ardour/livetrax
13
0
Fork 0
Code
155401b875
livetrax/libs/fluidsynth/src/fluid_rvoice.c

665 lines
22 KiB
C
Raw Blame History

/* 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
* 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"
/**
* @return -1 if voice has finished, 0 if it's currently quiet, 1 otherwise
*/
static 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_atten2amp (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_atten2amp (voice->dsp.attenuation)
* fluid_cb2amp (960.0f * (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_atten2amp (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 (!voice->dsp.check_sample_sanity_flag){
return;
}
#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);
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_dsp_float.c).
*/
int
fluid_rvoice_write (fluid_rvoice_t* voice, fluid_real_t *dsp_buf)
{
int ticks = voice->envlfo.ticks;
int count;
/******************* 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(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;
/******************* phase **********************/
/* 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 +
fluid_lfo_get_val(&voice->envlfo.modlfo) * voice->envlfo.modlfo_to_pitch
+ fluid_lfo_get_val(&voice->envlfo.viblfo) * voice->envlfo.viblfo_to_pitch
+ fluid_adsr_env_get_val(&voice->envlfo.modenv) * voice->envlfo.modenv_to_pitch)
/ voice->dsp.root_pitch_hz;
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->dsp.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. */
voice->dsp.dsp_buf = dsp_buf;
switch (voice->dsp.interp_method)
{
case FLUID_INTERP_NONE:
count = fluid_rvoice_dsp_interpolate_none (&voice->dsp);
break;
case FLUID_INTERP_LINEAR:
count = fluid_rvoice_dsp_interpolate_linear (&voice->dsp);
break;
case FLUID_INTERP_4THORDER:
default:
count = fluid_rvoice_dsp_interpolate_4th_order (&voice->dsp);
break;
case FLUID_INTERP_7THORDER:
count = fluid_rvoice_dsp_interpolate_7th_order (&voice->dsp);
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 +
fluid_adsr_env_get_val(&voice->envlfo.modenv) * voice->envlfo.modenv_to_fc);
fluid_iir_filter_apply(&voice->resonant_filter, dsp_buf, count);
return count;
}
static inline fluid_real_t*
get_dest_buf(fluid_rvoice_buffers_t* buffers, int index,
fluid_real_t** dest_bufs, int dest_bufcount)
{
int j = buffers->bufs[index].mapping;
if (j >= dest_bufcount || j < 0) return NULL;
return dest_bufs[j];
}
/**
* Mix data down to buffers
*
* @param buffers Destination buffer(s)
* @param dsp_buf Mono sample source
* @param samplecount Number of samples to process (no FLUID_BUFSIZE restriction)
* @param dest_bufs Array of buffers to mixdown to
* @param dest_bufcount Length of dest_bufs
*/
void
fluid_rvoice_buffers_mix(fluid_rvoice_buffers_t* buffers,
fluid_real_t* dsp_buf, int samplecount,
fluid_real_t** dest_bufs, int dest_bufcount)
{
int bufcount = buffers->count;
int i, dsp_i;
if (!samplecount || !bufcount || !dest_bufcount)
return;
for (i=0; i < bufcount; i++) {
fluid_real_t* buf = get_dest_buf(buffers, i, dest_bufs, dest_bufcount);
fluid_real_t* next_buf;
fluid_real_t amp = buffers->bufs[i].amp;
if (buf == NULL || amp == 0.0f)
continue;
/* Optimization for centered stereo samples - we can save one
multiplication per sample */
next_buf = (i+1 >= bufcount ? NULL : get_dest_buf(buffers, i+1, dest_bufs, dest_bufcount));
if (next_buf && buffers->bufs[i+1].amp == amp) {
for (dsp_i = 0; dsp_i < samplecount; dsp_i++) {
fluid_real_t samp = amp * dsp_buf[dsp_i];
buf[dsp_i] += samp;
next_buf[dsp_i] += samp;
}
i++;
}
else {
for (dsp_i = 0; dsp_i < samplecount; dsp_i++)
buf[dsp_i] += amp * dsp_buf[dsp_i];
}
}
}
/**
* 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[bufnum].amp = 0.0f;
buffers->bufs[bufnum].mapping = i;
}
buffers->count = bufnum+1;
return FLUID_OK;
}
void
fluid_rvoice_buffers_set_amp(fluid_rvoice_buffers_t* buffers,
unsigned int bufnum, fluid_real_t value)
{
if (fluid_rvoice_buffers_check_bufnum(buffers, bufnum) != FLUID_OK)
return;
buffers->bufs[bufnum].amp = value;
}
void
fluid_rvoice_buffers_set_mapping(fluid_rvoice_buffers_t* buffers,
unsigned int bufnum, int mapping)
{
if (fluid_rvoice_buffers_check_bufnum(buffers, bufnum) != FLUID_OK)
return;
buffers->bufs[bufnum].mapping = mapping;
}
void
fluid_rvoice_reset(fluid_rvoice_t* voice)
{
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 */
/* 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);
/* 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;
}
void
fluid_rvoice_noteoff(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 equievalent 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) * 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);
fluid_adsr_env_set_val(&voice->envlfo.volenv, env_value);
}
}
fluid_adsr_env_set_section(&voice->envlfo.volenv, FLUID_VOICE_ENVRELEASE);
fluid_adsr_env_set_section(&voice->envlfo.modenv, FLUID_VOICE_ENVRELEASE);
}
void
fluid_rvoice_set_output_rate(fluid_rvoice_t* voice, fluid_real_t value)
{
voice->dsp.output_rate = value;
}
void
fluid_rvoice_set_interp_method(fluid_rvoice_t* voice, int value)
{
voice->dsp.interp_method = value;
}
void
fluid_rvoice_set_root_pitch_hz(fluid_rvoice_t* voice, fluid_real_t value)
{
voice->dsp.root_pitch_hz = value;
}
void
fluid_rvoice_set_pitch(fluid_rvoice_t* voice, fluid_real_t value)
{
voice->dsp.pitch = value;
}
void
fluid_rvoice_set_attenuation(fluid_rvoice_t* voice, fluid_real_t value)
{
voice->dsp.attenuation = value;
}
void
fluid_rvoice_set_min_attenuation_cB(fluid_rvoice_t* voice, fluid_real_t value)
{
voice->dsp.min_attenuation_cB = value;
}
void
fluid_rvoice_set_viblfo_to_pitch(fluid_rvoice_t* voice, fluid_real_t value)
{
voice->envlfo.viblfo_to_pitch = value;
}
void fluid_rvoice_set_modlfo_to_pitch(fluid_rvoice_t* voice, fluid_real_t value)
{
voice->envlfo.modlfo_to_pitch = value;
}
void
fluid_rvoice_set_modlfo_to_vol(fluid_rvoice_t* voice, fluid_real_t value)
{
voice->envlfo.modlfo_to_vol = value;
}
void
fluid_rvoice_set_modlfo_to_fc(fluid_rvoice_t* voice, fluid_real_t value)
{
voice->envlfo.modlfo_to_fc = value;
}
void
fluid_rvoice_set_modenv_to_fc(fluid_rvoice_t* voice, fluid_real_t value)
{
voice->envlfo.modenv_to_fc = value;
}
void
fluid_rvoice_set_modenv_to_pitch(fluid_rvoice_t* voice, fluid_real_t value)
{
voice->envlfo.modenv_to_pitch = value;
}
void
fluid_rvoice_set_synth_gain(fluid_rvoice_t* voice, fluid_real_t value)
{
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;
}
void
fluid_rvoice_set_start(fluid_rvoice_t* voice, int value)
{
voice->dsp.start = value;
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_CHECK;
}
void
fluid_rvoice_set_end(fluid_rvoice_t* voice, int value)
{
voice->dsp.end = value;
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_CHECK;
}
void
fluid_rvoice_set_loopstart(fluid_rvoice_t* voice, int value)
{
voice->dsp.loopstart = value;
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_CHECK;
}
void fluid_rvoice_set_loopend(fluid_rvoice_t* voice, int value)
{
voice->dsp.loopend = value;
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_CHECK;
}
void fluid_rvoice_set_samplemode(fluid_rvoice_t* voice, enum fluid_loop value)
{
voice->dsp.samplemode = value;
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_CHECK;
}
void
fluid_rvoice_set_sample(fluid_rvoice_t* voice, fluid_sample_t* value)
{
voice->dsp.sample = value;
if (value) {
voice->dsp.check_sample_sanity_flag |= FLUID_SAMPLESANITY_STARTUP;
}
}
void
fluid_rvoice_voiceoff(fluid_rvoice_t* voice)
{
fluid_adsr_env_set_section(&voice->envlfo.volenv, FLUID_VOICE_ENVFINISHED);
fluid_adsr_env_set_section(&voice->envlfo.modenv, FLUID_VOICE_ENVFINISHED);
}
View Git Blame Copy Permalink