virtualx-engine/servers/audio/audio_mixer_sw.h

/*************************************************************************/
/*  audio_mixer_sw.h                                                     */
/*************************************************************************/
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/*                           GODOT ENGINE                                */
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/* Copyright (c) 2007-2015 Juan Linietsky, Ariel Manzur.                 */
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#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