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livetrax/libs/fluidsynth/src/fluid_rvoice.c

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/* 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 Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free
* Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301, USA
*/
#include "fluid_rvoice.h"
#include "fluid_conv.h"
#include "fluid_sys.h"
static void fluid_rvoice_noteoff_LOCAL(fluid_rvoice_t *voice, unsigned int min_ticks);
/**
* @return -1 if voice is quiet, 0 if voice has finished, 1 otherwise
*/
static FLUID_INLINE int
fluid_rvoice_calc_amp(fluid_rvoice_t *voice)
{
fluid_real_t target_amp; /* target amplitude */
if(fluid_adsr_env_get_section(&voice->envlfo.volenv) == FLUID_VOICE_ENVDELAY)
{
return -1; /* The volume amplitude is in hold phase. No sound is produced. */
}
if(fluid_adsr_env_get_section(&voice->envlfo.volenv) == 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_cb2amp(voice->dsp.attenuation)
* fluid_cb2amp(fluid_lfo_get_val(&voice->envlfo.modlfo) * -voice->envlfo.modlfo_to_vol)
* fluid_adsr_env_get_val(&voice->envlfo.volenv);
}
else
{
fluid_real_t amplitude_that_reaches_noise_floor;
fluid_real_t amp_max;
target_amp = fluid_cb2amp(voice->dsp.attenuation)
* fluid_cb2amp(FLUID_PEAK_ATTENUATION * (1.0f - fluid_adsr_env_get_val(&voice->envlfo.volenv))
+ fluid_lfo_get_val(&voice->envlfo.modlfo) * -voice->envlfo.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->dsp.has_looped)
{
amplitude_that_reaches_noise_floor = voice->dsp.amplitude_that_reaches_noise_floor_loop;
}
else
{
amplitude_that_reaches_noise_floor = voice->dsp.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_cb2amp(voice->dsp.min_attenuation_cB) *
fluid_adsr_env_get_val(&voice->envlfo.volenv);
/* 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)
{
return 0;
}
}
/* Volume increment to go from voice->amp to target_amp in FLUID_BUFSIZE steps */
voice->dsp.amp_incr = (target_amp - voice->dsp.amp) / FLUID_BUFSIZE;
fluid_check_fpe("voice_write amplitude calculation");
/* no volume and not changing? - No need to process */
if((voice->dsp.amp == 0.0f) && (voice->dsp.amp_incr == 0.0f))
{
return -1;
}
return 1;
}
/* these should be the absolute minimum that FluidSynth can deal with */
#define FLUID_MIN_LOOP_SIZE 2
#define FLUID_MIN_LOOP_PAD 0
#define FLUID_SAMPLESANITY_CHECK (1 << 0)
#define FLUID_SAMPLESANITY_STARTUP (1 << 1)
/* Purpose:
*
* Make sure, that sample start / end point and loop points are in
* proper order. When starting up, calculate the initial phase.
* TODO: Investigate whether this can be moved from rvoice to voice.
*/
static void
fluid_rvoice_check_sample_sanity(fluid_rvoice_t *voice)
{
int min_index_nonloop = (int) voice->dsp.sample->start;
int max_index_nonloop = (int) voice->dsp.sample->end;
/* make sure we have enough samples surrounding the loop */
int min_index_loop = (int) voice->dsp.sample->start + FLUID_MIN_LOOP_PAD;
int max_index_loop = (int) voice->dsp.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 0
printf("Sample from %i to %i\n", voice->dsp.sample->start, voice->dsp.sample->end);
printf("Sample loop from %i %i\n", voice->dsp.sample->loopstart, voice->dsp.sample->loopend);
printf("Playback from %i to %i\n", voice->dsp.start, voice->dsp.end);
printf("Playback loop from %i to %i\n", voice->dsp.loopstart, voice->dsp.loopend);
#endif
/* Keep the start point within the sample data */
if(voice->dsp.start < min_index_nonloop)
{
voice->dsp.start = min_index_nonloop;
}
else if(voice->dsp.start > max_index_nonloop)
{
voice->dsp.start = max_index_nonloop;
}
/* Keep the end point within the sample data */
if(voice->dsp.end < min_index_nonloop)
{
voice->dsp.end = min_index_nonloop;
}
else if(voice->dsp.end > max_index_nonloop)
{
voice->dsp.end = max_index_nonloop;
}
/* Keep start and end point in the right order */
if(voice->dsp.start > voice->dsp.end)
{
int temp = voice->dsp.start;
voice->dsp.start = voice->dsp.end;
voice->dsp.end = temp;
/*FLUID_LOG(FLUID_DBG, "Loop / sample sanity check: Changing order of start / end points!"); */
}
/* Zero length? */
if(voice->dsp.start == voice->dsp.end)
{
fluid_rvoice_voiceoff(voice, NULL);
return;
}
if((voice->dsp.samplemode == FLUID_LOOP_UNTIL_RELEASE)
|| (voice->dsp.samplemode == FLUID_LOOP_DURING_RELEASE))
{
/* Keep the loop start point within the sample data */
if(voice->dsp.loopstart < min_index_loop)
{
voice->dsp.loopstart = min_index_loop;
}
else if(voice->dsp.loopstart > max_index_loop)
{
voice->dsp.loopstart = max_index_loop;
}
/* Keep the loop end point within the sample data */
if(voice->dsp.loopend < min_index_loop)
{
voice->dsp.loopend = min_index_loop;
}
else if(voice->dsp.loopend > max_index_loop)
{
voice->dsp.loopend = max_index_loop;
}
/* Keep loop start and end point in the right order */
if(voice->dsp.loopstart > voice->dsp.loopend)
{
int temp = voice->dsp.loopstart;
voice->dsp.loopstart = voice->dsp.loopend;
voice->dsp.loopend = temp;
/*FLUID_LOG(FLUID_DBG, "Loop / sample sanity check: Changing order of loop points!"); */
}
/* Loop too short? Then don't loop. */
if(voice->dsp.loopend < voice->dsp.loopstart + FLUID_MIN_LOOP_SIZE)
{
voice->dsp.samplemode = 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->dsp.loopstart >= (int)voice->dsp.sample->loopstart
&& (int)voice->dsp.loopend <= (int)voice->dsp.sample->loopend)
{
/* Is there a valid peak amplitude available for the loop, and can we use it? */
if(voice->dsp.sample->amplitude_that_reaches_noise_floor_is_valid && voice->dsp.samplemode == FLUID_LOOP_DURING_RELEASE)
{
voice->dsp.amplitude_that_reaches_noise_floor_loop = voice->dsp.sample->amplitude_that_reaches_noise_floor / voice->dsp.synth_gain;
}
else
{
/* Worst case */
voice->dsp.amplitude_that_reaches_noise_floor_loop = voice->dsp.amplitude_that_reaches_noise_floor_nonloop;
};
};
} /* if sample mode is looped */
/* Run startup specific code (only once, when the voice is started) */
if(voice->dsp.check_sample_sanity_flag & FLUID_SAMPLESANITY_STARTUP)
{
if(max_index_loop - min_index_loop < FLUID_MIN_LOOP_SIZE)
{
if((voice->dsp.samplemode == FLUID_LOOP_UNTIL_RELEASE)
|| (voice->dsp.samplemode == FLUID_LOOP_DURING_RELEASE))
{
voice->dsp.samplemode = FLUID_UNLOOPED;
}
}
/* Set the initial phase of the voice (using the result from the
start offset modulators). */
fluid_phase_set_int(voice->dsp.phase, voice->dsp.start);
} /* if startup */
/* Is this voice run in loop mode, or does it run straight to the
end of the waveform data? */
if(((voice->dsp.samplemode == FLUID_LOOP_UNTIL_RELEASE) &&
(fluid_adsr_env_get_section(&voice->envlfo.volenv) < FLUID_VOICE_ENVRELEASE))
|| (voice->dsp.samplemode == 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->dsp.phase);
if(index_in_sample >= voice->dsp.loopend)
{
/* FLUID_LOG(FLUID_DBG, "Loop / sample sanity check: Phase after 2nd loop point!"); */
fluid_phase_set_int(voice->dsp.phase, voice->dsp.loopstart);
}
}
/* FLUID_LOG(FLUID_DBG, "Loop / sample sanity check: Sample from %i to %i, loop from %i to %i", voice->dsp.start, voice->dsp.end, voice->dsp.loopstart, voice->dsp.loopend); */
/* Sample sanity has been assured. Don't check again, until some
sample parameter is changed by modulation. */
voice->dsp.check_sample_sanity_flag = 0;
#if 0
printf("Sane? playback loop from %i to %i\n", voice->dsp.loopstart, voice->dsp.loopend);
#endif
fluid_check_fpe("voice_check_sample_sanity");
}
/**
* Synthesize a voice to a buffer.
*
* @param voice rvoice to synthesize
* @param dsp_buf Audio buffer to synthesize to (#FLUID_BUFSIZE in length)
* @return Count of samples written to dsp_buf. (-1 means voice is currently
* quiet, 0 .. #FLUID_BUFSIZE-1 means voice finished.)
*
* Panning, reverb and chorus are processed separately. The dsp interpolation
* routine is in (fluid_rvoice_dsp.c).
*/
int
fluid_rvoice_write(fluid_rvoice_t *voice, fluid_real_t *dsp_buf)
{
int ticks = voice->envlfo.ticks;
int count, is_looping;
fluid_real_t modenv_val;
/******************* sample sanity check **********/
if(!voice->dsp.sample)
{
return 0;
}
if(voice->dsp.check_sample_sanity_flag)
{
fluid_rvoice_check_sample_sanity(voice);
}
/******************* noteoff check ****************/
if(voice->envlfo.noteoff_ticks != 0 &&
voice->envlfo.ticks >= voice->envlfo.noteoff_ticks)
{
fluid_rvoice_noteoff_LOCAL(voice, 0);
}
voice->envlfo.ticks += FLUID_BUFSIZE;
/******************* vol env **********************/
fluid_adsr_env_calc(&voice->envlfo.volenv, 1);
fluid_check_fpe("voice_write vol env");
if(fluid_adsr_env_get_section(&voice->envlfo.volenv) == FLUID_VOICE_ENVFINISHED)
{
return 0;
}
/******************* mod env **********************/
fluid_adsr_env_calc(&voice->envlfo.modenv, 0);
fluid_check_fpe("voice_write mod env");
/******************* lfo **********************/
fluid_lfo_calc(&voice->envlfo.modlfo, ticks);
fluid_check_fpe("voice_write mod LFO");
fluid_lfo_calc(&voice->envlfo.viblfo, ticks);
fluid_check_fpe("voice_write vib LFO");
/******************* amplitude **********************/
count = fluid_rvoice_calc_amp(voice);
if(count <= 0)
{
return count; /* return -1 if voice is quiet, 0 if voice has finished */
}
/******************* phase **********************/
/* SF2.04 section 8.1.2 #26:
* attack of modEnv is convex ?!?
*/
modenv_val = (fluid_adsr_env_get_section(&voice->envlfo.modenv) == FLUID_VOICE_ENVATTACK)
? fluid_convex(127 * fluid_adsr_env_get_val(&voice->envlfo.modenv))
: fluid_adsr_env_get_val(&voice->envlfo.modenv);
/* 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->dsp.phase_incr = fluid_ct2hz_real(voice->dsp.pitch +
voice->dsp.pitchoffset +
fluid_lfo_get_val(&voice->envlfo.modlfo) * voice->envlfo.modlfo_to_pitch
+ fluid_lfo_get_val(&voice->envlfo.viblfo) * voice->envlfo.viblfo_to_pitch
+ modenv_val * voice->envlfo.modenv_to_pitch)
/ voice->dsp.root_pitch_hz;
/******************* portamento ****************/
/* pitchoffset is updated if enabled.
Pitchoffset will be added to dsp pitch at next phase calculation time */
/* In most cases portamento will be disabled. Thus first verify that portamento is
* enabled before updating pitchoffset and before disabling portamento when necessary,
* in order to keep the performance loss at minimum.
* If the algorithm would first update pitchoffset and then verify if portamento
* needs to be disabled, there would be a significant performance drop on a x87 FPU
*/
if(voice->dsp.pitchinc > 0.0f)
{
/* portamento is enabled, so update pitchoffset */
voice->dsp.pitchoffset += voice->dsp.pitchinc;
/* when pitchoffset reaches 0.0f, portamento is disabled */
if(voice->dsp.pitchoffset > 0.0f)
{
voice->dsp.pitchoffset = voice->dsp.pitchinc = 0.0f;
}
}
else if(voice->dsp.pitchinc < 0.0f)
{
/* portamento is enabled, so update pitchoffset */
voice->dsp.pitchoffset += voice->dsp.pitchinc;
/* when pitchoffset reaches 0.0f, portamento is disabled */
if(voice->dsp.pitchoffset < 0.0f)
{
voice->dsp.pitchoffset = voice->dsp.pitchinc = 0.0f;
}
}
fluid_check_fpe("voice_write phase calculation");
/* if phase_incr is not advancing, set it to the minimum fraction value (prevent stuckage) */
if(voice->dsp.phase_incr == 0)
{
voice->dsp.phase_incr = 1;
}
/* voice is currently looping? */
is_looping = voice->dsp.samplemode == FLUID_LOOP_DURING_RELEASE
|| (voice->dsp.samplemode == FLUID_LOOP_UNTIL_RELEASE
&& fluid_adsr_env_get_section(&voice->envlfo.volenv) < FLUID_VOICE_ENVRELEASE);
/*********************** 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. */
switch(voice->dsp.interp_method)
{
case FLUID_INTERP_NONE:
count = fluid_rvoice_dsp_interpolate_none(&voice->dsp, dsp_buf, is_looping);
break;
case FLUID_INTERP_LINEAR:
count = fluid_rvoice_dsp_interpolate_linear(&voice->dsp, dsp_buf, is_looping);
break;
case FLUID_INTERP_4THORDER:
default:
count = fluid_rvoice_dsp_interpolate_4th_order(&voice->dsp, dsp_buf, is_looping);
break;
case FLUID_INTERP_7THORDER:
count = fluid_rvoice_dsp_interpolate_7th_order(&voice->dsp, dsp_buf, is_looping);
break;
}
fluid_check_fpe("voice_write interpolation");
if(count == 0)
{
return count;
}
/*************** resonant filter ******************/
fluid_iir_filter_calc(&voice->resonant_filter, voice->dsp.output_rate,
fluid_lfo_get_val(&voice->envlfo.modlfo) * voice->envlfo.modlfo_to_fc +
modenv_val * voice->envlfo.modenv_to_fc);
fluid_iir_filter_apply(&voice->resonant_filter, dsp_buf, count);
/* additional custom filter - only uses the fixed modulator, no lfos... */
fluid_iir_filter_calc(&voice->resonant_custom_filter, voice->dsp.output_rate, 0);
fluid_iir_filter_apply(&voice->resonant_custom_filter, dsp_buf, count);
return count;
}
/**
* Initialize buffers up to (and including) bufnum
*/
static int
fluid_rvoice_buffers_check_bufnum(fluid_rvoice_buffers_t *buffers, unsigned int bufnum)
{
unsigned int i;
if(bufnum < buffers->count)
{
return FLUID_OK;
}
if(bufnum >= FLUID_RVOICE_MAX_BUFS)
{
return FLUID_FAILED;
}
for(i = buffers->count; i <= bufnum; i++)
{
buffers->bufs[i].target_amp = 0.0f;
buffers->bufs[i].current_amp = 0.0f;
}
buffers->count = bufnum + 1;
return FLUID_OK;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_buffers_set_amp)
{
fluid_rvoice_buffers_t *buffers = obj;
unsigned int bufnum = param[0].i;
fluid_real_t value = param[1].real;
if(fluid_rvoice_buffers_check_bufnum(buffers, bufnum) != FLUID_OK)
{
return;
}
buffers->bufs[bufnum].target_amp = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_buffers_set_mapping)
{
fluid_rvoice_buffers_t *buffers = obj;
unsigned int bufnum = param[0].i;
int mapping = param[1].i;
if(fluid_rvoice_buffers_check_bufnum(buffers, bufnum) != FLUID_OK)
{
return;
}
buffers->bufs[bufnum].mapping = mapping;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_reset)
{
fluid_rvoice_t *voice = obj;
voice->dsp.has_looped = 0;
voice->envlfo.ticks = 0;
voice->envlfo.noteoff_ticks = 0;
voice->dsp.amp = 0.0f; /* The last value of the volume envelope, used to
calculate the volume increment during
processing */
/* legato initialization */
voice->dsp.pitchoffset = 0.0; /* portamento initialization */
voice->dsp.pitchinc = 0.0;
/* mod env initialization*/
fluid_adsr_env_reset(&voice->envlfo.modenv);
/* vol env initialization */
fluid_adsr_env_reset(&voice->envlfo.volenv);
/* Fixme: Retrieve from any other existing
voice on this channel to keep LFOs in
unison? */
fluid_lfo_reset(&voice->envlfo.viblfo);
fluid_lfo_reset(&voice->envlfo.modlfo);
/* Clear sample history in filter */
fluid_iir_filter_reset(&voice->resonant_filter);
fluid_iir_filter_reset(&voice->resonant_custom_filter);
/* Force setting of the phase at the first DSP loop run
* This cannot be done earlier, because it depends on modulators.
[DH] Is that comment really true? */
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_STARTUP;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_noteoff)
{
fluid_rvoice_t *rvoice = obj;
unsigned int min_ticks = param[0].i;
fluid_rvoice_noteoff_LOCAL(rvoice, min_ticks);
}
static void
fluid_rvoice_noteoff_LOCAL(fluid_rvoice_t *voice, unsigned int min_ticks)
{
if(min_ticks > voice->envlfo.ticks)
{
/* Delay noteoff */
voice->envlfo.noteoff_ticks = min_ticks;
return;
}
voice->envlfo.noteoff_ticks = 0;
if(fluid_adsr_env_get_section(&voice->envlfo.volenv) == 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 equivalent amplitude during the release phase
* for seamless volume transition.
*/
if(fluid_adsr_env_get_val(&voice->envlfo.volenv) > 0)
{
fluid_real_t lfo = fluid_lfo_get_val(&voice->envlfo.modlfo) * -voice->envlfo.modlfo_to_vol;
fluid_real_t amp = fluid_adsr_env_get_val(&voice->envlfo.volenv) * fluid_cb2amp(lfo);
fluid_real_t env_value = - (((-200.f / FLUID_M_LN10) * FLUID_LOGF(amp) - lfo) / FLUID_PEAK_ATTENUATION - 1);
fluid_clip(env_value, 0.0f, 1.0f);
fluid_adsr_env_set_val(&voice->envlfo.volenv, env_value);
}
}
if(fluid_adsr_env_get_section(&voice->envlfo.modenv) == FLUID_VOICE_ENVATTACK)
{
/* A voice is turned off during the attack section of the modulation
* envelope. The attack section use convex scaling with pitch and filter
* frequency cutoff (see fluid_rvoice_write(): modenv_val = fluid_convex(127 * modenv.val)
* The other sections use linear scaling: modenv_val = modenv.val
*
* Calculate new modenv.val to achieve equivalent modenv_val during the release phase
* for seamless pitch and filter frequency cutoff transition.
*/
if(fluid_adsr_env_get_val(&voice->envlfo.modenv) > 0)
{
fluid_real_t env_value = fluid_convex(127 * fluid_adsr_env_get_val(&voice->envlfo.modenv));
fluid_clip(env_value, 0.0, 1.0);
fluid_adsr_env_set_val(&voice->envlfo.modenv, env_value);
}
}
fluid_adsr_env_set_section(&voice->envlfo.volenv, FLUID_VOICE_ENVRELEASE);
fluid_adsr_env_set_section(&voice->envlfo.modenv, FLUID_VOICE_ENVRELEASE);
}
/**
* skips to Attack section
*
* Updates vol and attack data
* Correction on volume val to achieve equivalent amplitude at noteOn legato
*
* @param voice the synthesis voice to be updated
*/
static FLUID_INLINE void fluid_rvoice_local_retrigger_attack(fluid_rvoice_t *voice)
{
/* skips to Attack section */
/* Once in Attack section, current count must be reset, to be sure
that the section will be not be prematurely finished. */
fluid_adsr_env_set_section(&voice->envlfo.volenv, FLUID_VOICE_ENVATTACK);
{
/* Correction on volume val to achieve equivalent amplitude at noteOn legato */
fluid_env_data_t *env_data;
fluid_real_t peak = fluid_cb2amp(voice->dsp.attenuation);
fluid_real_t prev_peak = fluid_cb2amp(voice->dsp.prev_attenuation);
voice->envlfo.volenv.val = (voice->envlfo.volenv.val * prev_peak) / peak;
/* Correction on slope direction for Attack section */
env_data = &voice->envlfo.volenv.data[FLUID_VOICE_ENVATTACK];
if(voice->envlfo.volenv.val <= 1.0f)
{
/* slope attack for legato note needs to be positive from val up to 1 */
env_data->increment = 1.0f / env_data->count;
env_data->min = -1.0f;
env_data->max = 1.0f;
}
else
{
/* slope attack for legato note needs to be negative: from val down to 1 */
env_data->increment = -voice->envlfo.volenv.val / env_data->count;
env_data->min = 1.0f;
env_data->max = voice->envlfo.volenv.val;
}
}
}
/**
* Used by legato Mode : multi_retrigger
* see fluid_synth_noteon_mono_legato_multi_retrigger()
* @param voice the synthesis voice to be updated
*/
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_multi_retrigger_attack)
{
fluid_rvoice_t *voice = obj;
int section; /* volume or modulation section */
/*-------------------------------------------------------------------------
Section skip for volume envelope
--------------------------------------------------------------------------*/
section = fluid_adsr_env_get_section(&voice->envlfo.volenv);
if(section >= FLUID_VOICE_ENVHOLD)
{
/* DECAY, SUSTAIN,RELEASE section use logarithmic scaling. Calculates new
volenv_val to achieve equivalent amplitude during the attack phase
for seamless volume transition. */
fluid_real_t amp_cb, env_value;
amp_cb = FLUID_PEAK_ATTENUATION *
(1.0f - fluid_adsr_env_get_val(&voice->envlfo.volenv));
env_value = fluid_cb2amp(amp_cb); /* a bit of optimization */
fluid_clip(env_value, 0.0, 1.0);
fluid_adsr_env_set_val(&voice->envlfo.volenv, env_value);
/* next, skips to Attack section */
}
/* skips to Attack section from any section */
/* Update vol and attack data */
fluid_rvoice_local_retrigger_attack(voice);
/*-------------------------------------------------------------------------
Section skip for modulation envelope
--------------------------------------------------------------------------*/
section = fluid_adsr_env_get_section(&voice->envlfo.modenv);
if(section >= FLUID_VOICE_ENVHOLD)
{
/* DECAY, SUSTAIN,RELEASE section use linear scaling.
Since v 2.1 , as recommended by soundfont 2.01/2.4 spec, ATTACK section
uses convex shape (see fluid_rvoice_write() - fluid_convex()).
Calculate new modenv value (new_value) for seamless attack transition.
Here we need the inverse of fluid_convex() function defined as:
new_value = pow(10, (1 - current_val) . FLUID_PEAK_ATTENUATION / -200 . 2.0)
For performance reason we use fluid_cb2amp(Val) = pow(10, val/-200) with
val = (1 - current_val) . FLUID_PEAK_ATTENUATION / 2.0
*/
fluid_real_t new_value; /* new modenv value */
new_value = fluid_cb2amp((1.0f - fluid_adsr_env_get_val(&voice->envlfo.modenv))
* FLUID_PEAK_ATTENUATION / 2.0);
fluid_clip(new_value, 0.0, 1.0);
fluid_adsr_env_set_val(&voice->envlfo.modenv, new_value);
}
/* Skips from any section to ATTACK section */
fluid_adsr_env_set_section(&voice->envlfo.modenv, FLUID_VOICE_ENVATTACK);
}
/**
* sets the portamento dsp parameters: dsp.pitchoffset, dsp.pitchinc
* @param voice rvoice to set portamento.
* @param countinc increment count number.
* @param pitchoffset pitch offset to apply to voice dsp.pitch.
*
* Notes:
* 1) To get continuous portamento between consecutive noteOn (n1,n2,n3...),
* pitchoffset is accumulated in current dsp pitchoffset.
* 2) And to get constant portamento duration, dsp pitch increment is updated.
*/
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_portamento)
{
fluid_rvoice_t *voice = obj;
unsigned int countinc = param[0].i;
fluid_real_t pitchoffset = param[1].real;
if(countinc)
{
voice->dsp.pitchoffset += pitchoffset;
voice->dsp.pitchinc = - voice->dsp.pitchoffset / countinc;
}
/* Then during the voice processing (in fluid_rvoice_write()),
dsp.pitchoffset will be incremented by dsp pitchinc. */
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_output_rate)
{
fluid_rvoice_t *voice = obj;
fluid_real_t value = param[0].real;
voice->dsp.output_rate = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_interp_method)
{
fluid_rvoice_t *voice = obj;
int value = param[0].i;
voice->dsp.interp_method = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_root_pitch_hz)
{
fluid_rvoice_t *voice = obj;
fluid_real_t value = param[0].real;
voice->dsp.root_pitch_hz = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_pitch)
{
fluid_rvoice_t *voice = obj;
fluid_real_t value = param[0].real;
voice->dsp.pitch = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_attenuation)
{
fluid_rvoice_t *voice = obj;
fluid_real_t value = param[0].real;
voice->dsp.prev_attenuation = voice->dsp.attenuation;
voice->dsp.attenuation = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_min_attenuation_cB)
{
fluid_rvoice_t *voice = obj;
fluid_real_t value = param[0].real;
voice->dsp.min_attenuation_cB = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_viblfo_to_pitch)
{
fluid_rvoice_t *voice = obj;
fluid_real_t value = param[0].real;
voice->envlfo.viblfo_to_pitch = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_modlfo_to_pitch)
{
fluid_rvoice_t *voice = obj;
fluid_real_t value = param[0].real;
voice->envlfo.modlfo_to_pitch = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_modlfo_to_vol)
{
fluid_rvoice_t *voice = obj;
fluid_real_t value = param[0].real;
voice->envlfo.modlfo_to_vol = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_modlfo_to_fc)
{
fluid_rvoice_t *voice = obj;
fluid_real_t value = param[0].real;
voice->envlfo.modlfo_to_fc = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_modenv_to_fc)
{
fluid_rvoice_t *voice = obj;
fluid_real_t value = param[0].real;
voice->envlfo.modenv_to_fc = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_modenv_to_pitch)
{
fluid_rvoice_t *voice = obj;
fluid_real_t value = param[0].real;
voice->envlfo.modenv_to_pitch = value;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_synth_gain)
{
fluid_rvoice_t *voice = obj;
fluid_real_t value = param[0].real;
voice->dsp.synth_gain = value;
/* 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->dsp.amplitude_that_reaches_noise_floor_nonloop = FLUID_NOISE_FLOOR / value;
voice->dsp.amplitude_that_reaches_noise_floor_loop = FLUID_NOISE_FLOOR / value;
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_CHECK;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_start)
{
fluid_rvoice_t *voice = obj;
int value = param[0].i;
voice->dsp.start = value;
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_CHECK;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_end)
{
fluid_rvoice_t *voice = obj;
int value = param[0].i;
voice->dsp.end = value;
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_CHECK;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_loopstart)
{
fluid_rvoice_t *voice = obj;
int value = param[0].i;
voice->dsp.loopstart = value;
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_CHECK;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_loopend)
{
fluid_rvoice_t *voice = obj;
int value = param[0].i;
voice->dsp.loopend = value;
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_CHECK;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_samplemode)
{
fluid_rvoice_t *voice = obj;
enum fluid_loop value = param[0].i;
voice->dsp.samplemode = value;
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_CHECK;
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_set_sample)
{
fluid_rvoice_t *voice = obj;
fluid_sample_t *value = param[0].ptr;
voice->dsp.sample = value;
if(value)
{
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_STARTUP;
}
}
DECLARE_FLUID_RVOICE_FUNCTION(fluid_rvoice_voiceoff)
{
fluid_rvoice_t *voice = obj;
fluid_adsr_env_set_section(&voice->envlfo.volenv, FLUID_VOICE_ENVFINISHED);
fluid_adsr_env_set_section(&voice->envlfo.modenv, FLUID_VOICE_ENVFINISHED);
}
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