merge FAAD2 - the GPL AAC decoder library.

xine_decoder.c is working, but demux_qt must send some needed
initialization data. currently it's hardcoded to play my test stream, so
it's not usable yet.

CVS patchset: 2267
CVS date: 2002/07/14 23:43:01

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;
+}
\ No newline at end of file