Mercurial > audlegacy
diff Input/aac/libfaad2/output.c @ 2:6efb9e514224 trunk
[svn] Import AAC stuff.
author | nenolod |
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date | Mon, 24 Oct 2005 10:44:27 -0700 |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/Input/aac/libfaad2/output.c Mon Oct 24 10:44:27 2005 -0700 @@ -0,0 +1,261 @@ +/* +** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding +** Copyright (C) 2003 M. Bakker, Ahead Software AG, http://www.nero.com +** +** 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. +** +** Any non-GPL usage of this software or parts of this software is strictly +** forbidden. +** +** Commercial non-GPL licensing of this software is possible. +** For more info contact Ahead Software through Mpeg4AAClicense@nero.com. +** +** $Id: output.c,v 1.29 2003/11/12 20:47:58 menno Exp $ +**/ + +#include "common.h" +#include "structs.h" + +#include "output.h" +#include "decoder.h" + +#ifndef FIXED_POINT + + +#define FLOAT_SCALE (1.0f/(1<<15)) + +#define DM_MUL ((real_t)1.0/((real_t)1.0+(real_t)sqrt(2.0))) + + +static INLINE real_t get_sample(real_t **input, uint8_t channel, uint16_t sample, + uint8_t downMatrix, uint8_t *internal_channel) +{ + if (!downMatrix) + return input[internal_channel[channel]][sample]; + + if (channel == 0) + { + return DM_MUL * (input[internal_channel[1]][sample] + + input[internal_channel[0]][sample]/(real_t)sqrt(2.) + + input[internal_channel[3]][sample]/(real_t)sqrt(2.)); + } else { + return DM_MUL * (input[internal_channel[2]][sample] + + input[internal_channel[0]][sample]/(real_t)sqrt(2.) + + input[internal_channel[4]][sample]/(real_t)sqrt(2.)); + } +} + +void* output_to_PCM(faacDecHandle hDecoder, + real_t **input, void *sample_buffer, uint8_t channels, + uint16_t frame_len, uint8_t format) +{ + uint8_t ch; + uint16_t i, j = 0; + uint8_t internal_channel; + + 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; + double *double_sample_buffer = (double*)sample_buffer; + + /* Copy output to a standard PCM buffer */ + for (ch = 0; ch < channels; ch++) + { + internal_channel = hDecoder->internal_channel[ch]; + + switch (format) + { + case FAAD_FMT_16BIT: + for(i = 0; i < frame_len; i++) + { + real_t inp = get_sample(input, ch, i, hDecoder->downMatrix, hDecoder->internal_channel); + if (inp >= 0.0f) + { +#ifndef HAS_LRINTF + inp += 0.5f; +#endif + if (inp >= 32768.0f) + { + inp = 32767.0f; + } + } else { +#ifndef HAS_LRINTF + inp += -0.5f; +#endif + if (inp <= -32769.0f) + { + inp = -32768.0f; + } + } + short_sample_buffer[(i*channels)+ch] = (int16_t)lrintf(inp); + } + break; + case FAAD_FMT_24BIT: + for(i = 0; i < frame_len; i++) + { + real_t inp = get_sample(input, ch, i, hDecoder->downMatrix, hDecoder->internal_channel); + inp *= 256.0f; + if (inp >= 0.0f) + { +#ifndef HAS_LRINTF + inp += 0.5f; +#endif + if (inp >= 8388608.0f) + { + inp = 8388607.0f; + } + } else { +#ifndef HAS_LRINTF + inp += -0.5f; +#endif + if (inp <= -8388609.0f) + { + inp = -8388608.0f; + } + } + int_sample_buffer[(i*channels)+ch] = lrintf(inp); + } + break; + case FAAD_FMT_32BIT: + for(i = 0; i < frame_len; i++) + { + real_t inp = get_sample(input, ch, i, hDecoder->downMatrix, hDecoder->internal_channel); + inp *= 65536.0f; + if (inp >= 0.0f) + { +#ifndef HAS_LRINTF + inp += 0.5f; +#endif + if (inp >= 2147483648.0f) + { + inp = 2147483647.0f; + } + } else { +#ifndef HAS_LRINTF + inp += -0.5f; +#endif + if (inp <= -2147483649.0f) + { + inp = -2147483648.0f; + } + } + int_sample_buffer[(i*channels)+ch] = lrintf(inp); + } + break; + case FAAD_FMT_FLOAT: + for(i = 0; i < frame_len; i++) + { + //real_t inp = input[internal_channel][i]; + real_t inp = get_sample(input, ch, i, hDecoder->downMatrix, hDecoder->internal_channel); + float_sample_buffer[(i*channels)+ch] = inp*FLOAT_SCALE; + } + break; + case FAAD_FMT_DOUBLE: + for(i = 0; i < frame_len; i++) + { + //real_t inp = input[internal_channel][i]; + real_t inp = get_sample(input, ch, i, hDecoder->downMatrix, hDecoder->internal_channel); + double_sample_buffer[(i*channels)+ch] = (double)inp*FLOAT_SCALE; + } + break; + } + } + + return sample_buffer; +} + +#else + +void* output_to_PCM(faacDecHandle hDecoder, + real_t **input, void *sample_buffer, uint8_t channels, + uint16_t frame_len, uint8_t format) +{ + uint8_t ch; + uint16_t i; + int16_t *short_sample_buffer = (int16_t*)sample_buffer; + int32_t *int_sample_buffer = (int32_t*)sample_buffer; + + /* Copy output to a standard PCM buffer */ + for (ch = 0; ch < channels; ch++) + { + switch (format) + { + case FAAD_FMT_16BIT: + for(i = 0; i < frame_len; i++) + { + int32_t tmp = input[ch][i]; + if (tmp >= 0) + { + tmp += (1 << (REAL_BITS-1)); + if (tmp >= REAL_CONST(32768)) + { + tmp = REAL_CONST(32767); + } + } else { + tmp += -(1 << (REAL_BITS-1)); + if (tmp <= REAL_CONST(-32769)) + { + tmp = REAL_CONST(-32768); + } + } + tmp >>= REAL_BITS; + short_sample_buffer[(i*channels)+ch] = (int16_t)tmp; + } + break; + case FAAD_FMT_24BIT: + for(i = 0; i < frame_len; i++) + { + int32_t tmp = input[ch][i]; + if (tmp >= 0) + { + tmp += (1 << (REAL_BITS-9)); + tmp >>= (REAL_BITS-8); + if (tmp >= 8388608) + { + tmp = 8388607; + } + } else { + tmp += -(1 << (REAL_BITS-9)); + tmp >>= (REAL_BITS-8); + if (tmp <= -8388609) + { + tmp = -8388608; + } + } + int_sample_buffer[(i*channels)+ch] = (int32_t)tmp; + } + break; + case FAAD_FMT_32BIT: + for(i = 0; i < frame_len; i++) + { + int32_t tmp = input[ch][i]; + if (tmp >= 0) + { + tmp += (1 << (16-REAL_BITS-1)); + tmp <<= (16-REAL_BITS); + } else { + tmp += -(1 << (16-REAL_BITS-1)); + tmp <<= (16-REAL_BITS); + } + int_sample_buffer[(i*channels)+ch] = (int32_t)tmp; + } + break; + } + } + + return sample_buffer; +} + +#endif