Mercurial > SDL_sound_CoreAudio
view decoders/libmpg123/decode.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.c: decoding samples... 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" /* 8bit functions silenced for FLOATOUT */ int synth_1to1_8bit(real *bandPtr,int channel, mpg123_handle *fr, int final) { sample_t samples_tmp[64]; sample_t *tmp1 = samples_tmp + channel; int i,ret; /* save buffer stuff, trick samples_tmp into there, decode, restore */ unsigned char *samples = fr->buffer.data; int pnt = fr->buffer.fill; fr->buffer.data = (unsigned char*) samples_tmp; fr->buffer.fill = 0; ret = synth_1to1(bandPtr, channel, fr, 0); fr->buffer.data = samples; /* restore original value */ samples += channel + pnt; for(i=0;i<32;i++) { #ifdef FLOATOUT *samples = 0; #else *samples = fr->conv16to8[*tmp1>>AUSHIFT]; #endif samples += 2; tmp1 += 2; } fr->buffer.fill = pnt + (final ? 64 : 0 ); return ret; } int synth_1to1_8bit_mono(real *bandPtr, mpg123_handle *fr) { sample_t samples_tmp[64]; sample_t *tmp1 = samples_tmp; int i,ret; /* save buffer stuff, trick samples_tmp into there, decode, restore */ unsigned char *samples = fr->buffer.data; int pnt = fr->buffer.fill; fr->buffer.data = (unsigned char*) samples_tmp; fr->buffer.fill = 0; ret = synth_1to1(bandPtr,0, fr, 0); fr->buffer.data = samples; /* restore original value */ samples += pnt; for(i=0;i<32;i++) { #ifdef FLOATOUT *samples++ = 0; #else *samples++ = fr->conv16to8[*tmp1>>AUSHIFT]; #endif tmp1 += 2; } fr->buffer.fill = pnt + 32; return ret; } int synth_1to1_8bit_mono2stereo(real *bandPtr, mpg123_handle *fr) { sample_t samples_tmp[64]; sample_t *tmp1 = samples_tmp; int i,ret; /* save buffer stuff, trick samples_tmp into there, decode, restore */ unsigned char *samples = fr->buffer.data; int pnt = fr->buffer.fill; fr->buffer.data = (unsigned char*) samples_tmp; fr->buffer.fill = 0; ret = synth_1to1(bandPtr, 0, fr, 0); fr->buffer.data = samples; /* restore original value */ samples += pnt; for(i=0;i<32;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 + 64; return ret; } int synth_1to1_mono(real *bandPtr, mpg123_handle *fr) { sample_t samples_tmp[64]; sample_t *tmp1 = samples_tmp; int i,ret; /* save buffer stuff, trick samples_tmp into there, decode, restore */ unsigned char *samples = fr->buffer.data; int pnt = fr->buffer.fill; fr->buffer.data = (unsigned char*) samples_tmp; fr->buffer.fill = 0; ret = synth_1to1(bandPtr, 0, fr, 0); /* decode into samples_tmp */ fr->buffer.data = samples; /* restore original value */ /* now append samples from samples_tmp */ samples += pnt; /* just the next mem in frame buffer */ for(i=0;i<32;i++){ *( (sample_t *)samples) = *tmp1; samples += sizeof(sample_t); tmp1 += 2; } fr->buffer.fill = pnt + 32*sizeof(sample_t); return ret; } int synth_1to1_mono2stereo(real *bandPtr, mpg123_handle *fr) { int i,ret; unsigned char *samples = fr->buffer.data; ret = synth_1to1(bandPtr,0,fr,1); samples += fr->buffer.fill - 64*sizeof(sample_t); for(i=0;i<32;i++) { ((sample_t *)samples)[1] = ((sample_t *)samples)[0]; samples+=2*sizeof(sample_t); } return ret; } int synth_1to1(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]; dct64(buf[1]+((fr->bo[0]+1)&0xf),buf[0]+fr->bo[0],bandPtr); } else { b0 = buf[1]; bo1 = fr->bo[0]+1; dct64(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,samples+=step) { 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); } { 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); b0-=0x10,window-=0x20,samples+=step; } window += bo1<<1; for (j=15;j;j--,b0-=0x20,window-=0x10,samples+=step) { 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); } } if(final) fr->buffer.fill += 64*sizeof(sample_t); return clip; }