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godot/servers/audio/audio_mixer_sw.h
Juan Linietsky b324ff7ea5 A bit of everything: 
-IMA-ADPCM support for samples, this means that sound effects can be compressed and use 4 timess less RAM.
-New 3D import workflow based on Wavefront OBJ. Import single objects as mesh resources instead of full scenes. Many people prefers to work this way. Just like the rest of the imported resources, these are updated in realtime if modified externally.
-Mesh resources now support naming surfaces. This helps reimporting to identify which user-created materials must be kept.
-Several fixes and improvements to SurfaceTool.
-Anti Aliasing added to WorldEnvironment effects (using FXAA)
-2D Physics bodies (RigidBody, KinematicBody, etc), Raycasts, Tilemap, etc support collision layers. This makes easy to group which objects collide against which.
-2D Trigger shapes can now also trigger collision reporting in other 2D bodies (it used to be in Area2D before)
-Viewport render target textures can now be filtered.
-Few fixes in GDscript make it easier to work with static functions and class members.
-Several and many bugfixes.
2014-05-14 01:22:15 -03:00

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/*************************************************************************/
/* audio_mixer_sw.h */
/*************************************************************************/
/* 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. */
/*************************************************************************/
#ifndef AUDIO_MIXER_SW_H
#define AUDIO_MIXER_SW_H
#include "servers/audio_server.h"
#include "servers/audio/sample_manager_sw.h"
#include "servers/audio/audio_filter_sw.h"
#include "servers/audio/reverb_sw.h"
class AudioMixerSW : public AudioMixer {
public:
enum InterpolationType {
INTERPOLATION_RAW,
INTERPOLATION_LINEAR,
INTERPOLATION_CUBIC
};
enum MixChannels {
MIX_STEREO=2,
MIX_QUAD=4
};
typedef void (*MixStepCallback)(void*);
private:
SampleManagerSW *sample_manager;
enum {
MAX_CHANNELS=64,
// fixed point defs
MIX_FRAC_BITS=13,
MIX_FRAC_LEN=(1<<MIX_FRAC_BITS),
MIX_FRAC_MASK=MIX_FRAC_LEN-1,
MIX_VOL_FRAC_BITS=12,
MIX_VOLRAMP_FRAC_BITS=16,
MIX_VOLRAMP_FRAC_LEN=(1<<MIX_VOLRAMP_FRAC_BITS),
MIX_VOLRAMP_FRAC_MASK=MIX_VOLRAMP_FRAC_LEN-1,
MIX_FILTER_FRAC_BITS=16,
MIX_FILTER_RAMP_FRAC_BITS=8,
MIX_VOL_MOVE_TO_24=4,
MAX_REVERBS=4
};
struct Channel {
RID sample;
struct Mix {
int64_t offset;
int32_t increment;
int32_t vol[4];
int32_t reverb_vol[4];
int32_t chorus_vol[4];
int32_t old_vol[4];
int32_t old_reverb_vol[4];
int32_t old_chorus_vol[4];
struct Filter { //history (stereo)
float ha[2],hb[2];
} filter_l,filter_r;
struct IMA_ADPCM_State {
int16_t step_index;
int32_t predictor;
/* values at loop point */
int16_t loop_step_index;
int32_t loop_predictor;
int32_t last_nibble;
int32_t loop_pos;
int32_t window_ofs;
const uint8_t *ptr;
} ima_adpcm;
} mix;
float vol;
float pan;
float depth;
float height;
float chorus_send;
ReverbRoomType reverb_room;
float reverb_send;
int speed;
int check;
bool positional;
bool had_prev_reverb;
bool had_prev_chorus;
bool had_prev_vol;
struct Filter {
bool dirty;
FilterType type;
float cutoff;
float resonance;
float gain;
struct Coefs {
float a1,a2,b0,b1,b2; // fixed point coefficients
} coefs,old_coefs;
} filter;
bool first_mix;
bool active;
Channel() { active=false; check=-1; first_mix=false; filter.dirty=true; filter.type=FILTER_NONE; filter.cutoff=8000; filter.resonance=0; filter.gain=0; }
};
Channel channels[MAX_CHANNELS];
uint32_t mix_rate;
bool fx_enabled;
InterpolationType interpolation_type;
int mix_chunk_bits;
int mix_chunk_size;
int mix_chunk_mask;
int32_t *mix_buffer;
int32_t *zero_buffer; // fx feed when no input was mixed
struct ResamplerState {
uint32_t amount;
int32_t increment;
int32_t pos;
int32_t vol[4];
int32_t reverb_vol[4];
int32_t chorus_vol[4];
int32_t vol_inc[4];
int32_t reverb_vol_inc[4];
int32_t chorus_vol_inc[4];
Channel::Mix::Filter *filter_l;
Channel::Mix::Filter *filter_r;
Channel::Filter::Coefs coefs;
Channel::Filter::Coefs coefs_inc;
Channel::Mix::IMA_ADPCM_State *ima_adpcm;
int32_t *reverb_buffer;
};
template<class Depth,bool is_stereo,bool use_filter,bool is_ima_adpcm,bool use_fx,InterpolationType type,MixChannels>
_FORCE_INLINE_ void do_resample(const Depth* p_src, int32_t *p_dst, ResamplerState *p_state);
MixChannels mix_channels;
void mix_channel(Channel& p_channel);
int mix_chunk_left;
void mix_chunk();
float channel_nrg;
int channel_id_count;
bool inside_mix;
MixStepCallback step_callback;
void *step_udata;
_FORCE_INLINE_ int _get_channel(ChannelID p_channel) const;
int max_reverbs;
struct ReverbState {
bool used_in_chunk;
bool enabled;
ReverbSW *reverb;
int frames_idle;
int32_t *buffer; //reverb is sent here
ReverbState() { enabled=false; frames_idle=0; used_in_chunk=false; }
};
ReverbState *reverb_state;
public:
virtual ChannelID channel_alloc(RID p_sample);
virtual void channel_set_volume(ChannelID p_channel, float p_gain);
virtual void channel_set_pan(ChannelID p_channel, float p_pan, float p_depth=0,float height=0); //pan and depth go from -1 to 1
virtual void channel_set_filter(ChannelID p_channel, FilterType p_type, float p_cutoff, float p_resonance, float p_gain=1.0);
virtual void channel_set_chorus(ChannelID p_channel, float p_chorus );
virtual void channel_set_reverb(ChannelID p_channel, ReverbRoomType p_room_type, float p_reverb);
virtual void channel_set_mix_rate(ChannelID p_channel, int p_mix_rate);
virtual void channel_set_positional(ChannelID p_channel, bool p_positional);
virtual float channel_get_volume(ChannelID p_channel) const;
virtual float channel_get_pan(ChannelID p_channel) const; //pan and depth go from -1 to 1
virtual float channel_get_pan_depth(ChannelID p_channel) const; //pan and depth go from -1 to 1
virtual float channel_get_pan_height(ChannelID p_channel) const; //pan and depth go from -1 to 1
virtual FilterType channel_get_filter_type(ChannelID p_channel) const;
virtual float channel_get_filter_cutoff(ChannelID p_channel) const;
virtual float channel_get_filter_resonance(ChannelID p_channel) const;
virtual float channel_get_filter_gain(ChannelID p_channel) const;
virtual float channel_get_chorus(ChannelID p_channel) const;
virtual ReverbRoomType channel_get_reverb_type(ChannelID p_channel) const;
virtual float channel_get_reverb(ChannelID p_channel) const;
virtual int channel_get_mix_rate(ChannelID p_channel) const;
virtual bool channel_is_positional(ChannelID p_channel) const;
virtual bool channel_is_valid(ChannelID p_channel) const;
virtual void channel_free(ChannelID p_channel);
int mix(int32_t *p_buffer,int p_frames); //return amount of mixsteps
uint64_t get_step_usecs() const;
virtual void set_mixer_volume(float p_volume);
AudioMixerSW(SampleManagerSW *p_sample_manager,int p_desired_latency_ms,int p_mix_rate,MixChannels p_mix_channels,bool p_use_fx=true,InterpolationType p_interp=INTERPOLATION_LINEAR,MixStepCallback p_step_callback=NULL,void *p_callback_udata=NULL);
~AudioMixerSW();
};
#endif // AUDIO_MIXER_SW_H
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