Mercurial > SDL_sound_CoreAudio
view decoders/libmpg123/decode_2to1.c @ 591:8faf61a640f0 tip
Resynced fixes for unit conversion bugs in the Ogg Tremor decoder from SoundDecoder/ALmixer.
Ogg Vorbis uses seconds and we multiply by 1000 to convert to milliseconds. But Ogg Tremor already uses milliseconds but I was still multiplying by 1000.
author | Eric Wing <ewing . public |-at-| gmail . com> |
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date | Thu, 25 Oct 2012 16:34:18 -0700 |
parents | 7e08477b0fc1 |
children |
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/* decode_2to1.c: ...with 2to1 downsampling copyright 1995-2006 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 */ #include "mpg123lib_intern.h" int synth_2to1_8bit(real *bandPtr, int channel, mpg123_handle *fr, int final) { sample_t samples_tmp[32]; sample_t *tmp1 = samples_tmp + channel; int i,ret; unsigned char *samples = fr->buffer.data; int pnt = fr->buffer.fill; fr->buffer.data = (unsigned char*) samples_tmp; fr->buffer.fill = 0; ret = synth_2to1(bandPtr,channel, fr, 0); fr->buffer.data = samples; samples += channel + pnt; for(i=0;i<16;i++) { #ifdef FLOATOUT *samples = 0; #else *samples = fr->conv16to8[*tmp1>>AUSHIFT]; #endif samples += 2; tmp1 += 2; } fr->buffer.fill = pnt + (final ? 32 : 0); return ret; } int synth_2to1_8bit_mono(real *bandPtr, mpg123_handle *fr) { sample_t samples_tmp[32]; sample_t *tmp1 = samples_tmp; int i,ret; unsigned char *samples = fr->buffer.data; int pnt = fr->buffer.fill; fr->buffer.data = (unsigned char*) samples_tmp; fr->buffer.fill = 0; ret = synth_2to1(bandPtr, 0, fr, 0); fr->buffer.data = samples; samples += pnt; for(i=0;i<16;i++) { #ifdef FLOATOUT *samples++ = 0; #else *samples++ = fr->conv16to8[*tmp1>>AUSHIFT]; #endif tmp1 += 2; } fr->buffer.fill = pnt + 16; return ret; } int synth_2to1_8bit_mono2stereo(real *bandPtr, mpg123_handle *fr) { sample_t samples_tmp[32]; sample_t *tmp1 = samples_tmp; int i,ret; unsigned char *samples = fr->buffer.data; int pnt = fr->buffer.fill; fr->buffer.data = (unsigned char*) samples_tmp; fr->buffer.fill = 0; ret = synth_2to1(bandPtr,0, fr, 0); fr->buffer.data = samples; samples += pnt; for(i=0;i<16;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 + 32; return ret; } int synth_2to1_mono(real *bandPtr, mpg123_handle *fr) { sample_t samples_tmp[32]; sample_t *tmp1 = samples_tmp; int i,ret; unsigned char *samples = fr->buffer.data; int pnt = fr->buffer.fill; fr->buffer.data = (unsigned char*) samples_tmp; fr->buffer.fill = 0; ret = synth_2to1(bandPtr, 0, fr, 0); fr->buffer.data = samples; samples += pnt; for(i=0;i<16;i++) { *( (sample_t *) samples) = *tmp1; samples += sizeof(sample_t); tmp1 += 2; } fr->buffer.fill = pnt + 16*sizeof(sample_t); return ret; } int synth_2to1_mono2stereo(real *bandPtr, mpg123_handle *fr) { int i,ret; unsigned char *samples = fr->buffer.data; ret = synth_2to1(bandPtr,0, fr, 1); samples += fr->buffer.fill - 32*sizeof(sample_t); for(i=0;i<16;i++) { ((sample_t *)samples)[1] = ((sample_t *)samples)[0]; samples+=2*sizeof(sample_t); } return ret; } int synth_2to1(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; 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]; } else { samples++; buf = fr->real_buffs[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=8;j;j--,b0+=0x10,window+=0x30) { real sum; 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++); WRITE_SAMPLE(samples,sum,clip); samples += step; #if 0 WRITE_SAMPLE(samples,sum,clip); samples += step; #endif } { 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]); WRITE_SAMPLE(samples,sum,clip); samples += step; #if 0 WRITE_SAMPLE(samples,sum,clip); samples += step; #endif b0-=0x20,window-=0x40; } window += bo1<<1; for (j=7;j;j--,b0-=0x30,window-=0x30) { real sum; 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++); WRITE_SAMPLE(samples,sum,clip); samples += step; #if 0 WRITE_SAMPLE(samples,sum,clip); samples += step; #endif } } if(final) fr->buffer.fill += 32*sizeof(sample_t); return clip; }