Non-Functional voice communication.
git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@16 fc73d0e0-1445-4013-8a0c-d673dee63da5
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288
engine/sndcodec/g711.c
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288
engine/sndcodec/g711.c
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#include "bothdefs.h"
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#ifdef VOICECHAT
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/*
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* This source code is a product of Sun Microsystems, Inc. and is provided
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* for unrestricted use. Users may copy or modify this source code without
|
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* charge.
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||||
*
|
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* SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
|
||||
* THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
|
||||
* PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
|
||||
*
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||||
* Sun source code is provided with no support and without any obligation on
|
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* the part of Sun Microsystems, Inc. to assist in its use, correction,
|
||||
* modification or enhancement.
|
||||
*
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* SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
|
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* INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
|
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* OR ANY PART THEREOF.
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*
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* In no event will Sun Microsystems, Inc. be liable for any lost revenue
|
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* or profits or other special, indirect and consequential damages, even if
|
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* Sun has been advised of the possibility of such damages.
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*
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* Sun Microsystems, Inc.
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* 2550 Garcia Avenue
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* Mountain View, California 94043
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*/
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/*
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* g711.c
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*
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* u-law, A-law and linear PCM conversions.
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*/
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#define SIGN_BIT (0x80) /* Sign bit for a A-law byte. */
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#define QUANT_MASK (0xf) /* Quantization field mask. */
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#define NSEGS (8) /* Number of A-law segments. */
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#define SEG_SHIFT (4) /* Left shift for segment number. */
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#define SEG_MASK (0x70) /* Segment field mask. */
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static short seg_end[8] = {0xFF, 0x1FF, 0x3FF, 0x7FF,
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0xFFF, 0x1FFF, 0x3FFF, 0x7FFF};
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/* copy from CCITT G.711 specifications */
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unsigned char _u2a[128] = { /* u- to A-law conversions */
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1, 1, 2, 2, 3, 3, 4, 4,
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5, 5, 6, 6, 7, 7, 8, 8,
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9, 10, 11, 12, 13, 14, 15, 16,
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17, 18, 19, 20, 21, 22, 23, 24,
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25, 27, 29, 31, 33, 34, 35, 36,
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37, 38, 39, 40, 41, 42, 43, 44,
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46, 48, 49, 50, 51, 52, 53, 54,
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55, 56, 57, 58, 59, 60, 61, 62,
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64, 65, 66, 67, 68, 69, 70, 71,
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72, 73, 74, 75, 76, 77, 78, 79,
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81, 82, 83, 84, 85, 86, 87, 88,
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89, 90, 91, 92, 93, 94, 95, 96,
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97, 98, 99, 100, 101, 102, 103, 104,
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105, 106, 107, 108, 109, 110, 111, 112,
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113, 114, 115, 116, 117, 118, 119, 120,
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121, 122, 123, 124, 125, 126, 127, 128};
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unsigned char _a2u[128] = { /* A- to u-law conversions */
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1, 3, 5, 7, 9, 11, 13, 15,
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16, 17, 18, 19, 20, 21, 22, 23,
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24, 25, 26, 27, 28, 29, 30, 31,
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32, 32, 33, 33, 34, 34, 35, 35,
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36, 37, 38, 39, 40, 41, 42, 43,
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44, 45, 46, 47, 48, 48, 49, 49,
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50, 51, 52, 53, 54, 55, 56, 57,
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58, 59, 60, 61, 62, 63, 64, 64,
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65, 66, 67, 68, 69, 70, 71, 72,
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73, 74, 75, 76, 77, 78, 79, 79,
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80, 81, 82, 83, 84, 85, 86, 87,
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88, 89, 90, 91, 92, 93, 94, 95,
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96, 97, 98, 99, 100, 101, 102, 103,
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104, 105, 106, 107, 108, 109, 110, 111,
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112, 113, 114, 115, 116, 117, 118, 119,
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120, 121, 122, 123, 124, 125, 126, 127};
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static int
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search(
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int val,
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short *table,
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int size)
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{
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int i;
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for (i = 0; i < size; i++) {
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if (val <= *table++)
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return (i);
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}
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return (size);
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}
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/*
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* linear2alaw() - Convert a 16-bit linear PCM value to 8-bit A-law
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*
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* linear2alaw() accepts an 16-bit integer and encodes it as A-law data.
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*
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* Linear Input Code Compressed Code
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* ------------------------ ---------------
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* 0000000wxyza 000wxyz
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* 0000001wxyza 001wxyz
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* 000001wxyzab 010wxyz
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* 00001wxyzabc 011wxyz
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* 0001wxyzabcd 100wxyz
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* 001wxyzabcde 101wxyz
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* 01wxyzabcdef 110wxyz
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* 1wxyzabcdefg 111wxyz
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*
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* For further information see John C. Bellamy's Digital Telephony, 1982,
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* John Wiley & Sons, pps 98-111 and 472-476.
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*/
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unsigned char
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linear2alaw(
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int pcm_val) /* 2's complement (16-bit range) */
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{
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int mask;
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int seg;
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unsigned char aval;
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if (pcm_val >= 0) {
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mask = 0xD5; /* sign (7th) bit = 1 */
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} else {
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mask = 0x55; /* sign bit = 0 */
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pcm_val = -pcm_val - 8;
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}
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/* Convert the scaled magnitude to segment number. */
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seg = search(pcm_val, seg_end, 8);
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/* Combine the sign, segment, and quantization bits. */
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if (seg >= 8) /* out of range, return maximum value. */
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return (0x7F ^ mask);
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else {
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aval = seg << SEG_SHIFT;
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if (seg < 2)
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aval |= (pcm_val >> 4) & QUANT_MASK;
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else
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aval |= (pcm_val >> (seg + 3)) & QUANT_MASK;
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return (aval ^ mask);
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}
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}
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/*
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* alaw2linear() - Convert an A-law value to 16-bit linear PCM
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*
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*/
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int
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alaw2linear(
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unsigned char a_val)
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{
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int t;
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int seg;
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a_val ^= 0x55;
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t = (a_val & QUANT_MASK) << 4;
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seg = ((unsigned)a_val & SEG_MASK) >> SEG_SHIFT;
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switch (seg) {
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case 0:
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t += 8;
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break;
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case 1:
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t += 0x108;
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break;
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default:
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t += 0x108;
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t <<= seg - 1;
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}
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return ((a_val & SIGN_BIT) ? t : -t);
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}
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#define BIAS (0x84) /* Bias for linear code. */
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/*
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* linear2ulaw() - Convert a linear PCM value to u-law
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*
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* In order to simplify the encoding process, the original linear magnitude
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* is biased by adding 33 which shifts the encoding range from (0 - 8158) to
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* (33 - 8191). The result can be seen in the following encoding table:
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*
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* Biased Linear Input Code Compressed Code
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* ------------------------ ---------------
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* 00000001wxyza 000wxyz
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* 0000001wxyzab 001wxyz
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* 000001wxyzabc 010wxyz
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* 00001wxyzabcd 011wxyz
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* 0001wxyzabcde 100wxyz
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* 001wxyzabcdef 101wxyz
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* 01wxyzabcdefg 110wxyz
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* 1wxyzabcdefgh 111wxyz
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*
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* Each biased linear code has a leading 1 which identifies the segment
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* number. The value of the segment number is equal to 7 minus the number
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* of leading 0's. The quantization interval is directly available as the
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* four bits wxyz. * The trailing bits (a - h) are ignored.
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*
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* Ordinarily the complement of the resulting code word is used for
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* transmission, and so the code word is complemented before it is returned.
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*
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* For further information see John C. Bellamy's Digital Telephony, 1982,
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* John Wiley & Sons, pps 98-111 and 472-476.
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*/
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unsigned char
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linear2ulaw(
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int pcm_val) /* 2's complement (16-bit range) */
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{
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int mask;
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int seg;
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unsigned char uval;
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/* Get the sign and the magnitude of the value. */
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if (pcm_val < 0) {
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pcm_val = BIAS - pcm_val;
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mask = 0x7F;
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} else {
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pcm_val += BIAS;
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mask = 0xFF;
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}
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/* Convert the scaled magnitude to segment number. */
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seg = search(pcm_val, seg_end, 8);
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/*
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* Combine the sign, segment, quantization bits;
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* and complement the code word.
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*/
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if (seg >= 8) /* out of range, return maximum value. */
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return (0x7F ^ mask);
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else {
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uval = (seg << 4) | ((pcm_val >> (seg + 3)) & 0xF);
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return (uval ^ mask);
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}
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}
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/*
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* ulaw2linear() - Convert a u-law value to 16-bit linear PCM
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*
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* First, a biased linear code is derived from the code word. An unbiased
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* output can then be obtained by subtracting 33 from the biased code.
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*
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* Note that this function expects to be passed the complement of the
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* original code word. This is in keeping with ISDN conventions.
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*/
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int
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ulaw2linear(
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unsigned char u_val)
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{
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int t;
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/* Complement to obtain normal u-law value. */
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u_val = ~u_val;
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/*
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* Extract and bias the quantization bits. Then
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* shift up by the segment number and subtract out the bias.
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*/
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t = ((u_val & QUANT_MASK) << 3) + BIAS;
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t <<= ((unsigned)u_val & SEG_MASK) >> SEG_SHIFT;
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return ((u_val & SIGN_BIT) ? (BIAS - t) : (t - BIAS));
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}
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/* A-law to u-law conversion */
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unsigned char
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alaw2ulaw(
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unsigned char aval)
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{
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aval &= 0xff;
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return ((aval & 0x80) ? (0xFF ^ _a2u[aval ^ 0xD5]) :
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(0x7F ^ _a2u[aval ^ 0x55]));
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}
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/* u-law to A-law conversion */
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unsigned char
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ulaw2alaw(
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unsigned char uval)
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{
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uval &= 0xff;
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return ((uval & 0x80) ? (0xD5 ^ (_u2a[0xFF ^ uval] - 1)) :
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(0x55 ^ (_u2a[0x7F ^ uval] - 1)));
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}
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#endif
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178
engine/sndcodec/g721.c
Normal file
178
engine/sndcodec/g721.c
Normal file
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#include "bothdefs.h"
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#ifdef VOICECHAT
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/*
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* This source code is a product of Sun Microsystems, Inc. and is provided
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* for unrestricted use. Users may copy or modify this source code without
|
||||
* charge.
|
||||
*
|
||||
* SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
|
||||
* THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
|
||||
* PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
|
||||
*
|
||||
* Sun source code is provided with no support and without any obligation on
|
||||
* the part of Sun Microsystems, Inc. to assist in its use, correction,
|
||||
* modification or enhancement.
|
||||
*
|
||||
* SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
|
||||
* INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
|
||||
* OR ANY PART THEREOF.
|
||||
*
|
||||
* In no event will Sun Microsystems, Inc. be liable for any lost revenue
|
||||
* or profits or other special, indirect and consequential damages, even if
|
||||
* Sun has been advised of the possibility of such damages.
|
||||
*
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* Sun Microsystems, Inc.
|
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* 2550 Garcia Avenue
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* Mountain View, California 94043
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*/
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/*
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* g721.c
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*
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* Description:
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*
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* g721_encoder(), g721_decoder()
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*
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* These routines comprise an implementation of the CCITT G.721 ADPCM
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* coding algorithm. Essentially, this implementation is identical to
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* the bit level description except for a few deviations which
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* take advantage of work station attributes, such as hardware 2's
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* complement arithmetic and large memory. Specifically, certain time
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* consuming operations such as multiplications are replaced
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* with lookup tables and software 2's complement operations are
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* replaced with hardware 2's complement.
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*
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* The deviation from the bit level specification (lookup tables)
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* preserves the bit level performance specifications.
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*
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* As outlined in the G.721 Recommendation, the algorithm is broken
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* down into modules. Each section of code below is preceded by
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* the name of the module which it is implementing.
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*
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*/
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#include "g72x.h"
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static short qtab_721[7] = {-124, 80, 178, 246, 300, 349, 400};
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/*
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* Maps G.721 code word to reconstructed scale factor normalized log
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* magnitude values.
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*/
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static short _dqlntab[16] = {-2048, 4, 135, 213, 273, 323, 373, 425,
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425, 373, 323, 273, 213, 135, 4, -2048};
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/* Maps G.721 code word to log of scale factor multiplier. */
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static short _witab[16] = {-12, 18, 41, 64, 112, 198, 355, 1122,
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1122, 355, 198, 112, 64, 41, 18, -12};
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/*
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* Maps G.721 code words to a set of values whose long and short
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* term averages are computed and then compared to give an indication
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* how stationary (steady state) the signal is.
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*/
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static short _fitab[16] = {0, 0, 0, 0x200, 0x200, 0x200, 0x600, 0xE00,
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0xE00, 0x600, 0x200, 0x200, 0x200, 0, 0, 0};
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/*
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* g721_encoder()
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*
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* Encodes the input vale of linear PCM, A-law or u-law data sl and returns
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* the resulting code. -1 is returned for unknown input coding value.
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*/
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int
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g721_encoder(
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int sl,
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int in_coding,
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struct g72x_state *state_ptr)
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{
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short sezi, se, sez; /* ACCUM */
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short d; /* SUBTA */
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short sr; /* ADDB */
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short y; /* MIX */
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short dqsez; /* ADDC */
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short dq, i;
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switch (in_coding) { /* linearize input sample to 14-bit PCM */
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case AUDIO_ENCODING_ALAW:
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sl = alaw2linear((unsigned char)sl) >> 2;
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break;
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case AUDIO_ENCODING_ULAW:
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sl = ulaw2linear((unsigned char)sl) >> 2;
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break;
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case AUDIO_ENCODING_LINEAR:
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sl >>= 2; /* 14-bit dynamic range */
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break;
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default:
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return (-1);
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}
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sezi = predictor_zero(state_ptr);
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sez = sezi >> 1;
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se = (sezi + predictor_pole(state_ptr)) >> 1; /* estimated signal */
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|
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d = sl - se; /* estimation difference */
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/* quantize the prediction difference */
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y = step_size(state_ptr); /* quantizer step size */
|
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i = quantize(d, y, qtab_721, 7); /* i = ADPCM code */
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dq = reconstruct(i & 8, _dqlntab[i], y); /* quantized est diff */
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sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; /* reconst. signal */
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dqsez = sr + sez - se; /* pole prediction diff. */
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update(4, y, _witab[i] << 5, _fitab[i], dq, sr, dqsez, state_ptr);
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return (i);
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}
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/*
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* g721_decoder()
|
||||
*
|
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* Description:
|
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*
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* Decodes a 4-bit code of G.721 encoded data of i and
|
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* returns the resulting linear PCM, A-law or u-law value.
|
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* return -1 for unknown out_coding value.
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*/
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int
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g721_decoder(
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int i,
|
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int out_coding,
|
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struct g72x_state *state_ptr)
|
||||
{
|
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short sezi, sei, sez, se; /* ACCUM */
|
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short y; /* MIX */
|
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short sr; /* ADDB */
|
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short dq;
|
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short dqsez;
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i &= 0x0f; /* mask to get proper bits */
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sezi = predictor_zero(state_ptr);
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sez = sezi >> 1;
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sei = sezi + predictor_pole(state_ptr);
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se = sei >> 1; /* se = estimated signal */
|
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|
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y = step_size(state_ptr); /* dynamic quantizer step size */
|
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|
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dq = reconstruct(i & 0x08, _dqlntab[i], y); /* quantized diff. */
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sr = (dq < 0) ? (se - (dq & 0x3FFF)) : se + dq; /* reconst. signal */
|
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|
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dqsez = sr - se + sez; /* pole prediction diff. */
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update(4, y, _witab[i] << 5, _fitab[i], dq, sr, dqsez, state_ptr);
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switch (out_coding) {
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case AUDIO_ENCODING_ALAW:
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return (tandem_adjust_alaw(sr, se, y, i, 8, qtab_721));
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||||
case AUDIO_ENCODING_ULAW:
|
||||
return (tandem_adjust_ulaw(sr, se, y, i, 8, qtab_721));
|
||||
case AUDIO_ENCODING_LINEAR:
|
||||
return (sr << 2); /* sr was 14-bit dynamic range */
|
||||
default:
|
||||
return (-1);
|
||||
}
|
||||
}
|
||||
|
||||
#endif
|
||||
163
engine/sndcodec/g723_24.c
Normal file
163
engine/sndcodec/g723_24.c
Normal file
|
|
@ -0,0 +1,163 @@
|
|||
#include "bothdefs.h"
|
||||
|
||||
#ifdef VOICECHAT
|
||||
/*
|
||||
* This source code is a product of Sun Microsystems, Inc. and is provided
|
||||
* for unrestricted use. Users may copy or modify this source code without
|
||||
* charge.
|
||||
*
|
||||
* SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
|
||||
* THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
|
||||
* PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
|
||||
*
|
||||
* Sun source code is provided with no support and without any obligation on
|
||||
* the part of Sun Microsystems, Inc. to assist in its use, correction,
|
||||
* modification or enhancement.
|
||||
*
|
||||
* SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
|
||||
* INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
|
||||
* OR ANY PART THEREOF.
|
||||
*
|
||||
* In no event will Sun Microsystems, Inc. be liable for any lost revenue
|
||||
* or profits or other special, indirect and consequential damages, even if
|
||||
* Sun has been advised of the possibility of such damages.
|
||||
*
|
||||
* Sun Microsystems, Inc.
|
||||
* 2550 Garcia Avenue
|
||||
* Mountain View, California 94043
|
||||
*/
|
||||
|
||||
/*
|
||||
* g723_24.c
|
||||
*
|
||||
* Description:
|
||||
*
|
||||
* g723_24_encoder(), g723_24_decoder()
|
||||
*
|
||||
* These routines comprise an implementation of the CCITT G.723 24 Kbps
|
||||
* ADPCM coding algorithm. Essentially, this implementation is identical to
|
||||
* the bit level description except for a few deviations which take advantage
|
||||
* of workstation attributes, such as hardware 2's complement arithmetic.
|
||||
*
|
||||
*/
|
||||
#include "g72x.h"
|
||||
|
||||
/*
|
||||
* Maps G.723_24 code word to reconstructed scale factor normalized log
|
||||
* magnitude values.
|
||||
*/
|
||||
static short _dqlntab[8] = {-2048, 135, 273, 373, 373, 273, 135, -2048};
|
||||
|
||||
/* Maps G.723_24 code word to log of scale factor multiplier. */
|
||||
static short _witab[8] = {-128, 960, 4384, 18624, 18624, 4384, 960, -128};
|
||||
|
||||
/*
|
||||
* Maps G.723_24 code words to a set of values whose long and short
|
||||
* term averages are computed and then compared to give an indication
|
||||
* how stationary (steady state) the signal is.
|
||||
*/
|
||||
static short _fitab[8] = {0, 0x200, 0x400, 0xE00, 0xE00, 0x400, 0x200, 0};
|
||||
|
||||
static short qtab_723_24[3] = {8, 218, 331};
|
||||
|
||||
/*
|
||||
* g723_24_encoder()
|
||||
*
|
||||
* Encodes a linear PCM, A-law or u-law input sample and returns its 3-bit code.
|
||||
* Returns -1 if invalid input coding value.
|
||||
*/
|
||||
int
|
||||
g723_24_encoder(
|
||||
int sl,
|
||||
int in_coding,
|
||||
struct g72x_state *state_ptr)
|
||||
{
|
||||
short sei, sezi, se, sez; /* ACCUM */
|
||||
short d; /* SUBTA */
|
||||
short y; /* MIX */
|
||||
short sr; /* ADDB */
|
||||
short dqsez; /* ADDC */
|
||||
short dq, i;
|
||||
|
||||
switch (in_coding) { /* linearize input sample to 14-bit PCM */
|
||||
case AUDIO_ENCODING_ALAW:
|
||||
sl = alaw2linear((unsigned char)sl) >> 2;
|
||||
break;
|
||||
case AUDIO_ENCODING_ULAW:
|
||||
sl = ulaw2linear((unsigned char)sl) >> 2;
|
||||
break;
|
||||
case AUDIO_ENCODING_LINEAR:
|
||||
sl >>= 2; /* sl of 14-bit dynamic range */
|
||||
break;
|
||||
default:
|
||||
return (-1);
|
||||
}
|
||||
|
||||
sezi = predictor_zero(state_ptr);
|
||||
sez = sezi >> 1;
|
||||
sei = sezi + predictor_pole(state_ptr);
|
||||
se = sei >> 1; /* se = estimated signal */
|
||||
|
||||
d = sl - se; /* d = estimation diff. */
|
||||
|
||||
/* quantize prediction difference d */
|
||||
y = step_size(state_ptr); /* quantizer step size */
|
||||
i = quantize(d, y, qtab_723_24, 3); /* i = ADPCM code */
|
||||
dq = reconstruct(i & 4, _dqlntab[i], y); /* quantized diff. */
|
||||
|
||||
sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; /* reconstructed signal */
|
||||
|
||||
dqsez = sr + sez - se; /* pole prediction diff. */
|
||||
|
||||
update(3, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
|
||||
|
||||
return (i);
|
||||
}
|
||||
|
||||
/*
|
||||
* g723_24_decoder()
|
||||
*
|
||||
* Decodes a 3-bit CCITT G.723_24 ADPCM code and returns
|
||||
* the resulting 16-bit linear PCM, A-law or u-law sample value.
|
||||
* -1 is returned if the output coding is unknown.
|
||||
*/
|
||||
int
|
||||
g723_24_decoder(
|
||||
int i,
|
||||
int out_coding,
|
||||
struct g72x_state *state_ptr)
|
||||
{
|
||||
short sezi, sei, sez, se; /* ACCUM */
|
||||
short y; /* MIX */
|
||||
short sr; /* ADDB */
|
||||
short dq;
|
||||
short dqsez;
|
||||
|
||||
i &= 0x07; /* mask to get proper bits */
|
||||
sezi = predictor_zero(state_ptr);
|
||||
sez = sezi >> 1;
|
||||
sei = sezi + predictor_pole(state_ptr);
|
||||
se = sei >> 1; /* se = estimated signal */
|
||||
|
||||
y = step_size(state_ptr); /* adaptive quantizer step size */
|
||||
dq = reconstruct(i & 0x04, _dqlntab[i], y); /* unquantize pred diff */
|
||||
|
||||
sr = (dq < 0) ? (se - (dq & 0x3FFF)) : (se + dq); /* reconst. signal */
|
||||
|
||||
dqsez = sr - se + sez; /* pole prediction diff. */
|
||||
|
||||
update(3, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
|
||||
|
||||
switch (out_coding) {
|
||||
case AUDIO_ENCODING_ALAW:
|
||||
return (tandem_adjust_alaw(sr, se, y, i, 4, qtab_723_24));
|
||||
case AUDIO_ENCODING_ULAW:
|
||||
return (tandem_adjust_ulaw(sr, se, y, i, 4, qtab_723_24));
|
||||
case AUDIO_ENCODING_LINEAR:
|
||||
return (sr << 2); /* sr was of 14-bit dynamic range */
|
||||
default:
|
||||
return (-1);
|
||||
}
|
||||
}
|
||||
|
||||
#endif
|
||||
183
engine/sndcodec/g723_40.c
Normal file
183
engine/sndcodec/g723_40.c
Normal file
|
|
@ -0,0 +1,183 @@
|
|||
#include "bothdefs.h"
|
||||
|
||||
#ifdef VOICECHAT
|
||||
/*
|
||||
* This source code is a product of Sun Microsystems, Inc. and is provided
|
||||
* for unrestricted use. Users may copy or modify this source code without
|
||||
* charge.
|
||||
*
|
||||
* SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
|
||||
* THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
|
||||
* PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
|
||||
*
|
||||
* Sun source code is provided with no support and without any obligation on
|
||||
* the part of Sun Microsystems, Inc. to assist in its use, correction,
|
||||
* modification or enhancement.
|
||||
*
|
||||
* SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
|
||||
* INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
|
||||
* OR ANY PART THEREOF.
|
||||
*
|
||||
* In no event will Sun Microsystems, Inc. be liable for any lost revenue
|
||||
* or profits or other special, indirect and consequential damages, even if
|
||||
* Sun has been advised of the possibility of such damages.
|
||||
*
|
||||
* Sun Microsystems, Inc.
|
||||
* 2550 Garcia Avenue
|
||||
* Mountain View, California 94043
|
||||
*/
|
||||
|
||||
/*
|
||||
* g723_40.c
|
||||
*
|
||||
* Description:
|
||||
*
|
||||
* g723_40_encoder(), g723_40_decoder()
|
||||
*
|
||||
* These routines comprise an implementation of the CCITT G.723 40Kbps
|
||||
* ADPCM coding algorithm. Essentially, this implementation is identical to
|
||||
* the bit level description except for a few deviations which
|
||||
* take advantage of workstation attributes, such as hardware 2's
|
||||
* complement arithmetic.
|
||||
*
|
||||
* The deviation from the bit level specification (lookup tables),
|
||||
* preserves the bit level performance specifications.
|
||||
*
|
||||
* As outlined in the G.723 Recommendation, the algorithm is broken
|
||||
* down into modules. Each section of code below is preceded by
|
||||
* the name of the module which it is implementing.
|
||||
*
|
||||
*/
|
||||
#include "g72x.h"
|
||||
|
||||
/*
|
||||
* Maps G.723_40 code word to ructeconstructed scale factor normalized log
|
||||
* magnitude values.
|
||||
*/
|
||||
static short _dqlntab[32] = {-2048, -66, 28, 104, 169, 224, 274, 318,
|
||||
358, 395, 429, 459, 488, 514, 539, 566,
|
||||
566, 539, 514, 488, 459, 429, 395, 358,
|
||||
318, 274, 224, 169, 104, 28, -66, -2048};
|
||||
|
||||
/* Maps G.723_40 code word to log of scale factor multiplier. */
|
||||
static short _witab[32] = {448, 448, 768, 1248, 1280, 1312, 1856, 3200,
|
||||
4512, 5728, 7008, 8960, 11456, 14080, 16928, 22272,
|
||||
22272, 16928, 14080, 11456, 8960, 7008, 5728, 4512,
|
||||
3200, 1856, 1312, 1280, 1248, 768, 448, 448};
|
||||
|
||||
/*
|
||||
* Maps G.723_40 code words to a set of values whose long and short
|
||||
* term averages are computed and then compared to give an indication
|
||||
* how stationary (steady state) the signal is.
|
||||
*/
|
||||
static short _fitab[32] = {0, 0, 0, 0, 0, 0x200, 0x200, 0x200,
|
||||
0x200, 0x200, 0x400, 0x600, 0x800, 0xA00, 0xC00, 0xC00,
|
||||
0xC00, 0xC00, 0xA00, 0x800, 0x600, 0x400, 0x200, 0x200,
|
||||
0x200, 0x200, 0x200, 0, 0, 0, 0, 0};
|
||||
|
||||
static short qtab_723_40[15] = {-122, -16, 68, 139, 198, 250, 298, 339,
|
||||
378, 413, 445, 475, 502, 528, 553};
|
||||
|
||||
/*
|
||||
* g723_40_encoder()
|
||||
*
|
||||
* Encodes a 16-bit linear PCM, A-law or u-law input sample and retuens
|
||||
* the resulting 5-bit CCITT G.723 40Kbps code.
|
||||
* Returns -1 if the input coding value is invalid.
|
||||
*/
|
||||
int
|
||||
g723_40_encoder(
|
||||
int sl,
|
||||
int in_coding,
|
||||
struct g72x_state *state_ptr)
|
||||
{
|
||||
short sei, sezi, se, sez; /* ACCUM */
|
||||
short d; /* SUBTA */
|
||||
short y; /* MIX */
|
||||
short sr; /* ADDB */
|
||||
short dqsez; /* ADDC */
|
||||
short dq, i;
|
||||
|
||||
switch (in_coding) { /* linearize input sample to 14-bit PCM */
|
||||
case AUDIO_ENCODING_ALAW:
|
||||
sl = alaw2linear((unsigned char)sl) >> 2;
|
||||
break;
|
||||
case AUDIO_ENCODING_ULAW:
|
||||
sl = ulaw2linear((unsigned char)sl) >> 2;
|
||||
break;
|
||||
case AUDIO_ENCODING_LINEAR:
|
||||
sl >>= 2; /* sl of 14-bit dynamic range */
|
||||
break;
|
||||
default:
|
||||
return (-1);
|
||||
}
|
||||
|
||||
sezi = predictor_zero(state_ptr);
|
||||
sez = sezi >> 1;
|
||||
sei = sezi + predictor_pole(state_ptr);
|
||||
se = sei >> 1; /* se = estimated signal */
|
||||
|
||||
d = sl - se; /* d = estimation difference */
|
||||
|
||||
/* quantize prediction difference */
|
||||
y = step_size(state_ptr); /* adaptive quantizer step size */
|
||||
i = quantize(d, y, qtab_723_40, 15); /* i = ADPCM code */
|
||||
|
||||
dq = reconstruct(i & 0x10, _dqlntab[i], y); /* quantized diff */
|
||||
|
||||
sr = (dq < 0) ? se - (dq & 0x7FFF) : se + dq; /* reconstructed signal */
|
||||
|
||||
dqsez = sr + sez - se; /* dqsez = pole prediction diff. */
|
||||
|
||||
update(5, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
|
||||
|
||||
return (i);
|
||||
}
|
||||
|
||||
/*
|
||||
* g723_40_decoder()
|
||||
*
|
||||
* Decodes a 5-bit CCITT G.723 40Kbps code and returns
|
||||
* the resulting 16-bit linear PCM, A-law or u-law sample value.
|
||||
* -1 is returned if the output coding is unknown.
|
||||
*/
|
||||
int
|
||||
g723_40_decoder(
|
||||
int i,
|
||||
int out_coding,
|
||||
struct g72x_state *state_ptr)
|
||||
{
|
||||
short sezi, sei, sez, se; /* ACCUM */
|
||||
short y; /* MIX */
|
||||
short sr; /* ADDB */
|
||||
short dq;
|
||||
short dqsez;
|
||||
|
||||
i &= 0x1f; /* mask to get proper bits */
|
||||
sezi = predictor_zero(state_ptr);
|
||||
sez = sezi >> 1;
|
||||
sei = sezi + predictor_pole(state_ptr);
|
||||
se = sei >> 1; /* se = estimated signal */
|
||||
|
||||
y = step_size(state_ptr); /* adaptive quantizer step size */
|
||||
dq = reconstruct(i & 0x10, _dqlntab[i], y); /* estimation diff. */
|
||||
|
||||
sr = (dq < 0) ? (se - (dq & 0x7FFF)) : (se + dq); /* reconst. signal */
|
||||
|
||||
dqsez = sr - se + sez; /* pole prediction diff. */
|
||||
|
||||
update(5, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
|
||||
|
||||
switch (out_coding) {
|
||||
case AUDIO_ENCODING_ALAW:
|
||||
return (tandem_adjust_alaw(sr, se, y, i, 0x10, qtab_723_40));
|
||||
case AUDIO_ENCODING_ULAW:
|
||||
return (tandem_adjust_ulaw(sr, se, y, i, 0x10, qtab_723_40));
|
||||
case AUDIO_ENCODING_LINEAR:
|
||||
return (sr << 2); /* sr was of 14-bit dynamic range */
|
||||
default:
|
||||
return (-1);
|
||||
}
|
||||
}
|
||||
|
||||
#endif
|
||||
573
engine/sndcodec/g72x.c
Normal file
573
engine/sndcodec/g72x.c
Normal file
|
|
@ -0,0 +1,573 @@
|
|||
#include "bothdefs.h"
|
||||
|
||||
#ifdef VOICECHAT
|
||||
|
||||
/*
|
||||
* This source code is a product of Sun Microsystems, Inc. and is provided
|
||||
* for unrestricted use. Users may copy or modify this source code without
|
||||
* charge.
|
||||
*
|
||||
* SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
|
||||
* THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
|
||||
* PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
|
||||
*
|
||||
* Sun source code is provided with no support and without any obligation on
|
||||
* the part of Sun Microsystems, Inc. to assist in its use, correction,
|
||||
* modification or enhancement.
|
||||
*
|
||||
* SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
|
||||
* INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
|
||||
* OR ANY PART THEREOF.
|
||||
*
|
||||
* In no event will Sun Microsystems, Inc. be liable for any lost revenue
|
||||
* or profits or other special, indirect and consequential damages, even if
|
||||
* Sun has been advised of the possibility of such damages.
|
||||
*
|
||||
* Sun Microsystems, Inc.
|
||||
* 2550 Garcia Avenue
|
||||
* Mountain View, California 94043
|
||||
*/
|
||||
|
||||
/*
|
||||
* g72x.c
|
||||
*
|
||||
* Common routines for G.721 and G.723 conversions.
|
||||
*/
|
||||
|
||||
#include "g72x.h"
|
||||
#include <math.h>
|
||||
#include <stdlib.h>
|
||||
|
||||
static short power2[15] = {1, 2, 4, 8, 0x10, 0x20, 0x40, 0x80,
|
||||
0x100, 0x200, 0x400, 0x800, 0x1000, 0x2000, 0x4000};
|
||||
|
||||
/*
|
||||
* quan()
|
||||
*
|
||||
* quantizes the input val against the table of size short integers.
|
||||
* It returns i if table[i - 1] <= val < table[i].
|
||||
*
|
||||
* Using linear search for simple coding.
|
||||
*/
|
||||
static int
|
||||
quan(
|
||||
int val,
|
||||
short *table,
|
||||
int size)
|
||||
{
|
||||
int i;
|
||||
|
||||
for (i = 0; i < size; i++)
|
||||
if (val < *table++)
|
||||
break;
|
||||
return (i);
|
||||
}
|
||||
|
||||
/*
|
||||
* fmult()
|
||||
*
|
||||
* returns the integer product of the 14-bit integer "an" and
|
||||
* "floating point" representation (4-bit exponent, 6-bit mantessa) "srn".
|
||||
*/
|
||||
static int
|
||||
fmult(
|
||||
int an,
|
||||
int srn)
|
||||
{
|
||||
short anmag, anexp, anmant;
|
||||
short wanexp, wanmant;
|
||||
short retval;
|
||||
|
||||
anmag = (an > 0) ? an : ((-an) & 0x1FFF);
|
||||
anexp = quan(anmag, power2, 15) - 6;
|
||||
anmant = (anmag == 0) ? 32 :
|
||||
(anexp >= 0) ? anmag >> anexp : anmag << -anexp;
|
||||
wanexp = anexp + ((srn >> 6) & 0xF) - 13;
|
||||
|
||||
wanmant = (anmant * (srn & 077) + 0x30) >> 4;
|
||||
retval = (wanexp >= 0) ? ((wanmant << wanexp) & 0x7FFF) :
|
||||
(wanmant >> -wanexp);
|
||||
|
||||
return (((an ^ srn) < 0) ? -retval : retval);
|
||||
}
|
||||
|
||||
/*
|
||||
* g72x_init_state()
|
||||
*
|
||||
* This routine initializes and/or resets the g72x_state structure
|
||||
* pointed to by 'state_ptr'.
|
||||
* All the initial state values are specified in the CCITT G.721 document.
|
||||
*/
|
||||
void
|
||||
g72x_init_state(
|
||||
struct g72x_state *state_ptr)
|
||||
{
|
||||
int cnta;
|
||||
|
||||
state_ptr->yl = 34816;
|
||||
state_ptr->yu = 544;
|
||||
state_ptr->dms = 0;
|
||||
state_ptr->dml = 0;
|
||||
state_ptr->ap = 0;
|
||||
for (cnta = 0; cnta < 2; cnta++) {
|
||||
state_ptr->a[cnta] = 0;
|
||||
state_ptr->pk[cnta] = 0;
|
||||
state_ptr->sr[cnta] = 32;
|
||||
}
|
||||
for (cnta = 0; cnta < 6; cnta++) {
|
||||
state_ptr->b[cnta] = 0;
|
||||
state_ptr->dq[cnta] = 32;
|
||||
}
|
||||
state_ptr->td = 0;
|
||||
}
|
||||
|
||||
/*
|
||||
* predictor_zero()
|
||||
*
|
||||
* computes the estimated signal from 6-zero predictor.
|
||||
*
|
||||
*/
|
||||
int
|
||||
predictor_zero(
|
||||
struct g72x_state *state_ptr)
|
||||
{
|
||||
int i;
|
||||
int sezi;
|
||||
|
||||
sezi = fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]);
|
||||
for (i = 1; i < 6; i++) /* ACCUM */
|
||||
sezi += fmult(state_ptr->b[i] >> 2, state_ptr->dq[i]);
|
||||
return (sezi);
|
||||
}
|
||||
/*
|
||||
* predictor_pole()
|
||||
*
|
||||
* computes the estimated signal from 2-pole predictor.
|
||||
*
|
||||
*/
|
||||
int
|
||||
predictor_pole(
|
||||
struct g72x_state *state_ptr)
|
||||
{
|
||||
return (fmult(state_ptr->a[1] >> 2, state_ptr->sr[1]) +
|
||||
fmult(state_ptr->a[0] >> 2, state_ptr->sr[0]));
|
||||
}
|
||||
/*
|
||||
* step_size()
|
||||
*
|
||||
* computes the quantization step size of the adaptive quantizer.
|
||||
*
|
||||
*/
|
||||
int
|
||||
step_size(
|
||||
struct g72x_state *state_ptr)
|
||||
{
|
||||
int y;
|
||||
int dif;
|
||||
int al;
|
||||
|
||||
if (state_ptr->ap >= 256)
|
||||
return (state_ptr->yu);
|
||||
else {
|
||||
y = state_ptr->yl >> 6;
|
||||
dif = state_ptr->yu - y;
|
||||
al = state_ptr->ap >> 2;
|
||||
if (dif > 0)
|
||||
y += (dif * al) >> 6;
|
||||
else if (dif < 0)
|
||||
y += (dif * al + 0x3F) >> 6;
|
||||
return (y);
|
||||
}
|
||||
}
|
||||
|
||||
/*
|
||||
* quantize()
|
||||
*
|
||||
* Given a raw sample, 'd', of the difference signal and a
|
||||
* quantization step size scale factor, 'y', this routine returns the
|
||||
* ADPCM codeword to which that sample gets quantized. The step
|
||||
* size scale factor division operation is done in the log base 2 domain
|
||||
* as a subtraction.
|
||||
*/
|
||||
int
|
||||
quantize(
|
||||
int d, /* Raw difference signal sample */
|
||||
int y, /* Step size multiplier */
|
||||
short *table, /* quantization table */
|
||||
int size) /* table size of short integers */
|
||||
{
|
||||
short dqm; /* Magnitude of 'd' */
|
||||
short exp; /* Integer part of base 2 log of 'd' */
|
||||
short mant; /* Fractional part of base 2 log */
|
||||
short dl; /* Log of magnitude of 'd' */
|
||||
short dln; /* Step size scale factor normalized log */
|
||||
int i;
|
||||
|
||||
/*
|
||||
* LOG
|
||||
*
|
||||
* Compute base 2 log of 'd', and store in 'dl'.
|
||||
*/
|
||||
dqm = abs(d);
|
||||
exp = quan(dqm >> 1, power2, 15);
|
||||
mant = ((dqm << 7) >> exp) & 0x7F; /* Fractional portion. */
|
||||
dl = (exp << 7) + mant;
|
||||
|
||||
/*
|
||||
* SUBTB
|
||||
*
|
||||
* "Divide" by step size multiplier.
|
||||
*/
|
||||
dln = dl - (y >> 2);
|
||||
|
||||
/*
|
||||
* QUAN
|
||||
*
|
||||
* Obtain codword i for 'd'.
|
||||
*/
|
||||
i = quan(dln, table, size);
|
||||
if (d < 0) /* take 1's complement of i */
|
||||
return ((size << 1) + 1 - i);
|
||||
else if (i == 0) /* take 1's complement of 0 */
|
||||
return ((size << 1) + 1); /* new in 1988 */
|
||||
else
|
||||
return (i);
|
||||
}
|
||||
/*
|
||||
* reconstruct()
|
||||
*
|
||||
* Returns reconstructed difference signal 'dq' obtained from
|
||||
* codeword 'i' and quantization step size scale factor 'y'.
|
||||
* Multiplication is performed in log base 2 domain as addition.
|
||||
*/
|
||||
int
|
||||
reconstruct(
|
||||
int sign, /* 0 for non-negative value */
|
||||
int dqln, /* G.72x codeword */
|
||||
int y) /* Step size multiplier */
|
||||
{
|
||||
short dql; /* Log of 'dq' magnitude */
|
||||
short dex; /* Integer part of log */
|
||||
short dqt;
|
||||
short dq; /* Reconstructed difference signal sample */
|
||||
|
||||
dql = dqln + (y >> 2); /* ADDA */
|
||||
|
||||
if (dql < 0) {
|
||||
return ((sign) ? -0x8000 : 0);
|
||||
} else { /* ANTILOG */
|
||||
dex = (dql >> 7) & 15;
|
||||
dqt = 128 + (dql & 127);
|
||||
dq = (dqt << 7) >> (14 - dex);
|
||||
return ((sign) ? (dq - 0x8000) : dq);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
/*
|
||||
* update()
|
||||
*
|
||||
* updates the state variables for each output code
|
||||
*/
|
||||
void
|
||||
update(
|
||||
int code_size, /* distinguish 723_40 with others */
|
||||
int y, /* quantizer step size */
|
||||
int wi, /* scale factor multiplier */
|
||||
int fi, /* for long/short term energies */
|
||||
int dq, /* quantized prediction difference */
|
||||
int sr, /* reconstructed signal */
|
||||
int dqsez, /* difference from 2-pole predictor */
|
||||
struct g72x_state *state_ptr) /* coder state pointer */
|
||||
{
|
||||
int cnt;
|
||||
short mag, exp; /* Adaptive predictor, FLOAT A */
|
||||
short a2p; /* LIMC */
|
||||
short a1ul; /* UPA1 */
|
||||
short pks1; /* UPA2 */
|
||||
short fa1;
|
||||
char tr; /* tone/transition detector */
|
||||
short ylint, thr2, dqthr;
|
||||
short ylfrac, thr1;
|
||||
short pk0;
|
||||
|
||||
pk0 = (dqsez < 0) ? 1 : 0; /* needed in updating predictor poles */
|
||||
|
||||
mag = dq & 0x7FFF; /* prediction difference magnitude */
|
||||
/* TRANS */
|
||||
ylint = (short)(state_ptr->yl >> 15); /* exponent part of yl */
|
||||
ylfrac = (state_ptr->yl >> 10) & 0x1F; /* fractional part of yl */
|
||||
thr1 = (32 + ylfrac) << ylint; /* threshold */
|
||||
thr2 = (ylint > 9) ? 31 << 10 : thr1; /* limit thr2 to 31 << 10 */
|
||||
dqthr = (thr2 + (thr2 >> 1)) >> 1; /* dqthr = 0.75 * thr2 */
|
||||
if (state_ptr->td == 0) /* signal supposed voice */
|
||||
tr = 0;
|
||||
else if (mag <= dqthr) /* supposed data, but small mag */
|
||||
tr = 0; /* treated as voice */
|
||||
else /* signal is data (modem) */
|
||||
tr = 1;
|
||||
|
||||
/*
|
||||
* Quantizer scale factor adaptation.
|
||||
*/
|
||||
|
||||
/* FUNCTW & FILTD & DELAY */
|
||||
/* update non-steady state step size multiplier */
|
||||
state_ptr->yu = y + ((wi - y) >> 5);
|
||||
|
||||
/* LIMB */
|
||||
if (state_ptr->yu < 544) /* 544 <= yu <= 5120 */
|
||||
state_ptr->yu = 544;
|
||||
else if (state_ptr->yu > 5120)
|
||||
state_ptr->yu = 5120;
|
||||
|
||||
/* FILTE & DELAY */
|
||||
/* update steady state step size multiplier */
|
||||
state_ptr->yl += state_ptr->yu + ((-state_ptr->yl) >> 6);
|
||||
|
||||
/*
|
||||
* Adaptive predictor coefficients.
|
||||
*/
|
||||
if (tr == 1) { /* reset a's and b's for modem signal */
|
||||
state_ptr->a[0] = 0;
|
||||
state_ptr->a[1] = 0;
|
||||
state_ptr->b[0] = 0;
|
||||
state_ptr->b[1] = 0;
|
||||
state_ptr->b[2] = 0;
|
||||
state_ptr->b[3] = 0;
|
||||
state_ptr->b[4] = 0;
|
||||
state_ptr->b[5] = 0;
|
||||
} else { /* update a's and b's */
|
||||
pks1 = pk0 ^ state_ptr->pk[0]; /* UPA2 */
|
||||
|
||||
/* update predictor pole a[1] */
|
||||
a2p = state_ptr->a[1] - (state_ptr->a[1] >> 7);
|
||||
if (dqsez != 0) {
|
||||
fa1 = (pks1) ? state_ptr->a[0] : -state_ptr->a[0];
|
||||
if (fa1 < -8191) /* a2p = function of fa1 */
|
||||
a2p -= 0x100;
|
||||
else if (fa1 > 8191)
|
||||
a2p += 0xFF;
|
||||
else
|
||||
a2p += fa1 >> 5;
|
||||
|
||||
if (pk0 ^ state_ptr->pk[1])
|
||||
/* LIMC */
|
||||
if (a2p <= -12160)
|
||||
a2p = -12288;
|
||||
else if (a2p >= 12416)
|
||||
a2p = 12288;
|
||||
else
|
||||
a2p -= 0x80;
|
||||
else if (a2p <= -12416)
|
||||
a2p = -12288;
|
||||
else if (a2p >= 12160)
|
||||
a2p = 12288;
|
||||
else
|
||||
a2p += 0x80;
|
||||
}
|
||||
|
||||
/* TRIGB & DELAY */
|
||||
state_ptr->a[1] = a2p;
|
||||
|
||||
/* UPA1 */
|
||||
/* update predictor pole a[0] */
|
||||
state_ptr->a[0] -= state_ptr->a[0] >> 8;
|
||||
if (dqsez != 0)
|
||||
{
|
||||
if (pks1 == 0)
|
||||
state_ptr->a[0] += 192;
|
||||
else
|
||||
state_ptr->a[0] -= 192;
|
||||
}
|
||||
|
||||
/* LIMD */
|
||||
a1ul = 15360 - a2p;
|
||||
if (state_ptr->a[0] < -a1ul)
|
||||
state_ptr->a[0] = -a1ul;
|
||||
else if (state_ptr->a[0] > a1ul)
|
||||
state_ptr->a[0] = a1ul;
|
||||
|
||||
/* UPB : update predictor zeros b[6] */
|
||||
for (cnt = 0; cnt < 6; cnt++) {
|
||||
if (code_size == 5) /* for 40Kbps G.723 */
|
||||
state_ptr->b[cnt] -= state_ptr->b[cnt] >> 9;
|
||||
else /* for G.721 and 24Kbps G.723 */
|
||||
state_ptr->b[cnt] -= state_ptr->b[cnt] >> 8;
|
||||
if (dq & 0x7FFF) { /* XOR */
|
||||
if ((dq ^ state_ptr->dq[cnt]) >= 0)
|
||||
state_ptr->b[cnt] += 128;
|
||||
else
|
||||
state_ptr->b[cnt] -= 128;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
for (cnt = 5; cnt > 0; cnt--)
|
||||
state_ptr->dq[cnt] = state_ptr->dq[cnt-1];
|
||||
/* FLOAT A : convert dq[0] to 4-bit exp, 6-bit mantissa f.p. */
|
||||
if (mag == 0) {
|
||||
state_ptr->dq[0] = (dq >= 0) ? 0x20 : 0xFC20;
|
||||
} else {
|
||||
exp = quan(mag, power2, 15);
|
||||
state_ptr->dq[0] = (dq >= 0) ?
|
||||
(exp << 6) + ((mag << 6) >> exp) :
|
||||
(exp << 6) + ((mag << 6) >> exp) - 0x400;
|
||||
}
|
||||
|
||||
state_ptr->sr[1] = state_ptr->sr[0];
|
||||
/* FLOAT B : convert sr to 4-bit exp., 6-bit mantissa f.p. */
|
||||
if (sr == 0) {
|
||||
state_ptr->sr[0] = 0x20;
|
||||
} else if (sr > 0) {
|
||||
exp = quan(sr, power2, 15);
|
||||
state_ptr->sr[0] = (exp << 6) + ((sr << 6) >> exp);
|
||||
} else if (sr > -32768) {
|
||||
mag = -sr;
|
||||
exp = quan(mag, power2, 15);
|
||||
state_ptr->sr[0] = (exp << 6) + ((mag << 6) >> exp) - 0x400;
|
||||
} else
|
||||
state_ptr->sr[0] = (short)0xFC20;
|
||||
|
||||
/* DELAY A */
|
||||
state_ptr->pk[1] = state_ptr->pk[0];
|
||||
state_ptr->pk[0] = pk0;
|
||||
|
||||
/* TONE */
|
||||
if (tr == 1) /* this sample has been treated as data */
|
||||
state_ptr->td = 0; /* next one will be treated as voice */
|
||||
else if (a2p < -11776) /* small sample-to-sample correlation */
|
||||
state_ptr->td = 1; /* signal may be data */
|
||||
else /* signal is voice */
|
||||
state_ptr->td = 0;
|
||||
|
||||
/*
|
||||
* Adaptation speed control.
|
||||
*/
|
||||
state_ptr->dms += (fi - state_ptr->dms) >> 5; /* FILTA */
|
||||
state_ptr->dml += (((fi << 2) - state_ptr->dml) >> 7); /* FILTB */
|
||||
|
||||
if (tr == 1)
|
||||
state_ptr->ap = 256;
|
||||
else if (y < 1536) /* SUBTC */
|
||||
state_ptr->ap += (0x200 - state_ptr->ap) >> 4;
|
||||
else if (state_ptr->td == 1)
|
||||
state_ptr->ap += (0x200 - state_ptr->ap) >> 4;
|
||||
else if (abs((state_ptr->dms << 2) - state_ptr->dml) >=
|
||||
(state_ptr->dml >> 3))
|
||||
state_ptr->ap += (0x200 - state_ptr->ap) >> 4;
|
||||
else
|
||||
state_ptr->ap += (-state_ptr->ap) >> 4;
|
||||
}
|
||||
|
||||
/*
|
||||
* tandem_adjust(sr, se, y, i, sign)
|
||||
*
|
||||
* At the end of ADPCM decoding, it simulates an encoder which may be receiving
|
||||
* the output of this decoder as a tandem process. If the output of the
|
||||
* simulated encoder differs from the input to this decoder, the decoder output
|
||||
* is adjusted by one level of A-law or u-law codes.
|
||||
*
|
||||
* Input:
|
||||
* sr decoder output linear PCM sample,
|
||||
* se predictor estimate sample,
|
||||
* y quantizer step size,
|
||||
* i decoder input code,
|
||||
* sign sign bit of code i
|
||||
*
|
||||
* Return:
|
||||
* adjusted A-law or u-law compressed sample.
|
||||
*/
|
||||
int
|
||||
tandem_adjust_alaw(
|
||||
int sr, /* decoder output linear PCM sample */
|
||||
int se, /* predictor estimate sample */
|
||||
int y, /* quantizer step size */
|
||||
int i, /* decoder input code */
|
||||
int sign,
|
||||
short *qtab)
|
||||
{
|
||||
unsigned char sp; /* A-law compressed 8-bit code */
|
||||
short dx; /* prediction error */
|
||||
char id; /* quantized prediction error */
|
||||
int sd; /* adjusted A-law decoded sample value */
|
||||
int im; /* biased magnitude of i */
|
||||
int imx; /* biased magnitude of id */
|
||||
|
||||
if (sr <= -32768)
|
||||
sr = -1;
|
||||
sp = linear2alaw((sr >> 1) << 3); /* short to A-law compression */
|
||||
dx = (alaw2linear(sp) >> 2) - se; /* 16-bit prediction error */
|
||||
id = quantize(dx, y, qtab, sign - 1);
|
||||
|
||||
if (id == i) { /* no adjustment on sp */
|
||||
return (sp);
|
||||
} else { /* sp adjustment needed */
|
||||
/* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */
|
||||
im = i ^ sign; /* 2's complement to biased unsigned */
|
||||
imx = id ^ sign;
|
||||
|
||||
if (imx > im) { /* sp adjusted to next lower value */
|
||||
if (sp & 0x80) {
|
||||
sd = (sp == 0xD5) ? 0x55 :
|
||||
((sp ^ 0x55) - 1) ^ 0x55;
|
||||
} else {
|
||||
sd = (sp == 0x2A) ? 0x2A :
|
||||
((sp ^ 0x55) + 1) ^ 0x55;
|
||||
}
|
||||
} else { /* sp adjusted to next higher value */
|
||||
if (sp & 0x80)
|
||||
sd = (sp == 0xAA) ? 0xAA :
|
||||
((sp ^ 0x55) + 1) ^ 0x55;
|
||||
else
|
||||
sd = (sp == 0x55) ? 0xD5 :
|
||||
((sp ^ 0x55) - 1) ^ 0x55;
|
||||
}
|
||||
return (sd);
|
||||
}
|
||||
}
|
||||
|
||||
int
|
||||
tandem_adjust_ulaw(
|
||||
int sr, /* decoder output linear PCM sample */
|
||||
int se, /* predictor estimate sample */
|
||||
int y, /* quantizer step size */
|
||||
int i, /* decoder input code */
|
||||
int sign,
|
||||
short *qtab)
|
||||
{
|
||||
unsigned char sp; /* u-law compressed 8-bit code */
|
||||
short dx; /* prediction error */
|
||||
char id; /* quantized prediction error */
|
||||
int sd; /* adjusted u-law decoded sample value */
|
||||
int im; /* biased magnitude of i */
|
||||
int imx; /* biased magnitude of id */
|
||||
|
||||
if (sr <= -32768)
|
||||
sr = 0;
|
||||
sp = linear2ulaw(sr << 2); /* short to u-law compression */
|
||||
dx = (ulaw2linear(sp) >> 2) - se; /* 16-bit prediction error */
|
||||
id = quantize(dx, y, qtab, sign - 1);
|
||||
if (id == i) {
|
||||
return (sp);
|
||||
} else {
|
||||
/* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */
|
||||
im = i ^ sign; /* 2's complement to biased unsigned */
|
||||
imx = id ^ sign;
|
||||
if (imx > im) { /* sp adjusted to next lower value */
|
||||
if (sp & 0x80)
|
||||
sd = (sp == 0xFF) ? 0x7E : sp + 1;
|
||||
else
|
||||
sd = (sp == 0) ? 0 : sp - 1;
|
||||
|
||||
} else { /* sp adjusted to next higher value */
|
||||
if (sp & 0x80)
|
||||
sd = (sp == 0x80) ? 0x80 : sp - 1;
|
||||
else
|
||||
sd = (sp == 0x7F) ? 0xFE : sp + 1;
|
||||
}
|
||||
return (sd);
|
||||
}
|
||||
}
|
||||
|
||||
#endif
|
||||
118
engine/sndcodec/g72x.h
Normal file
118
engine/sndcodec/g72x.h
Normal file
|
|
@ -0,0 +1,118 @@
|
|||
/*
|
||||
* This source code is a product of Sun Microsystems, Inc. and is provided
|
||||
* for unrestricted use. Users may copy or modify this source code without
|
||||
* charge.
|
||||
*
|
||||
* SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
|
||||
* THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
|
||||
* PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
|
||||
*
|
||||
* Sun source code is provided with no support and without any obligation on
|
||||
* the part of Sun Microsystems, Inc. to assist in its use, correction,
|
||||
* modification or enhancement.
|
||||
*
|
||||
* SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
|
||||
* INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
|
||||
* OR ANY PART THEREOF.
|
||||
*
|
||||
* In no event will Sun Microsystems, Inc. be liable for any lost revenue
|
||||
* or profits or other special, indirect and consequential damages, even if
|
||||
* Sun has been advised of the possibility of such damages.
|
||||
*
|
||||
* Sun Microsystems, Inc.
|
||||
* 2550 Garcia Avenue
|
||||
* Mountain View, California 94043
|
||||
*/
|
||||
|
||||
/*
|
||||
* g72x.h
|
||||
*
|
||||
* Header file for CCITT conversion routines.
|
||||
*
|
||||
*/
|
||||
#ifndef _G72X_H
|
||||
#define _G72X_H
|
||||
|
||||
#define AUDIO_ENCODING_ULAW (1) /* ISDN u-law */
|
||||
#define AUDIO_ENCODING_ALAW (2) /* ISDN A-law */
|
||||
#define AUDIO_ENCODING_LINEAR (3) /* PCM 2's-complement (0-center) */
|
||||
|
||||
/*
|
||||
* The following is the definition of the state structure
|
||||
* used by the G.721/G.723 encoder and decoder to preserve their internal
|
||||
* state between successive calls. The meanings of the majority
|
||||
* of the state structure fields are explained in detail in the
|
||||
* CCITT Recommendation G.721. The field names are essentially indentical
|
||||
* to variable names in the bit level description of the coding algorithm
|
||||
* included in this Recommendation.
|
||||
*/
|
||||
struct g72x_state {
|
||||
long yl; /* Locked or steady state step size multiplier. */
|
||||
short yu; /* Unlocked or non-steady state step size multiplier. */
|
||||
short dms; /* Short term energy estimate. */
|
||||
short dml; /* Long term energy estimate. */
|
||||
short ap; /* Linear weighting coefficient of 'yl' and 'yu'. */
|
||||
|
||||
short a[2]; /* Coefficients of pole portion of prediction filter. */
|
||||
short b[6]; /* Coefficients of zero portion of prediction filter. */
|
||||
short pk[2]; /*
|
||||
* Signs of previous two samples of a partially
|
||||
* reconstructed signal.
|
||||
*/
|
||||
short dq[6]; /*
|
||||
* Previous 6 samples of the quantized difference
|
||||
* signal represented in an internal floating point
|
||||
* format.
|
||||
*/
|
||||
short sr[2]; /*
|
||||
* Previous 2 samples of the quantized difference
|
||||
* signal represented in an internal floating point
|
||||
* format.
|
||||
*/
|
||||
char td; /* delayed tone detect, new in 1988 version */
|
||||
};
|
||||
|
||||
/* External function definitions. */
|
||||
|
||||
extern void g72x_init_state(struct g72x_state *);
|
||||
extern int g721_encoder(
|
||||
int sample,
|
||||
int in_coding,
|
||||
struct g72x_state *state_ptr);
|
||||
extern int g721_decoder(
|
||||
int code,
|
||||
int out_coding,
|
||||
struct g72x_state *state_ptr);
|
||||
extern int g723_24_encoder(
|
||||
int sample,
|
||||
int in_coding,
|
||||
struct g72x_state *state_ptr);
|
||||
extern int g723_24_decoder(
|
||||
int code,
|
||||
int out_coding,
|
||||
struct g72x_state *state_ptr);
|
||||
extern int g723_40_encoder(
|
||||
int sample,
|
||||
int in_coding,
|
||||
struct g72x_state *state_ptr);
|
||||
extern int g723_40_decoder(
|
||||
int code,
|
||||
int out_coding,
|
||||
struct g72x_state *state_ptr);
|
||||
|
||||
|
||||
//see the function bodies for a description
|
||||
unsigned char linear2alaw(int pcm_val);
|
||||
unsigned char linear2ulaw(int pcm_val);
|
||||
int ulaw2linear(unsigned char u_val);
|
||||
int alaw2linear(unsigned char a_val);
|
||||
int predictor_zero(struct g72x_state *state_ptr);
|
||||
int predictor_pole(struct g72x_state *state_ptr);
|
||||
int step_size(struct g72x_state *state_ptr);
|
||||
int quantize(int d, int y, short *table, int size);
|
||||
int reconstruct(int sign, int dqln, int y);
|
||||
void update(int code_size, int y, int wi, int fi, int dq, int sr, int dqsez, struct g72x_state *state_ptr);
|
||||
int tandem_adjust_ulaw(int sr, int se, int y, int i, int sign, short *qtab);
|
||||
int tandem_adjust_alaw(int sr, int se, int y, int i, int sign, short *qtab);
|
||||
|
||||
#endif /* !_G72X_H */
|
||||
436
engine/sndcodec/snd_vc.c
Normal file
436
engine/sndcodec/snd_vc.c
Normal file
|
|
@ -0,0 +1,436 @@
|
|||
//voice chat routines.
|
||||
|
||||
//needs quite a bit of work.
|
||||
|
||||
//it needs a proper protocol.
|
||||
//the server needs to be able to shutdown again.
|
||||
//options about who gets the sound data is also needed.
|
||||
|
||||
#include "bothdefs.h"
|
||||
|
||||
#ifdef VOICECHAT
|
||||
|
||||
#include "quakedef.h"
|
||||
#ifdef _WIN32
|
||||
#include "winquake.h"
|
||||
|
||||
#define EWOULDBLOCK WSAEWOULDBLOCK
|
||||
#define EMSGSIZE WSAEMSGSIZE
|
||||
#define ECONNRESET WSAECONNRESET
|
||||
#define ECONNABORTED WSAECONNABORTED
|
||||
#define ECONNREFUSED WSAECONNREFUSED
|
||||
#define EADDRNOTAVAIL WSAEADDRNOTAVAIL
|
||||
|
||||
#define qerrno WSAGetLastError()
|
||||
#else
|
||||
#define qerrno errno
|
||||
|
||||
#include <sys/types.h>
|
||||
#include <sys/socket.h>
|
||||
#include <netinet/in.h>
|
||||
#include <netdb.h>
|
||||
#include <sys/param.h>
|
||||
#include <sys/ioctl.h>
|
||||
#include <sys/uio.h>
|
||||
#include <arpa/inet.h>
|
||||
#include <errno.h>
|
||||
|
||||
#include <unistd.h>
|
||||
|
||||
#ifdef sun
|
||||
#include <sys/filio.h>
|
||||
#endif
|
||||
|
||||
#ifdef NeXT
|
||||
#include <libc.h>
|
||||
#endif
|
||||
|
||||
#define closesocket close
|
||||
#define ioctlsocket ioctl
|
||||
#endif
|
||||
|
||||
#include "voicechat.h"
|
||||
|
||||
static int CLVS_socket;
|
||||
static int SVVS_socket;
|
||||
static qboolean SVVS_inited;
|
||||
|
||||
static qbyte inputbuffer[44100];
|
||||
static int readpoint;
|
||||
static qbyte outputbuffer[44100];
|
||||
static int sendpoint;
|
||||
|
||||
/*
|
||||
Protocol:
|
||||
Sound chunk:
|
||||
(char) data begins with codec id.
|
||||
(short) followed by number of samples
|
||||
(short) then size in bytes of chunk.
|
||||
*/
|
||||
|
||||
#ifndef CLIENTONLY
|
||||
void SVVC_ServerInit(void)
|
||||
{
|
||||
netadr_t adr;
|
||||
struct sockaddr_in address;
|
||||
unsigned long _true = true;
|
||||
int i;
|
||||
int port = PORT_SERVER;
|
||||
|
||||
if ((SVVS_socket = socket (PF_INET, SOCK_STREAM, IPPROTO_TCP)) == -1)
|
||||
{
|
||||
Sys_Error ("FTP_TCP_OpenSocket: socket:", strerror(qerrno));
|
||||
}
|
||||
|
||||
if (ioctlsocket (SVVS_socket, FIONBIO, &_true) == -1)
|
||||
{
|
||||
Sys_Error ("FTP_TCP_OpenSocket: ioctl FIONBIO:", strerror(qerrno));
|
||||
}
|
||||
|
||||
address.sin_family = AF_INET;
|
||||
//ZOID -- check for interface binding option
|
||||
if ((i = COM_CheckParm("-ip")) != 0 && i < com_argc) {
|
||||
address.sin_addr.s_addr = inet_addr(com_argv[i+1]);
|
||||
Con_TPrintf(TL_NETBINDINTERFACE,
|
||||
inet_ntoa(address.sin_addr));
|
||||
} else
|
||||
address.sin_addr.s_addr = INADDR_ANY;
|
||||
|
||||
if (port == PORT_ANY)
|
||||
address.sin_port = 0;
|
||||
else
|
||||
address.sin_port = htons((short)port);
|
||||
|
||||
if( bind (SVVS_socket, (void *)&address, sizeof(address)) == -1)
|
||||
{
|
||||
closesocket(SVVS_socket);
|
||||
return;
|
||||
}
|
||||
|
||||
listen(SVVS_socket, 3);
|
||||
|
||||
SVVS_inited = true;
|
||||
|
||||
|
||||
Con_Printf("VC server is running\n");
|
||||
SockadrToNetadr((struct sockaddr_qstorage*)&address, &adr);
|
||||
Info_SetValueForKey(svs.info, "voiceaddress", NET_AdrToString(adr), MAX_SERVERINFO_STRING);
|
||||
return;
|
||||
}
|
||||
|
||||
//currently a dum forwarding/broadcasting mechanism
|
||||
void SVVC_Frame (qboolean running)
|
||||
{
|
||||
struct sockaddr_in frm;
|
||||
int size = sizeof(frm);
|
||||
int newcl;
|
||||
int i, j;
|
||||
char buffer[1400];
|
||||
|
||||
if (!running)
|
||||
return;
|
||||
|
||||
if (!SVVS_socket)
|
||||
{
|
||||
SVVC_ServerInit();
|
||||
return;
|
||||
}
|
||||
newcl = accept(SVVS_socket, (struct sockaddr *)&frm, &size);
|
||||
if (newcl != INVALID_SOCKET)
|
||||
{
|
||||
for (i = 0; i < MAX_CLIENTS; i++)
|
||||
{
|
||||
if (!svs.clients[i].voicechat.socket) //this really isn't the right way...
|
||||
{
|
||||
svs.clients[i].voicechat.socket = newcl;
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
for (i = 0; i < MAX_CLIENTS; i++)
|
||||
{
|
||||
host_client = &svs.clients[i];
|
||||
|
||||
if (host_client->voicechat.socket)
|
||||
{
|
||||
size = recv(host_client->voicechat.socket, buffer, sizeof(buffer), 0);
|
||||
if (size > 0)
|
||||
{
|
||||
for (j = 0; j < MAX_CLIENTS; j++)
|
||||
{
|
||||
if (j != i && svs.clients[j].voicechat.socket) //gotta be capable of receiving, and not the sender (that would be dumb).
|
||||
send(svs.clients[j].voicechat.socket, buffer, size, 0);
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
#endif
|
||||
|
||||
#ifndef SERVERONLY
|
||||
|
||||
sfxcache_t *voicesoundbuffer[2];
|
||||
sfx_t recordaudio[2] = {
|
||||
{"recordaudio1",
|
||||
{NULL, false},
|
||||
NULL},
|
||||
{"recordaudio2",
|
||||
{NULL, false},
|
||||
NULL}
|
||||
};
|
||||
int audiobuffer;
|
||||
void SNDVC_Submit(int codec, qbyte *buffer, int samples, int freq, int width)
|
||||
{
|
||||
S_RawAudio(0, buffer, freq, samples, 1, width);
|
||||
/*
|
||||
soundcardinfo_t *cursndcard;
|
||||
audiobuffer=0;
|
||||
if (!recordaudio[audiobuffer].cache.data)
|
||||
{
|
||||
voicesoundbuffer[audiobuffer] = BZ_Malloc(44100*2+sizeof(sfxcache_t));
|
||||
recordaudio[audiobuffer].cache.data = voicesoundbuffer[audiobuffer];
|
||||
recordaudio[audiobuffer].cache.fake = true;
|
||||
}
|
||||
|
||||
cursndcard = sndcardinfo;
|
||||
if (!cursndcard)
|
||||
{
|
||||
Con_Printf("Accepting voice chat with no sound card\n");
|
||||
return;
|
||||
}
|
||||
voicesoundbuffer[audiobuffer]->length = samples;
|
||||
voicesoundbuffer[audiobuffer]->stereo = false;
|
||||
voicesoundbuffer[audiobuffer]->speed = sndcardinfo->sn.speed;
|
||||
voicesoundbuffer[audiobuffer]->width = width;
|
||||
voicesoundbuffer[audiobuffer]->loopstart=-1;
|
||||
|
||||
// Con_DPrintf("Submit %i\n", (int)samples);
|
||||
|
||||
if (codec == 0) //codec 0 is special. (A straight copy)
|
||||
ResampleSfx(&recordaudio[audiobuffer], freq, width, buffer);
|
||||
else
|
||||
{
|
||||
qbyte *temp = BZ_Malloc(samples*width);
|
||||
audiocodecs[codec].decode(buffer, (short*)temp, samples);
|
||||
ResampleSfx(&recordaudio[audiobuffer], freq, width, temp);
|
||||
BZ_Free(temp);
|
||||
}
|
||||
|
||||
for (cursndcard = sndcardinfo; cursndcard; cursndcard=cursndcard->next)
|
||||
{
|
||||
if (0&&cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+1-audiobuffer].sfx == &recordaudio[1-audiobuffer]) //other buffer is playing.
|
||||
{
|
||||
cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+audiobuffer].sfx = &recordaudio[audiobuffer];
|
||||
cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+audiobuffer].pos = cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+1-audiobuffer].end - voicesoundbuffer[1-audiobuffer]->length- cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+1-audiobuffer].pos;
|
||||
cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+audiobuffer].end = cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+audiobuffer].end + samples;
|
||||
|
||||
if (cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+audiobuffer].pos >= voicesoundbuffer[audiobuffer]->length)
|
||||
{
|
||||
cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+1-audiobuffer].pos = voicesoundbuffer[1-audiobuffer]->length;
|
||||
cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+audiobuffer].pos = 0;
|
||||
Con_Printf("Sound out of sync\n");
|
||||
}
|
||||
}
|
||||
else
|
||||
{
|
||||
cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+audiobuffer].sfx = &recordaudio[audiobuffer];
|
||||
cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+audiobuffer].pos = 0;
|
||||
cursndcard->channel[MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS+audiobuffer].end = cursndcard->paintedtime + samples;
|
||||
}
|
||||
}
|
||||
audiobuffer = 1-audiobuffer;*/
|
||||
}
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
void CLVC_SetServer (char *addy)
|
||||
{
|
||||
unsigned long _true = true;
|
||||
struct sockaddr_qstorage from;
|
||||
|
||||
|
||||
if ((CLVS_socket = socket (PF_INET, SOCK_STREAM, IPPROTO_TCP)) == -1)
|
||||
{
|
||||
Sys_Error ("FTP_UDP_OpenSocket: socket: %s\n", strerror(qerrno));
|
||||
}
|
||||
|
||||
|
||||
{//quake routines using dns and stuff (Really, I wanna keep quake and ftp fairly seperate)
|
||||
netadr_t qaddy;
|
||||
if (!NET_StringToAdr (addy, &qaddy)) //server doesn't exist.
|
||||
return;
|
||||
if (qaddy.type != NA_IP) //Only TCP is supported.
|
||||
return;
|
||||
if (!qaddy.port)
|
||||
qaddy.port = htons(PORT_SERVER);
|
||||
NetadrToSockadr(&qaddy, &from);
|
||||
|
||||
}
|
||||
|
||||
//not yet non blocking.
|
||||
if (connect(CLVS_socket, (struct sockaddr*)&from, sizeof(from)) == -1)
|
||||
{
|
||||
Con_Printf ("FTP_TCP_OpenSocket: connect: %i %s\n", qerrno, strerror(qerrno));
|
||||
closesocket(CLVS_socket);
|
||||
CLVS_socket = 0;
|
||||
return;
|
||||
}
|
||||
|
||||
if (ioctlsocket (CLVS_socket, FIONBIO, &_true) == -1) //now make it non blocking.
|
||||
{
|
||||
Sys_Error ("FTP_TCP_OpenSocket: ioctl FIONBIO: %s\n", strerror(qerrno));
|
||||
}
|
||||
}
|
||||
|
||||
void CLVC_Disconnect (void)
|
||||
{
|
||||
closesocket(CLVS_socket);
|
||||
CLVS_socket = 0;
|
||||
}
|
||||
|
||||
void CLVC_Poll (void)
|
||||
{
|
||||
int codec;
|
||||
int size;
|
||||
|
||||
if (!CLVS_socket)
|
||||
return;
|
||||
|
||||
while (1)
|
||||
{
|
||||
size = recv(CLVS_socket, &inputbuffer[readpoint], sizeof(inputbuffer) - readpoint, 0);
|
||||
if (size>0)
|
||||
{
|
||||
int samps;
|
||||
int bytes;
|
||||
readpoint+=size;
|
||||
if (readpoint >= 1)
|
||||
{
|
||||
codec = *inputbuffer;
|
||||
|
||||
if (codec >= 0 && codec <= 127 && readpoint>=5) //just in case.
|
||||
{
|
||||
samps = LittleLong(*(signed short *)(inputbuffer+1));
|
||||
bytes = LittleLong(*(unsigned short *)(inputbuffer+3));
|
||||
// Con_DPrintf("read %i\n", size);
|
||||
if (samps < 0) //something special
|
||||
{
|
||||
readpoint=0;
|
||||
}
|
||||
else
|
||||
{
|
||||
if (readpoint >= bytes+5)
|
||||
{
|
||||
if (codec == 1)
|
||||
Con_Printf("Reading\n");
|
||||
if (codec < audionumcodecs && audiocodecs[codec].decode)
|
||||
{
|
||||
SNDVC_Submit(codec, ((qbyte *)inputbuffer)+5, samps, 11025, 2);
|
||||
readpoint -= bytes+5;
|
||||
memmove(inputbuffer, &inputbuffer[readpoint+bytes+5], readpoint);
|
||||
}
|
||||
else
|
||||
{
|
||||
Con_Printf("Bad codec %i\n", (int)codec);
|
||||
readpoint=0;
|
||||
closesocket(CLVS_socket);
|
||||
CLVS_socket = 0;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
else if (readpoint >= sizeof(inputbuffer) || readpoint < 0)
|
||||
{
|
||||
Con_Printf("Too small buffer or extended client %i\n", (int)readpoint);
|
||||
readpoint=0;
|
||||
closesocket(CLVS_socket);
|
||||
CLVS_socket = 0;
|
||||
}
|
||||
else
|
||||
{
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
void SNDVC_MicInput(qbyte *buffer, int samples, int freq, int width) //this correctly buffers data ready to be sent.
|
||||
{
|
||||
int sent;
|
||||
qbyte codec;
|
||||
unsigned short *sampleswritten;
|
||||
samples/=width;
|
||||
if (!CLVS_socket)
|
||||
{
|
||||
if (cls.state)
|
||||
{
|
||||
char *server;
|
||||
server = Info_ValueForKey(cl.serverinfo, "voiceaddress");
|
||||
if (*server)
|
||||
CLVC_SetServer(server);
|
||||
}
|
||||
|
||||
SNDVC_Submit(0, buffer, samples, freq, width);
|
||||
return;
|
||||
}
|
||||
else if (!cls.state)
|
||||
{
|
||||
readpoint = 0;
|
||||
sendpoint= 0;
|
||||
SNDVC_Submit(0, buffer, samples, freq, width);
|
||||
return;
|
||||
}
|
||||
|
||||
SNDVC_Submit(0, buffer, samples, freq, width); //remembering at all times that we will not be allowed to hear it ourselves.
|
||||
|
||||
//add to send buffer.
|
||||
if (samples > 0x7ffe)
|
||||
samples = 0x7ffe;
|
||||
|
||||
if (sendpoint + samples*width+sizeof(unsigned char)+sizeof(short)+sizeof(*sampleswritten) < sizeof(outputbuffer))
|
||||
{
|
||||
// Con_DPrintf("sending %i\n", (int)samples);
|
||||
codec = 1;
|
||||
outputbuffer[sendpoint] = codec; sendpoint += sizeof(unsigned char);
|
||||
*(unsigned short*)(&outputbuffer[sendpoint]) = samples; sendpoint += sizeof(unsigned short);
|
||||
sampleswritten = (short *)&outputbuffer[sendpoint]; sendpoint += sizeof(*sampleswritten);
|
||||
*sampleswritten = audiocodecs[codec].encode((short *)buffer, &outputbuffer[sendpoint], samples);
|
||||
sendpoint += *sampleswritten;
|
||||
}
|
||||
else
|
||||
{
|
||||
Con_Printf("Connection overflowing\n");
|
||||
}
|
||||
|
||||
//try and send it
|
||||
sent = send(CLVS_socket, outputbuffer, sendpoint, 0);
|
||||
if (sent > 0)
|
||||
{
|
||||
// Con_DPrintf("sent %i\n", (int)sent);
|
||||
sendpoint -= sent;
|
||||
}
|
||||
else
|
||||
{
|
||||
CLVS_socket=0;
|
||||
}
|
||||
|
||||
|
||||
// SNDVC_Submit(buffer, samples, freq, width);
|
||||
}
|
||||
|
||||
#endif
|
||||
#endif
|
||||
189
engine/sndcodec/snd_voicecodecs.c
Normal file
189
engine/sndcodec/snd_voicecodecs.c
Normal file
|
|
@ -0,0 +1,189 @@
|
|||
#include "bothdefs.h"
|
||||
|
||||
#ifdef VOICECHAT
|
||||
|
||||
#include "quakedef.h"
|
||||
#include "voicechat.h"
|
||||
|
||||
|
||||
|
||||
#include "g72x.h"
|
||||
|
||||
int VC_g72x_encoder (short *in, unsigned char *out, int samples, int (*g72x_encoder)(int,int,struct g72x_state *state), int bits)
|
||||
{
|
||||
struct g72x_state state;
|
||||
unsigned int out_buffer = 0;
|
||||
int out_bits = 0;
|
||||
unsigned char out_byte;
|
||||
int code;
|
||||
int written=0;
|
||||
|
||||
g72x_init_state(&state);
|
||||
|
||||
while (samples)
|
||||
{
|
||||
code = g72x_encoder(*in, AUDIO_ENCODING_LINEAR, &state);
|
||||
|
||||
out_buffer |= (code << out_bits);
|
||||
out_bits += bits;
|
||||
if (out_bits >= 8)
|
||||
{
|
||||
out_byte = out_buffer & 0xff;
|
||||
out_bits -= 8;
|
||||
out_buffer >>= 8;
|
||||
*out = out_byte;
|
||||
out++;
|
||||
written++;
|
||||
}
|
||||
in++;
|
||||
|
||||
samples--;
|
||||
}
|
||||
code=0;
|
||||
while (out_bits > 0)
|
||||
{
|
||||
out_buffer |= (code << out_bits);
|
||||
out_bits += bits;
|
||||
if (out_bits >= 8)
|
||||
{
|
||||
out_byte = out_buffer & 0xff;
|
||||
out_bits -= 8;
|
||||
out_buffer >>= 8;
|
||||
*out = out_byte;
|
||||
out++;
|
||||
written++;
|
||||
}
|
||||
}
|
||||
|
||||
return written;
|
||||
}
|
||||
|
||||
int VC_g723_24_encoder (short *in, unsigned char *out, int samples)
|
||||
{
|
||||
return VC_g72x_encoder(in, out, samples, g723_24_encoder, 3);
|
||||
}
|
||||
|
||||
int VC_g721_encoder (short *in, unsigned char *out, int samples)
|
||||
{
|
||||
return VC_g72x_encoder(in, out, samples, g721_encoder, 4);
|
||||
}
|
||||
|
||||
int VC_g723_40_encoder (short *in, unsigned char *out, int samples)
|
||||
{
|
||||
return VC_g72x_encoder(in, out, samples, g723_40_encoder, 5);
|
||||
}
|
||||
|
||||
int VC_g72x_decoder (unsigned char *in, short *out, int samples, int (*g72x_decoder)(int,int,struct g72x_state *state), int bits)
|
||||
{
|
||||
struct g72x_state state;
|
||||
unsigned int in_buffer = 0;
|
||||
int in_bits = 0;
|
||||
unsigned char in_byte;
|
||||
int code;
|
||||
int read=0;
|
||||
|
||||
g72x_init_state(&state);
|
||||
|
||||
while (samples)
|
||||
{
|
||||
if (in_bits < bits)
|
||||
{
|
||||
in_byte = *in++;
|
||||
read++;
|
||||
in_buffer |= (in_byte << in_bits);
|
||||
in_bits += 8;
|
||||
}
|
||||
code = in_buffer & ((1 << bits) - 1);
|
||||
in_buffer >>= bits;
|
||||
in_bits -= bits;
|
||||
|
||||
*out = g72x_decoder(code, AUDIO_ENCODING_LINEAR, &state);
|
||||
out++;
|
||||
|
||||
samples--;
|
||||
}
|
||||
|
||||
return read;
|
||||
}
|
||||
|
||||
int VC_g723_24_decoder (unsigned char *in, short *out, int samples)
|
||||
{
|
||||
return VC_g72x_decoder(in, out, samples, g723_24_decoder, 3);
|
||||
}
|
||||
|
||||
int VC_g721_decoder (unsigned char *in, short *out, int samples)
|
||||
{
|
||||
return VC_g72x_decoder(in, out, samples, g721_decoder, 4);
|
||||
}
|
||||
|
||||
int VC_g723_40_decoder (unsigned char *in, short *out, int samples)
|
||||
{
|
||||
return VC_g72x_decoder(in, out, samples, g723_40_decoder, 5);
|
||||
}
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
int VS_Raw_enc (short *in, unsigned char *out, int samples)
|
||||
{
|
||||
memcpy(out, in, samples*2);
|
||||
return samples*2;
|
||||
}
|
||||
|
||||
int VS_Raw_dec (unsigned char *in, short *out, int samples)
|
||||
{
|
||||
memcpy(out, in, samples*2);
|
||||
return samples*2;
|
||||
}
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
audiocodec_t audiocodecs[] = {
|
||||
{"Raw 11025sps 16bit", VS_Raw_enc, VS_Raw_dec},
|
||||
#ifdef _G72X_H
|
||||
{"G.723.24", VC_g723_24_encoder, VC_g723_24_decoder},
|
||||
{"G.721 32", VC_g721_encoder, VC_g721_decoder},
|
||||
{"G.723.40", VC_g723_40_encoder, VC_g723_40_decoder},
|
||||
#else
|
||||
{"Non-implemented codec (G.723.24)"},
|
||||
{"Non-implemented codec (G.721 32)"},
|
||||
{"Non-implemented codec (G.723.40)"},
|
||||
#endif
|
||||
{0}
|
||||
};
|
||||
|
||||
const int audionumcodecs = sizeof(audiocodecs)/sizeof(audiocodec_t);
|
||||
#endif
|
||||
9
engine/sndcodec/voicechat.h
Normal file
9
engine/sndcodec/voicechat.h
Normal file
|
|
@ -0,0 +1,9 @@
|
|||
//raw data is 16 bps
|
||||
typedef struct {
|
||||
char *name;
|
||||
int (*encode) (short *in, unsigned char *out, int numsamps); //returns number of bytes.
|
||||
int (*decode) (unsigned char *in, short *out, int numsamps); //returns number of 16bps samples.
|
||||
} audiocodec_t;
|
||||
|
||||
extern audiocodec_t audiocodecs[];
|
||||
extern const int audionumcodecs;
|
||||
Loading…
Reference in a new issue