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I've split up the dsp core file in three
files: fluid_dsp_simple.c, fluid_dsp_float.c, and fluid_dsp_sse.c. This improves the readability.
This commit is contained in:
Peter Hanappe
parent
dd092ac0ad
commit
654ec72119
4 changed files with 794 additions and 0 deletions
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@ -1,3 +1,9 @@
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2004-03-30 Peter Hanappe <peter@hanappe.com>
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* src/fluid_dsp_core.c: I've split up the dsp core file in three
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files: fluid_dsp_simple.c, fluid_dsp_float.c, and
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fluid_dsp_sse.c. This improves the readability.
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2004-03-29 Peter Hanappe <peter@hanappe.com>
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2004-03-29 Peter Hanappe <peter@hanappe.com>
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* src/fluid_jack.c (new_fluid_jack_audio_driver2): Testing the
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* src/fluid_jack.c (new_fluid_jack_audio_driver2): Testing the
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281
fluidsynth/src/fluid_dsp_float.c
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281
fluidsynth/src/fluid_dsp_float.c
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/* FluidSynth - A Software Synthesizer
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*
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* Copyright (C) 2003 Peter Hanappe and others.
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public License
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* as published by the Free Software Foundation; either version 2 of
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* the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the Free
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA
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* 02111-1307, USA
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*/
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/* Purpose:
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* Low-level voice processing:
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*
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* - interpolates (obtains values between the samples of the original waveform data)
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* - filters (applies a lowpass filter with variable cutoff frequency and quality factor)
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* - mixes the processed sample to left and right output using the pan setting
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* - sends the processed sample to chorus and reverb
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*
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*
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* This file does -not- generate an object file.
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* Instead, it is #included in several places in fluid_voice.c.
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* The motivation for this is
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* - Calling it as a subroutine may be time consuming, especially with optimization off
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* - The previous implementation as a macro was clumsy to handle
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*
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*
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* Fluid_voice.c sets a couple of variables before #including this:
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* - dsp_data: Pointer to the original waveform data
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* - dsp_left_buf: The generated signal goes here, left channel
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* - dsp_right_buf: right channel
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* - dsp_reverb_buf: Send to reverb unit
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* - dsp_chorus_buf: Send to chorus unit
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* - dsp_start: Start processing at this output buffer index
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* - dsp_end: End processing just before this output buffer index
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* - dsp_a1: Coefficient for the filter
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* - dsp_a2: same
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* - dsp_b0: same
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* - dsp_b1: same
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* - dsp_b2: same
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* - dsp_filter_flag: Set, the filter is needed (many sound fonts don't use
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* the filter at all. If it is left at its default setting
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* of roughly 20 kHz, there is no need to apply filterling.)
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* - dsp_interp_method: Which interpolation method to use.
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* - voice holds the voice structure
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*
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* Some variables are set and modified:
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* - dsp_phase: The position in the original waveform data.
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* This has an integer and a fractional part (between samples).
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* - dsp_phase_incr: For each output sample, the position in the original
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* waveform advances by dsp_phase_incr. This also has an integer
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* part and a fractional part.
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* If a sample is played at root pitch (no pitch change),
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* dsp_phase_incr is integer=1 and fractional=0.
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* - dsp_amp: The current amplitude envelope value.
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* - dsp_amp_incr: The changing rate of the amplitude envelope.
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*
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* A couple of variables are used internally, their results are discarded:
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* - dsp_i: Index through the output buffer
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* - dsp_phase_fractional: The fractional part of dsp_phase
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* - dsp_coeff: A table of four coefficients, depending on the fractional phase.
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* Used to interpolate between samples.
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* - dsp_process_buffer: Holds the processed signal between stages
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* - dsp_centernode: delay line for the IIR filter
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* - dsp_hist1: same
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* - dsp_hist2: same
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*
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*/
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/* Purpose:
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* zap_almost_zero will return a number, as long as its
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* absolute value is over a certain threshold. Otherwise 0. See
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* fluid_rev.c for documentation (denormal numbers)
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*/
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# if defined(WITH_FLOAT)
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# define zap_almost_zero(_sample) \
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((((*(unsigned int*)&(_sample))&0x7f800000) < 0x08000000)? 0.0f : (_sample))
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# else
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/* 1e-20 was chosen as an arbitrary (small) threshold. */
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#define zap_almost_zero(_sample) ((abs(_sample) < 1e-20)? 0.0f : (_sample))
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#endif
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/* Interpolation (find a value between two samples of the original waveform) */
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if ((fluid_phase_fract(dsp_phase) == 0)
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&& (fluid_phase_fract(dsp_phase_incr) == 0)
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&& (fluid_phase_index(dsp_phase_incr) == 1)) {
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/* Check for a special case: The current phase falls directly on an
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* original sample. Also, the stepsize per output sample is exactly
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* one sample, no fractional part. In other words: The sample is
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* played back at normal phase and root pitch. => No interpolation
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* needed.
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*/
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for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
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/* Mix to the buffer and advance the phase by one sample */
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dsp_buf[dsp_i] = dsp_amp * dsp_data[fluid_phase_index_plusplus(dsp_phase)];
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dsp_amp += dsp_amp_incr;
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}
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} else {
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/* wave table interpolation: Choose the interpolation method */
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switch(dsp_interp_method){
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case FLUID_INTERP_NONE:
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/* No interpolation. Just take the sample, which is closest to
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* the playback pointer. Questionable quality, but very
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* efficient. */
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for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
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dsp_phase_index = fluid_phase_index(dsp_phase);
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dsp_buf[dsp_i] = dsp_amp * dsp_data[dsp_phase_index];
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/* increment phase and amplitude */
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fluid_phase_incr(dsp_phase, dsp_phase_incr);
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dsp_amp += dsp_amp_incr;
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};
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break;
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case FLUID_INTERP_LINEAR:
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/* Straight line interpolation. */
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for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
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dsp_coeff = &interp_coeff_linear[fluid_phase_fract_to_tablerow(dsp_phase)];
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dsp_phase_index = fluid_phase_index(dsp_phase);
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dsp_buf[dsp_i] = (dsp_amp *
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(dsp_coeff->a0 * dsp_data[dsp_phase_index]
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+ dsp_coeff->a1 * dsp_data[dsp_phase_index+1]));
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/* increment phase and amplitude */
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fluid_phase_incr(dsp_phase, dsp_phase_incr);
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dsp_amp += dsp_amp_incr;
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};
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break;
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case FLUID_INTERP_4THORDER:
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default:
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/* Default interpolation loop using floats */
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for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
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dsp_coeff = &interp_coeff[fluid_phase_fract_to_tablerow(dsp_phase)];
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dsp_phase_index = fluid_phase_index(dsp_phase);
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dsp_buf[dsp_i] = (dsp_amp *
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(dsp_coeff->a0 * dsp_data[dsp_phase_index]
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+ dsp_coeff->a1 * dsp_data[dsp_phase_index+1]
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+ dsp_coeff->a2 * dsp_data[dsp_phase_index+2]
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+ dsp_coeff->a3 * dsp_data[dsp_phase_index+3]));
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/* increment phase and amplitude */
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fluid_phase_incr(dsp_phase, dsp_phase_incr);
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dsp_amp += dsp_amp_incr;
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}
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break;
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case FLUID_INTERP_7THORDER:
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for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
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int fract = fluid_phase_fract_to_tablerow(dsp_phase);
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dsp_phase_index = fluid_phase_index(dsp_phase);
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dsp_buf[dsp_i] = (dsp_amp *
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(sinc_table7[0][fract] * (fluid_real_t) dsp_data[dsp_phase_index]
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+ sinc_table7[1][fract] * (fluid_real_t) dsp_data[dsp_phase_index+1]
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+ sinc_table7[2][fract] * (fluid_real_t) dsp_data[dsp_phase_index+2]
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+ sinc_table7[3][fract] * (fluid_real_t) dsp_data[dsp_phase_index+3]
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+ sinc_table7[4][fract] * (fluid_real_t) dsp_data[dsp_phase_index+4]
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+ sinc_table7[5][fract] * (fluid_real_t) dsp_data[dsp_phase_index+5]
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+ sinc_table7[6][fract] * (fluid_real_t) dsp_data[dsp_phase_index+6]));
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/* increment phase and amplitude */
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fluid_phase_incr(dsp_phase, dsp_phase_incr);
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dsp_amp += dsp_amp_incr;
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}
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break;
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} /* switch interpolation method */
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} /* If interpolation is needed */
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/* filter (implement the voice filter according to Soundfont standard) */
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if (dsp_use_filter_flag) {
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/* Check for denormal number (too close to zero) once in a
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* while. This is not a big concern here - why would someone play a
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* sample with an empty tail? */
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dsp_hist1 = zap_almost_zero(dsp_hist1);
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/* Two versions of the filter loop. One, while the filter is
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* changing towards its new setting. The other, if the filter
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* doesn't change.
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*/
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if (dsp_filter_coeff_incr_count > 0) {
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/* The increment is added to each filter coefficient
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filter_coeff_incr_count times. */
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for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
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/* The filter is implemented in Direct-II form. */
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dsp_centernode = dsp_buf[dsp_i] - dsp_a1 * dsp_hist1 - dsp_a2 * dsp_hist2;
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dsp_buf[dsp_i] = dsp_b02 * (dsp_centernode + dsp_hist2) + dsp_b1 * dsp_hist1;
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dsp_hist2 = dsp_hist1;
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dsp_hist1 = dsp_centernode;
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if (dsp_filter_coeff_incr_count-- > 0){
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dsp_a1 += dsp_a1_incr;
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dsp_a2 += dsp_a2_incr;
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dsp_b02 += dsp_b02_incr;
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dsp_b1 += dsp_b1_incr;
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}
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} /* for dsp_i */
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} else {
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/* The filter parameters are constant. This is duplicated to save
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* time. */
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for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
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/* The filter is implemented in Direct-II form. */
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dsp_centernode = dsp_buf[dsp_i] - dsp_a1 * dsp_hist1 - dsp_a2 * dsp_hist2;
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dsp_buf[dsp_i] = dsp_b02 * (dsp_centernode + dsp_hist2) + dsp_b1 * dsp_hist1;
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dsp_hist2 = dsp_hist1;
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dsp_hist1 = dsp_centernode;
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}
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} /* if filter is fixed */
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} /* if filter is enabled */
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/* pan (Copy the signal to the left and right output buffer) The voice
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* panning generator has a range of -500 .. 500. If it is centered,
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* it's close to 0. voice->amp_left and voice->amp_right are then the
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* same, and we can save one multiplication per voice and sample.
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*/
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if ((-0.5 < voice->pan) && (voice->pan < 0.5)) {
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/* The voice is centered. Use voice->amp_left twice. */
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for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
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float v = voice->amp_left * dsp_buf[dsp_i];
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dsp_left_buf[dsp_i] += v;
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dsp_right_buf[dsp_i] += v;
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}
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} else {
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/* The voice is not centered. For stereo samples, one of the
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* amplitudes will be zero. */
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if (voice->amp_left != 0.0){
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for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
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dsp_left_buf[dsp_i] += voice->amp_left * dsp_buf[dsp_i];
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}
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}
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if (voice->amp_right != 0.0){
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for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
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dsp_right_buf[dsp_i] += voice->amp_right * dsp_buf[dsp_i];
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}
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}
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}
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/* reverb send. Buffer may be NULL. */
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if ((dsp_reverb_buf != NULL) && (voice->amp_reverb != 0.0)) {
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for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
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dsp_reverb_buf[dsp_i] += voice->amp_reverb * dsp_buf[dsp_i];
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}
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}
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/* chorus send. Buffer may be NULL. */
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if ((dsp_chorus_buf != NULL) && (voice->amp_chorus != 0)) {
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for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
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dsp_chorus_buf[dsp_i] += voice->amp_chorus * dsp_buf[dsp_i];
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}
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}
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120
fluidsynth/src/fluid_dsp_simple.c
Normal file
120
fluidsynth/src/fluid_dsp_simple.c
Normal file
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@ -0,0 +1,120 @@
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/* FluidSynth - A Software Synthesizer
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*
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* Copyright (C) 2003 Peter Hanappe and others.
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public License
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* as published by the Free Software Foundation; either version 2 of
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* the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful, but
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||||||
|
* 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
|
||||||
|
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA
|
||||||
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* 02111-1307, USA
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||||||
|
*/
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||||||
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|
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|
||||||
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/* Purpose:
|
||||||
|
* Low-level voice processing:
|
||||||
|
*
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||||||
|
* - interpolates (obtains values between the samples of the original waveform data)
|
||||||
|
* - filters (applies a lowpass filter with variable cutoff frequency and quality factor)
|
||||||
|
* - mixes the processed sample to left and right output using the pan setting
|
||||||
|
* - sends the processed sample to chorus and reverb
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*
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||||||
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*
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||||||
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* This file does -not- generate an object file.
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||||||
|
* Instead, it is #included in several places in fluid_voice.c.
|
||||||
|
* The motivation for this is
|
||||||
|
* - Calling it as a subroutine may be time consuming, especially with optimization off
|
||||||
|
* - The previous implementation as a macro was clumsy to handle
|
||||||
|
*
|
||||||
|
*
|
||||||
|
* Fluid_voice.c sets a couple of variables before #including this:
|
||||||
|
* - dsp_data: Pointer to the original waveform data
|
||||||
|
* - dsp_left_buf: The generated signal goes here, left channel
|
||||||
|
* - dsp_right_buf: right channel
|
||||||
|
* - dsp_reverb_buf: Send to reverb unit
|
||||||
|
* - dsp_chorus_buf: Send to chorus unit
|
||||||
|
* - dsp_start: Start processing at this output buffer index
|
||||||
|
* - dsp_end: End processing just before this output buffer index
|
||||||
|
* - dsp_a1: Coefficient for the filter
|
||||||
|
* - dsp_a2: same
|
||||||
|
* - dsp_b0: same
|
||||||
|
* - dsp_b1: same
|
||||||
|
* - dsp_b2: same
|
||||||
|
* - dsp_filter_flag: Set, the filter is needed (many sound fonts don't use
|
||||||
|
* the filter at all. If it is left at its default setting
|
||||||
|
* of roughly 20 kHz, there is no need to apply filterling.)
|
||||||
|
* - dsp_interp_method: Which interpolation method to use.
|
||||||
|
* - voice holds the voice structure
|
||||||
|
*
|
||||||
|
* Some variables are set and modified:
|
||||||
|
* - dsp_phase: The position in the original waveform data.
|
||||||
|
* This has an integer and a fractional part (between samples).
|
||||||
|
* - dsp_phase_incr: For each output sample, the position in the original
|
||||||
|
* waveform advances by dsp_phase_incr. This also has an integer
|
||||||
|
* part and a fractional part.
|
||||||
|
* If a sample is played at root pitch (no pitch change),
|
||||||
|
* dsp_phase_incr is integer=1 and fractional=0.
|
||||||
|
* - dsp_amp: The current amplitude envelope value.
|
||||||
|
* - dsp_amp_incr: The changing rate of the amplitude envelope.
|
||||||
|
*
|
||||||
|
* A couple of variables are used internally, their results are discarded:
|
||||||
|
* - dsp_i: Index through the output buffer
|
||||||
|
* - dsp_phase_fractional: The fractional part of dsp_phase
|
||||||
|
* - dsp_coeff: A table of four coefficients, depending on the fractional phase.
|
||||||
|
* Used to interpolate between samples.
|
||||||
|
* - dsp_process_buffer: Holds the processed signal between stages
|
||||||
|
* - dsp_centernode: delay line for the IIR filter
|
||||||
|
* - dsp_hist1: same
|
||||||
|
* - dsp_hist2: same
|
||||||
|
*
|
||||||
|
*/
|
||||||
|
|
||||||
|
|
||||||
|
/* Nonoptimized DSP loop */
|
||||||
|
#warning "This code is meant for experiments only.";
|
||||||
|
|
||||||
|
/* wave table interpolation */
|
||||||
|
for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
|
||||||
|
|
||||||
|
dsp_coeff = &interp_coeff[fluid_phase_fract_to_tablerow(dsp_phase)];
|
||||||
|
dsp_phase_index = fluid_phase_index(dsp_phase);
|
||||||
|
dsp_sample = (dsp_amp *
|
||||||
|
(dsp_coeff->a0 * dsp_data[dsp_phase_index]
|
||||||
|
+ dsp_coeff->a1 * dsp_data[dsp_phase_index+1]
|
||||||
|
+ dsp_coeff->a2 * dsp_data[dsp_phase_index+2]
|
||||||
|
+ dsp_coeff->a3 * dsp_data[dsp_phase_index+3]));
|
||||||
|
|
||||||
|
/* increment phase and amplitude */
|
||||||
|
fluid_phase_incr(dsp_phase, dsp_phase_incr);
|
||||||
|
dsp_amp += dsp_amp_incr;
|
||||||
|
|
||||||
|
/* filter */
|
||||||
|
/* The filter is implemented in Direct-II form. */
|
||||||
|
dsp_centernode = dsp_sample - dsp_a1 * dsp_hist1 - dsp_a2 * dsp_hist2;
|
||||||
|
dsp_sample = dsp_b0 * dsp_centernode + dsp_b1 * dsp_hist1 + dsp_b2 * dsp_hist2;
|
||||||
|
dsp_hist2 = dsp_hist1;
|
||||||
|
dsp_hist1 = dsp_centernode;
|
||||||
|
|
||||||
|
/* pan */
|
||||||
|
dsp_left_buf[dsp_i] += voice->amp_left * dsp_sample;
|
||||||
|
dsp_right_buf[dsp_i] += voice->amp_right * dsp_sample;
|
||||||
|
|
||||||
|
/* reverb */
|
||||||
|
if (dsp_reverb_buf){
|
||||||
|
dsp_reverb_buf[dsp_i] += voice->amp_reverb * dsp_sample;
|
||||||
|
}
|
||||||
|
|
||||||
|
/* chorus */
|
||||||
|
if (dsp_chorus_buf){
|
||||||
|
dsp_chorus_buf[dsp_i] += voice->amp_chorus * dsp_sample;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
387
fluidsynth/src/fluid_dsp_sse.c
Normal file
387
fluidsynth/src/fluid_dsp_sse.c
Normal file
|
|
@ -0,0 +1,387 @@
|
||||||
|
/* 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
|
||||||
|
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA
|
||||||
|
* 02111-1307, USA
|
||||||
|
*/
|
||||||
|
|
||||||
|
|
||||||
|
/* Purpose:
|
||||||
|
* Low-level voice processing:
|
||||||
|
*
|
||||||
|
* - interpolates (obtains values between the samples of the original waveform data)
|
||||||
|
* - filters (applies a lowpass filter with variable cutoff frequency and quality factor)
|
||||||
|
* - mixes the processed sample to left and right output using the pan setting
|
||||||
|
* - sends the processed sample to chorus and reverb
|
||||||
|
*
|
||||||
|
*
|
||||||
|
* This file does -not- generate an object file.
|
||||||
|
* Instead, it is #included in several places in fluid_voice.c.
|
||||||
|
* The motivation for this is
|
||||||
|
* - Calling it as a subroutine may be time consuming, especially with optimization off
|
||||||
|
* - The previous implementation as a macro was clumsy to handle
|
||||||
|
*
|
||||||
|
*
|
||||||
|
* Fluid_voice.c sets a couple of variables before #including this:
|
||||||
|
* - dsp_data: Pointer to the original waveform data
|
||||||
|
* - dsp_left_buf: The generated signal goes here, left channel
|
||||||
|
* - dsp_right_buf: right channel
|
||||||
|
* - dsp_reverb_buf: Send to reverb unit
|
||||||
|
* - dsp_chorus_buf: Send to chorus unit
|
||||||
|
* - dsp_start: Start processing at this output buffer index
|
||||||
|
* - dsp_end: End processing just before this output buffer index
|
||||||
|
* - dsp_a1: Coefficient for the filter
|
||||||
|
* - dsp_a2: same
|
||||||
|
* - dsp_b0: same
|
||||||
|
* - dsp_b1: same
|
||||||
|
* - dsp_b2: same
|
||||||
|
* - dsp_filter_flag: Set, the filter is needed (many sound fonts don't use
|
||||||
|
* the filter at all. If it is left at its default setting
|
||||||
|
* of roughly 20 kHz, there is no need to apply filterling.)
|
||||||
|
* - dsp_interp_method: Which interpolation method to use.
|
||||||
|
* - voice holds the voice structure
|
||||||
|
*
|
||||||
|
* Some variables are set and modified:
|
||||||
|
* - dsp_phase: The position in the original waveform data.
|
||||||
|
* This has an integer and a fractional part (between samples).
|
||||||
|
* - dsp_phase_incr: For each output sample, the position in the original
|
||||||
|
* waveform advances by dsp_phase_incr. This also has an integer
|
||||||
|
* part and a fractional part.
|
||||||
|
* If a sample is played at root pitch (no pitch change),
|
||||||
|
* dsp_phase_incr is integer=1 and fractional=0.
|
||||||
|
* - dsp_amp: The current amplitude envelope value.
|
||||||
|
* - dsp_amp_incr: The changing rate of the amplitude envelope.
|
||||||
|
*
|
||||||
|
* A couple of variables are used internally, their results are discarded:
|
||||||
|
* - dsp_i: Index through the output buffer
|
||||||
|
* - dsp_phase_fractional: The fractional part of dsp_phase
|
||||||
|
* - dsp_coeff: A table of four coefficients, depending on the fractional phase.
|
||||||
|
* Used to interpolate between samples.
|
||||||
|
* - dsp_process_buffer: Holds the processed signal between stages
|
||||||
|
* - dsp_centernode: delay line for the IIR filter
|
||||||
|
* - dsp_hist1: same
|
||||||
|
* - dsp_hist2: same
|
||||||
|
*
|
||||||
|
*/
|
||||||
|
|
||||||
|
/* Purpose:
|
||||||
|
* zap_almost_zero will return a number, as long as its
|
||||||
|
* absolute value is over a certain threshold. Otherwise 0. See
|
||||||
|
* fluid_rev.c for documentation (denormal numbers)
|
||||||
|
*/
|
||||||
|
|
||||||
|
# if defined(WITH_FLOAT)
|
||||||
|
# define zap_almost_zero(_sample) \
|
||||||
|
((((*(unsigned int*)&(_sample))&0x7f800000) < 0x08000000)? 0.0f : (_sample))
|
||||||
|
# else
|
||||||
|
/* 1e-20 was chosen as an arbitrary (small) threshold. */
|
||||||
|
#define zap_almost_zero(_sample) ((abs(_sample) < 1e-20)? 0.0f : (_sample))
|
||||||
|
#endif
|
||||||
|
|
||||||
|
/* Interpolation (find a value between two samples of the original waveform) */
|
||||||
|
|
||||||
|
if ((fluid_phase_fract(dsp_phase) == 0)
|
||||||
|
&& (fluid_phase_fract(dsp_phase_incr) == 0)
|
||||||
|
&& (fluid_phase_index(dsp_phase_incr) == 1)) {
|
||||||
|
|
||||||
|
/* Check for a special case: The current phase falls directly on an
|
||||||
|
* original sample. Also, the stepsize per output sample is exactly
|
||||||
|
* one sample, no fractional part. In other words: The sample is
|
||||||
|
* played back at normal phase and root pitch. => No interpolation
|
||||||
|
* needed.
|
||||||
|
*/
|
||||||
|
for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
|
||||||
|
/* Mix to the buffer and advance the phase by one sample */
|
||||||
|
dsp_buf[dsp_i] = dsp_amp * dsp_data[fluid_phase_index_plusplus(dsp_phase)];
|
||||||
|
dsp_amp += dsp_amp_incr;
|
||||||
|
}
|
||||||
|
|
||||||
|
} else {
|
||||||
|
|
||||||
|
/* wave table interpolation: Choose the interpolation method */
|
||||||
|
/* !!! SSE interpolation is less efficient that normal interpolation. */
|
||||||
|
|
||||||
|
|
||||||
|
/* Initialize amplitude increase */
|
||||||
|
sse_b->sf[0] = sse_b->sf[1] = sse_b->sf[2] = sse_b->sf[3] = 4.*dsp_amp_incr;
|
||||||
|
|
||||||
|
/* Initialize amplitude => xmm7
|
||||||
|
* The amplitude is kept in xmm7 throughout the whole process
|
||||||
|
*/
|
||||||
|
|
||||||
|
sse_a->sf[0]=sse_a->sf[1]=sse_a->sf[2]=sse_a->sf[3]=dsp_amp;
|
||||||
|
sse_a->sf[1] += dsp_amp_incr;
|
||||||
|
sse_a->sf[2] += 2.* dsp_amp_incr;
|
||||||
|
sse_a->sf[3] += 3.* dsp_amp_incr;
|
||||||
|
movaps_m2r(*sse_a,xmm7);
|
||||||
|
|
||||||
|
/* Where to store the result */
|
||||||
|
sse_dest=(sse_t*)&dsp_buf[0];
|
||||||
|
for (dsp_i = 0; dsp_i < FLUID_BUFSIZE; dsp_i += 4) {
|
||||||
|
/* Note / fixme: The coefficients are first copied to
|
||||||
|
* sse_c, then to the xmm register. Can't get it through
|
||||||
|
* the compiler differently... */
|
||||||
|
|
||||||
|
/* Load the four source samples for the 1st output sample */
|
||||||
|
dsp_phase_index = fluid_phase_index(dsp_phase);
|
||||||
|
sse_a->sf[0]=(fluid_real_t)dsp_data[dsp_phase_index];
|
||||||
|
sse_a->sf[1]=(fluid_real_t)dsp_data[dsp_phase_index+1];
|
||||||
|
sse_a->sf[2]=(fluid_real_t)dsp_data[dsp_phase_index+2];
|
||||||
|
sse_a->sf[3]=(fluid_real_t)dsp_data[dsp_phase_index+3];
|
||||||
|
movaps_m2r(*sse_a,xmm0);
|
||||||
|
*sse_c=interp_coeff_sse[fluid_phase_fract_to_tablerow(dsp_phase)];
|
||||||
|
mulps_m2r(*sse_c,xmm0);
|
||||||
|
|
||||||
|
fluid_phase_incr(dsp_phase, dsp_phase_incr);
|
||||||
|
|
||||||
|
/* Load the four source samples for the 2nd output sample */
|
||||||
|
dsp_phase_index = fluid_phase_index(dsp_phase);
|
||||||
|
sse_a->sf[0]=(fluid_real_t)dsp_data[dsp_phase_index];
|
||||||
|
sse_a->sf[1]=(fluid_real_t)dsp_data[dsp_phase_index+1];
|
||||||
|
sse_a->sf[2]=(fluid_real_t)dsp_data[dsp_phase_index+2];
|
||||||
|
sse_a->sf[3]=(fluid_real_t)dsp_data[dsp_phase_index+3];
|
||||||
|
movaps_m2r(*sse_a,xmm1);
|
||||||
|
*sse_c=interp_coeff_sse[fluid_phase_fract_to_tablerow(dsp_phase)];
|
||||||
|
mulps_m2r(*sse_c,xmm1);
|
||||||
|
|
||||||
|
fluid_phase_incr(dsp_phase, dsp_phase_incr);
|
||||||
|
|
||||||
|
/* Load the four source samples for the 3rd output sample */
|
||||||
|
dsp_phase_index = fluid_phase_index(dsp_phase);
|
||||||
|
sse_a->sf[0]=(fluid_real_t)dsp_data[dsp_phase_index];
|
||||||
|
sse_a->sf[1]=(fluid_real_t)dsp_data[dsp_phase_index+1];
|
||||||
|
sse_a->sf[2]=(fluid_real_t)dsp_data[dsp_phase_index+2];
|
||||||
|
sse_a->sf[3]=(fluid_real_t)dsp_data[dsp_phase_index+3];
|
||||||
|
movaps_m2r(*sse_a,xmm2);
|
||||||
|
*sse_c=interp_coeff_sse[fluid_phase_fract_to_tablerow(dsp_phase)];
|
||||||
|
mulps_m2r(*sse_c,xmm2);
|
||||||
|
|
||||||
|
fluid_phase_incr(dsp_phase, dsp_phase_incr);
|
||||||
|
|
||||||
|
/* Load the four source samples for the 4th output sample */
|
||||||
|
dsp_phase_index = fluid_phase_index(dsp_phase);
|
||||||
|
sse_a->sf[0]=(fluid_real_t)dsp_data[dsp_phase_index];
|
||||||
|
sse_a->sf[1]=(fluid_real_t)dsp_data[dsp_phase_index+1];
|
||||||
|
sse_a->sf[2]=(fluid_real_t)dsp_data[dsp_phase_index+2];
|
||||||
|
sse_a->sf[3]=(fluid_real_t)dsp_data[dsp_phase_index+3];
|
||||||
|
movaps_m2r(*sse_a,xmm3);
|
||||||
|
*sse_c=interp_coeff_sse[fluid_phase_fract_to_tablerow(dsp_phase)];
|
||||||
|
mulps_m2r(*sse_c,xmm3);
|
||||||
|
|
||||||
|
fluid_phase_incr(dsp_phase, dsp_phase_incr);
|
||||||
|
|
||||||
|
#if 0
|
||||||
|
/*Testcase for horizontal add */
|
||||||
|
sse_a->sf[0]=0.1;sse_a->sf[1]=0.01;sse_a->sf[2]=0.001;sse_a->sf[3]=0.0001;
|
||||||
|
movaps_m2r(*sse_a,xmm0);
|
||||||
|
sse_a->sf[0]=0.2;sse_a->sf[1]=0.02;sse_a->sf[2]=0.002;sse_a->sf[3]=0.0002;
|
||||||
|
movaps_m2r(*sse_a,xmm1);
|
||||||
|
sse_a->sf[0]=0.3;sse_a->sf[1]=0.03;sse_a->sf[2]=0.003;sse_a->sf[3]=0.0003;
|
||||||
|
movaps_m2r(*sse_a,xmm2);
|
||||||
|
sse_a->sf[0]=0.4;sse_a->sf[1]=0.04;sse_a->sf[2]=0.004;sse_a->sf[3]=0.0004;
|
||||||
|
movaps_m2r(*sse_a,xmm3);
|
||||||
|
#endif /* #if 0 */
|
||||||
|
|
||||||
|
#if 1
|
||||||
|
/* Horizontal add
|
||||||
|
* xmm4[0]:=xmm0[0]+xmm1[0]+xmm2[0]+xmm3[0]
|
||||||
|
* xmm4[1]:=xmm0[1]+xmm1[1]+xmm2[1]+xmm3[1]
|
||||||
|
* etc.
|
||||||
|
* The only register, which is unused, is xmm7.
|
||||||
|
*/
|
||||||
|
movaps_r2r(xmm0,xmm5);
|
||||||
|
movaps_r2r(xmm2,xmm6);
|
||||||
|
movlhps_r2r(xmm1,xmm5);
|
||||||
|
movlhps_r2r(xmm3,xmm6);
|
||||||
|
movhlps_r2r(xmm0,xmm1);
|
||||||
|
movhlps_r2r(xmm2,xmm3);
|
||||||
|
addps_r2r(xmm1,xmm5);
|
||||||
|
addps_r2r(xmm3,xmm6);
|
||||||
|
movaps_r2r(xmm5,xmm4);
|
||||||
|
shufps_r2r(xmm6,xmm5,0xDD);
|
||||||
|
shufps_r2r(xmm6,xmm4,0x88);
|
||||||
|
addps_r2r(xmm5,xmm4);
|
||||||
|
|
||||||
|
/* movaps_r2m(xmm4,*sse_a); */
|
||||||
|
/* printf("xmm4 (Result): %f %f %f %f\n", */
|
||||||
|
/* sse_a->sf[0], sse_a->sf[1], */
|
||||||
|
/* sse_a->sf[2], sse_a->sf[3]); */
|
||||||
|
#else
|
||||||
|
/* Add using normal FPU */
|
||||||
|
movaps_r2m(xmm0,*sse_a);
|
||||||
|
sse_c->sf[0]=sse_a->sf[0]+sse_a->sf[1]+sse_a->sf[2]+sse_a->sf[3];
|
||||||
|
movaps_r2m(xmm1,*sse_a);
|
||||||
|
sse_c->sf[1]=sse_a->sf[0]+sse_a->sf[1]+sse_a->sf[2]+sse_a->sf[3];
|
||||||
|
movaps_r2m(xmm2,*sse_a);
|
||||||
|
sse_c->sf[2]=sse_a->sf[0]+sse_a->sf[1]+sse_a->sf[2]+sse_a->sf[3];
|
||||||
|
movaps_r2m(xmm3,*sse_a);
|
||||||
|
sse_c->sf[3]=sse_a->sf[0]+sse_a->sf[1]+sse_a->sf[2]+sse_a->sf[3];
|
||||||
|
movaps_m2r(*sse_c,xmm4);
|
||||||
|
#endif /* #if 1 */
|
||||||
|
/* end horizontal add. Result in xmm6. */
|
||||||
|
|
||||||
|
|
||||||
|
/* Multiply xmm4 with amplitude */
|
||||||
|
mulps_r2r(xmm7,xmm4);
|
||||||
|
|
||||||
|
/* Store the result */
|
||||||
|
movaps_r2m(xmm4,*sse_dest); // ++
|
||||||
|
|
||||||
|
/* Advance the position in the output buffer */
|
||||||
|
sse_dest++;
|
||||||
|
|
||||||
|
/* Change the amplitude */
|
||||||
|
addps_m2r(*sse_b,xmm7);
|
||||||
|
|
||||||
|
} /* for dsp_i in steps of four */
|
||||||
|
|
||||||
|
movaps_r2m(xmm7,*sse_a);
|
||||||
|
|
||||||
|
/* Retrieve the last amplitude value. */
|
||||||
|
dsp_amp=sse_a->sf[3];
|
||||||
|
|
||||||
|
|
||||||
|
} /* If interpolation is needed */
|
||||||
|
|
||||||
|
/* filter (implement the voice filter according to Soundfont standard) */
|
||||||
|
if (dsp_use_filter_flag) {
|
||||||
|
|
||||||
|
/* Check for denormal number (too close to zero) once in a
|
||||||
|
* while. This is not a big concern here - why would someone play a
|
||||||
|
* sample with an empty tail? */
|
||||||
|
dsp_hist1 = zap_almost_zero(dsp_hist1);
|
||||||
|
|
||||||
|
/* Two versions of the filter loop. One, while the filter is
|
||||||
|
* changing towards its new setting. The other, if the filter
|
||||||
|
* doesn't change.
|
||||||
|
*/
|
||||||
|
|
||||||
|
if (dsp_filter_coeff_incr_count > 0) {
|
||||||
|
/* The increment is added to each filter coefficient
|
||||||
|
filter_coeff_incr_count times. */
|
||||||
|
|
||||||
|
for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
|
||||||
|
/* The filter is implemented in Direct-II form. */
|
||||||
|
dsp_centernode = dsp_buf[dsp_i] - dsp_a1 * dsp_hist1 - dsp_a2 * dsp_hist2;
|
||||||
|
dsp_buf[dsp_i] = dsp_b02 * (dsp_centernode + dsp_hist2) + dsp_b1 * dsp_hist1;
|
||||||
|
dsp_hist2 = dsp_hist1;
|
||||||
|
dsp_hist1 = dsp_centernode;
|
||||||
|
|
||||||
|
if (dsp_filter_coeff_incr_count-- > 0){
|
||||||
|
dsp_a1 += dsp_a1_incr;
|
||||||
|
dsp_a2 += dsp_a2_incr;
|
||||||
|
dsp_b02 += dsp_b02_incr;
|
||||||
|
dsp_b1 += dsp_b1_incr;
|
||||||
|
}
|
||||||
|
} /* for dsp_i */
|
||||||
|
|
||||||
|
} else {
|
||||||
|
|
||||||
|
/* The filter parameters are constant. This is duplicated to save
|
||||||
|
* time. */
|
||||||
|
|
||||||
|
for (dsp_i = dsp_start; dsp_i < dsp_end; dsp_i++) {
|
||||||
|
/* The filter is implemented in Direct-II form. */
|
||||||
|
dsp_centernode = dsp_buf[dsp_i] - dsp_a1 * dsp_hist1 - dsp_a2 * dsp_hist2;
|
||||||
|
dsp_buf[dsp_i] = dsp_b02 * (dsp_centernode + dsp_hist2) + dsp_b1 * dsp_hist1;
|
||||||
|
dsp_hist2 = dsp_hist1;
|
||||||
|
dsp_hist1 = dsp_centernode;
|
||||||
|
}
|
||||||
|
} /* if filter is fixed */
|
||||||
|
} /* if filter is enabled */
|
||||||
|
|
||||||
|
|
||||||
|
/* The following optimization will process a whole buffer using the
|
||||||
|
* SSE extension of the Pentium processor.
|
||||||
|
*/
|
||||||
|
|
||||||
|
if (voice->amp_left != 0.0) {
|
||||||
|
sse_a->sf[0]=voice->amp_left;
|
||||||
|
sse_a->sf[1]=voice->amp_left;
|
||||||
|
sse_a->sf[2]=voice->amp_left;
|
||||||
|
sse_a->sf[3]=voice->amp_left;
|
||||||
|
movaps_m2r(*sse_a,xmm0);
|
||||||
|
|
||||||
|
sse_src=(sse_t*)dsp_buf;
|
||||||
|
sse_dest=(sse_t*)&dsp_left_buf[0];
|
||||||
|
for (dsp_i = 0; dsp_i < FLUID_BUFSIZE; dsp_i+=4) {
|
||||||
|
movaps_m2r(*sse_src,xmm4); /* Load original sample */
|
||||||
|
mulps_r2r(xmm0,xmm4); /* Gain */
|
||||||
|
sse_src++;
|
||||||
|
addps_m2r(*sse_dest,xmm4); /* Mix with buf */
|
||||||
|
movaps_r2m(xmm4,*sse_dest); /* Store in buf */
|
||||||
|
sse_dest++;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
if (voice->amp_right != 0.0){
|
||||||
|
sse_a->sf[0]=voice->amp_right;
|
||||||
|
sse_a->sf[1]=voice->amp_right;
|
||||||
|
sse_a->sf[2]=voice->amp_right;
|
||||||
|
sse_a->sf[3]=voice->amp_right;
|
||||||
|
movaps_m2r(*sse_a,xmm0);
|
||||||
|
|
||||||
|
sse_src=(sse_t*)dsp_buf;
|
||||||
|
sse_dest=(sse_t*)&dsp_right_buf[0];
|
||||||
|
for (dsp_i = 0; dsp_i < FLUID_BUFSIZE; dsp_i+=4) {
|
||||||
|
movaps_m2r(*sse_src,xmm4); /* Load original sample */
|
||||||
|
sse_src++;
|
||||||
|
mulps_r2r(xmm0,xmm4); /* Gain */
|
||||||
|
addps_m2r(*sse_dest,xmm4); /* Mix with buf */
|
||||||
|
movaps_r2m(xmm4,*sse_dest); /* Store in buf */
|
||||||
|
sse_dest++;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
/* reverb send. Buffer may be NULL. */
|
||||||
|
if (dsp_reverb_buf && voice->amp_reverb != 0.0){
|
||||||
|
sse_a->sf[0]=voice->amp_reverb;
|
||||||
|
sse_a->sf[1]=voice->amp_reverb;
|
||||||
|
sse_a->sf[2]=voice->amp_reverb;
|
||||||
|
sse_a->sf[3]=voice->amp_reverb;
|
||||||
|
movaps_m2r(*sse_a,xmm0);
|
||||||
|
|
||||||
|
sse_src=(sse_t*)dsp_buf;
|
||||||
|
sse_dest=(sse_t*)&dsp_reverb_buf[0];
|
||||||
|
for (dsp_i = 0; dsp_i < FLUID_BUFSIZE; dsp_i+=4) {
|
||||||
|
movaps_m2r(*sse_src,xmm4); /* Load original sample */
|
||||||
|
sse_src++;
|
||||||
|
mulps_r2r(xmm0,xmm4); /* Gain */
|
||||||
|
addps_m2r(*sse_dest,xmm4); /* Mix with buf */
|
||||||
|
movaps_r2m(xmm4,*sse_dest); /* Store in buf */
|
||||||
|
sse_dest++;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
/* chorus send. Buffer may be NULL. */
|
||||||
|
if (dsp_chorus_buf && voice->amp_chorus != 0){
|
||||||
|
sse_a->sf[0]=voice->amp_chorus;
|
||||||
|
sse_a->sf[1]=voice->amp_chorus;
|
||||||
|
sse_a->sf[2]=voice->amp_chorus;
|
||||||
|
sse_a->sf[3]=voice->amp_chorus;
|
||||||
|
movaps_m2r(*sse_a,xmm0);
|
||||||
|
|
||||||
|
sse_src=(sse_t*)dsp_buf;
|
||||||
|
sse_dest=(sse_t*)&dsp_chorus_buf[0];
|
||||||
|
for (dsp_i = 0; dsp_i < FLUID_BUFSIZE; dsp_i+=4) {
|
||||||
|
movaps_m2r(*sse_src,xmm4); /* Load original sample */
|
||||||
|
sse_src++;
|
||||||
|
mulps_r2r(xmm0,xmm4); /* Gain */
|
||||||
|
addps_m2r(*sse_dest,xmm4); /* Mix with buf */
|
||||||
|
movaps_r2m(xmm4,*sse_dest); /* Store in buf */
|
||||||
|
sse_dest++;
|
||||||
|
}
|
||||||
|
}
|
||||||
Loading…
Reference in a new issue