Add Interplay MVE format demuxer/decoder and muxer/encoder. Demuxer doesn't support seeking yet, but seems to work fi...

Original commit message from CVS:
Patch by: Jens Granseuer  <jensgr at gmx net>
* configure.ac:
* gst/mve/Makefile.am:
* gst/mve/TODO:
* gst/mve/gstmve.c:
* gst/mve/gstmvedemux.c:
* gst/mve/gstmvedemux.h:
* gst/mve/gstmvemux.c:
* gst/mve/gstmvemux.h:
* gst/mve/mve.h:
* gst/mve/mveaudiodec.c:
* gst/mve/mveaudioenc.c:
* gst/mve/mvevideodec16.c:
* gst/mve/mvevideodec8.c:
* gst/mve/mvevideoenc16.c:
* gst/mve/mvevideoenc8.c:
Add Interplay MVE format demuxer/decoder and muxer/encoder. Demuxer
doesn't support seeking yet, but seems to work fine otherwise.
Closes #348973.

1 files changed, 120 insertions, 0 deletions

diff --git a/gst/mve/mveaudioenc.c b/gst/mve/mveaudioenc.c
new file mode 100644
index 00000000..1de73753
--- /dev/null
+++ b/gst/mve/mveaudioenc.c
@@ -0,0 +1,120 @@
+/* 
+ * Interplay MVE audio compressor
+ * Copyright (C) 2003, 2004 Alexander Belyakov <abel@krasu.ru>
+ * Copyright (C) 2006 Jens Granseuer <jensgr@gmx.net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <math.h>
+#include <gst/gst.h>
+
+static const gint32 dec_table[256] =
+    { 
0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 
16, 17, 18, 19,
+  20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 
32, 33, 34, 35, 36, 37,
+  38, 39, 40, 41, 42, 43, 47, 51, 56, 61, 
66, 72, 79, 86, 94, 102, 112,
+  122, 133, 145, 158, 173, 189, 206, 225, 245, 
267, 292, 318, 348, 379,
+  414, 452, 493, 538, 587, 640, 699, 763, 832, 908, 991, 
1081, 1180, 1288,
+  1405, 1534, 1673, 1826, 1993, 2175, 2373, 2590, 2826, 3084, 3365, 3672,
+  4008, 
4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059, 8794, 9597, 10472,
+  11428, 12471, 13609, 14851, 16206, 
17685, 19298, 21060, 22981, 25078,
+  27367, 29864, 32589, 35563, 38808, 42350, 46214, 50431, 55033, 60055,
+  65535, 
1, -65535, -60055, -55033, -50431, -46214, -42350, -38808, -35563,
+  -32589, -29864, -27367, -25078, -22981, -21060, -19298, 
-17685, -16206,
+  -14851, -13609, -12471, -11428, -10472, -9597, -8794, -8059, -7385, -6767,
+  -6202, -5683, -5208, -4772, 
-4373, -4008, -3672, -3365, -3084, -2826,
+  -2590, -2373, -2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
+  
-1081, -991, -908, -832, -763, -699, -640, -587, -538, -493, -452, -414,
+  -379, -348, -318, -292, 
-267, -245, -225, -206, -189, -173, -158, -145,
+  -133, -122, -112, -102, -94, -86, -79, -72, 
-66, -61, -56, -51, -47, -43,
+  -42, -41, -40, -39, -38, -37, -36, -35, -34, -33, 
-32, -31, -30, -29,
+  -28, -27, -26, -25, -24, -23, -22, -21, -20, -19, -18, -17, 
-16, -15,
+  -14, -13, -12, -11, -10, -9, -8, -7, -6, -5, -4, -3, -2, -1 

+};
+
+


+/* This value could be non-optimal. Without knowledge of the value

+   distribution in the real signal, the actual optimum cannot be evaluated.

+   Should be somewhere between 11.458 and 11.542. */ 

+static const gdouble DPCM_SCALE = 11.5131;
+
static gint8
+mve_enc_delta (guint n)
+{
+  if (n < 44)
+    return n;
+  return floor (DPCM_SCALE * log (n));
+}
+
+gint
+mve_compress_audio (guint8 * dest, const guint8 * src, guint16 len,
+    guint8 channels)
+{
+  gint16 prev[2], s;
+  gint delta, real_res;
+  gint cur_chan;
+  guint8 v;
+
+  for (cur_chan = 0; cur_chan < channels; ++cur_chan) {
+    prev[cur_chan] = GST_READ_UINT16_LE (src);
+    GST_WRITE_UINT16_LE (dest, prev[cur_chan]);
+    src += 2;
+    dest += 2;
+    len -= 2;
+  }
+
+  cur_chan = 0;
+  while (len > 0) {
+    s = GST_READ_UINT16_LE (src);
+    src += 2;
+
+    delta = s - prev[cur_chan];
+    
if (delta >= 0)
+      
v = mve_enc_delta (delta);
+    

+    else
+      
v = 256 - mve_enc_delta (-delta);
+    

real_res = dec_table[v] + prev[cur_chan];
+    
if (real_res < -32768 || real_res > 32767) {
+      

+          /* correct overflow */ 

+          /* GST_DEBUG ("co:%d + %d = %d -> new v:%d, dec_table:%d will be %d",
+             prev[cur_chan], dec_table[v], real_res,
+             v, dec_table[v], prev[cur_chan]+dec_table[v]); */
+          if (s > 0) {
+        
if (real_res > 32767)
+          --v;
+      
} else {
+        
if (real_res < -32768)
+          ++v;
+      
}
+
+      real_res = dec_table[v] + prev[cur_chan];
+    
}
+
+    if (G_UNLIKELY (abs (real_res - s) > 32767)) {
+      GST_ERROR ("sign loss left unfixed in audio stream, deviation:%ld",
+          real_res - s);
+      return -1;
+    }
+    

*dest++ = v;
+    
--len;
+    /* use previous output instead of input. That way output will not go too far from input. */ 

+        prev[cur_chan] += dec_table[v];
+    cur_chan = channels - 1 - cur_chan;
+  
}
+
+  return 0;
+}