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/*************************************************************************/
/* audio_mixer_sw.h */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* http://www.godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2016 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[2];
} 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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