUT_QUAD:
case AudioDriverSW::OUTPUT_5_1:
mix_chans=AudioMixerSW::MIX_QUAD;
break;
}
mixer = memnew( AudioMixerSW( sample_manager, latency, AudioDriverSW::get_singleton()->get_mix_rate(),mix_chans,mixer_use_fx,mixer_interp,_mixer_callback,this ) );
mixer_step_usecs=mixer->get_step_usecs();
3D Import Import & UDP -=-=-=-=-=-=-=-=-=-=- -Animation Import filter support -Animation Clip import support -Animation Optimizer Fixes, Improvements and Visibile Options -Extremely Experimental UDP support.
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_output_delay=0;
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stream_volume=0.3;
// start the audio driver
if (AudioDriverSW::get_singleton())
AudioDriverSW::get_singleton()->start();
-another approach to solving the deadlock problem :|
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#ifndef NO_THREADS
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exit_update_thread=false;
thread = Thread::create(_thread_func,this);
#endif
}
void AudioServerSW::finish() {
-another approach to solving the deadlock problem :|
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#ifndef NO_THREADS
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exit_update_thread=true;
Thread::wait_to_finish(thread);
memdelete(thread);
#endif
if (AudioDriverSW::get_singleton())
AudioDriverSW::get_singleton()->finish();
memdelete_arr(internal_buffer);
memdelete_arr(stream_buffer);
memdelete(mixer);
}
void AudioServerSW::_update_streams(bool p_thread) {
_THREAD_SAFE_METHOD_
Rewrite of the AudioStream API -Fixes long-standing issues regarding to playing a single stream multiple times simultanteously -Fixes wrong-looping, starting, caching, etc. Issues resulting from bad original design -Allows more interesting kinds of streams (stream graphs with streams inside streams!) in the future
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for(List<Stream*>::Element *E=active_audio_streams.front();E;) { //stream might be removed durnig this callback
List<Stream*>::Element *N=E->next();
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if (E->get()->audio_stream && p_thread == E->get()->audio_stream->can_update_mt())
E->get()->audio_stream->update();
Rewrite of the AudioStream API -Fixes long-standing issues regarding to playing a single stream multiple times simultanteously -Fixes wrong-looping, starting, caching, etc. Issues resulting from bad original design -Allows more interesting kinds of streams (stream graphs with streams inside streams!) in the future
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E=N;
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}
}
void AudioServerSW::update() {
_update_streams(false);
-another approach to solving the deadlock problem :|
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#ifdef NO_THREADS
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_update_streams(true);
#endif
}
void AudioServerSW::lock() {
AudioDriverSW::get_singleton()->lock();
}
void AudioServerSW::unlock() {
AudioDriverSW::get_singleton()->unlock();
}
int AudioServerSW::get_default_mix_rate() const {
return AudioDriverSW::get_singleton()->get_mix_rate();
}
int AudioServerSW::get_default_channel_count() const {
return internal_buffer_channels;
}
void AudioServerSW::set_mixer_params(AudioMixerSW::InterpolationType p_interp, bool p_use_fx) {
mixer_interp=p_interp;
mixer_use_fx=p_use_fx;
}
void AudioServerSW::set_stream_global_volume_scale(float p_volume) {
stream_volume_scale=p_volume;
}
float AudioServerSW::get_stream_global_volume_scale() const {
return stream_volume_scale;
}
void AudioServerSW::set_fx_global_volume_scale(float p_volume) {
fx_volume_scale=p_volume;
//mixer->set_mixer_volume(fx_volume_scale);
VoiceRBSW::Command cmd;
cmd.type=VoiceRBSW::Command::CMD_CHANGE_ALL_FX_VOLUMES;
cmd.voice=RID();
cmd.volume.volume=p_volume;
voice_rb.push_command(cmd);
}
float AudioServerSW::get_fx_global_volume_scale() const {
return fx_volume_scale;
}
void AudioServerSW::set_event_voice_global_volume_scale(float p_volume) {
event_voice_volume_scale=p_volume;
//mixer->set_mixer_volume(event_voice_volume_scale);
}
float AudioServerSW::get_event_voice_global_volume_scale() const {
return event_voice_volume_scale;
}
3D Import Import & UDP -=-=-=-=-=-=-=-=-=-=- -Animation Import filter support -Animation Clip import support -Animation Optimizer Fixes, Improvements and Visibile Options -Extremely Experimental UDP support.
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double AudioServerSW::get_output_delay() const {
Fixed theora playback. Removed theoraplayer. Still need to get proper audio output latency in some platforms.
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return _output_delay+AudioDriverSW::get_singleton()->get_latency();
3D Import Import & UDP -=-=-=-=-=-=-=-=-=-=- -Animation Import filter support -Animation Clip import support -Animation Optimizer Fixes, Improvements and Visibile Options -Extremely Experimental UDP support.
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}
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double AudioServerSW::get_mix_time() const {
return AudioDriverSW::get_singleton()->get_mix_time();
}
uint32_t AudioServerSW::read_output_peak() const {
uint32_t val = max_peak;
uint32_t *p = (uint32_t*)&max_peak;
*p=0;
return val;
}
AudioServerSW::AudioServerSW(SampleManagerSW *p_sample_manager) {
sample_manager=p_sample_manager;
String interp = GLOBAL_DEF("audio/mixer_interp","linear");
-Changed most project settings in the engine, so they have major and minor categories. -Changed SectionedPropertyEditor to support this -Renamed Globals singleton to GlobalConfig, makes more sense. -Changed the logic behind persisten global settings, instead of the persist checkbox, a revert button is now available
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GlobalConfig::get_singleton()->set_custom_property_info("audio/mixer_interp",PropertyInfo(Variant::STRING,"audio/mixer_interp",PROPERTY_HINT_ENUM,"raw,linear,cubic"));
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if (interp=="raw")
mixer_interp=AudioMixerSW::INTERPOLATION_RAW;
else if (interp=="cubic")
mixer_interp=AudioMixerSW::INTERPOLATION_CUBIC;
else
mixer_interp=AudioMixerSW::INTERPOLATION_LINEAR;
mixer_use_fx = GLOBAL_DEF("audio/use_chorus_reverb",true);
stream_volume_scale=GLOBAL_DEF("audio/stream_volume_scale",1.0);
fx_volume_scale=GLOBAL_DEF("audio/fx_volume_scale",1.0);
event_voice_volume_scale=GLOBAL_DEF("audio/event_voice_volume_scale",0.5);
max_peak=0;
}
AudioServerSW::~AudioServerSW() {
}
AudioDriverSW *AudioDriverSW::singleton=NULL;
AudioDriverSW *AudioDriverSW::get_singleton() {
return singleton;
}
void AudioDriverSW::set_singleton() {
singleton=this;
}
void AudioDriverSW::audio_server_process(int p_frames,int32_t *p_buffer,bool p_update_mix_time) {
AudioServerSW * audio_server = static_cast<AudioServerSW*>(AudioServer::get_singleton());
if (p_update_mix_time)
update_mix_time(p_frames);
audio_server->driver_process(p_frames,p_buffer);
}
void AudioDriverSW::update_mix_time(int p_frames) {
_mix_amount+=p_frames;
_last_mix_time=OS::get_singleton()->get_ticks_usec();
}
double AudioDriverSW::get_mix_time() const {
double total = (OS::get_singleton()->get_ticks_usec() - _last_mix_time) / 1000000.0;
total+=_mix_amount/(double)get_mix_rate();
return total;
}
AudioDriverSW::AudioDriverSW() {
_last_mix_time=0;
_mix_amount=0;
}
AudioDriverSW *AudioDriverManagerSW::drivers[MAX_DRIVERS];
int AudioDriverManagerSW::driver_count=0;
void AudioDriverManagerSW::add_driver(AudioDriverSW *p_driver) {
ERR_FAIL_COND(driver_count>=MAX_DRIVERS);
drivers[driver_count++]=p_driver;
}
int AudioDriverManagerSW::get_driver_count() {
return driver_count;
}
AudioDriverSW *AudioDriverManagerSW::get_driver(int p_driver) {
ERR_FAIL_INDEX_V(p_driver,driver_count,NULL);
return drivers[p_driver];
}