Mercurial > audlegacy
view Plugins/Input/aac/libfaad2/output.c @ 184:740c072f8e23 trunk
[svn] List sndfile extensions in the plugin configuration table.
author | chainsaw |
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date | Sun, 13 Nov 2005 14:27:07 -0800 |
parents | fa848bd484d8 |
children | 0a2ad94e8607 |
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/* ** 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