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Diffstat (limited to 'src/libfaad/output.c')
| -rw-r--r-- | src/libfaad/output.c | 107 |
1 files changed, 107 insertions, 0 deletions
diff --git a/src/libfaad/output.c b/src/libfaad/output.c new file mode 100644 index 000000000..f6e8c1382 --- /dev/null +++ b/src/libfaad/output.c @@ -0,0 +1,107 @@ +/* +** FAAD - Freeware Advanced Audio Decoder +** Copyright (C) 2002 M. Bakker +** +** This program is free software; you can redistribute it and/or modify +** it under the terms of the GNU General Public License as published by +** the Free Software Foundation; either version 2 of the License, or +** (at your option) any later version. +** +** This program 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 General Public License for more details. +** +** You should have received a copy of the GNU General Public License +** along with this program; if not, write to the Free Software +** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. +** +** $Id: output.c,v 1.1 2002/07/14 23:43:01 miguelfreitas Exp $ +**/ + +#include "common.h" + +#include "output.h" +#include "decoder.h" + + +#define ftol(A,B) {tmp = *(int32_t*) & A - 0x4B7F8000; \ + B = (int16_t)((tmp==(int16_t)tmp) ? tmp : (tmp>>31)^0x7FFF);} + +#define ROUND(x) ((x >= 0) ? (int32_t)floor((x) + 0.5) : (int32_t)ceil((x) + 0.5)) + +#define ROUND32(x) ROUND(x) + +#define FLOAT_SCALE (1.0f/(1<<15)) + + +void* output_to_PCM(real_t **input, void *sample_buffer, uint8_t channels, + uint16_t frame_len, uint8_t format) +{ + uint8_t ch; + uint16_t i; + + uint8_t *p = (uint8_t*)sample_buffer; + int16_t *short_sample_buffer = (int16_t*)sample_buffer; + int32_t *int_sample_buffer = (int32_t*)sample_buffer; + float32_t *float_sample_buffer = (float32_t*)sample_buffer; + + /* Copy output to a standard PCM buffer */ + switch (format) + { + case FAAD_FMT_16BIT: + for (ch = 0; ch < channels; ch++) + { + for(i = 0; i < frame_len; i++) + { + int32_t tmp; + real_t ftemp; + + ftemp = input[ch][i] + 0xff8000; + ftol(ftemp, short_sample_buffer[(i*channels)+ch]); + } + } + break; + case FAAD_FMT_24BIT: + for (ch = 0; ch < channels; ch++) + { + for(i = 0; i < frame_len; i++) + { + if (input[ch][i] > (1<<15)-1) + input[ch][i] = (1<<15)-1; + else if (input[ch][i] < -(1<<15)) + input[ch][i] = -(1<<15); + int_sample_buffer[(i*channels)+ch] = ROUND(input[ch][i]*(1<<8)); + } + } + break; + case FAAD_FMT_32BIT: + for (ch = 0; ch < channels; ch++) + { + for(i = 0; i < frame_len; i++) + { + if (input[ch][i] > (1<<15)-1) + input[ch][i] = (1<<15)-1; + else if (input[ch][i] < -(1<<15)) + input[ch][i] = -(1<<15); + int_sample_buffer[(i*channels)+ch] = ROUND32(input[ch][i]*(1<<16)); + } + } + break; + case FAAD_FMT_FLOAT: + for (ch = 0; ch < channels; ch++) + { + for(i = 0; i < frame_len; i++) + { + if (input[ch][i] > (1<<15)-1) + input[ch][i] = (1<<15)-1; + else if (input[ch][i] < -(1<<15)) + input[ch][i] = -(1<<15); + float_sample_buffer[(i*channels)+ch] = input[ch][i]*FLOAT_SCALE; + } + } + break; + } + + return sample_buffer; +}
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