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Added missing changes to the OpenAL-Soft update.

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This commit is contained in:
Alex Szpakowski
2015-12-01 13:40:34 -04:00
parent f8dae3ea09
commit e9d77ef766
94 changed files with 9219 additions and 6099 deletions
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libs/openal-soft/Alc/effects/autowah.c
+12 -14
View File
@@ -13,8 +13,8 @@
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
@@ -31,15 +31,15 @@
/* Auto-wah is simply a low-pass filter with a cutoff frequency that shifts up
* or down depending on the input signal, and a resonant peak at the cutoff.
*
* Currently, we assume a cutoff frequency range of 500hz (no amplitude) to
* 3khz (peak gain). Peak gain is assumed to be in normalized scale.
* Currently, we assume a cutoff frequency range of 20hz (no amplitude) to
* 20khz (peak gain). Peak gain is assumed to be in normalized scale.
*/
typedef struct ALautowahState {
DERIVE_FROM_TYPE(ALeffectState);
/* Effect gains for each channel */
ALfloat Gain[MaxChannels];
ALfloat Gain[MAX_OUTPUT_CHANNELS];
/* Effect parameters */
ALfloat AttackRate;
@@ -66,7 +66,6 @@ static ALboolean ALautowahState_deviceUpdate(ALautowahState *state, ALCdevice *d
static ALvoid ALautowahState_update(ALautowahState *state, ALCdevice *device, const ALeffectslot *slot)
{
ALfloat attackTime, releaseTime;
ALfloat gain;
attackTime = slot->EffectProps.Autowah.AttackTime * state->Frequency;
releaseTime = slot->EffectProps.Autowah.ReleaseTime * state->Frequency;
@@ -76,19 +75,18 @@ static ALvoid ALautowahState_update(ALautowahState *state, ALCdevice *device, co
state->PeakGain = slot->EffectProps.Autowah.PeakGain;
state->Resonance = slot->EffectProps.Autowah.Resonance;
gain = sqrtf(1.0f / device->NumChan) * slot->Gain;
SetGains(device, gain, state->Gain);
ComputeAmbientGains(device, slot->Gain, state->Gain);
}
static ALvoid ALautowahState_process(ALautowahState *state, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[BUFFERSIZE])
static ALvoid ALautowahState_process(ALautowahState *state, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[BUFFERSIZE], ALuint NumChannels)
{
ALuint it, kt;
ALuint base;
for(base = 0;base < SamplesToDo;)
{
ALfloat temps[64];
ALuint td = minu(SamplesToDo-base, 64);
ALfloat temps[256];
ALuint td = minu(256, SamplesToDo-base);
ALfloat gain = state->GainCtrl;
for(it = 0;it < td;it++)
@@ -114,7 +112,7 @@ static ALvoid ALautowahState_process(ALautowahState *state, ALuint SamplesToDo,
* ALfilterType_LowPass. However, instead of passing a bandwidth,
* we use the resonance property for Q. This also inlines the call.
*/
w0 = F_2PI * cutoff / state->Frequency;
w0 = F_TAU * cutoff / state->Frequency;
/* FIXME: Resonance controls the resonant peak, or Q. How? Not sure
* that Q = resonance*0.1. */
@@ -137,10 +135,10 @@ static ALvoid ALautowahState_process(ALautowahState *state, ALuint SamplesToDo,
}
state->GainCtrl = gain;
for(kt = 0;kt < MaxChannels;kt++)
for(kt = 0;kt < NumChannels;kt++)
{
ALfloat gain = state->Gain[kt];
if(!(gain > GAIN_SILENCE_THRESHOLD))
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(it = 0;it < td;it++)
libs/openal-soft/Alc/effects/chorus.c
+14 -12
View File
@@ -13,8 +13,8 @@
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
@@ -46,7 +46,7 @@ typedef struct ALchorusState {
ALint lfo_disp;
/* Gains for left and right sides */
ALfloat Gain[2][MaxChannels];
ALfloat Gain[2][MAX_OUTPUT_CHANNELS];
/* effect parameters */
enum ChorusWaveForm waveform;
@@ -93,6 +93,8 @@ static ALboolean ALchorusState_deviceUpdate(ALchorusState *state, ALCdevice *Dev
static ALvoid ALchorusState_update(ALchorusState *state, ALCdevice *Device, const ALeffectslot *Slot)
{
static const ALfloat left_dir[3] = { -1.0f, 0.0f, 0.0f };
static const ALfloat right_dir[3] = { 1.0f, 0.0f, 0.0f };
ALfloat frequency = (ALfloat)Device->Frequency;
ALfloat rate;
ALint phase;
@@ -111,8 +113,8 @@ static ALvoid ALchorusState_update(ALchorusState *state, ALCdevice *Device, cons
state->delay = fastf2i(Slot->EffectProps.Chorus.Delay * frequency);
/* Gains for left and right sides */
ComputeAngleGains(Device, atan2f(-1.0f, 0.0f), 0.0f, Slot->Gain, state->Gain[0]);
ComputeAngleGains(Device, atan2f(+1.0f, 0.0f), 0.0f, Slot->Gain, state->Gain[1]);
ComputeDirectionalGains(Device, left_dir, Slot->Gain, state->Gain[0]);
ComputeDirectionalGains(Device, right_dir, Slot->Gain, state->Gain[1]);
phase = Slot->EffectProps.Chorus.Phase;
rate = Slot->EffectProps.Chorus.Rate;
@@ -132,7 +134,7 @@ static ALvoid ALchorusState_update(ALchorusState *state, ALCdevice *Device, cons
state->lfo_scale = 4.0f / state->lfo_range;
break;
case CWF_Sinusoid:
state->lfo_scale = F_2PI / state->lfo_range;
state->lfo_scale = F_TAU / state->lfo_range;
break;
}
@@ -201,15 +203,15 @@ DECL_TEMPLATE(Sinusoid)
#undef DECL_TEMPLATE
static ALvoid ALchorusState_process(ALchorusState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
static ALvoid ALchorusState_process(ALchorusState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
{
ALuint it, kt;
ALuint base;
for(base = 0;base < SamplesToDo;)
{
ALfloat temps[64][2];
ALuint td = minu(SamplesToDo-base, 64);
ALfloat temps[128][2];
ALuint td = minu(128, SamplesToDo-base);
switch(state->waveform)
{
@@ -221,17 +223,17 @@ static ALvoid ALchorusState_process(ALchorusState *state, ALuint SamplesToDo, co
break;
}
for(kt = 0;kt < MaxChannels;kt++)
for(kt = 0;kt < NumChannels;kt++)
{
ALfloat gain = state->Gain[0][kt];
if(gain > GAIN_SILENCE_THRESHOLD)
if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
{
for(it = 0;it < td;it++)
SamplesOut[kt][it+base] += temps[it][0] * gain;
}
gain = state->Gain[1][kt];
if(gain > GAIN_SILENCE_THRESHOLD)
if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
{
for(it = 0;it < td;it++)
SamplesOut[kt][it+base] += temps[it][1] * gain;
libs/openal-soft/Alc/effects/compressor.c
+11 -14
View File
@@ -13,8 +13,8 @@
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
@@ -31,7 +31,7 @@ typedef struct ALcompressorState {
DERIVE_FROM_TYPE(ALeffectState);
/* Effect gains for each channel */
ALfloat Gain[MaxChannels];
ALfloat Gain[MAX_OUTPUT_CHANNELS];
/* Effect parameters */
ALboolean Enabled;
@@ -55,25 +55,22 @@ static ALboolean ALcompressorState_deviceUpdate(ALcompressorState *state, ALCdev
return AL_TRUE;
}
static ALvoid ALcompressorState_update(ALcompressorState *state, ALCdevice *Device, const ALeffectslot *Slot)
static ALvoid ALcompressorState_update(ALcompressorState *state, ALCdevice *device, const ALeffectslot *slot)
{
ALfloat gain;
state->Enabled = slot->EffectProps.Compressor.OnOff;
state->Enabled = Slot->EffectProps.Compressor.OnOff;
gain = sqrtf(1.0f / Device->NumChan) * Slot->Gain;
SetGains(Device, gain, state->Gain);
ComputeAmbientGains(device, slot->Gain, state->Gain);
}
static ALvoid ALcompressorState_process(ALcompressorState *state, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[BUFFERSIZE])
static ALvoid ALcompressorState_process(ALcompressorState *state, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[BUFFERSIZE], ALuint NumChannels)
{
ALuint it, kt;
ALuint base;
for(base = 0;base < SamplesToDo;)
{
ALfloat temps[64];
ALuint td = minu(SamplesToDo-base, 64);
ALfloat temps[256];
ALuint td = minu(256, SamplesToDo-base);
if(state->Enabled)
{
@@ -119,10 +116,10 @@ static ALvoid ALcompressorState_process(ALcompressorState *state, ALuint Samples
}
for(kt = 0;kt < MaxChannels;kt++)
for(kt = 0;kt < NumChannels;kt++)
{
ALfloat gain = state->Gain[kt];
if(!(gain > GAIN_SILENCE_THRESHOLD))
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(it = 0;it < td;it++)
libs/openal-soft/Alc/effects/dedicated.c
+28 -14
View File
@@ -13,8 +13,8 @@
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
@@ -32,7 +32,7 @@
typedef struct ALdedicatedState {
DERIVE_FROM_TYPE(ALeffectState);
ALfloat gains[MaxChannels];
ALfloat gains[MAX_OUTPUT_CHANNELS];
} ALdedicatedState;
@@ -48,27 +48,41 @@ static ALboolean ALdedicatedState_deviceUpdate(ALdedicatedState *UNUSED(state),
static ALvoid ALdedicatedState_update(ALdedicatedState *state, ALCdevice *device, const ALeffectslot *Slot)
{
ALfloat Gain;
ALsizei s;
ALuint i;
for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
state->gains[i] = 0.0f;
Gain = Slot->Gain * Slot->EffectProps.Dedicated.Gain;
if(Slot->EffectType == AL_EFFECT_DEDICATED_DIALOGUE)
ComputeAngleGains(device, atan2f(0.0f, 1.0f), 0.0f, Gain, state->gains);
else if(Slot->EffectType == AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT)
if(Slot->EffectType == AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT)
{
for(s = 0;s < MaxChannels;s++)
state->gains[s] = 0.0f;
state->gains[LFE] = Gain;
int idx;
if((idx=GetChannelIdxByName(device, LFE)) != -1)
state->gains[idx] = Gain;
}
else if(Slot->EffectType == AL_EFFECT_DEDICATED_DIALOGUE)
{
int idx;
/* Dialog goes to the front-center speaker if it exists, otherwise it
* plays from the front-center location. */
if((idx=GetChannelIdxByName(device, FrontCenter)) != -1)
state->gains[idx] = Gain;
else
{
static const ALfloat front_dir[3] = { 0.0f, 0.0f, -1.0f };
ComputeDirectionalGains(device, front_dir, Gain, state->gains);
}
}
}
static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
{
const ALfloat *gains = state->gains;
ALuint i, c;
for(c = 0;c < MaxChannels;c++)
for(c = 0;c < NumChannels;c++)
{
if(!(gains[c] > GAIN_SILENCE_THRESHOLD))
if(!(fabsf(gains[c]) > GAIN_SILENCE_THRESHOLD))
continue;
for(i = 0;i < SamplesToDo;i++)
@@ -94,7 +108,7 @@ ALeffectState *ALdedicatedStateFactory_create(ALdedicatedStateFactory *UNUSED(fa
if(!state) return NULL;
SET_VTABLE2(ALdedicatedState, ALeffectState, state);
for(s = 0;s < MaxChannels;s++)
for(s = 0;s < MAX_OUTPUT_CHANNELS;s++)
state->gains[s] = 0.0f;
return STATIC_CAST(ALeffectState, state);
libs/openal-soft/Alc/effects/distortion.c
+18 -20
View File
@@ -13,8 +13,8 @@
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
@@ -34,7 +34,7 @@ typedef struct ALdistortionState {
DERIVE_FROM_TYPE(ALeffectState);
/* Effect gains for each channel */
ALfloat Gain[MaxChannels];
ALfloat Gain[MAX_OUTPUT_CHANNELS];
/* Effect parameters */
ALfilterState lowpass;
@@ -58,7 +58,6 @@ static ALvoid ALdistortionState_update(ALdistortionState *state, ALCdevice *Devi
ALfloat bandwidth;
ALfloat cutoff;
ALfloat edge;
ALfloat gain;
/* Store distorted signal attenuation settings */
state->attenuation = Slot->EffectProps.Distortion.Gain;
@@ -73,23 +72,23 @@ static ALvoid ALdistortionState_update(ALdistortionState *state, ALCdevice *Devi
/* Bandwidth value is constant in octaves */
bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f);
ALfilterState_setParams(&state->lowpass, ALfilterType_LowPass, 1.0f,
cutoff / (frequency*4.0f), bandwidth);
cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
);
/* Bandpass filter */
cutoff = Slot->EffectProps.Distortion.EQCenter;
/* Convert bandwidth in Hz to octaves */
bandwidth = Slot->EffectProps.Distortion.EQBandwidth / (cutoff * 0.67f);
ALfilterState_setParams(&state->bandpass, ALfilterType_BandPass, 1.0f,
cutoff / (frequency*4.0f), bandwidth);
cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
);
gain = sqrtf(1.0f / Device->NumChan) * Slot->Gain;
SetGains(Device, gain, state->Gain);
ComputeAmbientGains(Device, Slot->Gain, state->Gain);
}
static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
{
const ALfloat fc = state->edge_coeff;
float oversample_buffer[64][4];
ALuint base;
ALuint it;
ALuint ot;
@@ -97,8 +96,8 @@ static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint Samples
for(base = 0;base < SamplesToDo;)
{
ALfloat temps[64];
ALuint td = minu(SamplesToDo-base, 64);
float oversample_buffer[64][4];
ALuint td = minu(64, SamplesToDo-base);
/* Perform 4x oversampling to avoid aliasing. */
/* Oversampling greatly improves distortion */
@@ -150,20 +149,19 @@ static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint Samples
smp = ALfilterState_processSingle(&state->bandpass, smp);
oversample_buffer[it][ot] = smp;
}
/* Fourth step, final, do attenuation and perform decimation, */
/* store only one sample out of 4. */
temps[it] = oversample_buffer[it][0] * state->attenuation;
}
for(kt = 0;kt < MaxChannels;kt++)
for(kt = 0;kt < NumChannels;kt++)
{
ALfloat gain = state->Gain[kt];
if(!(gain > GAIN_SILENCE_THRESHOLD))
/* Fourth step, final, do attenuation and perform decimation,
* store only one sample out of 4.
*/
ALfloat gain = state->Gain[kt] * state->attenuation;
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(it = 0;it < td;it++)
SamplesOut[kt][base+it] += gain * temps[it];
SamplesOut[kt][base+it] += gain * oversample_buffer[it][0];
}
base += td;
libs/openal-soft/Alc/effects/echo.c
+18 -16
View File
@@ -13,8 +13,8 @@
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
@@ -43,7 +43,7 @@ typedef struct ALechoState {
} Tap[2];
ALuint Offset;
/* The panning gains for the two taps */
ALfloat Gain[2][MaxChannels];
ALfloat Gain[2][MAX_OUTPUT_CHANNELS];
ALfloat FeedGain;
@@ -83,9 +83,9 @@ static ALboolean ALechoState_deviceUpdate(ALechoState *state, ALCdevice *Device)
static ALvoid ALechoState_update(ALechoState *state, ALCdevice *Device, const ALeffectslot *Slot)
{
ALfloat pandir[3] = { 0.0f, 0.0f, 0.0f };
ALuint frequency = Device->Frequency;
ALfloat lrpan, gain;
ALfloat dirGain;
ALfloat gain, lrpan;
state->Tap[0].delay = fastf2u(Slot->EffectProps.Echo.Delay * frequency) + 1;
state->Tap[1].delay = fastf2u(Slot->EffectProps.Echo.LRDelay * frequency);
@@ -95,21 +95,23 @@ static ALvoid ALechoState_update(ALechoState *state, ALCdevice *Device, const AL
state->FeedGain = Slot->EffectProps.Echo.Feedback;
gain = minf(1.0f - Slot->EffectProps.Echo.Damping, 0.01f);
ALfilterState_setParams(&state->Filter, ALfilterType_HighShelf,
1.0f - Slot->EffectProps.Echo.Damping,
LOWPASSFREQREF/frequency, 0.0f);
gain, LOWPASSFREQREF/frequency,
calc_rcpQ_from_slope(gain, 0.75f));
gain = Slot->Gain;
dirGain = fabsf(lrpan);
/* First tap panning */
ComputeAngleGains(Device, atan2f(-lrpan, 0.0f), (1.0f-dirGain)*F_PI, gain, state->Gain[0]);
pandir[0] = -lrpan;
ComputeDirectionalGains(Device, pandir, gain, state->Gain[0]);
/* Second tap panning */
ComputeAngleGains(Device, atan2f(+lrpan, 0.0f), (1.0f-dirGain)*F_PI, gain, state->Gain[1]);
pandir[0] = +lrpan;
ComputeDirectionalGains(Device, pandir, gain, state->Gain[1]);
}
static ALvoid ALechoState_process(ALechoState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
static ALvoid ALechoState_process(ALechoState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
{
const ALuint mask = state->BufferLength-1;
const ALuint tap1 = state->Tap[0].delay;
@@ -121,8 +123,8 @@ static ALvoid ALechoState_process(ALechoState *state, ALuint SamplesToDo, const
for(base = 0;base < SamplesToDo;)
{
ALfloat temps[64][2];
ALuint td = minu(SamplesToDo-base, 64);
ALfloat temps[128][2];
ALuint td = minu(128, SamplesToDo-base);
for(i = 0;i < td;i++)
{
@@ -138,17 +140,17 @@ static ALvoid ALechoState_process(ALechoState *state, ALuint SamplesToDo, const
offset++;
}
for(k = 0;k < MaxChannels;k++)
for(k = 0;k < NumChannels;k++)
{
ALfloat gain = state->Gain[0][k];
if(gain > GAIN_SILENCE_THRESHOLD)
if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
{
for(i = 0;i < td;i++)
SamplesOut[k][i+base] += temps[i][0] * gain;
}
gain = state->Gain[1][k];
if(gain > GAIN_SILENCE_THRESHOLD)
if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
{
for(i = 0;i < td;i++)
SamplesOut[k][i+base] += temps[i][1] * gain;
libs/openal-soft/Alc/effects/equalizer.c
+30 -23
View File
@@ -13,8 +13,8 @@
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
@@ -75,7 +75,7 @@ typedef struct ALequalizerState {
DERIVE_FROM_TYPE(ALeffectState);
/* Effect gains for each channel */
ALfloat Gain[MaxChannels];
ALfloat Gain[MAX_OUTPUT_CHANNELS];
/* Effect parameters */
ALfilterState filter[4];
@@ -93,33 +93,40 @@ static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *UNUSED(state),
static ALvoid ALequalizerState_update(ALequalizerState *state, ALCdevice *device, const ALeffectslot *slot)
{
ALfloat frequency = (ALfloat)device->Frequency;
ALfloat gain = sqrtf(1.0f / device->NumChan) * slot->Gain;
ALfloat gain, freq_mult;
SetGains(device, gain, state->Gain);
ComputeAmbientGains(device, slot->Gain, state->Gain);
/* Calculate coefficients for the each type of filter */
/* Calculate coefficients for the each type of filter. Note that the shelf
* filters' gain is for the reference frequency, which is the centerpoint
* of the transition band.
*/
gain = sqrtf(slot->EffectProps.Equalizer.LowGain);
freq_mult = slot->EffectProps.Equalizer.LowCutoff/frequency;
ALfilterState_setParams(&state->filter[0], ALfilterType_LowShelf,
sqrtf(slot->EffectProps.Equalizer.LowGain),
slot->EffectProps.Equalizer.LowCutoff/frequency,
0.0f);
gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f)
);
gain = slot->EffectProps.Equalizer.Mid1Gain;
freq_mult = slot->EffectProps.Equalizer.Mid1Center/frequency;
ALfilterState_setParams(&state->filter[1], ALfilterType_Peaking,
sqrtf(slot->EffectProps.Equalizer.Mid1Gain),
slot->EffectProps.Equalizer.Mid1Center/frequency,
slot->EffectProps.Equalizer.Mid1Width);
gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid1Width)
);
gain = slot->EffectProps.Equalizer.Mid2Gain;
freq_mult = slot->EffectProps.Equalizer.Mid2Center/frequency;
ALfilterState_setParams(&state->filter[2], ALfilterType_Peaking,
sqrtf(slot->EffectProps.Equalizer.Mid2Gain),
slot->EffectProps.Equalizer.Mid2Center/frequency,
slot->EffectProps.Equalizer.Mid2Width);
gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid2Width)
);
gain = sqrtf(slot->EffectProps.Equalizer.HighGain);
freq_mult = slot->EffectProps.Equalizer.HighCutoff/frequency;
ALfilterState_setParams(&state->filter[3], ALfilterType_HighShelf,
sqrtf(slot->EffectProps.Equalizer.HighGain),
slot->EffectProps.Equalizer.HighCutoff/frequency,
0.0f);
gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f)
);
}
static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
{
ALuint base;
ALuint it;
@@ -128,8 +135,8 @@ static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesTo
for(base = 0;base < SamplesToDo;)
{
ALfloat temps[64];
ALuint td = minu(SamplesToDo-base, 64);
ALfloat temps[256];
ALuint td = minu(256, SamplesToDo-base);
for(it = 0;it < td;it++)
{
@@ -141,10 +148,10 @@ static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesTo
temps[it] = smp;
}
for(kt = 0;kt < MaxChannels;kt++)
for(kt = 0;kt < NumChannels;kt++)
{
ALfloat gain = state->Gain[kt];
if(!(gain > GAIN_SILENCE_THRESHOLD))
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(it = 0;it < td;it++)
libs/openal-soft/Alc/effects/flanger.c
+14 -12
View File
@@ -13,8 +13,8 @@
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
@@ -46,7 +46,7 @@ typedef struct ALflangerState {
ALint lfo_disp;
/* Gains for left and right sides */
ALfloat Gain[2][MaxChannels];
ALfloat Gain[2][MAX_OUTPUT_CHANNELS];
/* effect parameters */
enum FlangerWaveForm waveform;
@@ -93,6 +93,8 @@ static ALboolean ALflangerState_deviceUpdate(ALflangerState *state, ALCdevice *D
static ALvoid ALflangerState_update(ALflangerState *state, ALCdevice *Device, const ALeffectslot *Slot)
{
static const ALfloat left_dir[3] = { -1.0f, 0.0f, 0.0f };
static const ALfloat right_dir[3] = { 1.0f, 0.0f, 0.0f };
ALfloat frequency = (ALfloat)Device->Frequency;
ALfloat rate;
ALint phase;
@@ -111,8 +113,8 @@ static ALvoid ALflangerState_update(ALflangerState *state, ALCdevice *Device, co
state->delay = fastf2i(Slot->EffectProps.Flanger.Delay * frequency);
/* Gains for left and right sides */
ComputeAngleGains(Device, atan2f(-1.0f, 0.0f), 0.0f, Slot->Gain, state->Gain[0]);
ComputeAngleGains(Device, atan2f(+1.0f, 0.0f), 0.0f, Slot->Gain, state->Gain[1]);
ComputeDirectionalGains(Device, left_dir, Slot->Gain, state->Gain[0]);
ComputeDirectionalGains(Device, right_dir, Slot->Gain, state->Gain[1]);
phase = Slot->EffectProps.Flanger.Phase;
rate = Slot->EffectProps.Flanger.Rate;
@@ -132,7 +134,7 @@ static ALvoid ALflangerState_update(ALflangerState *state, ALCdevice *Device, co
state->lfo_scale = 4.0f / state->lfo_range;
break;
case FWF_Sinusoid:
state->lfo_scale = F_2PI / state->lfo_range;
state->lfo_scale = F_TAU / state->lfo_range;
break;
}
@@ -201,15 +203,15 @@ DECL_TEMPLATE(Sinusoid)
#undef DECL_TEMPLATE
static ALvoid ALflangerState_process(ALflangerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
static ALvoid ALflangerState_process(ALflangerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
{
ALuint it, kt;
ALuint base;
for(base = 0;base < SamplesToDo;)
{
ALfloat temps[64][2];
ALuint td = minu(SamplesToDo-base, 64);
ALfloat temps[128][2];
ALuint td = minu(128, SamplesToDo-base);
switch(state->waveform)
{
@@ -221,17 +223,17 @@ static ALvoid ALflangerState_process(ALflangerState *state, ALuint SamplesToDo,
break;
}
for(kt = 0;kt < MaxChannels;kt++)
for(kt = 0;kt < NumChannels;kt++)
{
ALfloat gain = state->Gain[0][kt];
if(gain > GAIN_SILENCE_THRESHOLD)
if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
{
for(it = 0;it < td;it++)
SamplesOut[kt][it+base] += temps[it][0] * gain;
}
gain = state->Gain[1][kt];
if(gain > GAIN_SILENCE_THRESHOLD)
if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
{
for(it = 0;it < td;it++)
SamplesOut[kt][it+base] += temps[it][1] * gain;
libs/openal-soft/Alc/effects/modulator.c
+16 -17
View File
@@ -13,8 +13,8 @@
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
@@ -42,7 +42,7 @@ typedef struct ALmodulatorState {
ALuint index;
ALuint step;
ALfloat Gain[MaxChannels];
ALfloat Gain[MAX_OUTPUT_CHANNELS];
ALfilterState Filter;
} ALmodulatorState;
@@ -53,7 +53,7 @@ typedef struct ALmodulatorState {
static inline ALfloat Sin(ALuint index)
{
return sinf(index*(F_2PI/WAVEFORM_FRACONE) - F_PI)*0.5f + 0.5f;
return sinf(index*(F_TAU/WAVEFORM_FRACONE) - F_PI)*0.5f + 0.5f;
}
static inline ALfloat Saw(ALuint index)
@@ -69,7 +69,7 @@ static inline ALfloat Square(ALuint index)
#define DECL_TEMPLATE(func) \
static void Process##func(ALmodulatorState *state, ALuint SamplesToDo, \
const ALfloat *restrict SamplesIn, \
ALfloat (*restrict SamplesOut)[BUFFERSIZE]) \
ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) \
{ \
const ALuint step = state->step; \
ALuint index = state->index; \
@@ -77,8 +77,8 @@ static void Process##func(ALmodulatorState *state, ALuint SamplesToDo, \
\
for(base = 0;base < SamplesToDo;) \
{ \
ALfloat temps[64]; \
ALuint td = minu(SamplesToDo-base, 64); \
ALfloat temps[256]; \
ALuint td = minu(256, SamplesToDo-base); \
ALuint i, k; \
\
for(i = 0;i < td;i++) \
@@ -92,10 +92,10 @@ static void Process##func(ALmodulatorState *state, ALuint SamplesToDo, \
temps[i] = samp * func(index); \
} \
\
for(k = 0;k < MaxChannels;k++) \
for(k = 0;k < NumChannels;k++) \
{ \
ALfloat gain = state->Gain[k]; \
if(!(gain > GAIN_SILENCE_THRESHOLD)) \
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) \
continue; \
\
for(i = 0;i < td;i++) \
@@ -125,7 +125,7 @@ static ALboolean ALmodulatorState_deviceUpdate(ALmodulatorState *UNUSED(state),
static ALvoid ALmodulatorState_update(ALmodulatorState *state, ALCdevice *Device, const ALeffectslot *Slot)
{
ALfloat gain, cw, a;
ALfloat cw, a;
if(Slot->EffectProps.Modulator.Waveform == AL_RING_MODULATOR_SINUSOID)
state->Waveform = SINUSOID;
@@ -139,7 +139,7 @@ static ALvoid ALmodulatorState_update(ALmodulatorState *state, ALCdevice *Device
if(state->step == 0) state->step = 1;
/* Custom filter coeffs, which match the old version instead of a low-shelf. */
cw = cosf(F_2PI * Slot->EffectProps.Modulator.HighPassCutoff / Device->Frequency);
cw = cosf(F_TAU * Slot->EffectProps.Modulator.HighPassCutoff / Device->Frequency);
a = (2.0f-cw) - sqrtf(powf(2.0f-cw, 2.0f) - 1.0f);
state->Filter.b[0] = a;
@@ -149,24 +149,23 @@ static ALvoid ALmodulatorState_update(ALmodulatorState *state, ALCdevice *Device
state->Filter.a[1] = -a;
state->Filter.a[2] = 0.0f;
gain = sqrtf(1.0f/Device->NumChan) * Slot->Gain;
SetGains(Device, gain, state->Gain);
ComputeAmbientGains(Device, Slot->Gain, state->Gain);
}
static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
{
switch(state->Waveform)
{
case SINUSOID:
ProcessSin(state, SamplesToDo, SamplesIn, SamplesOut);
ProcessSin(state, SamplesToDo, SamplesIn, SamplesOut, NumChannels);
break;
case SAWTOOTH:
ProcessSaw(state, SamplesToDo, SamplesIn, SamplesOut);
ProcessSaw(state, SamplesToDo, SamplesIn, SamplesOut, NumChannels);
break;
case SQUARE:
ProcessSquare(state, SamplesToDo, SamplesIn, SamplesOut);
ProcessSquare(state, SamplesToDo, SamplesIn, SamplesOut, NumChannels);
break;
}
}
libs/openal-soft/Alc/effects/null.c
+1 -4
View File
@@ -41,11 +41,8 @@ static ALvoid ALnullState_update(ALnullState* UNUSED(state), ALCdevice* UNUSED(d
* input to the output buffer. The result should be added to the output buffer,
* not replace it.
*/
static ALvoid ALnullState_process(ALnullState* UNUSED(state), ALuint UNUSED(samplesToDo), const ALfloat *restrict UNUSED(samplesIn), ALfloat (*restrict samplesOut)[BUFFERSIZE])
static ALvoid ALnullState_process(ALnullState* UNUSED(state), ALuint UNUSED(samplesToDo), const ALfloat *restrict UNUSED(samplesIn), ALfloatBUFFERSIZE*restrict UNUSED(samplesOut), ALuint UNUSED(NumChannels))
{
/* NOTE: Couldn't use the UNUSED macro on samplesOut due to the way GCC's
* __attribute__ declaration interacts with the parenthesis. */
(void)samplesOut;
}
/* This allocates memory to store the object, before it gets constructed.
libs/openal-soft/Alc/effects/reverb.c
+425 -401
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