/* FluidSynth - A Software Synthesizer * * Copyright (C) 2003 Peter Hanappe and others. * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public License * as published by the Free Software Foundation; either version 2 of * the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, but * WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the Free * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA * 02111-1307, USA */ #include "fluidsynth_priv.h" #include "fluid_voice.h" #include "fluid_mod.h" #include "fluid_chan.h" #include "fluid_conv.h" #include "fluid_synth.h" #include "fluid_sys.h" #include "fluid_sfont.h" /* used for filter turn off optimization - if filter cutoff is above the specified value and filter q is below the other value, turn filter off */ #define FLUID_MAX_AUDIBLE_FILTER_FC 19000.0f #define FLUID_MIN_AUDIBLE_FILTER_Q 1.2f /* Smallest amplitude that can be perceived (full scale is +/- 0.5) * 16 bits => 96+4=100 dB dynamic range => 0.00001 * 0.00001 * 2 is approximately 0.00003 :) */ #define FLUID_NOISE_FLOOR 0.00003 /* these should be the absolute minimum that FluidSynth can deal with */ #define FLUID_MIN_LOOP_SIZE 2 #define FLUID_MIN_LOOP_PAD 0 /* min vol envelope release (to stop clicks) in SoundFont timecents */ #define FLUID_MIN_VOLENVRELEASE -7200.0f /* ~16ms */ static inline void fluid_voice_effects (fluid_voice_t *voice, int count, fluid_real_t* dsp_left_buf, fluid_real_t* dsp_right_buf, fluid_real_t* dsp_reverb_buf, fluid_real_t* dsp_chorus_buf); /* * new_fluid_voice */ fluid_voice_t* new_fluid_voice(fluid_real_t output_rate) { fluid_voice_t* voice; voice = FLUID_NEW(fluid_voice_t); if (voice == NULL) { FLUID_LOG(FLUID_ERR, "Out of memory"); return NULL; } voice->status = FLUID_VOICE_CLEAN; voice->chan = NO_CHANNEL; voice->key = 0; voice->vel = 0; voice->channel = NULL; voice->sample = NULL; voice->output_rate = output_rate; /* The 'sustain' and 'finished' segments of the volume / modulation * envelope are constant. They are never affected by any modulator * or generator. Therefore it is enough to initialize them once * during the lifetime of the synth. */ voice->volenv_data[FLUID_VOICE_ENVSUSTAIN].count = 0xffffffff; voice->volenv_data[FLUID_VOICE_ENVSUSTAIN].coeff = 1.0f; voice->volenv_data[FLUID_VOICE_ENVSUSTAIN].incr = 0.0f; voice->volenv_data[FLUID_VOICE_ENVSUSTAIN].min = -1.0f; voice->volenv_data[FLUID_VOICE_ENVSUSTAIN].max = 2.0f; voice->volenv_data[FLUID_VOICE_ENVFINISHED].count = 0xffffffff; voice->volenv_data[FLUID_VOICE_ENVFINISHED].coeff = 0.0f; voice->volenv_data[FLUID_VOICE_ENVFINISHED].incr = 0.0f; voice->volenv_data[FLUID_VOICE_ENVFINISHED].min = -1.0f; voice->volenv_data[FLUID_VOICE_ENVFINISHED].max = 1.0f; voice->modenv_data[FLUID_VOICE_ENVSUSTAIN].count = 0xffffffff; voice->modenv_data[FLUID_VOICE_ENVSUSTAIN].coeff = 1.0f; voice->modenv_data[FLUID_VOICE_ENVSUSTAIN].incr = 0.0f; voice->modenv_data[FLUID_VOICE_ENVSUSTAIN].min = -1.0f; voice->modenv_data[FLUID_VOICE_ENVSUSTAIN].max = 2.0f; voice->modenv_data[FLUID_VOICE_ENVFINISHED].count = 0xffffffff; voice->modenv_data[FLUID_VOICE_ENVFINISHED].coeff = 0.0f; voice->modenv_data[FLUID_VOICE_ENVFINISHED].incr = 0.0f; voice->modenv_data[FLUID_VOICE_ENVFINISHED].min = -1.0f; voice->modenv_data[FLUID_VOICE_ENVFINISHED].max = 1.0f; return voice; } /* * delete_fluid_voice */ int delete_fluid_voice(fluid_voice_t* voice) { if (voice == NULL) { return FLUID_OK; } FLUID_FREE(voice); return FLUID_OK; } /* fluid_voice_init * * Initialize the synthesis process */ int fluid_voice_init(fluid_voice_t* voice, fluid_sample_t* sample, fluid_channel_t* channel, int key, int vel, unsigned int id, unsigned int start_time, fluid_real_t gain) { /* Note: The voice parameters will be initialized later, when the * generators have been retrieved from the sound font. Here, only * the 'working memory' of the voice (position in envelopes, history * of IIR filters, position in sample etc) is initialized. */ voice->id = id; voice->chan = fluid_channel_get_num(channel); voice->key = (unsigned char) key; voice->vel = (unsigned char) vel; voice->channel = channel; voice->mod_count = 0; voice->sample = sample; voice->start_time = start_time; voice->ticks = 0; voice->debug = 0; voice->has_looped = 0; /* Will be set during voice_write when the 2nd loop point is reached */ voice->last_fres = -1; /* The filter coefficients have to be calculated later in the DSP loop. */ voice->filter_startup = 1; /* Set the filter immediately, don't fade between old and new settings */ voice->interp_method = fluid_channel_get_interp_method(voice->channel); /* vol env initialization */ voice->volenv_count = 0; voice->volenv_section = 0; voice->volenv_val = 0.0f; voice->amp = 0.0f; /* The last value of the volume envelope, used to calculate the volume increment during processing */ /* mod env initialization*/ voice->modenv_count = 0; voice->modenv_section = 0; voice->modenv_val = 0.0f; /* mod lfo */ voice->modlfo_val = 0.0;/* Fixme: Retrieve from any other existing voice on this channel to keep LFOs in unison? */ /* vib lfo */ voice->viblfo_val = 0.0f; /* Fixme: See mod lfo */ /* Clear sample history in filter */ voice->hist1 = 0; voice->hist2 = 0; /* Set all the generators to their default value, according to SF * 2.01 section 8.1.3 (page 48). The value of NRPN messages are * copied from the channel to the voice's generators. The sound font * loader overwrites them. The generator values are later converted * into voice parameters in * fluid_voice_calculate_runtime_synthesis_parameters. */ fluid_gen_init(&voice->gen[0], channel); voice->synth_gain = gain; /* avoid division by zero later*/ if (voice->synth_gain < 0.0000001){ voice->synth_gain = 0.0000001; } /* For a looped sample, this value will be overwritten as soon as the * loop parameters are initialized (they may depend on modulators). * This value can be kept, it is a worst-case estimate. */ voice->amplitude_that_reaches_noise_floor_nonloop = FLUID_NOISE_FLOOR / voice->synth_gain; voice->amplitude_that_reaches_noise_floor_loop = FLUID_NOISE_FLOOR / voice->synth_gain; /* Increment the reference count of the sample to prevent the unloading of the soundfont while this voice is playing. */ fluid_sample_incr_ref(voice->sample); return FLUID_OK; } void fluid_voice_gen_set(fluid_voice_t* voice, int i, float val) { voice->gen[i].val = val; voice->gen[i].flags = GEN_SET; } void fluid_voice_gen_incr(fluid_voice_t* voice, int i, float val) { voice->gen[i].val += val; voice->gen[i].flags = GEN_SET; } float fluid_voice_gen_get(fluid_voice_t* voice, int gen) { return voice->gen[gen].val; } fluid_real_t fluid_voice_gen_value(fluid_voice_t* voice, int num) { /* This is an extension to the SoundFont standard. More * documentation is available at the fluid_synth_set_gen2() * function. */ if (voice->gen[num].flags == GEN_ABS_NRPN) { return (fluid_real_t) voice->gen[num].nrpn; } else { return (fluid_real_t) (voice->gen[num].val + voice->gen[num].mod + voice->gen[num].nrpn); } } /* * fluid_voice_write * * This is where it all happens. This function is called by the * synthesizer to generate the sound samples. The synthesizer passes * four audio buffers: left, right, reverb out, and chorus out. * * The biggest part of this function sets the correct values for all * the dsp parameters (all the control data boil down to only a few * dsp parameters). The dsp routine is #included in several places (fluid_dsp_core.c). */ int fluid_voice_write(fluid_voice_t* voice, fluid_real_t* dsp_left_buf, fluid_real_t* dsp_right_buf, fluid_real_t* dsp_reverb_buf, fluid_real_t* dsp_chorus_buf) { unsigned int i; fluid_real_t incr; fluid_real_t fres; fluid_real_t target_amp; /* target amplitude */ int count; int dsp_interp_method = voice->interp_method; fluid_real_t dsp_buf[FLUID_BUFSIZE]; fluid_env_data_t* env_data; fluid_real_t x; /* make sure we're playing and that we have sample data */ if (!_PLAYING(voice)) return FLUID_OK; /******************* sample **********************/ if (voice->sample == NULL) { fluid_voice_off(voice); return FLUID_OK; } fluid_check_fpe ("voice_write startup"); /* Range checking for sample- and loop-related parameters * Initial phase is calculated here*/ fluid_voice_check_sample_sanity (voice); /******************* vol env **********************/ env_data = &voice->volenv_data[voice->volenv_section]; /* skip to the next section of the envelope if necessary */ while (voice->volenv_count >= env_data->count) { // If we're switching envelope stages from decay to sustain, force the value to be the end value of the previous stage if (env_data && voice->volenv_section == FLUID_VOICE_ENVDECAY) voice->volenv_val = env_data->min * env_data->coeff; env_data = &voice->volenv_data[++voice->volenv_section]; voice->volenv_count = 0; } /* calculate the envelope value and check for valid range */ x = env_data->coeff * voice->volenv_val + env_data->incr; if (x < env_data->min) { x = env_data->min; voice->volenv_section++; voice->volenv_count = 0; } else if (x > env_data->max) { x = env_data->max; voice->volenv_section++; voice->volenv_count = 0; } voice->volenv_val = x; voice->volenv_count++; if (voice->volenv_section == FLUID_VOICE_ENVFINISHED) { fluid_profile (FLUID_PROF_VOICE_RELEASE, voice->ref); fluid_voice_off (voice); return FLUID_OK; } fluid_check_fpe ("voice_write vol env"); /******************* mod env **********************/ env_data = &voice->modenv_data[voice->modenv_section]; /* skip to the next section of the envelope if necessary */ while (voice->modenv_count >= env_data->count) { env_data = &voice->modenv_data[++voice->modenv_section]; voice->modenv_count = 0; } /* calculate the envelope value and check for valid range */ x = env_data->coeff * voice->modenv_val + env_data->incr; if (x < env_data->min) { x = env_data->min; voice->modenv_section++; voice->modenv_count = 0; } else if (x > env_data->max) { x = env_data->max; voice->modenv_section++; voice->modenv_count = 0; } voice->modenv_val = x; voice->modenv_count++; fluid_check_fpe ("voice_write mod env"); /******************* mod lfo **********************/ if (voice->ticks >= voice->modlfo_delay) { voice->modlfo_val += voice->modlfo_incr; if (voice->modlfo_val > 1.0) { voice->modlfo_incr = -voice->modlfo_incr; voice->modlfo_val = (fluid_real_t) 2.0 - voice->modlfo_val; } else if (voice->modlfo_val < -1.0) { voice->modlfo_incr = -voice->modlfo_incr; voice->modlfo_val = (fluid_real_t) -2.0 - voice->modlfo_val; } } fluid_check_fpe ("voice_write mod LFO"); /******************* vib lfo **********************/ if (voice->ticks >= voice->viblfo_delay) { voice->viblfo_val += voice->viblfo_incr; if (voice->viblfo_val > (fluid_real_t) 1.0) { voice->viblfo_incr = -voice->viblfo_incr; voice->viblfo_val = (fluid_real_t) 2.0 - voice->viblfo_val; } else if (voice->viblfo_val < -1.0) { voice->viblfo_incr = -voice->viblfo_incr; voice->viblfo_val = (fluid_real_t) -2.0 - voice->viblfo_val; } } fluid_check_fpe ("voice_write Vib LFO"); /******************* amplitude **********************/ /* calculate final amplitude * - initial gain * - amplitude envelope */ if (voice->volenv_section == FLUID_VOICE_ENVDELAY) goto post_process; /* The volume amplitude is in hold phase. No sound is produced. */ if (voice->volenv_section == FLUID_VOICE_ENVATTACK) { /* the envelope is in the attack section: ramp linearly to max value. * A positive modlfo_to_vol should increase volume (negative attenuation). */ target_amp = fluid_atten2amp (voice->attenuation) * fluid_cb2amp (voice->modlfo_val * -voice->modlfo_to_vol) * voice->volenv_val; } else { fluid_real_t amplitude_that_reaches_noise_floor; fluid_real_t amp_max; target_amp = fluid_atten2amp (voice->attenuation) * fluid_cb2amp (960.0f * (1.0f - voice->volenv_val) + voice->modlfo_val * -voice->modlfo_to_vol); /* We turn off a voice, if the volume has dropped low enough. */ /* A voice can be turned off, when an estimate for the volume * (upper bound) falls below that volume, that will drop the * sample below the noise floor. */ /* If the loop amplitude is known, we can use it if the voice loop is within * the sample loop */ /* Is the playing pointer already in the loop? */ if (voice->has_looped) amplitude_that_reaches_noise_floor = voice->amplitude_that_reaches_noise_floor_loop; else amplitude_that_reaches_noise_floor = voice->amplitude_that_reaches_noise_floor_nonloop; /* voice->attenuation_min is a lower boundary for the attenuation * now and in the future (possibly 0 in the worst case). Now the * amplitude of sample and volenv cannot exceed amp_max (since * volenv_val can only drop): */ amp_max = fluid_atten2amp (voice->min_attenuation_cB) * voice->volenv_val; /* And if amp_max is already smaller than the known amplitude, * which will attenuate the sample below the noise floor, then we * can safely turn off the voice. Duh. */ if (amp_max < amplitude_that_reaches_noise_floor) { fluid_profile (FLUID_PROF_VOICE_RELEASE, voice->ref); fluid_voice_off (voice); goto post_process; } } /* Volume increment to go from voice->amp to target_amp in FLUID_BUFSIZE steps */ voice->amp_incr = (target_amp - voice->amp) / FLUID_BUFSIZE; fluid_check_fpe ("voice_write amplitude calculation"); /* no volume and not changing? - No need to process */ if ((voice->amp == 0.0f) && (voice->amp_incr == 0.0f)) goto post_process; /* Calculate the number of samples, that the DSP loop advances * through the original waveform with each step in the output * buffer. It is the ratio between the frequencies of original * waveform and output waveform.*/ voice->phase_incr = fluid_ct2hz_real (voice->pitch + voice->modlfo_val * voice->modlfo_to_pitch + voice->viblfo_val * voice->viblfo_to_pitch + voice->modenv_val * voice->modenv_to_pitch) / voice->root_pitch; fluid_check_fpe ("voice_write phase calculation"); /* if phase_incr is not advancing, set it to the minimum fraction value (prevent stuckage) */ if (voice->phase_incr == 0) voice->phase_incr = 1; /*************** resonant filter ******************/ /* calculate the frequency of the resonant filter in Hz */ fres = fluid_ct2hz(voice->fres + voice->modlfo_val * voice->modlfo_to_fc + voice->modenv_val * voice->modenv_to_fc); /* FIXME - Still potential for a click during turn on, can we interpolate between 20khz cutoff and 0 Q? */ /* I removed the optimization of turning the filter off when the * resonance frequence is above the maximum frequency. Instead, the * filter frequency is set to a maximum of 0.45 times the sampling * rate. For a 44100 kHz sampling rate, this amounts to 19845 * Hz. The reason is that there were problems with anti-aliasing when the * synthesizer was run at lower sampling rates. Thanks to Stephan * Tassart for pointing me to this bug. By turning the filter on and * clipping the maximum filter frequency at 0.45*srate, the filter * is used as an anti-aliasing filter. */ if (fres > 0.45f * voice->output_rate) fres = 0.45f * voice->output_rate; else if (fres < 5) fres = 5; /* if filter enabled and there is a significant frequency change.. */ if ((abs (fres - voice->last_fres) > 0.01)) { /* The filter coefficients have to be recalculated (filter * parameters have changed). Recalculation for various reasons is * forced by setting last_fres to -1. The flag filter_startup * indicates, that the DSP loop runs for the first time, in this * case, the filter is set directly, instead of smoothly fading * between old and new settings. * * Those equations from Robert Bristow-Johnson's `Cookbook * formulae for audio EQ biquad filter coefficients', obtained * from Harmony-central.com / Computer / Programming. They are * the result of the bilinear transform on an analogue filter * prototype. To quote, `BLT frequency warping has been taken * into account for both significant frequency relocation and for * bandwidth readjustment'. */ fluid_real_t omega = (fluid_real_t) (2.0 * M_PI * (fres / ((float) voice->output_rate))); fluid_real_t sin_coeff = (fluid_real_t) sin(omega); fluid_real_t cos_coeff = (fluid_real_t) cos(omega); fluid_real_t alpha_coeff = sin_coeff / (2.0f * voice->q_lin); fluid_real_t a0_inv = 1.0f / (1.0f + alpha_coeff); /* Calculate the filter coefficients. All coefficients are * normalized by a0. Think of `a1' as `a1/a0'. * * Here a couple of multiplications are saved by reusing common expressions. * The original equations should be: * voice->b0=(1.-cos_coeff)*a0_inv*0.5*voice->filter_gain; * voice->b1=(1.-cos_coeff)*a0_inv*voice->filter_gain; * voice->b2=(1.-cos_coeff)*a0_inv*0.5*voice->filter_gain; */ fluid_real_t a1_temp = -2.0f * cos_coeff * a0_inv; fluid_real_t a2_temp = (1.0f - alpha_coeff) * a0_inv; fluid_real_t b1_temp = (1.0f - cos_coeff) * a0_inv * voice->filter_gain; /* both b0 -and- b2 */ fluid_real_t b02_temp = b1_temp * 0.5f; if (voice->filter_startup) { /* The filter is calculated, because the voice was started up. * In this case set the filter coefficients without delay. */ voice->a1 = a1_temp; voice->a2 = a2_temp; voice->b02 = b02_temp; voice->b1 = b1_temp; voice->filter_coeff_incr_count = 0; voice->filter_startup = 0; // printf("Setting initial filter coefficients.\n"); } else { /* The filter frequency is changed. Calculate an increment * factor, so that the new setting is reached after one buffer * length. x_incr is added to the current value FLUID_BUFSIZE * times. The length is arbitrarily chosen. Longer than one * buffer will sacrifice some performance, though. Note: If * the filter is still too 'grainy', then increase this number * at will. */ #define FILTER_TRANSITION_SAMPLES (FLUID_BUFSIZE) voice->a1_incr = (a1_temp - voice->a1) / FILTER_TRANSITION_SAMPLES; voice->a2_incr = (a2_temp - voice->a2) / FILTER_TRANSITION_SAMPLES; voice->b02_incr = (b02_temp - voice->b02) / FILTER_TRANSITION_SAMPLES; voice->b1_incr = (b1_temp - voice->b1) / FILTER_TRANSITION_SAMPLES; /* Have to add the increments filter_coeff_incr_count times. */ voice->filter_coeff_incr_count = FILTER_TRANSITION_SAMPLES; } voice->last_fres = fres; fluid_check_fpe ("voice_write filter calculation"); } fluid_check_fpe ("voice_write DSP coefficients"); /*********************** run the dsp chain ************************ * The sample is mixed with the output buffer. * The buffer has to be filled from 0 to FLUID_BUFSIZE-1. * Depending on the position in the loop and the loop size, this * may require several runs. */ voice->dsp_buf = dsp_buf; switch (voice->interp_method) { case FLUID_INTERP_NONE: count = fluid_dsp_float_interpolate_none (voice); break; case FLUID_INTERP_LINEAR: count = fluid_dsp_float_interpolate_linear (voice); break; case FLUID_INTERP_4THORDER: default: count = fluid_dsp_float_interpolate_4th_order (voice); break; case FLUID_INTERP_7THORDER: count = fluid_dsp_float_interpolate_7th_order (voice); break; } fluid_check_fpe ("voice_write interpolation"); if (count > 0) fluid_voice_effects (voice, count, dsp_left_buf, dsp_right_buf, dsp_reverb_buf, dsp_chorus_buf); /* turn off voice if short count (sample ended and not looping) */ if (count < FLUID_BUFSIZE) { fluid_profile(FLUID_PROF_VOICE_RELEASE, voice->ref); fluid_voice_off(voice); } post_process: voice->ticks += FLUID_BUFSIZE; fluid_check_fpe ("voice_write postprocess"); return FLUID_OK; } /* Purpose: * * - filters (applies a lowpass filter with variable cutoff frequency and quality factor) * - mixes the processed sample to left and right output using the pan setting * - sends the processed sample to chorus and reverb * * Variable description: * - dsp_data: Pointer to the original waveform data * - dsp_left_buf: The generated signal goes here, left channel * - dsp_right_buf: right channel * - dsp_reverb_buf: Send to reverb unit * - dsp_chorus_buf: Send to chorus unit * - dsp_a1: Coefficient for the filter * - dsp_a2: same * - dsp_b0: same * - dsp_b1: same * - dsp_b2: same * - voice holds the voice structure * * A couple of variables are used internally, their results are discarded: * - dsp_i: Index through the output buffer * - dsp_phase_fractional: The fractional part of dsp_phase * - dsp_coeff: A table of four coefficients, depending on the fractional phase. * Used to interpolate between samples. * - dsp_process_buffer: Holds the processed signal between stages * - dsp_centernode: delay line for the IIR filter * - dsp_hist1: same * - dsp_hist2: same * */ static inline void fluid_voice_effects (fluid_voice_t *voice, int count, fluid_real_t* dsp_left_buf, fluid_real_t* dsp_right_buf, fluid_real_t* dsp_reverb_buf, fluid_real_t* dsp_chorus_buf) { /* IIR filter sample history */ fluid_real_t dsp_hist1 = voice->hist1; fluid_real_t dsp_hist2 = voice->hist2; /* IIR filter coefficients */ fluid_real_t dsp_a1 = voice->a1; fluid_real_t dsp_a2 = voice->a2; fluid_real_t dsp_b02 = voice->b02; fluid_real_t dsp_b1 = voice->b1; fluid_real_t dsp_a1_incr = voice->a1_incr; fluid_real_t dsp_a2_incr = voice->a2_incr; fluid_real_t dsp_b02_incr = voice->b02_incr; fluid_real_t dsp_b1_incr = voice->b1_incr; int dsp_filter_coeff_incr_count = voice->filter_coeff_incr_count; fluid_real_t *dsp_buf = voice->dsp_buf; fluid_real_t dsp_centernode; int dsp_i; float v; /* filter (implement the voice filter according to SoundFont standard) */ /* Check for denormal number (too close to zero). */ if (fabs (dsp_hist1) < 1e-20) dsp_hist1 = 0.0f; /* FIXME JMG - Is this even needed? */ /* Two versions of the filter loop. One, while the filter is * changing towards its new setting. The other, if the filter * doesn't change. */ if (dsp_filter_coeff_incr_count > 0) { /* Increment is added to each filter coefficient filter_coeff_incr_count times. */ for (dsp_i = 0; dsp_i < count; dsp_i++) { /* The filter is implemented in Direct-II form. */ dsp_centernode = dsp_buf[dsp_i] - dsp_a1 * dsp_hist1 - dsp_a2 * dsp_hist2; dsp_buf[dsp_i] = dsp_b02 * (dsp_centernode + dsp_hist2) + dsp_b1 * dsp_hist1; dsp_hist2 = dsp_hist1; dsp_hist1 = dsp_centernode; if (dsp_filter_coeff_incr_count-- > 0) { dsp_a1 += dsp_a1_incr; dsp_a2 += dsp_a2_incr; dsp_b02 += dsp_b02_incr; dsp_b1 += dsp_b1_incr; } } /* for dsp_i */ } else /* The filter parameters are constant. This is duplicated to save time. */ { for (dsp_i = 0; dsp_i < count; dsp_i++) { /* The filter is implemented in Direct-II form. */ dsp_centernode = dsp_buf[dsp_i] - dsp_a1 * dsp_hist1 - dsp_a2 * dsp_hist2; dsp_buf[dsp_i] = dsp_b02 * (dsp_centernode + dsp_hist2) + dsp_b1 * dsp_hist1; dsp_hist2 = dsp_hist1; dsp_hist1 = dsp_centernode; } } /* pan (Copy the signal to the left and right output buffer) The voice * panning generator has a range of -500 .. 500. If it is centered, * it's close to 0. voice->amp_left and voice->amp_right are then the * same, and we can save one multiplication per voice and sample. */ if ((-0.5 < voice->pan) && (voice->pan < 0.5)) { /* The voice is centered. Use voice->amp_left twice. */ for (dsp_i = 0; dsp_i < count; dsp_i++) { v = voice->amp_left * dsp_buf[dsp_i]; dsp_left_buf[dsp_i] += v; dsp_right_buf[dsp_i] += v; } } else /* The voice is not centered. Stereo samples have one side zero. */ { if (voice->amp_left != 0.0) { for (dsp_i = 0; dsp_i < count; dsp_i++) dsp_left_buf[dsp_i] += voice->amp_left * dsp_buf[dsp_i]; } if (voice->amp_right != 0.0) { for (dsp_i = 0; dsp_i < count; dsp_i++) dsp_right_buf[dsp_i] += voice->amp_right * dsp_buf[dsp_i]; } } /* reverb send. Buffer may be NULL. */ if ((dsp_reverb_buf != NULL) && (voice->amp_reverb != 0.0)) { for (dsp_i = 0; dsp_i < count; dsp_i++) dsp_reverb_buf[dsp_i] += voice->amp_reverb * dsp_buf[dsp_i]; } /* chorus send. Buffer may be NULL. */ if ((dsp_chorus_buf != NULL) && (voice->amp_chorus != 0)) { for (dsp_i = 0; dsp_i < count; dsp_i++) dsp_chorus_buf[dsp_i] += voice->amp_chorus * dsp_buf[dsp_i]; } voice->hist1 = dsp_hist1; voice->hist2 = dsp_hist2; voice->a1 = dsp_a1; voice->a2 = dsp_a2; voice->b02 = dsp_b02; voice->b1 = dsp_b1; voice->filter_coeff_incr_count = dsp_filter_coeff_incr_count; fluid_check_fpe ("voice_effects"); } /* * fluid_voice_get_channel */ fluid_channel_t* fluid_voice_get_channel(fluid_voice_t* voice) { return voice->channel; } /* * fluid_voice_start */ void fluid_voice_start(fluid_voice_t* voice) { /* The maximum volume of the loop is calculated and cached once for each * sample with its nominal loop settings. This happens, when the sample is used * for the first time.*/ fluid_voice_calculate_runtime_synthesis_parameters(voice); /* Force setting of the phase at the first DSP loop run * This cannot be done earlier, because it depends on modulators.*/ voice->check_sample_sanity_flag=FLUID_SAMPLESANITY_STARTUP; voice->ref = fluid_profile_ref(); voice->status = FLUID_VOICE_ON; } /* * fluid_voice_calculate_runtime_synthesis_parameters * * in this function we calculate the values of all the parameters. the * parameters are converted to their most useful unit for the DSP * algorithm, for example, number of samples instead of * timecents. Some parameters keep their "perceptual" unit and * conversion will be done in the DSP function. This is the case, for * example, for the pitch since it is modulated by the controllers in * cents. */ int fluid_voice_calculate_runtime_synthesis_parameters(fluid_voice_t* voice) { fluid_real_t x; fluid_real_t q_db; int i; int list_of_generators_to_initialize[35] = { GEN_STARTADDROFS, /* SF2.01 page 48 #0 */ GEN_ENDADDROFS, /* #1 */ GEN_STARTLOOPADDROFS, /* #2 */ GEN_ENDLOOPADDROFS, /* #3 */ /* GEN_STARTADDRCOARSEOFS see comment below [1] #4 */ GEN_MODLFOTOPITCH, /* #5 */ GEN_VIBLFOTOPITCH, /* #6 */ GEN_MODENVTOPITCH, /* #7 */ GEN_FILTERFC, /* #8 */ GEN_FILTERQ, /* #9 */ GEN_MODLFOTOFILTERFC, /* #10 */ GEN_MODENVTOFILTERFC, /* #11 */ /* GEN_ENDADDRCOARSEOFS [1] #12 */ GEN_MODLFOTOVOL, /* #13 */ /* not defined #14 */ GEN_CHORUSSEND, /* #15 */ GEN_REVERBSEND, /* #16 */ GEN_PAN, /* #17 */ /* not defined #18 */ /* not defined #19 */ /* not defined #20 */ GEN_MODLFODELAY, /* #21 */ GEN_MODLFOFREQ, /* #22 */ GEN_VIBLFODELAY, /* #23 */ GEN_VIBLFOFREQ, /* #24 */ GEN_MODENVDELAY, /* #25 */ GEN_MODENVATTACK, /* #26 */ GEN_MODENVHOLD, /* #27 */ GEN_MODENVDECAY, /* #28 */ /* GEN_MODENVSUSTAIN [1] #29 */ GEN_MODENVRELEASE, /* #30 */ /* GEN_KEYTOMODENVHOLD [1] #31 */ /* GEN_KEYTOMODENVDECAY [1] #32 */ GEN_VOLENVDELAY, /* #33 */ GEN_VOLENVATTACK, /* #34 */ GEN_VOLENVHOLD, /* #35 */ GEN_VOLENVDECAY, /* #36 */ /* GEN_VOLENVSUSTAIN [1] #37 */ GEN_VOLENVRELEASE, /* #38 */ /* GEN_KEYTOVOLENVHOLD [1] #39 */ /* GEN_KEYTOVOLENVDECAY [1] #40 */ /* GEN_STARTLOOPADDRCOARSEOFS [1] #45 */ GEN_KEYNUM, /* #46 */ GEN_VELOCITY, /* #47 */ GEN_ATTENUATION, /* #48 */ /* GEN_ENDLOOPADDRCOARSEOFS [1] #50 */ /* GEN_COARSETUNE [1] #51 */ /* GEN_FINETUNE [1] #52 */ GEN_OVERRIDEROOTKEY, /* #58 */ GEN_PITCH, /* --- */ -1}; /* end-of-list marker */ /* When the voice is made ready for the synthesis process, a lot of * voice-internal parameters have to be calculated. * * At this point, the sound font has already set the -nominal- value * for all generators (excluding GEN_PITCH). Most generators can be * modulated - they include a nominal value and an offset (which * changes with velocity, note number, channel parameters like * aftertouch, mod wheel...) Now this offset will be calculated as * follows: * * - Process each modulator once. * - Calculate its output value. * - Find the target generator. * - Add the output value to the modulation value of the generator. * * Note: The generators have been initialized with * fluid_gen_set_default_values. */ for (i = 0; i < voice->mod_count; i++) { fluid_mod_t* mod = &voice->mod[i]; fluid_real_t modval = fluid_mod_get_value(mod, voice->channel, voice); int dest_gen_index = mod->dest; fluid_gen_t* dest_gen = &voice->gen[dest_gen_index]; dest_gen->mod += modval; /* fluid_dump_modulator(mod); */ } /* The GEN_PITCH is a hack to fit the pitch bend controller into the * modulator paradigm. Now the nominal pitch of the key is set. * Note about SCALETUNE: SF2.01 8.1.3 says, that this generator is a * non-realtime parameter. So we don't allow modulation (as opposed * to _GEN(voice, GEN_SCALETUNE) When the scale tuning is varied, * one key remains fixed. Here C3 (MIDI number 60) is used. */ if (fluid_channel_has_tuning(voice->channel)) { /* pitch(60) + scale * (pitch(key) - pitch(60)) */ #define __pitch(_k) fluid_tuning_get_pitch(tuning, _k) fluid_tuning_t* tuning = fluid_channel_get_tuning(voice->channel); voice->gen[GEN_PITCH].val = (__pitch(60) + (voice->gen[GEN_SCALETUNE].val / 100.0f * (__pitch(voice->key) - __pitch(60)))); } else { voice->gen[GEN_PITCH].val = (voice->gen[GEN_SCALETUNE].val * (voice->key - 60.0f) + 100.0f * 60.0f); } /* Now the generators are initialized, nominal and modulation value. * The voice parameters (which depend on generators) are calculated * with fluid_voice_update_param. Processing the list of generator * changes will calculate each voice parameter once. * * Note [1]: Some voice parameters depend on several generators. For * example, the pitch depends on GEN_COARSETUNE, GEN_FINETUNE and * GEN_PITCH. voice->pitch. Unnecessary recalculation is avoided * by removing all but one generator from the list of voice * parameters. Same with GEN_XXX and GEN_XXXCOARSE: the * initialisation list contains only GEN_XXX. */ /* Calculate the voice parameter(s) dependent on each generator. */ for (i = 0; list_of_generators_to_initialize[i] != -1; i++) { fluid_voice_update_param(voice, list_of_generators_to_initialize[i]); } /* Make an estimate on how loud this voice can get at any time (attenuation). */ voice->min_attenuation_cB = fluid_voice_get_lower_boundary_for_attenuation(voice); return FLUID_OK; } /* * calculate_hold_decay_buffers */ int calculate_hold_decay_buffers(fluid_voice_t* voice, int gen_base, int gen_key2base, int is_decay) { /* Purpose: * * Returns the number of DSP loops, that correspond to the hold * (is_decay=0) or decay (is_decay=1) time. * gen_base=GEN_VOLENVHOLD, GEN_VOLENVDECAY, GEN_MODENVHOLD, * GEN_MODENVDECAY gen_key2base=GEN_KEYTOVOLENVHOLD, * GEN_KEYTOVOLENVDECAY, GEN_KEYTOMODENVHOLD, GEN_KEYTOMODENVDECAY */ fluid_real_t timecents; fluid_real_t seconds; int buffers; /* SF2.01 section 8.4.3 # 31, 32, 39, 40 * GEN_KEYTOxxxENVxxx uses key 60 as 'origin'. * The unit of the generator is timecents per key number. * If KEYTOxxxENVxxx is 100, a key one octave over key 60 (72) * will cause (60-72)*100=-1200 timecents of time variation. * The time is cut in half. */ timecents = (_GEN(voice, gen_base) + _GEN(voice, gen_key2base) * (60.0 - voice->key)); /* Range checking */ if (is_decay){ /* SF 2.01 section 8.1.3 # 28, 36 */ if (timecents > 8000.0) { timecents = 8000.0; } } else { /* SF 2.01 section 8.1.3 # 27, 35 */ if (timecents > 5000) { timecents = 5000.0; } /* SF 2.01 section 8.1.2 # 27, 35: * The most negative number indicates no hold time */ if (timecents <= -32768.) { return 0; } } /* SF 2.01 section 8.1.3 # 27, 28, 35, 36 */ if (timecents < -12000.0) { timecents = -12000.0; } seconds = fluid_tc2sec(timecents); /* Each DSP loop processes FLUID_BUFSIZE samples. */ /* round to next full number of buffers */ buffers = (int)(((fluid_real_t)voice->output_rate * seconds) / (fluid_real_t)FLUID_BUFSIZE +0.5); return buffers; } /* * fluid_voice_update_param * * Purpose: * * The value of a generator (gen) has changed. (The different * generators are listed in fluidsynth.h, or in SF2.01 page 48-49) * Now the dependent 'voice' parameters are calculated. * * fluid_voice_update_param can be called during the setup of the * voice (to calculate the initial value for a voice parameter), or * during its operation (a generator has been changed due to * real-time parameter modifications like pitch-bend). * * Note: The generator holds three values: The base value .val, an * offset caused by modulators .mod, and an offset caused by the * NRPN system. _GEN(voice, generator_enumerator) returns the sum * of all three. */ void fluid_voice_update_param(fluid_voice_t* voice, int gen) { double q_dB; fluid_real_t x; fluid_real_t y; unsigned int count; // Alternate attenuation scale used by EMU10K1 cards when setting the attenuation at the preset or instrument level within the SoundFont bank. static const float ALT_ATTENUATION_SCALE = 0.4; switch (gen) { case GEN_PAN: /* range checking is done in the fluid_pan function */ voice->pan = _GEN(voice, GEN_PAN); voice->amp_left = fluid_pan(voice->pan, 1) * voice->synth_gain / 32768.0f; voice->amp_right = fluid_pan(voice->pan, 0) * voice->synth_gain / 32768.0f; break; case GEN_ATTENUATION: voice->attenuation = ((fluid_real_t)(voice)->gen[GEN_ATTENUATION].val*ALT_ATTENUATION_SCALE) + (fluid_real_t)(voice)->gen[GEN_ATTENUATION].mod + (fluid_real_t)(voice)->gen[GEN_ATTENUATION].nrpn; /* Range: SF2.01 section 8.1.3 # 48 * Motivation for range checking: * OHPiano.SF2 sets initial attenuation to a whooping -96 dB */ fluid_clip(voice->attenuation, 0.0, 1440.0); break; /* The pitch is calculated from three different generators. * Read comment in fluidsynth.h about GEN_PITCH. */ case GEN_PITCH: case GEN_COARSETUNE: case GEN_FINETUNE: /* The testing for allowed range is done in 'fluid_ct2hz' */ voice->pitch = (_GEN(voice, GEN_PITCH) + 100.0f * _GEN(voice, GEN_COARSETUNE) + _GEN(voice, GEN_FINETUNE)); break; case GEN_REVERBSEND: /* The generator unit is 'tenths of a percent'. */ voice->reverb_send = _GEN(voice, GEN_REVERBSEND) / 1000.0f; fluid_clip(voice->reverb_send, 0.0, 1.0); voice->amp_reverb = voice->reverb_send * voice->synth_gain / 32768.0f; break; case GEN_CHORUSSEND: /* The generator unit is 'tenths of a percent'. */ voice->chorus_send = _GEN(voice, GEN_CHORUSSEND) / 1000.0f; fluid_clip(voice->chorus_send, 0.0, 1.0); voice->amp_chorus = voice->chorus_send * voice->synth_gain / 32768.0f; break; case GEN_OVERRIDEROOTKEY: /* This is a non-realtime parameter. Therefore the .mod part of the generator * can be neglected. * NOTE: origpitch sets MIDI root note while pitchadj is a fine tuning amount * which offsets the original rate. This means that the fine tuning is * inverted with respect to the root note (so subtract it, not add). */ if (voice->gen[GEN_OVERRIDEROOTKEY].val > -1) { //FIXME: use flag instead of -1 voice->root_pitch = voice->gen[GEN_OVERRIDEROOTKEY].val * 100.0f - voice->sample->pitchadj; } else { voice->root_pitch = voice->sample->origpitch * 100.0f - voice->sample->pitchadj; } voice->root_pitch = fluid_ct2hz(voice->root_pitch); if (voice->sample != NULL) { voice->root_pitch *= (fluid_real_t) voice->output_rate / voice->sample->samplerate; } break; case GEN_FILTERFC: /* The resonance frequency is converted from absolute cents to * midicents .val and .mod are both used, this permits real-time * modulation. The allowed range is tested in the 'fluid_ct2hz' * function [PH,20021214] */ voice->fres = _GEN(voice, GEN_FILTERFC); /* The synthesis loop will have to recalculate the filter * coefficients. */ voice->last_fres = -1.0f; break; case GEN_FILTERQ: /* The generator contains 'centibels' (1/10 dB) => divide by 10 to * obtain dB */ q_dB = _GEN(voice, GEN_FILTERQ) / 10.0f; /* Range: SF2.01 section 8.1.3 # 8 (convert from cB to dB => /10) */ fluid_clip(q_dB, 0.0f, 96.0f); /* Short version: Modify the Q definition in a way, that a Q of 0 * dB leads to no resonance hump in the freq. response. * * Long version: From SF2.01, page 39, item 9 (initialFilterQ): * "The gain at the cutoff frequency may be less than zero when * zero is specified". Assume q_dB=0 / q_lin=1: If we would leave * q as it is, then this results in a 3 dB hump slightly below * fc. At fc, the gain is exactly the DC gain (0 dB). What is * (probably) meant here is that the filter does not show a * resonance hump for q_dB=0. In this case, the corresponding * q_lin is 1/sqrt(2)=0.707. The filter should have 3 dB of * attenuation at fc now. In this case Q_dB is the height of the * resonance peak not over the DC gain, but over the frequency * response of a non-resonant filter. This idea is implemented as * follows: */ q_dB -= 3.01f; /* The 'sound font' Q is defined in dB. The filter needs a linear q. Convert. */ voice->q_lin = (fluid_real_t) (pow(10.0f, q_dB / 20.0f)); /* SF 2.01 page 59: * * The SoundFont specs ask for a gain reduction equal to half the * height of the resonance peak (Q). For example, for a 10 dB * resonance peak, the gain is reduced by 5 dB. This is done by * multiplying the total gain with sqrt(1/Q). `Sqrt' divides dB * by 2 (100 lin = 40 dB, 10 lin = 20 dB, 3.16 lin = 10 dB etc) * The gain is later factored into the 'b' coefficients * (numerator of the filter equation). This gain factor depends * only on Q, so this is the right place to calculate it. */ voice->filter_gain = (fluid_real_t) (1.0 / sqrt(voice->q_lin)); /* The synthesis loop will have to recalculate the filter coefficients. */ voice->last_fres = -1.; break; case GEN_MODLFOTOPITCH: voice->modlfo_to_pitch = _GEN(voice, GEN_MODLFOTOPITCH); fluid_clip(voice->modlfo_to_pitch, -12000.0, 12000.0); break; case GEN_MODLFOTOVOL: voice->modlfo_to_vol = _GEN(voice, GEN_MODLFOTOVOL); fluid_clip(voice->modlfo_to_vol, -960.0, 960.0); break; case GEN_MODLFOTOFILTERFC: voice->modlfo_to_fc = _GEN(voice, GEN_MODLFOTOFILTERFC); fluid_clip(voice->modlfo_to_fc, -12000, 12000); break; case GEN_MODLFODELAY: x = _GEN(voice, GEN_MODLFODELAY); fluid_clip(x, -12000.0f, 5000.0f); voice->modlfo_delay = (unsigned int) (voice->output_rate * fluid_tc2sec_delay(x)); break; case GEN_MODLFOFREQ: /* - the frequency is converted into a delta value, per buffer of FLUID_BUFSIZE samples * - the delay into a sample delay */ x = _GEN(voice, GEN_MODLFOFREQ); fluid_clip(x, -16000.0f, 4500.0f); voice->modlfo_incr = (4.0f * FLUID_BUFSIZE * fluid_act2hz(x) / voice->output_rate); break; case GEN_VIBLFOFREQ: /* vib lfo * * - the frequency is converted into a delta value, per buffer of FLUID_BUFSIZE samples * - the delay into a sample delay */ x = _GEN(voice, GEN_VIBLFOFREQ); fluid_clip(x, -16000.0f, 4500.0f); voice->viblfo_incr = (4.0f * FLUID_BUFSIZE * fluid_act2hz(x) / voice->output_rate); break; case GEN_VIBLFODELAY: x = _GEN(voice,GEN_VIBLFODELAY); fluid_clip(x, -12000.0f, 5000.0f); voice->viblfo_delay = (unsigned int) (voice->output_rate * fluid_tc2sec_delay(x)); break; case GEN_VIBLFOTOPITCH: voice->viblfo_to_pitch = _GEN(voice, GEN_VIBLFOTOPITCH); fluid_clip(voice->viblfo_to_pitch, -12000.0, 12000.0); break; case GEN_KEYNUM: /* GEN_KEYNUM: SF2.01 page 46, item 46 * * If this generator is active, it forces the key number to its * value. Non-realtime controller. * * There is a flag, which should indicate, whether a generator is * enabled or not. But here we rely on the default value of -1. * */ x = _GEN(voice, GEN_KEYNUM); if (x >= 0){ voice->key = x; } break; case GEN_VELOCITY: /* GEN_VELOCITY: SF2.01 page 46, item 47 * * If this generator is active, it forces the velocity to its * value. Non-realtime controller. * * There is a flag, which should indicate, whether a generator is * enabled or not. But here we rely on the default value of -1. */ x = _GEN(voice, GEN_VELOCITY); if (x > 0) { voice->vel = x; } break; case GEN_MODENVTOPITCH: voice->modenv_to_pitch = _GEN(voice, GEN_MODENVTOPITCH); fluid_clip(voice->modenv_to_pitch, -12000.0, 12000.0); break; case GEN_MODENVTOFILTERFC: voice->modenv_to_fc = _GEN(voice,GEN_MODENVTOFILTERFC); /* Range: SF2.01 section 8.1.3 # 1 * Motivation for range checking: * Filter is reported to make funny noises now and then */ fluid_clip(voice->modenv_to_fc, -12000.0, 12000.0); break; /* sample start and ends points * * Range checking is initiated via the * voice->check_sample_sanity flag, * because it is impossible to check here: * During the voice setup, all modulators are processed, while * the voice is inactive. Therefore, illegal settings may * occur during the setup (for example: First move the loop * end point ahead of the loop start point => invalid, then * move the loop start point forward => valid again. */ case GEN_STARTADDROFS: /* SF2.01 section 8.1.3 # 0 */ case GEN_STARTADDRCOARSEOFS: /* SF2.01 section 8.1.3 # 4 */ if (voice->sample != NULL) { voice->start = (voice->sample->start + (int) _GEN(voice, GEN_STARTADDROFS) + 32768 * (int) _GEN(voice, GEN_STARTADDRCOARSEOFS)); voice->check_sample_sanity_flag = FLUID_SAMPLESANITY_CHECK; } break; case GEN_ENDADDROFS: /* SF2.01 section 8.1.3 # 1 */ case GEN_ENDADDRCOARSEOFS: /* SF2.01 section 8.1.3 # 12 */ if (voice->sample != NULL) { voice->end = (voice->sample->end + (int) _GEN(voice, GEN_ENDADDROFS) + 32768 * (int) _GEN(voice, GEN_ENDADDRCOARSEOFS)); voice->check_sample_sanity_flag = FLUID_SAMPLESANITY_CHECK; } break; case GEN_STARTLOOPADDROFS: /* SF2.01 section 8.1.3 # 2 */ case GEN_STARTLOOPADDRCOARSEOFS: /* SF2.01 section 8.1.3 # 45 */ if (voice->sample != NULL) { voice->loopstart = (voice->sample->loopstart + (int) _GEN(voice, GEN_STARTLOOPADDROFS) + 32768 * (int) _GEN(voice, GEN_STARTLOOPADDRCOARSEOFS)); voice->check_sample_sanity_flag = FLUID_SAMPLESANITY_CHECK; } break; case GEN_ENDLOOPADDROFS: /* SF2.01 section 8.1.3 # 3 */ case GEN_ENDLOOPADDRCOARSEOFS: /* SF2.01 section 8.1.3 # 50 */ if (voice->sample != NULL) { voice->loopend = (voice->sample->loopend + (int) _GEN(voice, GEN_ENDLOOPADDROFS) + 32768 * (int) _GEN(voice, GEN_ENDLOOPADDRCOARSEOFS)); voice->check_sample_sanity_flag = FLUID_SAMPLESANITY_CHECK; } break; /* Conversion functions differ in range limit */ #define NUM_BUFFERS_DELAY(_v) (unsigned int) (voice->output_rate * fluid_tc2sec_delay(_v) / FLUID_BUFSIZE) #define NUM_BUFFERS_ATTACK(_v) (unsigned int) (voice->output_rate * fluid_tc2sec_attack(_v) / FLUID_BUFSIZE) #define NUM_BUFFERS_RELEASE(_v) (unsigned int) (voice->output_rate * fluid_tc2sec_release(_v) / FLUID_BUFSIZE) /* volume envelope * * - delay and hold times are converted to absolute number of samples * - sustain is converted to its absolute value * - attack, decay and release are converted to their increment per sample */ case GEN_VOLENVDELAY: /* SF2.01 section 8.1.3 # 33 */ x = _GEN(voice, GEN_VOLENVDELAY); fluid_clip(x, -12000.0f, 5000.0f); count = NUM_BUFFERS_DELAY(x); voice->volenv_data[FLUID_VOICE_ENVDELAY].count = count; voice->volenv_data[FLUID_VOICE_ENVDELAY].coeff = 0.0f; voice->volenv_data[FLUID_VOICE_ENVDELAY].incr = 0.0f; voice->volenv_data[FLUID_VOICE_ENVDELAY].min = -1.0f; voice->volenv_data[FLUID_VOICE_ENVDELAY].max = 1.0f; break; case GEN_VOLENVATTACK: /* SF2.01 section 8.1.3 # 34 */ x = _GEN(voice, GEN_VOLENVATTACK); fluid_clip(x, -12000.0f, 8000.0f); count = 1 + NUM_BUFFERS_ATTACK(x); voice->volenv_data[FLUID_VOICE_ENVATTACK].count = count; voice->volenv_data[FLUID_VOICE_ENVATTACK].coeff = 1.0f; voice->volenv_data[FLUID_VOICE_ENVATTACK].incr = count ? 1.0f / count : 0.0f; voice->volenv_data[FLUID_VOICE_ENVATTACK].min = -1.0f; voice->volenv_data[FLUID_VOICE_ENVATTACK].max = 1.0f; break; case GEN_VOLENVHOLD: /* SF2.01 section 8.1.3 # 35 */ case GEN_KEYTOVOLENVHOLD: /* SF2.01 section 8.1.3 # 39 */ count = calculate_hold_decay_buffers(voice, GEN_VOLENVHOLD, GEN_KEYTOVOLENVHOLD, 0); /* 0 means: hold */ voice->volenv_data[FLUID_VOICE_ENVHOLD].count = count; voice->volenv_data[FLUID_VOICE_ENVHOLD].coeff = 1.0f; voice->volenv_data[FLUID_VOICE_ENVHOLD].incr = 0.0f; voice->volenv_data[FLUID_VOICE_ENVHOLD].min = -1.0f; voice->volenv_data[FLUID_VOICE_ENVHOLD].max = 2.0f; break; case GEN_VOLENVDECAY: /* SF2.01 section 8.1.3 # 36 */ case GEN_VOLENVSUSTAIN: /* SF2.01 section 8.1.3 # 37 */ case GEN_KEYTOVOLENVDECAY: /* SF2.01 section 8.1.3 # 40 */ y = 1.0f - 0.001f * _GEN(voice, GEN_VOLENVSUSTAIN); fluid_clip(y, 0.0f, 1.0f); count = calculate_hold_decay_buffers(voice, GEN_VOLENVDECAY, GEN_KEYTOVOLENVDECAY, 1); /* 1 for decay */ voice->volenv_data[FLUID_VOICE_ENVDECAY].count = count; voice->volenv_data[FLUID_VOICE_ENVDECAY].coeff = 1.0f; voice->volenv_data[FLUID_VOICE_ENVDECAY].incr = count ? -1.0f / count : 0.0f; voice->volenv_data[FLUID_VOICE_ENVDECAY].min = y; voice->volenv_data[FLUID_VOICE_ENVDECAY].max = 2.0f; break; case GEN_VOLENVRELEASE: /* SF2.01 section 8.1.3 # 38 */ x = _GEN(voice, GEN_VOLENVRELEASE); fluid_clip(x, FLUID_MIN_VOLENVRELEASE, 8000.0f); count = 1 + NUM_BUFFERS_RELEASE(x); voice->volenv_data[FLUID_VOICE_ENVRELEASE].count = count; voice->volenv_data[FLUID_VOICE_ENVRELEASE].coeff = 1.0f; voice->volenv_data[FLUID_VOICE_ENVRELEASE].incr = count ? -1.0f / count : 0.0f; voice->volenv_data[FLUID_VOICE_ENVRELEASE].min = 0.0f; voice->volenv_data[FLUID_VOICE_ENVRELEASE].max = 1.0f; break; /* Modulation envelope */ case GEN_MODENVDELAY: /* SF2.01 section 8.1.3 # 25 */ x = _GEN(voice, GEN_MODENVDELAY); fluid_clip(x, -12000.0f, 5000.0f); voice->modenv_data[FLUID_VOICE_ENVDELAY].count = NUM_BUFFERS_DELAY(x); voice->modenv_data[FLUID_VOICE_ENVDELAY].coeff = 0.0f; voice->modenv_data[FLUID_VOICE_ENVDELAY].incr = 0.0f; voice->modenv_data[FLUID_VOICE_ENVDELAY].min = -1.0f; voice->modenv_data[FLUID_VOICE_ENVDELAY].max = 1.0f; break; case GEN_MODENVATTACK: /* SF2.01 section 8.1.3 # 26 */ x = _GEN(voice, GEN_MODENVATTACK); fluid_clip(x, -12000.0f, 8000.0f); count = 1 + NUM_BUFFERS_ATTACK(x); voice->modenv_data[FLUID_VOICE_ENVATTACK].count = count; voice->modenv_data[FLUID_VOICE_ENVATTACK].coeff = 1.0f; voice->modenv_data[FLUID_VOICE_ENVATTACK].incr = count ? 1.0f / count : 0.0f; voice->modenv_data[FLUID_VOICE_ENVATTACK].min = -1.0f; voice->modenv_data[FLUID_VOICE_ENVATTACK].max = 1.0f; break; case GEN_MODENVHOLD: /* SF2.01 section 8.1.3 # 27 */ case GEN_KEYTOMODENVHOLD: /* SF2.01 section 8.1.3 # 31 */ count = calculate_hold_decay_buffers(voice, GEN_MODENVHOLD, GEN_KEYTOMODENVHOLD, 0); /* 1 means: hold */ voice->modenv_data[FLUID_VOICE_ENVHOLD].count = count; voice->modenv_data[FLUID_VOICE_ENVHOLD].coeff = 1.0f; voice->modenv_data[FLUID_VOICE_ENVHOLD].incr = 0.0f; voice->modenv_data[FLUID_VOICE_ENVHOLD].min = -1.0f; voice->modenv_data[FLUID_VOICE_ENVHOLD].max = 2.0f; break; case GEN_MODENVDECAY: /* SF 2.01 section 8.1.3 # 28 */ case GEN_MODENVSUSTAIN: /* SF 2.01 section 8.1.3 # 29 */ case GEN_KEYTOMODENVDECAY: /* SF 2.01 section 8.1.3 # 32 */ count = calculate_hold_decay_buffers(voice, GEN_MODENVDECAY, GEN_KEYTOMODENVDECAY, 1); /* 1 for decay */ y = 1.0f - 0.001f * _GEN(voice, GEN_MODENVSUSTAIN); fluid_clip(y, 0.0f, 1.0f); voice->modenv_data[FLUID_VOICE_ENVDECAY].count = count; voice->modenv_data[FLUID_VOICE_ENVDECAY].coeff = 1.0f; voice->modenv_data[FLUID_VOICE_ENVDECAY].incr = count ? -1.0f / count : 0.0f; voice->modenv_data[FLUID_VOICE_ENVDECAY].min = y; voice->modenv_data[FLUID_VOICE_ENVDECAY].max = 2.0f; break; case GEN_MODENVRELEASE: /* SF 2.01 section 8.1.3 # 30 */ x = _GEN(voice, GEN_MODENVRELEASE); fluid_clip(x, -12000.0f, 8000.0f); count = 1 + NUM_BUFFERS_RELEASE(x); voice->modenv_data[FLUID_VOICE_ENVRELEASE].count = count; voice->modenv_data[FLUID_VOICE_ENVRELEASE].coeff = 1.0f; voice->modenv_data[FLUID_VOICE_ENVRELEASE].incr = count ? -1.0f / count : 0.0; voice->modenv_data[FLUID_VOICE_ENVRELEASE].min = 0.0f; voice->modenv_data[FLUID_VOICE_ENVRELEASE].max = 2.0f; break; } /* switch gen */ } /** * fluid_voice_modulate * * In this implementation, I want to make sure that all controllers * are event based: the parameter values of the DSP algorithm should * only be updates when a controller event arrived and not at every * iteration of the audio cycle (which would probably be feasible if * the synth was made in silicon). * * The update is done in three steps: * * - first, we look for all the modulators that have the changed * controller as a source. This will yield a list of generators that * will be changed because of the controller event. * * - For every changed generator, calculate its new value. This is the * sum of its original value plus the values of al the attached * modulators. * * - For every changed generator, convert its value to the correct * unit of the corresponding DSP parameter * * @fn int fluid_voice_modulate(fluid_voice_t* voice, int cc, int ctrl, int val) * @param voice the synthesis voice * @param cc flag to distinguish between a continous control and a channel control (pitch bend, ...) * @param ctrl the control number * */ int fluid_voice_modulate(fluid_voice_t* voice, int cc, int ctrl) { int i, k; fluid_mod_t* mod; int gen; fluid_real_t modval; /* printf("Chan=%d, CC=%d, Src=%d, Val=%d\n", voice->channel->channum, cc, ctrl, val); */ for (i = 0; i < voice->mod_count; i++) { mod = &voice->mod[i]; /* step 1: find all the modulators that have the changed controller * as input source. */ if (fluid_mod_has_source(mod, cc, ctrl)) { gen = fluid_mod_get_dest(mod); modval = 0.0; /* step 2: for every changed modulator, calculate the modulation * value of its associated generator */ for (k = 0; k < voice->mod_count; k++) { if (fluid_mod_has_dest(&voice->mod[k], gen)) { modval += fluid_mod_get_value(&voice->mod[k], voice->channel, voice); } } fluid_gen_set_mod(&voice->gen[gen], modval); /* step 3: now that we have the new value of the generator, * recalculate the parameter values that are derived from the * generator */ fluid_voice_update_param(voice, gen); } } return FLUID_OK; } /** * fluid_voice_modulate_all * * Update all the modulators. This function is called after a * ALL_CTRL_OFF MIDI message has been received (CC 121). * */ int fluid_voice_modulate_all(fluid_voice_t* voice) { fluid_mod_t* mod; int i, k, gen; fluid_real_t modval; /* Loop through all the modulators. FIXME: we should loop through the set of generators instead of the set of modulators. We risk to call 'fluid_voice_update_param' several times for the same generator if several modulators have that generator as destination. It's not an error, just a wast of energy (think polution, global warming, unhappy musicians, ...) */ for (i = 0; i < voice->mod_count; i++) { mod = &voice->mod[i]; gen = fluid_mod_get_dest(mod); modval = 0.0; /* Accumulate the modulation values of all the modulators with * destination generator 'gen' */ for (k = 0; k < voice->mod_count; k++) { if (fluid_mod_has_dest(&voice->mod[k], gen)) { modval += fluid_mod_get_value(&voice->mod[k], voice->channel, voice); } } fluid_gen_set_mod(&voice->gen[gen], modval); /* Update the parameter values that are depend on the generator * 'gen' */ fluid_voice_update_param(voice, gen); } return FLUID_OK; } /* * fluid_voice_noteoff */ int fluid_voice_noteoff(fluid_voice_t* voice) { fluid_profile(FLUID_PROF_VOICE_NOTE, voice->ref); if (voice->channel && fluid_channel_sustained(voice->channel)) { voice->status = FLUID_VOICE_SUSTAINED; } else { if (voice->volenv_section == FLUID_VOICE_ENVATTACK) { /* A voice is turned off during the attack section of the volume * envelope. The attack section ramps up linearly with * amplitude. The other sections use logarithmic scaling. Calculate new * volenv_val to achieve equievalent amplitude during the release phase * for seamless volume transition. */ if (voice->volenv_val > 0){ fluid_real_t lfo = voice->modlfo_val * -voice->modlfo_to_vol; fluid_real_t amp = voice->volenv_val * pow (10.0, lfo / -200); fluid_real_t env_value = - ((-200 * log (amp) / log (10.0) - lfo) / 960.0 - 1); fluid_clip (env_value, 0.0, 1.0); voice->volenv_val = env_value; } } voice->volenv_section = FLUID_VOICE_ENVRELEASE; voice->volenv_count = 0; voice->modenv_section = FLUID_VOICE_ENVRELEASE; voice->modenv_count = 0; } return FLUID_OK; } /* * fluid_voice_kill_excl * * Percussion sounds can be mutually exclusive: for example, a 'closed * hihat' sound will terminate an 'open hihat' sound ringing at the * same time. This behaviour is modeled using 'exclusive classes', * turning on a voice with an exclusive class other than 0 will kill * all other voices having that exclusive class within the same preset * or channel. fluid_voice_kill_excl gets called, when 'voice' is to * be killed for that reason. */ int fluid_voice_kill_excl(fluid_voice_t* voice){ if (!_PLAYING(voice)) { return FLUID_OK; } /* Turn off the exclusive class information for this voice, so that it doesn't get killed twice */ fluid_voice_gen_set(voice, GEN_EXCLUSIVECLASS, 0); /* If the voice is not yet in release state, put it into release state */ if (voice->volenv_section != FLUID_VOICE_ENVRELEASE){ voice->volenv_section = FLUID_VOICE_ENVRELEASE; voice->volenv_count = 0; voice->modenv_section = FLUID_VOICE_ENVRELEASE; voice->modenv_count = 0; } /* Speed up the volume envelope */ /* The value was found through listening tests with hi-hat samples. */ fluid_voice_gen_set(voice, GEN_VOLENVRELEASE, -200); fluid_voice_update_param(voice, GEN_VOLENVRELEASE); /* Speed up the modulation envelope */ fluid_voice_gen_set(voice, GEN_MODENVRELEASE, -200); fluid_voice_update_param(voice, GEN_MODENVRELEASE); return FLUID_OK; } /* * fluid_voice_off * * Purpose: * Turns off a voice, meaning that it is not processed * anymore by the DSP loop. */ int fluid_voice_off(fluid_voice_t* voice) { fluid_profile(FLUID_PROF_VOICE_RELEASE, voice->ref); voice->chan = NO_CHANNEL; voice->volenv_section = FLUID_VOICE_ENVFINISHED; voice->volenv_count = 0; voice->modenv_section = FLUID_VOICE_ENVFINISHED; voice->modenv_count = 0; voice->status = FLUID_VOICE_OFF; /* Decrement the reference count of the sample. */ if (voice->sample) { fluid_sample_decr_ref(voice->sample); voice->sample = NULL; } return FLUID_OK; } /* * fluid_voice_add_mod * * Adds a modulator to the voice. "mode" indicates, what to do, if * an identical modulator exists already. * * mode == FLUID_VOICE_ADD: Identical modulators on preset level are added * mode == FLUID_VOICE_OVERWRITE: Identical modulators on instrument level are overwritten * mode == FLUID_VOICE_DEFAULT: This is a default modulator, there can be no identical modulator. * Don't check. */ void fluid_voice_add_mod(fluid_voice_t* voice, fluid_mod_t* mod, int mode) { int i; /* * Some soundfonts come with a huge number of non-standard * controllers, because they have been designed for one particular * sound card. Discard them, maybe print a warning. */ if (((mod->flags1 & FLUID_MOD_CC) == 0) && ((mod->src1 != 0) /* SF2.01 section 8.2.1: Constant value */ && (mod->src1 != 2) /* Note-on velocity */ && (mod->src1 != 3) /* Note-on key number */ && (mod->src1 != 10) /* Poly pressure */ && (mod->src1 != 13) /* Channel pressure */ && (mod->src1 != 14) /* Pitch wheel */ && (mod->src1 != 16))) { /* Pitch wheel sensitivity */ FLUID_LOG(FLUID_WARN, "Ignoring invalid controller, using non-CC source %i.", mod->src1); return; } if (mode == FLUID_VOICE_ADD) { /* if identical modulator exists, add them */ for (i = 0; i < voice->mod_count; i++) { if (fluid_mod_test_identity(&voice->mod[i], mod)) { // printf("Adding modulator...\n"); voice->mod[i].amount += mod->amount; return; } } } else if (mode == FLUID_VOICE_OVERWRITE) { /* if identical modulator exists, replace it (only the amount has to be changed) */ for (i = 0; i < voice->mod_count; i++) { if (fluid_mod_test_identity(&voice->mod[i], mod)) { // printf("Replacing modulator...amount is %f\n",mod->amount); voice->mod[i].amount = mod->amount; return; } } } /* Add a new modulator (No existing modulator to add / overwrite). Also, default modulators (FLUID_VOICE_DEFAULT) are added without checking, if the same modulator already exists. */ if (voice->mod_count < FLUID_NUM_MOD) { fluid_mod_clone(&voice->mod[voice->mod_count++], mod); } } unsigned int fluid_voice_get_id(fluid_voice_t* voice) { return voice->id; } int fluid_voice_is_playing(fluid_voice_t* voice) { return _PLAYING(voice); } /* * fluid_voice_get_lower_boundary_for_attenuation * * Purpose: * * A lower boundary for the attenuation (as in 'the minimum * attenuation of this voice, with volume pedals, modulators * etc. resulting in minimum attenuation, cannot fall below x cB) is * calculated. This has to be called during fluid_voice_init, after * all modulators have been run on the voice once. Also, * voice->attenuation has to be initialized. */ fluid_real_t fluid_voice_get_lower_boundary_for_attenuation(fluid_voice_t* voice) { int i; fluid_mod_t* mod; fluid_real_t possible_att_reduction_cB=0; fluid_real_t lower_bound; for (i = 0; i < voice->mod_count; i++) { mod = &voice->mod[i]; /* Modulator has attenuation as target and can change over time? */ if ((mod->dest == GEN_ATTENUATION) && ((mod->flags1 & FLUID_MOD_CC) || (mod->flags2 & FLUID_MOD_CC))) { fluid_real_t current_val = fluid_mod_get_value(mod, voice->channel, voice); fluid_real_t v = fabs(mod->amount); if ((mod->src1 == FLUID_MOD_PITCHWHEEL) || (mod->flags1 & FLUID_MOD_BIPOLAR) || (mod->flags2 & FLUID_MOD_BIPOLAR) || (mod->amount < 0)) { /* Can this modulator produce a negative contribution? */ v *= -1.0; } else { /* No negative value possible. But still, the minimum contribution is 0. */ v = 0; } /* For example: * - current_val=100 * - min_val=-4000 * - possible_att_reduction_cB += 4100 */ if (current_val > v){ possible_att_reduction_cB += (current_val - v); } } } lower_bound = voice->attenuation-possible_att_reduction_cB; /* SF2.01 specs do not allow negative attenuation */ if (lower_bound < 0) { lower_bound = 0; } return lower_bound; } /* Purpose: * * Make sure, that sample start / end point and loop points are in * proper order. When starting up, calculate the initial phase. */ void fluid_voice_check_sample_sanity(fluid_voice_t* voice) { int min_index_nonloop=(int) voice->sample->start; int max_index_nonloop=(int) voice->sample->end; /* make sure we have enough samples surrounding the loop */ int min_index_loop=(int) voice->sample->start + FLUID_MIN_LOOP_PAD; int max_index_loop=(int) voice->sample->end - FLUID_MIN_LOOP_PAD + 1; /* 'end' is last valid sample, loopend can be + 1 */ fluid_check_fpe("voice_check_sample_sanity start"); if (!voice->check_sample_sanity_flag){ return; } #if 0 printf("Sample from %i to %i\n",voice->sample->start, voice->sample->end); printf("Sample loop from %i %i\n",voice->sample->loopstart, voice->sample->loopend); printf("Playback from %i to %i\n", voice->start, voice->end); printf("Playback loop from %i to %i\n",voice->loopstart, voice->loopend); #endif /* Keep the start point within the sample data */ if (voice->start < min_index_nonloop){ voice->start = min_index_nonloop; } else if (voice->start > max_index_nonloop){ voice->start = max_index_nonloop; } /* Keep the end point within the sample data */ if (voice->end < min_index_nonloop){ voice->end = min_index_nonloop; } else if (voice->end > max_index_nonloop){ voice->end = max_index_nonloop; } /* Keep start and end point in the right order */ if (voice->start > voice->end){ int temp = voice->start; voice->start = voice->end; voice->end = temp; /*FLUID_LOG(FLUID_DBG, "Loop / sample sanity check: Changing order of start / end points!"); */ } /* Zero length? */ if (voice->start == voice->end){ fluid_voice_off(voice); return; } if ((_SAMPLEMODE(voice) == FLUID_LOOP_UNTIL_RELEASE) || (_SAMPLEMODE(voice) == FLUID_LOOP_DURING_RELEASE)) { /* Keep the loop start point within the sample data */ if (voice->loopstart < min_index_loop){ voice->loopstart = min_index_loop; } else if (voice->loopstart > max_index_loop){ voice->loopstart = max_index_loop; } /* Keep the loop end point within the sample data */ if (voice->loopend < min_index_loop){ voice->loopend = min_index_loop; } else if (voice->loopend > max_index_loop){ voice->loopend = max_index_loop; } /* Keep loop start and end point in the right order */ if (voice->loopstart > voice->loopend){ int temp = voice->loopstart; voice->loopstart = voice->loopend; voice->loopend = temp; /*FLUID_LOG(FLUID_DBG, "Loop / sample sanity check: Changing order of loop points!"); */ } /* Loop too short? Then don't loop. */ if (voice->loopend < voice->loopstart + FLUID_MIN_LOOP_SIZE){ voice->gen[GEN_SAMPLEMODE].val = FLUID_UNLOOPED; } /* The loop points may have changed. Obtain a new estimate for the loop volume. */ /* Is the voice loop within the sample loop? */ if ((int)voice->loopstart >= (int)voice->sample->loopstart && (int)voice->loopend <= (int)voice->sample->loopend){ /* Is there a valid peak amplitude available for the loop? */ if (voice->sample->amplitude_that_reaches_noise_floor_is_valid){ voice->amplitude_that_reaches_noise_floor_loop=voice->sample->amplitude_that_reaches_noise_floor / voice->synth_gain; } else { /* Worst case */ voice->amplitude_that_reaches_noise_floor_loop=voice->amplitude_that_reaches_noise_floor_nonloop; }; }; } /* if sample mode is looped */ /* Run startup specific code (only once, when the voice is started) */ if (voice->check_sample_sanity_flag & FLUID_SAMPLESANITY_STARTUP){ if (max_index_loop - min_index_loop < FLUID_MIN_LOOP_SIZE){ if ((_SAMPLEMODE(voice) == FLUID_LOOP_UNTIL_RELEASE) || (_SAMPLEMODE(voice) == FLUID_LOOP_DURING_RELEASE)){ voice->gen[GEN_SAMPLEMODE].val = FLUID_UNLOOPED; } } /* Set the initial phase of the voice (using the result from the start offset modulators). */ fluid_phase_set_int(voice->phase, voice->start); } /* if startup */ /* Is this voice run in loop mode, or does it run straight to the end of the waveform data? */ if (((_SAMPLEMODE(voice) == FLUID_LOOP_UNTIL_RELEASE) && (voice->volenv_section < FLUID_VOICE_ENVRELEASE)) || (_SAMPLEMODE(voice) == FLUID_LOOP_DURING_RELEASE)) { /* Yes, it will loop as soon as it reaches the loop point. In * this case we must prevent, that the playback pointer (phase) * happens to end up beyond the 2nd loop point, because the * point has moved. The DSP algorithm is unable to cope with * that situation. So if the phase is beyond the 2nd loop * point, set it to the start of the loop. No way to avoid some * noise here. Note: If the sample pointer ends up -before the * first loop point- instead, then the DSP loop will just play * the sample, enter the loop and proceed as expected => no * actions required. */ int index_in_sample = fluid_phase_index(voice->phase); if (index_in_sample >= voice->loopend){ /* FLUID_LOG(FLUID_DBG, "Loop / sample sanity check: Phase after 2nd loop point!"); */ fluid_phase_set_int(voice->phase, voice->loopstart); } } /* FLUID_LOG(FLUID_DBG, "Loop / sample sanity check: Sample from %i to %i, loop from %i to %i", voice->start, voice->end, voice->loopstart, voice->loopend); */ /* Sample sanity has been assured. Don't check again, until some sample parameter is changed by modulation. */ voice->check_sample_sanity_flag=0; #if 0 printf("Sane? playback loop from %i to %i\n", voice->loopstart, voice->loopend); #endif fluid_check_fpe("voice_check_sample_sanity"); } int fluid_voice_set_param(fluid_voice_t* voice, int gen, fluid_real_t nrpn_value, int abs) { voice->gen[gen].nrpn = nrpn_value; voice->gen[gen].flags = (abs)? GEN_ABS_NRPN : GEN_SET; fluid_voice_update_param(voice, gen); return FLUID_OK; } int fluid_voice_set_gain(fluid_voice_t* voice, fluid_real_t gain) { /* avoid division by zero*/ if (gain < 0.0000001){ gain = 0.0000001; } voice->synth_gain = gain; voice->amp_left = fluid_pan(voice->pan, 1) * gain / 32768.0f; voice->amp_right = fluid_pan(voice->pan, 0) * gain / 32768.0f; voice->amp_reverb = voice->reverb_send * gain / 32768.0f; voice->amp_chorus = voice->chorus_send * gain / 32768.0f; return FLUID_OK; } /* - Scan the loop * - determine the peak level * - Calculate, what factor will make the loop inaudible * - Store in sample */ int fluid_voice_optimize_sample(fluid_sample_t* s) { signed short peak_max = 0; signed short peak_min = 0; signed short peak; fluid_real_t normalized_amplitude_during_loop; double result; int i; /* ignore ROM and other(?) invalid samples */ if (!s->valid) return (FLUID_OK); if (!s->amplitude_that_reaches_noise_floor_is_valid){ /* Only once */ /* Scan the loop */ for (i = (int)s->loopstart; i < (int) s->loopend; i ++){ signed short val = s->data[i]; if (val > peak_max) { peak_max = val; } else if (val < peak_min) { peak_min = val; } } /* Determine the peak level */ if (peak_max >- peak_min){ peak = peak_max; } else { peak =- peak_min; }; if (peak == 0){ /* Avoid division by zero */ peak = 1; }; /* Calculate what factor will make the loop inaudible * For example: Take a peak of 3277 (10 % of 32768). The * normalized amplitude is 0.1 (10 % of 32768). An amplitude * factor of 0.0001 (as opposed to the default 0.00001) will * drop this sample to the noise floor. */ /* 16 bits => 96+4=100 dB dynamic range => 0.00001 */ normalized_amplitude_during_loop = ((fluid_real_t)peak)/32768.; result = FLUID_NOISE_FLOOR / normalized_amplitude_during_loop; /* Store in sample */ s->amplitude_that_reaches_noise_floor = (double)result; s->amplitude_that_reaches_noise_floor_is_valid = 1; #if 0 printf("Sample peak detection: factor %f\n", (double)result); #endif }; return FLUID_OK; }