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599 lines
13 KiB
C
599 lines
13 KiB
C
/*
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synth_s32.c: The functions for synthesizing real (float) samples, at the end of decoding.
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copyright 1995-2008 by the mpg123 project - free software under the terms of the LGPL 2.1
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see COPYING and AUTHORS files in distribution or http://mpg123.org
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initially written by Michael Hipp, heavily dissected and rearranged by Thomas Orgis
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*/
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#include "mpg123lib_intern.h"
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#include "sample.h"
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#include "debug.h"
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#ifdef REAL_IS_FIXED
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#error "Do not build this file with fixed point math!"
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#else
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/*
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Part 4: All synth functions that produce signed 32 bit output.
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What we need is just a special WRITE_SAMPLE.
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*/
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#define SAMPLE_T int32_t
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#define WRITE_SAMPLE(samples,sum,clip) WRITE_S32_SAMPLE(samples,sum,clip)
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/* Part 4a: All straight 1to1 decoding functions */
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#define BLOCK 0x40 /* One decoding block is 64 samples. */
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#define SYNTH_NAME synth_1to1_s32
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#include "synth.h"
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#undef SYNTH_NAME
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/* Mono-related synths; they wrap over _some_ synth_1to1_s32 (could be generic, could be i386). */
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#define SYNTH_NAME fr->synths.plain[r_1to1][f_32]
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#define MONO_NAME synth_1to1_s32_mono
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#define MONO2STEREO_NAME synth_1to1_s32_m2s
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#include "synth_mono.h"
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#undef SYNTH_NAME
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#undef MONO_NAME
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#undef MONO2STEREO_NAME
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#ifdef OPT_X86
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#define NO_AUTOINCREMENT
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#define SYNTH_NAME synth_1to1_s32_i386
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#include "synth.h"
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#undef SYNTH_NAME
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/* i386 uses the normal mono functions. */
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#undef NO_AUTOINCREMENT
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#endif
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#ifdef OPT_X86_64
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/* Assembler routines. */
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int synth_1to1_s32_x86_64_asm(real *window, real *b0, int32_t *samples, int bo1);
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int synth_1to1_s32_s_x86_64_asm(real *window, real *b0l, real *b0r, int32_t *samples, int bo1);
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void dct64_real_x86_64(real *out0, real *out1, real *samples);
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/* Hull for C mpg123 API */
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int synth_1to1_s32_x86_64(real *bandPtr,int channel, mpg123_handle *fr, int final)
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{
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int32_t *samples = (int32_t *) (fr->buffer.data+fr->buffer.fill);
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real *b0, **buf;
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int bo1;
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int clip;
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#ifndef NO_EQUALIZER
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if(fr->have_eq_settings) do_equalizer(bandPtr,channel,fr->equalizer);
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#endif
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if(!channel)
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{
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fr->bo--;
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fr->bo &= 0xf;
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buf = fr->real_buffs[0];
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}
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else
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{
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samples++;
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buf = fr->real_buffs[1];
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}
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if(fr->bo & 0x1)
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{
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b0 = buf[0];
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bo1 = fr->bo;
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dct64_real_x86_64(buf[1]+((fr->bo+1)&0xf),buf[0]+fr->bo,bandPtr);
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}
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else
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{
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b0 = buf[1];
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bo1 = fr->bo+1;
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dct64_real_x86_64(buf[0]+fr->bo,buf[1]+fr->bo+1,bandPtr);
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}
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clip = synth_1to1_s32_x86_64_asm(fr->decwin, b0, samples, bo1);
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if(final) fr->buffer.fill += 256;
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return clip;
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}
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int synth_1to1_s32_stereo_x86_64(real *bandPtr_l, real *bandPtr_r, mpg123_handle *fr)
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{
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int32_t *samples = (int32_t *) (fr->buffer.data+fr->buffer.fill);
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real *b0l, *b0r, **bufl, **bufr;
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int bo1;
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int clip;
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#ifndef NO_EQUALIZER
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if(fr->have_eq_settings)
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{
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do_equalizer(bandPtr_l,0,fr->equalizer);
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do_equalizer(bandPtr_r,1,fr->equalizer);
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}
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#endif
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fr->bo--;
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fr->bo &= 0xf;
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bufl = fr->real_buffs[0];
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bufr = fr->real_buffs[1];
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if(fr->bo & 0x1)
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{
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b0l = bufl[0];
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b0r = bufr[0];
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bo1 = fr->bo;
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dct64_real_x86_64(bufl[1]+((fr->bo+1)&0xf),bufl[0]+fr->bo,bandPtr_l);
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dct64_real_x86_64(bufr[1]+((fr->bo+1)&0xf),bufr[0]+fr->bo,bandPtr_r);
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}
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else
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{
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b0l = bufl[1];
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b0r = bufr[1];
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bo1 = fr->bo+1;
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dct64_real_x86_64(bufl[0]+fr->bo,bufl[1]+fr->bo+1,bandPtr_l);
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dct64_real_x86_64(bufr[0]+fr->bo,bufr[1]+fr->bo+1,bandPtr_r);
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}
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clip = synth_1to1_s32_s_x86_64_asm(fr->decwin, b0l, b0r, samples, bo1);
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fr->buffer.fill += 256;
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return clip;
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}
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#endif
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#ifdef OPT_AVX
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/* Assembler routines. */
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#ifndef OPT_x86_64
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int synth_1to1_s32_x86_64_asm(real *window, real *b0, int32_t *samples, int bo1);
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#endif
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int synth_1to1_s32_s_avx_asm(real *window, real *b0l, real *b0r, int32_t *samples, int bo1);
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void dct64_real_avx(real *out0, real *out1, real *samples);
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/* Hull for C mpg123 API */
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int synth_1to1_s32_avx(real *bandPtr,int channel, mpg123_handle *fr, int final)
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{
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int32_t *samples = (int32_t *) (fr->buffer.data+fr->buffer.fill);
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real *b0, **buf;
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int bo1;
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int clip;
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#ifndef NO_EQUALIZER
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if(fr->have_eq_settings) do_equalizer(bandPtr,channel,fr->equalizer);
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#endif
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if(!channel)
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{
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fr->bo--;
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fr->bo &= 0xf;
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buf = fr->real_buffs[0];
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}
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else
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{
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samples++;
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buf = fr->real_buffs[1];
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}
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if(fr->bo & 0x1)
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{
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b0 = buf[0];
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bo1 = fr->bo;
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dct64_real_avx(buf[1]+((fr->bo+1)&0xf),buf[0]+fr->bo,bandPtr);
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}
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else
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{
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b0 = buf[1];
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bo1 = fr->bo+1;
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dct64_real_avx(buf[0]+fr->bo,buf[1]+fr->bo+1,bandPtr);
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}
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clip = synth_1to1_s32_x86_64_asm(fr->decwin, b0, samples, bo1);
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if(final) fr->buffer.fill += 256;
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return clip;
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}
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int synth_1to1_s32_stereo_avx(real *bandPtr_l, real *bandPtr_r, mpg123_handle *fr)
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{
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int32_t *samples = (int32_t *) (fr->buffer.data+fr->buffer.fill);
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real *b0l, *b0r, **bufl, **bufr;
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int bo1;
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int clip;
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#ifndef NO_EQUALIZER
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if(fr->have_eq_settings)
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{
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do_equalizer(bandPtr_l,0,fr->equalizer);
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do_equalizer(bandPtr_r,1,fr->equalizer);
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}
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#endif
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fr->bo--;
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fr->bo &= 0xf;
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bufl = fr->real_buffs[0];
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bufr = fr->real_buffs[1];
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if(fr->bo & 0x1)
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{
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b0l = bufl[0];
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b0r = bufr[0];
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bo1 = fr->bo;
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dct64_real_avx(bufl[1]+((fr->bo+1)&0xf),bufl[0]+fr->bo,bandPtr_l);
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dct64_real_avx(bufr[1]+((fr->bo+1)&0xf),bufr[0]+fr->bo,bandPtr_r);
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}
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else
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{
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b0l = bufl[1];
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b0r = bufr[1];
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bo1 = fr->bo+1;
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dct64_real_avx(bufl[0]+fr->bo,bufl[1]+fr->bo+1,bandPtr_l);
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dct64_real_avx(bufr[0]+fr->bo,bufr[1]+fr->bo+1,bandPtr_r);
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}
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clip = synth_1to1_s32_s_avx_asm(fr->decwin, b0l, b0r, samples, bo1);
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fr->buffer.fill += 256;
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return clip;
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}
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#endif
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#if defined(OPT_SSE) || defined(OPT_SSE_VINTAGE)
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/* Assembler routines. */
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int synth_1to1_s32_sse_asm(real *window, real *b0, int32_t *samples, int bo1);
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int synth_1to1_s32_s_sse_asm(real *window, real *b0l, real *b0r, int32_t *samples, int bo1);
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void dct64_real_sse(real *out0, real *out1, real *samples);
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/* Hull for C mpg123 API */
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int synth_1to1_s32_sse(real *bandPtr,int channel, mpg123_handle *fr, int final)
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{
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int32_t *samples = (int32_t *) (fr->buffer.data+fr->buffer.fill);
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real *b0, **buf;
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int bo1;
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int clip;
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#ifndef NO_EQUALIZER
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if(fr->have_eq_settings) do_equalizer(bandPtr,channel,fr->equalizer);
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#endif
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if(!channel)
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{
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fr->bo--;
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fr->bo &= 0xf;
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buf = fr->real_buffs[0];
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}
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else
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{
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samples++;
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buf = fr->real_buffs[1];
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}
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if(fr->bo & 0x1)
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{
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b0 = buf[0];
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bo1 = fr->bo;
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dct64_real_sse(buf[1]+((fr->bo+1)&0xf),buf[0]+fr->bo,bandPtr);
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}
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else
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{
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b0 = buf[1];
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bo1 = fr->bo+1;
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dct64_real_sse(buf[0]+fr->bo,buf[1]+fr->bo+1,bandPtr);
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}
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clip = synth_1to1_s32_sse_asm(fr->decwin, b0, samples, bo1);
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if(final) fr->buffer.fill += 256;
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return clip;
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}
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int synth_1to1_s32_stereo_sse(real *bandPtr_l, real *bandPtr_r, mpg123_handle *fr)
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{
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int32_t *samples = (int32_t *) (fr->buffer.data+fr->buffer.fill);
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real *b0l, *b0r, **bufl, **bufr;
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int bo1;
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int clip;
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#ifndef NO_EQUALIZER
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if(fr->have_eq_settings)
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{
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do_equalizer(bandPtr_l,0,fr->equalizer);
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do_equalizer(bandPtr_r,1,fr->equalizer);
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}
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#endif
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fr->bo--;
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fr->bo &= 0xf;
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bufl = fr->real_buffs[0];
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bufr = fr->real_buffs[1];
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if(fr->bo & 0x1)
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{
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b0l = bufl[0];
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b0r = bufr[0];
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bo1 = fr->bo;
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dct64_real_sse(bufl[1]+((fr->bo+1)&0xf),bufl[0]+fr->bo,bandPtr_l);
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dct64_real_sse(bufr[1]+((fr->bo+1)&0xf),bufr[0]+fr->bo,bandPtr_r);
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}
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else
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{
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b0l = bufl[1];
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b0r = bufr[1];
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bo1 = fr->bo+1;
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dct64_real_sse(bufl[0]+fr->bo,bufl[1]+fr->bo+1,bandPtr_l);
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dct64_real_sse(bufr[0]+fr->bo,bufr[1]+fr->bo+1,bandPtr_r);
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}
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clip = synth_1to1_s32_s_sse_asm(fr->decwin, b0l, b0r, samples, bo1);
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fr->buffer.fill += 256;
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return clip;
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}
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#endif
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#ifdef OPT_NEON
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/* Assembler routines. */
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int synth_1to1_s32_neon_asm(real *window, real *b0, int32_t *samples, int bo1);
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int synth_1to1_s32_s_neon_asm(real *window, real *b0l, real *b0r, int32_t *samples, int bo1);
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void dct64_real_neon(real *out0, real *out1, real *samples);
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/* Hull for C mpg123 API */
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int synth_1to1_s32_neon(real *bandPtr,int channel, mpg123_handle *fr, int final)
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{
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int32_t *samples = (int32_t *) (fr->buffer.data+fr->buffer.fill);
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real *b0, **buf;
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int bo1;
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int clip;
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#ifndef NO_EQUALIZER
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if(fr->have_eq_settings) do_equalizer(bandPtr,channel,fr->equalizer);
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#endif
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if(!channel)
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{
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fr->bo--;
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fr->bo &= 0xf;
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buf = fr->real_buffs[0];
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}
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else
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{
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samples++;
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buf = fr->real_buffs[1];
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}
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if(fr->bo & 0x1)
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{
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b0 = buf[0];
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bo1 = fr->bo;
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dct64_real_neon(buf[1]+((fr->bo+1)&0xf),buf[0]+fr->bo,bandPtr);
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}
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else
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{
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b0 = buf[1];
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bo1 = fr->bo+1;
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dct64_real_neon(buf[0]+fr->bo,buf[1]+fr->bo+1,bandPtr);
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}
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clip = synth_1to1_s32_neon_asm(fr->decwin, b0, samples, bo1);
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if(final) fr->buffer.fill += 256;
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return clip;
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}
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int synth_1to1_s32_stereo_neon(real *bandPtr_l, real *bandPtr_r, mpg123_handle *fr)
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{
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int32_t *samples = (int32_t *) (fr->buffer.data+fr->buffer.fill);
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real *b0l, *b0r, **bufl, **bufr;
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int bo1;
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int clip;
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#ifndef NO_EQUALIZER
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if(fr->have_eq_settings)
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{
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do_equalizer(bandPtr_l,0,fr->equalizer);
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do_equalizer(bandPtr_r,1,fr->equalizer);
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}
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#endif
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fr->bo--;
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fr->bo &= 0xf;
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bufl = fr->real_buffs[0];
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bufr = fr->real_buffs[1];
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if(fr->bo & 0x1)
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{
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b0l = bufl[0];
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b0r = bufr[0];
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bo1 = fr->bo;
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dct64_real_neon(bufl[1]+((fr->bo+1)&0xf),bufl[0]+fr->bo,bandPtr_l);
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dct64_real_neon(bufr[1]+((fr->bo+1)&0xf),bufr[0]+fr->bo,bandPtr_r);
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}
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else
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{
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b0l = bufl[1];
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b0r = bufr[1];
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bo1 = fr->bo+1;
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dct64_real_neon(bufl[0]+fr->bo,bufl[1]+fr->bo+1,bandPtr_l);
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dct64_real_neon(bufr[0]+fr->bo,bufr[1]+fr->bo+1,bandPtr_r);
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}
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clip = synth_1to1_s32_s_neon_asm(fr->decwin, b0l, b0r, samples, bo1);
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fr->buffer.fill += 256;
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return clip;
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}
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#endif
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#ifdef OPT_NEON64
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/* Assembler routines. */
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int synth_1to1_s32_neon64_asm(real *window, real *b0, int32_t *samples, int bo1);
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int synth_1to1_s32_s_neon64_asm(real *window, real *b0l, real *b0r, int32_t *samples, int bo1);
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void dct64_real_neon64(real *out0, real *out1, real *samples);
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/* Hull for C mpg123 API */
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int synth_1to1_s32_neon64(real *bandPtr,int channel, mpg123_handle *fr, int final)
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{
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int32_t *samples = (int32_t *) (fr->buffer.data+fr->buffer.fill);
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real *b0, **buf;
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int bo1;
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int clip;
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#ifndef NO_EQUALIZER
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if(fr->have_eq_settings) do_equalizer(bandPtr,channel,fr->equalizer);
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#endif
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if(!channel)
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{
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fr->bo--;
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fr->bo &= 0xf;
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buf = fr->real_buffs[0];
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}
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else
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{
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samples++;
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buf = fr->real_buffs[1];
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}
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if(fr->bo & 0x1)
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{
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b0 = buf[0];
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bo1 = fr->bo;
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dct64_real_neon64(buf[1]+((fr->bo+1)&0xf),buf[0]+fr->bo,bandPtr);
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}
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else
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{
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b0 = buf[1];
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bo1 = fr->bo+1;
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dct64_real_neon64(buf[0]+fr->bo,buf[1]+fr->bo+1,bandPtr);
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}
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clip = synth_1to1_s32_neon64_asm(fr->decwin, b0, samples, bo1);
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if(final) fr->buffer.fill += 256;
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return clip;
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}
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int synth_1to1_s32st_neon64(real *bandPtr_l, real *bandPtr_r, mpg123_handle *fr)
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{
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int32_t *samples = (int32_t *) (fr->buffer.data+fr->buffer.fill);
|
|
|
|
real *b0l, *b0r, **bufl, **bufr;
|
|
int bo1;
|
|
int clip;
|
|
#ifndef NO_EQUALIZER
|
|
if(fr->have_eq_settings)
|
|
{
|
|
do_equalizer(bandPtr_l,0,fr->equalizer);
|
|
do_equalizer(bandPtr_r,1,fr->equalizer);
|
|
}
|
|
#endif
|
|
fr->bo--;
|
|
fr->bo &= 0xf;
|
|
bufl = fr->real_buffs[0];
|
|
bufr = fr->real_buffs[1];
|
|
|
|
if(fr->bo & 0x1)
|
|
{
|
|
b0l = bufl[0];
|
|
b0r = bufr[0];
|
|
bo1 = fr->bo;
|
|
dct64_real_neon64(bufl[1]+((fr->bo+1)&0xf),bufl[0]+fr->bo,bandPtr_l);
|
|
dct64_real_neon64(bufr[1]+((fr->bo+1)&0xf),bufr[0]+fr->bo,bandPtr_r);
|
|
}
|
|
else
|
|
{
|
|
b0l = bufl[1];
|
|
b0r = bufr[1];
|
|
bo1 = fr->bo+1;
|
|
dct64_real_neon64(bufl[0]+fr->bo,bufl[1]+fr->bo+1,bandPtr_l);
|
|
dct64_real_neon64(bufr[0]+fr->bo,bufr[1]+fr->bo+1,bandPtr_r);
|
|
}
|
|
|
|
clip = synth_1to1_s32_s_neon64_asm(fr->decwin, b0l, b0r, samples, bo1);
|
|
|
|
fr->buffer.fill += 256;
|
|
|
|
return clip;
|
|
}
|
|
#endif
|
|
|
|
#undef BLOCK
|
|
|
|
#ifndef NO_DOWNSAMPLE
|
|
|
|
/*
|
|
Part 4b: 2to1 synth. Only generic and i386.
|
|
*/
|
|
#define BLOCK 0x20 /* One decoding block is 32 samples. */
|
|
|
|
#define SYNTH_NAME synth_2to1_s32
|
|
#include "synth.h"
|
|
#undef SYNTH_NAME
|
|
|
|
/* Mono-related synths; they wrap over _some_ synth_2to1_s32 (could be generic, could be i386). */
|
|
#define SYNTH_NAME fr->synths.plain[r_2to1][f_32]
|
|
#define MONO_NAME synth_2to1_s32_mono
|
|
#define MONO2STEREO_NAME synth_2to1_s32_m2s
|
|
#include "synth_mono.h"
|
|
#undef SYNTH_NAME
|
|
#undef MONO_NAME
|
|
#undef MONO2STEREO_NAME
|
|
|
|
#ifdef OPT_X86
|
|
#define NO_AUTOINCREMENT
|
|
#define SYNTH_NAME synth_2to1_s32_i386
|
|
#include "synth.h"
|
|
#undef SYNTH_NAME
|
|
/* i386 uses the normal mono functions. */
|
|
#undef NO_AUTOINCREMENT
|
|
#endif
|
|
|
|
#undef BLOCK
|
|
|
|
/*
|
|
Part 4c: 4to1 synth. Only generic and i386.
|
|
*/
|
|
#define BLOCK 0x10 /* One decoding block is 16 samples. */
|
|
|
|
#define SYNTH_NAME synth_4to1_s32
|
|
#include "synth.h"
|
|
#undef SYNTH_NAME
|
|
|
|
/* Mono-related synths; they wrap over _some_ synth_4to1_s32 (could be generic, could be i386). */
|
|
#define SYNTH_NAME fr->synths.plain[r_4to1][f_32]
|
|
#define MONO_NAME synth_4to1_s32_mono
|
|
#define MONO2STEREO_NAME synth_4to1_s32_m2s
|
|
#include "synth_mono.h"
|
|
#undef SYNTH_NAME
|
|
#undef MONO_NAME
|
|
#undef MONO2STEREO_NAME
|
|
|
|
#ifdef OPT_X86
|
|
#define NO_AUTOINCREMENT
|
|
#define SYNTH_NAME synth_4to1_s32_i386
|
|
#include "synth.h"
|
|
#undef SYNTH_NAME
|
|
/* i386 uses the normal mono functions. */
|
|
#undef NO_AUTOINCREMENT
|
|
#endif
|
|
|
|
#undef BLOCK
|
|
|
|
#endif /* NO_DOWNSAMPLE */
|
|
|
|
#ifndef NO_NTOM
|
|
/*
|
|
Part 4d: ntom synth.
|
|
Same procedure as above... Just no extra play anymore, straight synth that may use an optimized dct64.
|
|
*/
|
|
|
|
/* These are all in one header, there's no flexibility to gain. */
|
|
#define SYNTH_NAME synth_ntom_s32
|
|
#define MONO_NAME synth_ntom_s32_mono
|
|
#define MONO2STEREO_NAME synth_ntom_s32_m2s
|
|
#include "synth_ntom.h"
|
|
#undef SYNTH_NAME
|
|
#undef MONO_NAME
|
|
#undef MONO2STEREO_NAME
|
|
|
|
#endif
|
|
|
|
#undef SAMPLE_T
|
|
#undef WRITE_SAMPLE
|
|
|
|
#endif /* non-fixed type */
|