void
Spatializer_Module::process ( nframes_t nframes )
{
float azimuth = control_input[0].control_value();
float elevation = control_input[1].control_value();
float radius = control_input[2].control_value();
float highpass_freq = control_input[3].control_value();
float width = control_input[4].control_value();
float angle = control_input[5].control_value();
// bool more_options = control_input[6].control_value();
bool speed_of_sound = control_input[7].control_value() > 0.5f;
float late_gain = DB_CO( control_input[8].control_value() );
float early_gain = DB_CO( control_input[9].control_value() );
control_input[3].hints.visible = highpass_freq != 0.0f;
float delay_seconds = 0.0f;
if ( speed_of_sound && radius > 1.0f )
delay_seconds = ( radius - 1.0f ) / 340.29f;
/* direct sound follows inverse square law */
/* but it's just the inverse as far as SPL goes */
/* let's not go nuts... */
if ( radius < 0.01f )
radius = 0.01f;
float gain = 1.0f / radius;
/* float cutoff_frequency = gain * LOWPASS_FREQ; */
sample_t gainbuf[nframes];
sample_t delaybuf[nframes];
bool use_gainbuf = false;
bool use_delaybuf = delay_smoothing.apply( delaybuf, nframes, delay_seconds );
for ( unsigned int i = 0; i < audio_input.size(); i++ )
{
sample_t *buf = (sample_t*) audio_input[i].buffer();
/* frequency effects */
_highpass[i]->run_highpass( buf, highpass_freq, nframes );
/* send to late reverb */
if ( i == 0 )
buffer_copy( (sample_t*)aux_audio_output[0].jack_port()->buffer(nframes), buf, nframes );
else
buffer_mix( (sample_t*)aux_audio_output[0].jack_port()->buffer(nframes), buf, nframes );
}
{
use_gainbuf = late_gain_smoothing.apply( gainbuf, nframes, late_gain );
/* gain effects */
if ( unlikely( use_gainbuf ) )
buffer_apply_gain_buffer( (sample_t*)aux_audio_output[0].jack_port()->buffer(nframes), gainbuf, nframes );
else
buffer_apply_gain( (sample_t*)aux_audio_output[0].jack_port()->buffer(nframes), nframes, late_gain );
}
float early_angle = azimuth - angle;
if ( early_angle > 180.0f )
early_angle = -180 - ( early_angle - 180 );
else if ( early_angle < -180.0f )
early_angle = 180 - ( early_angle + 180 );
/* send to early reverb */
if ( audio_input.size() == 1 )
{
_early_panner->run_mono( (sample_t*)audio_input[0].buffer(),
(sample_t*)aux_audio_output[1].jack_port()->buffer(nframes),
(sample_t*)aux_audio_output[2].jack_port()->buffer(nframes),
(sample_t*)aux_audio_output[3].jack_port()->buffer(nframes),
(sample_t*)aux_audio_output[4].jack_port()->buffer(nframes),
azimuth + angle,
elevation,
nframes );
}
else
{
_early_panner->run_stereo( (sample_t*)audio_input[0].buffer(),
(sample_t*)audio_input[1].buffer(),
(sample_t*)aux_audio_output[1].jack_port()->buffer(nframes),
(sample_t*)aux_audio_output[2].jack_port()->buffer(nframes),
(sample_t*)aux_audio_output[3].jack_port()->buffer(nframes),
(sample_t*)aux_audio_output[4].jack_port()->buffer(nframes),
azimuth + angle,
elevation,
width,
nframes );
}
{
use_gainbuf = early_gain_smoothing.apply( gainbuf, nframes, early_gain );
for ( int i = 1; i < 5; i++ )
{
/* gain effects */
if ( unlikely( use_gainbuf ) )
buffer_apply_gain_buffer( (sample_t*)aux_audio_output[i].jack_port()->buffer(nframes), gainbuf, nframes );
else
buffer_apply_gain( (sample_t*)aux_audio_output[i].jack_port()->buffer(nframes), nframes, early_gain );
}
}
float corrected_angle = fabs( angle ) - (fabs( width ) * 0.5f);
if ( corrected_angle < 0.0f )
corrected_angle = 0.0f;
float cutoff_frequency = ( 1.0f / ( 1.0f + corrected_angle ) ) * 300000.0f;
use_gainbuf = gain_smoothing.apply( gainbuf, nframes, gain );
for ( unsigned int i = 0; i < audio_input.size(); i++ )
{
/* gain effects */
if ( unlikely( use_gainbuf ) )
buffer_apply_gain_buffer( (sample_t*)audio_input[i].buffer(), gainbuf, nframes );
else
buffer_apply_gain( (sample_t*)audio_input[i].buffer(), nframes, gain );
/* frequency effects */
_lowpass[i]->run_lowpass( (sample_t*)audio_input[i].buffer(), cutoff_frequency, nframes );
/* delay effects */
if ( likely( speed_of_sound ) )
{
if ( unlikely( use_delaybuf ) )
_delay[i]->run( (sample_t*)audio_input[i].buffer(), delaybuf, 0, nframes );
else
_delay[i]->run( (sample_t*)audio_input[i].buffer(), 0, delay_seconds, nframes );
}
}
/* now do direct outputs */
if ( audio_input.size() == 1 )
{
_panner->run_mono( (sample_t*)audio_input[0].buffer(),
(sample_t*)audio_output[0].buffer(),
(sample_t*)audio_output[1].buffer(),
(sample_t*)audio_output[2].buffer(),
(sample_t*)audio_output[3].buffer(),
azimuth,
elevation,
nframes );
}
else
{
_panner->run_stereo( (sample_t*)audio_input[0].buffer(),
(sample_t*)audio_input[1].buffer(),
(sample_t*)audio_output[0].buffer(),
(sample_t*)audio_output[1].buffer(),
(sample_t*)audio_output[2].buffer(),
(sample_t*)audio_output[3].buffer(),
azimuth,
elevation,
width,
nframes );
}
}
/* FIXME: it is far more efficient to read all the channels from a
multichannel source at once... But how should we handle the case of a
mismatch between the number of channels in this region's source and
the number of channels on the track/buffer this data is being read
for? Would it not be better to simply buffer and deinterlace the
frames in the Audio_File class instead, so that sequential requests
for different channels at the same position avoid hitting the disk
again? */
nframes_t
Audio_Region::read ( sample_t *buf, nframes_t pos, nframes_t nframes, int channel ) const
{
THREAD_ASSERT( Playback );
const Range r = _range;
/* do nothing if we aren't covered by this frame range */
if ( pos > r.start + r.length || pos + nframes < r.start )
return 0;
/* calculate offsets into file and sample buffer */
nframes_t sofs, /* offset into source */
ofs, /* offset into buffer */
cnt; /* number of frames to read */
cnt = nframes;
if ( pos < r.start )
{
/* region starts somewhere after the beginning of this buffer */
sofs = 0;
ofs = r.start - pos;
cnt -= ofs;
}
else
{
/* region started before this buffer */
ofs = 0;
sofs = pos - r.start;
}
if ( ofs >= nframes )
return 0;
// const nframes_t start = ofs + r.start + sofs;
const nframes_t start = r.offset + sofs;
const nframes_t len = cnt;
if ( len == 0 )
return 0;
/* now that we know how much and where to read, get on with it */
// printf( "reading region ofs = %lu, sofs = %lu, %lu-%lu\n", ofs, sofs, start, end );
/* FIXME: keep the declick defults someplace else */
Fade declick;
declick.length = 256;
declick.type = Fade::Sigmoid;
if ( _loop )
{
nframes_t lofs = sofs % _loop;
nframes_t lstart = r.offset + lofs;
if ( lofs + len > _loop )
{
/* this buffer covers a loop binary */
/* read the first part */
cnt = _clip->read( buf + ofs, channel, lstart, len - ( ( lofs + len ) - _loop ) );
/* read the second part */
cnt += _clip->read( buf + ofs + cnt, channel, lstart + cnt, len - cnt );
/* TODO: declick/crossfade transition? */
assert( cnt == len );
}
else
cnt = _clip->read( buf + ofs, channel, lstart, len );
/* this buffer is inside declicking proximity to the loop boundary */
if ( lofs + cnt + declick.length > _loop /* buffer ends within declick length of the end of loop */
&&
sofs + declick.length < r.length /* not the last loop */
)
{
/* */
/* fixme: what if loop is shorter than declick? */
const nframes_t declick_start = _loop - declick.length;
/* when the buffer covers the beginning of the
* declick, how many frames between the beginning of
* the buffer and the beginning of the declick */
const nframes_t declick_onset_offset = declick_start > lofs ? declick_start - lofs : 0;
/* how far into the declick we are */
const nframes_t declick_offset = lofs > declick_start ? lofs - declick_start : 0;
/* this is the end side of the loop boundary */
const nframes_t fl = cnt - declick_onset_offset;
declick.apply( buf + ofs + declick_onset_offset,
Fade::Out,
declick_offset, fl );
}
if ( lofs < declick.length /* buffer begins within declick length of beginning of loop */
&&
sofs > _loop ) /* not the first loop */
{
const nframes_t declick_end = declick.length;
const nframes_t click_len = lofs + cnt > declick_end ? declick_end - lofs : cnt;
/* this is the beginning of the loop next boundary */
declick.apply( buf + ofs, Fade::In, lofs, click_len );
}
}
else
cnt = _clip->read( buf + ofs, channel, start, len );
if ( ! cnt )
return 0;
/* apply gain */
buffer_apply_gain( buf + ofs, cnt, _scale );
/* perform declicking if necessary */
{
assert( cnt <= nframes );
Fade fade;
fade = declick < _fade_in ? _fade_in : declick;
/* do fade in if necessary */
if ( sofs < fade.length )
fade.apply( buf + ofs, Fade::In, sofs, cnt );
fade = declick < _fade_out ? _fade_out : declick;
/* do fade out if necessary */
if ( start + fade.length > r.offset + r.length )
fade.apply( buf, Fade::Out, ( start + fade.length ) - ( r.offset + r.length ), cnt );
}
return cnt;
}
