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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/distortion.c
+18 -20
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@@ -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;