f8db8a3faa
Applies the clang-format style to the 2.1 branch as done for master in
5dbf1809c6
.
265 lines
7.9 KiB
C++
265 lines
7.9 KiB
C++
/*************************************************************************/
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/* audio_mixer_sw.h */
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/*************************************************************************/
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/* This file is part of: */
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/* GODOT ENGINE */
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/* http://www.godotengine.org */
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/*************************************************************************/
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/* Copyright (c) 2007-2017 Juan Linietsky, Ariel Manzur. */
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/* */
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/* Permission is hereby granted, free of charge, to any person obtaining */
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/* a copy of this software and associated documentation files (the */
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/* "Software"), to deal in the Software without restriction, including */
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/* without limitation the rights to use, copy, modify, merge, publish, */
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/* distribute, sublicense, and/or sell copies of the Software, and to */
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/* permit persons to whom the Software is furnished to do so, subject to */
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/* the following conditions: */
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/* */
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/* The above copyright notice and this permission notice shall be */
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/* included in all copies or substantial portions of the Software. */
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/* */
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/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
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/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
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/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
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/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
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/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
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/*************************************************************************/
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#ifndef AUDIO_MIXER_SW_H
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#define AUDIO_MIXER_SW_H
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#include "servers/audio/audio_filter_sw.h"
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#include "servers/audio/reverb_sw.h"
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#include "servers/audio/sample_manager_sw.h"
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#include "servers/audio_server.h"
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class AudioMixerSW : public AudioMixer {
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public:
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enum InterpolationType {
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INTERPOLATION_RAW,
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INTERPOLATION_LINEAR,
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INTERPOLATION_CUBIC
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};
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enum MixChannels {
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MIX_STEREO = 2,
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MIX_QUAD = 4
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};
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typedef void (*MixStepCallback)(void *);
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private:
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SampleManagerSW *sample_manager;
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enum {
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MAX_CHANNELS = 64,
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// fixed point defs
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MIX_FRAC_BITS = 13,
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MIX_FRAC_LEN = (1 << MIX_FRAC_BITS),
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MIX_FRAC_MASK = MIX_FRAC_LEN - 1,
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MIX_VOL_FRAC_BITS = 12,
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MIX_VOLRAMP_FRAC_BITS = 16,
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MIX_VOLRAMP_FRAC_LEN = (1 << MIX_VOLRAMP_FRAC_BITS),
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MIX_VOLRAMP_FRAC_MASK = MIX_VOLRAMP_FRAC_LEN - 1,
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MIX_FILTER_FRAC_BITS = 16,
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MIX_FILTER_RAMP_FRAC_BITS = 8,
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MIX_VOL_MOVE_TO_24 = 4
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};
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struct Channel {
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RID sample;
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struct Mix {
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int64_t offset;
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int32_t increment;
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int32_t vol[4];
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int32_t reverb_vol[4];
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int32_t chorus_vol[4];
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int32_t old_vol[4];
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int32_t old_reverb_vol[4];
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int32_t old_chorus_vol[4];
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struct Filter { //history (stereo)
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float ha[2], hb[2];
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} filter_l, filter_r;
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struct IMA_ADPCM_State {
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int16_t step_index;
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int32_t predictor;
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/* values at loop point */
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int16_t loop_step_index;
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int32_t loop_predictor;
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int32_t last_nibble;
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int32_t loop_pos;
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int32_t window_ofs;
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const uint8_t *ptr;
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} ima_adpcm[2];
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} mix;
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float vol;
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float pan;
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float depth;
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float height;
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float chorus_send;
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ReverbRoomType reverb_room;
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float reverb_send;
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int speed;
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int check;
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bool positional;
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bool had_prev_reverb;
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bool had_prev_chorus;
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bool had_prev_vol;
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struct Filter {
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bool dirty;
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FilterType type;
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float cutoff;
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float resonance;
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float gain;
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struct Coefs {
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float a1, a2, b0, b1, b2; // fixed point coefficients
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} coefs, old_coefs;
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} filter;
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bool first_mix;
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bool active;
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Channel() {
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active = false;
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check = -1;
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first_mix = false;
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filter.dirty = true;
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filter.type = FILTER_NONE;
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filter.cutoff = 8000;
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filter.resonance = 0;
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filter.gain = 0;
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}
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};
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Channel channels[MAX_CHANNELS];
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uint32_t mix_rate;
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bool fx_enabled;
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InterpolationType interpolation_type;
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int mix_chunk_bits;
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int mix_chunk_size;
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int mix_chunk_mask;
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int32_t *mix_buffer;
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int32_t *zero_buffer; // fx feed when no input was mixed
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struct ResamplerState {
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uint32_t amount;
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int32_t increment;
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int32_t pos;
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int32_t vol[4];
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int32_t reverb_vol[4];
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int32_t chorus_vol[4];
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int32_t vol_inc[4];
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int32_t reverb_vol_inc[4];
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int32_t chorus_vol_inc[4];
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Channel::Mix::Filter *filter_l;
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Channel::Mix::Filter *filter_r;
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Channel::Filter::Coefs coefs;
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Channel::Filter::Coefs coefs_inc;
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Channel::Mix::IMA_ADPCM_State *ima_adpcm;
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int32_t *reverb_buffer;
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};
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template <class Depth, bool is_stereo, bool use_filter, bool is_ima_adpcm, bool use_fx, InterpolationType type, MixChannels>
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_FORCE_INLINE_ void do_resample(const Depth *p_src, int32_t *p_dst, ResamplerState *p_state);
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MixChannels mix_channels;
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void mix_channel(Channel &p_channel);
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int mix_chunk_left;
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void mix_chunk();
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float channel_nrg;
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int channel_id_count;
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bool inside_mix;
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MixStepCallback step_callback;
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void *step_udata;
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_FORCE_INLINE_ int _get_channel(ChannelID p_channel) const;
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int max_reverbs;
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struct ReverbState {
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bool used_in_chunk;
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bool enabled;
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ReverbSW *reverb;
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int frames_idle;
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int32_t *buffer; //reverb is sent here
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ReverbState() {
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enabled = false;
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frames_idle = 0;
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used_in_chunk = false;
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}
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};
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ReverbState *reverb_state;
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public:
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virtual ChannelID channel_alloc(RID p_sample);
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virtual void channel_set_volume(ChannelID p_channel, float p_gain);
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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
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virtual void channel_set_filter(ChannelID p_channel, FilterType p_type, float p_cutoff, float p_resonance, float p_gain = 1.0);
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virtual void channel_set_chorus(ChannelID p_channel, float p_chorus);
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virtual void channel_set_reverb(ChannelID p_channel, ReverbRoomType p_room_type, float p_reverb);
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virtual void channel_set_mix_rate(ChannelID p_channel, int p_mix_rate);
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virtual void channel_set_positional(ChannelID p_channel, bool p_positional);
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virtual float channel_get_volume(ChannelID p_channel) const;
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virtual float channel_get_pan(ChannelID p_channel) const; //pan and depth go from -1 to 1
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virtual float channel_get_pan_depth(ChannelID p_channel) const; //pan and depth go from -1 to 1
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virtual float channel_get_pan_height(ChannelID p_channel) const; //pan and depth go from -1 to 1
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virtual FilterType channel_get_filter_type(ChannelID p_channel) const;
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virtual float channel_get_filter_cutoff(ChannelID p_channel) const;
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virtual float channel_get_filter_resonance(ChannelID p_channel) const;
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virtual float channel_get_filter_gain(ChannelID p_channel) const;
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virtual float channel_get_chorus(ChannelID p_channel) const;
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virtual ReverbRoomType channel_get_reverb_type(ChannelID p_channel) const;
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virtual float channel_get_reverb(ChannelID p_channel) const;
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virtual int channel_get_mix_rate(ChannelID p_channel) const;
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virtual bool channel_is_positional(ChannelID p_channel) const;
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virtual bool channel_is_valid(ChannelID p_channel) const;
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virtual void channel_free(ChannelID p_channel);
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int mix(int32_t *p_buffer, int p_frames); //return amount of mixsteps
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uint64_t get_step_usecs() const;
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virtual void set_mixer_volume(float p_volume);
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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);
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~AudioMixerSW();
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};
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#endif // AUDIO_MIXER_SW_H
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