Mercurial > SDL_sound_CoreAudio
view decoders/libmpg123/decode_ntom.c @ 590:1c8414cd5839
Introduced new Ogg Tremor decoder, based on the existing Ogg Vorbis decoder.
author | Eric Wing <ewing@anscamobile.com> |
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date | Wed, 27 Apr 2011 19:37:16 -0700 |
parents | 7e08477b0fc1 |
children |
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/* decode_ntom.c: N->M down/up sampling. Not optimized for speed. copyright 1995-2008 by the mpg123 project - free software under the terms of the LGPL 2.1 see COPYING and AUTHORS files in distribution or http://mpg123.org initially written by Michael Hipp */ #define SAFE_NTOM /* Do not depend on off_t*off_t with big values still being in the range... */ #include "mpg123lib_intern.h" #include "debug.h" int synth_ntom_set_step(mpg123_handle *fr) { long m,n; m = frame_freq(fr); n = fr->af.rate; if(VERBOSE2) fprintf(stderr,"Init rate converter: %ld->%ld\n",m,n); if(n > NTOM_MAX_FREQ || m > NTOM_MAX_FREQ || m <= 0 || n <= 0) { if(NOQUIET) error("NtoM converter: illegal rates"); fr->err = MPG123_BAD_RATE; return -1; } n *= NTOM_MUL; fr->ntom_step = (unsigned long) n / m; if(fr->ntom_step > (unsigned long)NTOM_MAX*NTOM_MUL) { if(NOQUIET) error3("max. 1:%i conversion allowed (%lu vs %lu)!", NTOM_MAX, fr->ntom_step, (unsigned long)8*NTOM_MUL); fr->err = MPG123_BAD_RATE; return -1; } fr->ntom_val[0] = fr->ntom_val[1] = ntom_val(fr, fr->num); return 0; } /* The SAFE_NTOM does iterative loops instead of straight multiplication. The safety is not just about the algorithm closely mimicking the decoder instead of applying some formula, it is more about avoiding multiplication of possibly big sample offsets (a 32bit off_t could overflow too easily). */ unsigned long ntom_val(mpg123_handle *fr, off_t frame) { off_t ntm; #ifdef SAFE_NTOM /* Carry out the loop, without the threatening integer overflow. */ off_t f; ntm = NTOM_MUL>>1; /* for frame 0 */ for(f=0; f<frame; ++f) /* for frame > 0 */ { ntm += spf(fr)*fr->ntom_step; ntm -= (ntm/NTOM_MUL)*NTOM_MUL; } #else /* Just make one computation with overall sample offset. */ ntm = (NTOM_MUL>>1) + spf(fr)*frame*fr->ntom_step; ntm -= (ntm/NTOM_MUL)*NTOM_MUL; #endif return (unsigned long) ntm; } /* Set the ntom value for next expected frame to be decoded. This is for keeping output consistent across seeks. */ void ntom_set_ntom(mpg123_handle *fr, off_t num) { fr->ntom_val[1] = fr->ntom_val[0] = ntom_val(fr, num); } /* Convert frame offset to unadjusted output sample offset. */ off_t ntom_frmouts(mpg123_handle *fr, off_t frame) { #ifdef SAFE_NTOM off_t f; #endif off_t soff = 0; off_t ntm = ntom_val(fr,0); #ifdef SAFE_NTOM if(frame <= 0) return 0; for(f=0; f<frame; ++f) { ntm += spf(fr)*fr->ntom_step; soff += ntm/NTOM_MUL; ntm -= (ntm/NTOM_MUL)*NTOM_MUL; } #else soff = (ntm + frame*(off_t)spf(fr)*(off_t)fr->ntom_step)/(off_t)NTOM_MUL; #endif return soff; } /* Convert input samples to unadjusted output samples. */ off_t ntom_ins2outs(mpg123_handle *fr, off_t ins) { off_t soff = 0; off_t ntm = ntom_val(fr,0); #ifdef SAFE_NTOM { off_t block = spf(fr); if(ins <= 0) return 0; do { off_t nowblock = ins > block ? block : ins; ntm += nowblock*fr->ntom_step; soff += ntm/NTOM_MUL; ntm -= (ntm/NTOM_MUL)*NTOM_MUL; ins -= nowblock; } while(ins > 0); } #else /* Beware of overflows: when off_t is 32bits, the multiplication blows too easily. Of course, it blows for 64bits, too, in theory, but that's for _really_ large files. */ soff = ((off_t)ntm + (off_t)ins*(off_t)fr->ntom_step)/(off_t)NTOM_MUL; #endif return soff; } /* Determine frame offset from unadjusted output sample offset. */ off_t ntom_frameoff(mpg123_handle *fr, off_t soff) { off_t ioff = 0; /* frames or samples */ off_t ntm = ntom_val(fr,0); #ifdef SAFE_NTOM if(soff <= 0) return 0; for(ioff=0; 1; ++ioff) { ntm += spf(fr)*fr->ntom_step; if(ntm/NTOM_MUL > soff) break; soff -= ntm/NTOM_MUL; ntm -= (ntm/NTOM_MUL)*NTOM_MUL; } return ioff; #else ioff = (soff*(off_t)NTOM_MUL-ntm)/(off_t)fr->ntom_step; return ioff/(off_t)spf(fr); #endif } /* Now to the actual decoding/synth functions... */ int synth_ntom_8bit(real *bandPtr,int channel, mpg123_handle *fr, int final) { sample_t samples_tmp[8*64]; sample_t *tmp1 = samples_tmp + channel; int i,ret; int pnt = fr->buffer.fill; unsigned char *samples = fr->buffer.data; fr->buffer.data = (unsigned char*) samples_tmp; fr->buffer.fill = 0; ret = synth_ntom(bandPtr, channel, fr, 1); fr->buffer.data = samples; samples += channel + pnt; for(i=0;i<(fr->buffer.fill>>2);i++) { #ifdef FLOATOUT *samples = 0; #else *samples = fr->conv16to8[*tmp1>>AUSHIFT]; #endif samples += 2; tmp1 += 2; } fr->buffer.fill = pnt + (final ? fr->buffer.fill>>1 : 0); return ret; } int synth_ntom_8bit_mono(real *bandPtr, mpg123_handle *fr) { sample_t samples_tmp[8*64]; sample_t *tmp1 = samples_tmp; int i,ret; int pnt = fr->buffer.fill; unsigned char *samples = fr->buffer.data; fr->buffer.data = (unsigned char*) samples_tmp; fr->buffer.fill = 0; ret = synth_ntom(bandPtr, 0, fr, 1); fr->buffer.data = samples; samples += pnt; for(i=0;i<(fr->buffer.fill>>2);i++) { #ifdef FLOATOUT *samples++ = 0; #else *samples++ = fr->conv16to8[*tmp1>>AUSHIFT]; #endif tmp1 += 2; } fr->buffer.fill = pnt + (fr->buffer.fill>>2); return ret; } int synth_ntom_8bit_mono2stereo(real *bandPtr, mpg123_handle *fr) { sample_t samples_tmp[8*64]; sample_t *tmp1 = samples_tmp; int i,ret; int pnt = fr->buffer.fill; unsigned char *samples = fr->buffer.data; fr->buffer.data = (unsigned char*) samples_tmp; fr->buffer.fill = 0; ret = synth_ntom(bandPtr, 0, fr, 1); fr->buffer.data = samples; samples += pnt; for(i=0;i<(fr->buffer.fill>>2);i++) { #ifdef FLOATOUT *samples++ = 0; *samples++ = 0; #else *samples++ = fr->conv16to8[*tmp1>>AUSHIFT]; *samples++ = fr->conv16to8[*tmp1>>AUSHIFT]; #endif tmp1 += 2; } fr->buffer.fill = pnt + (fr->buffer.fill>>1); return ret; } int synth_ntom_mono(real *bandPtr, mpg123_handle *fr) { sample_t samples_tmp[8*64]; sample_t *tmp1 = samples_tmp; int i,ret; int pnt = fr->buffer.fill; unsigned char *samples = fr->buffer.data; fr->buffer.data = (unsigned char*) samples_tmp; fr->buffer.fill = 0; ret = synth_ntom(bandPtr, 0, fr, 1); fr->buffer.data = samples; samples += pnt; for(i=0;i<(fr->buffer.fill>>2);i++) { *( (sample_t *)samples) = *tmp1; samples += sizeof(sample_t); tmp1 += 2; } fr->buffer.fill = pnt + (fr->buffer.fill>>2)*sizeof(sample_t); return ret; } int synth_ntom_mono2stereo(real *bandPtr, mpg123_handle *fr) { int i,ret; int pnt1 = fr->buffer.fill; unsigned char *samples = fr->buffer.data + pnt1; ret = synth_ntom(bandPtr, 0, fr, 1); for(i=0;i<((fr->buffer.fill-pnt1)>>2);i++) { ((sample_t *)samples)[1] = ((sample_t *)samples)[0]; samples+=2*sizeof(sample_t); } return ret; } int synth_ntom(real *bandPtr,int channel, mpg123_handle *fr, int final) { static const int step = 2; sample_t *samples = (sample_t *) (fr->buffer.data + fr->buffer.fill); real *b0, **buf; /* (*buf)[0x110]; */ int clip = 0; int bo1; int ntom; if(fr->have_eq_settings) do_equalizer(bandPtr,channel,fr->equalizer); if(!channel) { fr->bo[0]--; fr->bo[0] &= 0xf; buf = fr->real_buffs[0]; ntom = fr->ntom_val[1] = fr->ntom_val[0]; } else { samples++; buf = fr->real_buffs[1]; ntom = fr->ntom_val[1]; } if(fr->bo[0] & 0x1) { b0 = buf[0]; bo1 = fr->bo[0]; opt_dct64(fr)(buf[1]+((fr->bo[0]+1)&0xf),buf[0]+fr->bo[0],bandPtr); } else { b0 = buf[1]; bo1 = fr->bo[0]+1; opt_dct64(fr)(buf[0]+fr->bo[0],buf[1]+fr->bo[0]+1,bandPtr); } { register int j; real *window = opt_decwin(fr) + 16 - bo1; for (j=16;j;j--,window+=0x10) { real sum; ntom += fr->ntom_step; if(ntom < NTOM_MUL) { window += 16; b0 += 16; continue; } sum = REAL_MUL(*window++, *b0++); sum -= REAL_MUL(*window++, *b0++); sum += REAL_MUL(*window++, *b0++); sum -= REAL_MUL(*window++, *b0++); sum += REAL_MUL(*window++, *b0++); sum -= REAL_MUL(*window++, *b0++); sum += REAL_MUL(*window++, *b0++); sum -= REAL_MUL(*window++, *b0++); sum += REAL_MUL(*window++, *b0++); sum -= REAL_MUL(*window++, *b0++); sum += REAL_MUL(*window++, *b0++); sum -= REAL_MUL(*window++, *b0++); sum += REAL_MUL(*window++, *b0++); sum -= REAL_MUL(*window++, *b0++); sum += REAL_MUL(*window++, *b0++); sum -= REAL_MUL(*window++, *b0++); while(ntom >= NTOM_MUL) { WRITE_SAMPLE(samples,sum,clip); samples += step; ntom -= NTOM_MUL; } } ntom += fr->ntom_step; if(ntom >= NTOM_MUL) { real sum; sum = REAL_MUL(window[0x0], b0[0x0]); sum += REAL_MUL(window[0x2], b0[0x2]); sum += REAL_MUL(window[0x4], b0[0x4]); sum += REAL_MUL(window[0x6], b0[0x6]); sum += REAL_MUL(window[0x8], b0[0x8]); sum += REAL_MUL(window[0xA], b0[0xA]); sum += REAL_MUL(window[0xC], b0[0xC]); sum += REAL_MUL(window[0xE], b0[0xE]); while(ntom >= NTOM_MUL) { WRITE_SAMPLE(samples,sum,clip); samples += step; ntom -= NTOM_MUL; } } b0-=0x10,window-=0x20; window += bo1<<1; for (j=15;j;j--,b0-=0x20,window-=0x10) { real sum; ntom += fr->ntom_step; if(ntom < NTOM_MUL) { window -= 16; b0 += 16; continue; } sum = REAL_MUL(-*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); sum -= REAL_MUL(*(--window), *b0++); while(ntom >= NTOM_MUL) { WRITE_SAMPLE(samples,sum,clip); samples += step; ntom -= NTOM_MUL; } } } fr->ntom_val[channel] = ntom; if(final) fr->buffer.fill = ((unsigned char *) samples - fr->buffer.data - (channel ? 2 : 0)); return clip; }