fix #119731: fluidsynth optimizations
This commit is contained in:
Johannes Wegener
parent
95bcc1e5cb
commit
65fad51543
3 changed files with 109 additions and 91 deletions
106
fluid/dsp.cpp
106
fluid/dsp.cpp
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@ -46,13 +46,29 @@ namespace FluidS {
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* - dsp_buf: Output buffer of floating point values (FLUID_BUFSIZE in length)
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*/
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inline void updateAmpInc(unsigned int &nextNewAmpInc,std::map<int, qreal>::iterator &curSample2AmpInc, qreal &dsp_amp_incr, unsigned int dsp_i)
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inline bool Voice::updateAmpInc(unsigned int &nextNewAmpInc, std::map<int, qreal>::iterator &curSample2AmpInc, qreal &dsp_amp_incr, unsigned int &dsp_i)
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{
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if (positionToTurnOff > 0 && dsp_i >= (unsigned int) positionToTurnOff)
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return false;
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// if volume is zero skip all phases that do not change that!
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if (amp == 0.0f) {
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while (dsp_amp_incr == 0.0f && curSample2AmpInc != Sample2AmpInc.end()) {
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dsp_i = curSample2AmpInc->first;
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curSample2AmpInc++;
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nextNewAmpInc = curSample2AmpInc->first;
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dsp_amp_incr = curSample2AmpInc->second;
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}
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if (curSample2AmpInc == Sample2AmpInc.end())
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return false;
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}
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if (dsp_i >= nextNewAmpInc) {
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curSample2AmpInc++;
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nextNewAmpInc = curSample2AmpInc->first;
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dsp_amp_incr = curSample2AmpInc->second;
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}
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return true;
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}
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@ -131,7 +147,6 @@ int Voice::dsp_float_interpolate_none(unsigned n)
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Phase dsp_phase_incr; // end_phase;
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short int *dsp_data = voice->sample->data;
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float *dsp_buf = voice->dsp_buf;
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float dsp_amp = voice->amp;
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auto curSample2AmpInc = Sample2AmpInc.begin();
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qreal dsp_amp_incr = curSample2AmpInc->second;
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unsigned int nextNewAmpInc = curSample2AmpInc->first;
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@ -155,13 +170,14 @@ int Voice::dsp_float_interpolate_none(unsigned n)
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/* interpolate sequence of sample points */
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for ( ; dsp_i < n && dsp_phase_index <= end_index; dsp_i++) {
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dsp_buf[dsp_i] = dsp_amp * dsp_data[dsp_phase_index];
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dsp_buf[dsp_i] = amp * dsp_data[dsp_phase_index];
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index_round(); /* round to nearest point */
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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dsp_amp += dsp_amp_incr;
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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/* break out if not looping (buffer may not be full) */
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@ -180,7 +196,6 @@ int Voice::dsp_float_interpolate_none(unsigned n)
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}
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voice->phase = dsp_phase;
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voice->amp = dsp_amp;
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return dsp_i;
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}
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@ -198,7 +213,6 @@ int Voice::dsp_float_interpolate_linear(unsigned n)
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Phase dsp_phase_incr; // end_phase;
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short int *dsp_data = voice->sample->data;
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float *dsp_buf = voice->dsp_buf;
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float dsp_amp = voice->amp;
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auto curSample2AmpInc = Sample2AmpInc.begin();
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qreal dsp_amp_incr = curSample2AmpInc->second;
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unsigned int nextNewAmpInc = curSample2AmpInc->first;
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@ -232,14 +246,15 @@ int Voice::dsp_float_interpolate_linear(unsigned n)
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/* interpolate the sequence of sample points */
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for ( ; dsp_i < n && dsp_phase_index <= end_index; dsp_i++) {
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coeffs = interp_coeff_linear[fluid_phase_fract_to_tablerow (dsp_phase)];
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dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * dsp_data[dsp_phase_index]
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dsp_buf[dsp_i] = amp * (coeffs[0] * dsp_data[dsp_phase_index]
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+ coeffs[1] * dsp_data[dsp_phase_index+1]);
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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dsp_amp += dsp_amp_incr;
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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/* break out if buffer filled */
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@ -251,14 +266,15 @@ int Voice::dsp_float_interpolate_linear(unsigned n)
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/* interpolate within last point */
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for (; dsp_phase_index <= end_index && dsp_i < n; dsp_i++) {
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coeffs = interp_coeff_linear[fluid_phase_fract_to_tablerow (dsp_phase)];
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dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * dsp_data[dsp_phase_index]
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dsp_buf[dsp_i] = amp * (coeffs[0] * dsp_data[dsp_phase_index]
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+ coeffs[1] * point);
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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dsp_amp += dsp_amp_incr; /* increment amplitude */
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr; /* increment amplitude */
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}
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if (!looping)
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@ -277,7 +293,6 @@ int Voice::dsp_float_interpolate_linear(unsigned n)
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}
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voice->phase = dsp_phase;
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voice->amp = dsp_amp;
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return dsp_i;
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}
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@ -345,7 +360,8 @@ int Voice::dsp_float_interpolate_4th_order(unsigned n)
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/* increment phase and amplitude */
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phase += dsp_phase_incr;
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dsp_phase_index = phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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@ -360,7 +376,8 @@ int Voice::dsp_float_interpolate_4th_order(unsigned n)
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/* increment phase and amplitude */
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phase += dsp_phase_incr;
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dsp_phase_index = phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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@ -381,7 +398,8 @@ int Voice::dsp_float_interpolate_4th_order(unsigned n)
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/* increment phase and amplitude */
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phase += dsp_phase_incr;
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dsp_phase_index = phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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@ -398,7 +416,8 @@ int Voice::dsp_float_interpolate_4th_order(unsigned n)
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/* increment phase and amplitude */
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phase += dsp_phase_incr;
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dsp_phase_index = phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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@ -437,7 +456,6 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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Phase dsp_phase_incr; // end_phase;
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short int *dsp_data = voice->sample->data;
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float *dsp_buf = voice->dsp_buf;
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float dsp_amp = voice->amp;
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auto curSample2AmpInc = Sample2AmpInc.begin();
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qreal dsp_amp_incr = curSample2AmpInc->second;
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unsigned int nextNewAmpInc = curSample2AmpInc->first;
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@ -496,7 +514,7 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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for ( ; dsp_phase_index == start_index && dsp_i < n; dsp_i++) {
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coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
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dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)start_points[2]
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dsp_buf[dsp_i] = amp * (coeffs[0] * (float)start_points[2]
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+ coeffs[1] * (float)start_points[1]
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+ coeffs[2] * (float)start_points[0]
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+ coeffs[3] * (float)dsp_data[dsp_phase_index]
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@ -507,8 +525,9 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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dsp_amp += dsp_amp_incr;
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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start_index++;
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@ -517,7 +536,7 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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for ( ; dsp_phase_index == start_index && dsp_i < n; dsp_i++) {
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coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
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dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)start_points[1]
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dsp_buf[dsp_i] = amp * (coeffs[0] * (float)start_points[1]
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+ coeffs[1] * (float)start_points[0]
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+ coeffs[2] * (float)dsp_data[dsp_phase_index-1]
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+ coeffs[3] * (float)dsp_data[dsp_phase_index]
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@ -528,8 +547,9 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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dsp_amp += dsp_amp_incr;
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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start_index++;
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@ -538,7 +558,7 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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for ( ; dsp_phase_index == start_index && dsp_i < n; dsp_i++) {
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coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
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dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)start_points[0]
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dsp_buf[dsp_i] = amp * (coeffs[0] * (float)start_points[0]
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+ coeffs[1] * (float)dsp_data[dsp_phase_index-2]
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+ coeffs[2] * (float)dsp_data[dsp_phase_index-1]
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+ coeffs[3] * (float)dsp_data[dsp_phase_index]
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@ -549,8 +569,9 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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dsp_amp += dsp_amp_incr;
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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start_index -= 2; /* set back to original start index */
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@ -559,7 +580,7 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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for ( ; dsp_i < n && dsp_phase_index <= end_index; dsp_i++) {
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coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
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dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)dsp_data[dsp_phase_index-3]
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dsp_buf[dsp_i] = amp * (coeffs[0] * (float)dsp_data[dsp_phase_index-3]
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+ coeffs[1] * (float)dsp_data[dsp_phase_index-2]
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+ coeffs[2] * (float)dsp_data[dsp_phase_index-1]
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+ coeffs[3] * (float)dsp_data[dsp_phase_index]
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@ -570,8 +591,9 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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dsp_amp += dsp_amp_incr;
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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/* break out if buffer filled */
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@ -584,7 +606,7 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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for (; dsp_phase_index <= end_index && dsp_i < n; dsp_i++) {
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coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
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dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)dsp_data[dsp_phase_index-3]
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dsp_buf[dsp_i] = amp * (coeffs[0] * (float)dsp_data[dsp_phase_index-3]
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+ coeffs[1] * (float)dsp_data[dsp_phase_index-2]
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+ coeffs[2] * (float)dsp_data[dsp_phase_index-1]
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+ coeffs[3] * (float)dsp_data[dsp_phase_index]
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@ -595,8 +617,9 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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dsp_amp += dsp_amp_incr;
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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end_index++; /* we're now interpolating the 2nd to last point */
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@ -605,7 +628,7 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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for (; dsp_phase_index <= end_index && dsp_i < n; dsp_i++) {
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coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
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dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)dsp_data[dsp_phase_index-3]
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dsp_buf[dsp_i] = amp * (coeffs[0] * (float)dsp_data[dsp_phase_index-3]
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+ coeffs[1] * (float)dsp_data[dsp_phase_index-2]
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+ coeffs[2] * (float)dsp_data[dsp_phase_index-1]
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+ coeffs[3] * (float)dsp_data[dsp_phase_index]
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@ -616,8 +639,9 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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dsp_amp += dsp_amp_incr;
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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end_index++; /* we're now interpolating the last point */
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@ -626,7 +650,7 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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for (; dsp_phase_index <= end_index && dsp_i < n; dsp_i++) {
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coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
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dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)dsp_data[dsp_phase_index-3]
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dsp_buf[dsp_i] = amp * (coeffs[0] * (float)dsp_data[dsp_phase_index-3]
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+ coeffs[1] * (float)dsp_data[dsp_phase_index-2]
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+ coeffs[2] * (float)dsp_data[dsp_phase_index-1]
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+ coeffs[3] * (float)dsp_data[dsp_phase_index]
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@ -637,8 +661,9 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index();
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updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i);
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dsp_amp += dsp_amp_incr;
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if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i))
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return dsp_i;
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amp += dsp_amp_incr;
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}
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if (!looping)
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@ -668,7 +693,6 @@ int Voice::dsp_float_interpolate_7th_order(unsigned n)
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dsp_phase -= (Phase)0x80000000;
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voice->phase = dsp_phase;
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voice->amp = dsp_amp;
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return dsp_i;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -332,21 +332,23 @@ void Voice::write(unsigned n, float* out, float* reverb, float* chorus)
|
|||
|
||||
int modLfoStart = -1;
|
||||
|
||||
if (ticks >= modlfo_delay)
|
||||
modLfoStart = 0;
|
||||
else if (n >= modlfo_delay)
|
||||
modLfoStart = modlfo_delay;
|
||||
if (fabs(modlfo_to_vol) > 0) {
|
||||
if (ticks >= modlfo_delay)
|
||||
modLfoStart = 0;
|
||||
else if (n >= modlfo_delay)
|
||||
modLfoStart = modlfo_delay;
|
||||
|
||||
if (modLfoStart >= 0) {
|
||||
if (modLfoStart > 0)
|
||||
volumeChanges.insert(modLfoStart);
|
||||
if (modLfoStart >= 0) {
|
||||
if (modLfoStart > 0)
|
||||
volumeChanges.insert(modLfoStart);
|
||||
|
||||
unsigned int modLfoNextTurn = samplesToNextTurningPoint(modlfo_dur, modlfo_pos);
|
||||
unsigned int modLfoNextTurn = samplesToNextTurningPoint(modlfo_dur, modlfo_pos);
|
||||
|
||||
while (modLfoNextTurn+modLfoStart < n) {
|
||||
volumeChanges.insert(modLfoNextTurn+modLfoStart);
|
||||
modLfoNextTurn++;
|
||||
modLfoNextTurn += samplesToNextTurningPoint(modlfo_dur, modLfoNextTurn);
|
||||
while (modLfoNextTurn+modLfoStart < n) {
|
||||
volumeChanges.insert(modLfoNextTurn+modLfoStart);
|
||||
modLfoNextTurn++;
|
||||
modLfoNextTurn += samplesToNextTurningPoint(modlfo_dur, modLfoNextTurn);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
|
|
@ -425,12 +427,6 @@ void Voice::write(unsigned n, float* out, float* reverb, float* chorus)
|
|||
* fluid_cb2amp (960.0f * (1.0f - volenv_val)
|
||||
+ modlfo_val * -modlfo_to_vol);
|
||||
|
||||
// REMOVED DUE TO CHANGE IN ENVELOPE CALCULATION
|
||||
// is an optimization
|
||||
// TODO: Make it work again with sample based volume
|
||||
|
||||
/* 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.
|
||||
|
|
@ -440,28 +436,28 @@ void Voice::write(unsigned n, float* out, float* reverb, float* chorus)
|
|||
* the sample loop
|
||||
*/
|
||||
|
||||
/* Is the playing pointer already in the loop? */
|
||||
/*if (has_looped)
|
||||
amplitude_that_reaches_noise_floor = amplitude_that_reaches_noise_floor_loop;
|
||||
else
|
||||
amplitude_that_reaches_noise_floor = amplitude_that_reaches_noise_floor_nonloop;*/
|
||||
float amplitude_that_reaches_noise_floor;
|
||||
/* Is the playing pointer already in the loop? */
|
||||
if (has_looped)
|
||||
amplitude_that_reaches_noise_floor = amplitude_that_reaches_noise_floor_loop;
|
||||
else
|
||||
amplitude_that_reaches_noise_floor = 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):
|
||||
*/
|
||||
/* 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 (min_attenuation_cB) * volenv_val;
|
||||
float amp_max = fluid_atten2amp (min_attenuation_cB) * volenv_val;
|
||||
|
||||
/* 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) {
|
||||
positionToTurnOff = curPos;
|
||||
}
|
||||
|
||||
/* 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) {
|
||||
off();
|
||||
ticks += n;
|
||||
return;
|
||||
}*/
|
||||
}
|
||||
|
||||
if (curVolEnvSection->second != oldVolEnvSection->second) {
|
||||
|
|
@ -476,6 +472,11 @@ void Voice::write(unsigned n, float* out, float* reverb, float* chorus)
|
|||
amp_incr = (target_amp - oldTargetAmp) / (curPos - lastPos);
|
||||
lastPos = curPos;
|
||||
Sample2AmpInc.insert(std::pair<int, qreal>(curPos, amp_incr));
|
||||
|
||||
// if voice is turned off after this no need to calculate any more values
|
||||
if (positionToTurnOff > 0)
|
||||
break;
|
||||
|
||||
oldTargetAmp = target_amp;
|
||||
}
|
||||
|
||||
|
|
@ -486,17 +487,6 @@ void Voice::write(unsigned n, float* out, float* reverb, float* chorus)
|
|||
|
||||
fluid_check_fpe ("voice_write amplitude calculation");
|
||||
|
||||
// REMOVED DUE TO CHANGE IN ENVELOPE CALCULATION
|
||||
// is an optimization
|
||||
// TODO: Make it work again with sample based volume
|
||||
|
||||
/* no volume and not changing? - No need to process */
|
||||
|
||||
//if ((amp == 0.0f) && (amp_incr == 0.0f)) {
|
||||
// ticks += n;
|
||||
// return;
|
||||
// }
|
||||
|
||||
/* 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
|
||||
|
|
@ -622,6 +612,7 @@ void Voice::write(unsigned n, float* out, float* reverb, float* chorus)
|
|||
|
||||
float l_dsp_buf[n];
|
||||
dsp_buf = l_dsp_buf;
|
||||
memset(dsp_buf, 0, n*sizeof(float)); // init the arry with zeros so we can skip silent parts
|
||||
unsigned count;
|
||||
switch (interp_method) {
|
||||
case FLUID_INTERP_NONE:
|
||||
|
|
@ -642,8 +633,8 @@ void Voice::write(unsigned n, float* out, float* reverb, float* chorus)
|
|||
if (count > 0)
|
||||
effects(count, out, reverb, chorus);
|
||||
|
||||
/* turn off voice if short count (sample ended and not looping) */
|
||||
if (count < n)
|
||||
/* turn off voice if short count (sample ended and not looping) or voice reached noise floor*/
|
||||
if (count < n || positionToTurnOff > 0)
|
||||
off();
|
||||
|
||||
ticks += n;
|
||||
|
|
@ -792,6 +783,7 @@ void Voice::voice_start()
|
|||
* This cannot be done earlier, because it depends on modulators.
|
||||
*/
|
||||
check_sample_sanity_flag = FLUID_SAMPLESANITY_STARTUP;
|
||||
positionToTurnOff = -1;
|
||||
|
||||
status = FLUID_VOICE_ON;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -120,6 +120,7 @@ class Voice
|
|||
float volenv_val;
|
||||
float amplitude_that_reaches_noise_floor_nonloop;
|
||||
float amplitude_that_reaches_noise_floor_loop;
|
||||
int positionToTurnOff; // this is the sample accurate position where the sample reaches the noise floor
|
||||
|
||||
/* mod env */
|
||||
fluid_env_data_t modenv_data[FLUID_VOICE_ENVLAST];
|
||||
|
|
@ -234,6 +235,7 @@ class Voice
|
|||
void add_mod(const Mod* mod, int mode);
|
||||
|
||||
static void dsp_float_config();
|
||||
bool updateAmpInc(unsigned int &nextNewAmpInc, std::map<int, qreal>::iterator &curSample2AmpInc, qreal &dsp_amp_incr, unsigned int &dsp_i);
|
||||
int dsp_float_interpolate_none(unsigned);
|
||||
int dsp_float_interpolate_linear(unsigned);
|
||||
int dsp_float_interpolate_4th_order(unsigned);
|
||||
|
|
|
|||
Loading…
Reference in a new issue