diff Plugins/Input/aac/libfaad2/output.c @ 61:fa848bd484d8 trunk

[svn] Move plugins to Plugins/

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/Plugins/Input/aac/libfaad2/output.c	Fri Oct 28 22:58:11 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 [e-mail].
+**
+** $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