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Added G.722 codec from the WebRTC code. 

git-svn-id: http://yate.null.ro/svn/yate/trunk@6363 acf43c95-373e-0410-b603-e72c3f656dc1
This commit is contained in:
paulc 2019-11-06 18:31:54 +00:00
parent 2066fb6e37
commit 828ea4ddfa
14 changed files with 1631 additions and 3 deletions
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17
configure.ac
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@ -1082,6 +1082,23 @@ fi
AC_SUBST(HAVE_ISAC)
AC_SUBST(ISAC_INC)
HAVE_G722WR=no
G722WR_INC=""
AC_ARG_ENABLE(g722-webrtc,AC_HELP_STRING([--enable-g722-webrtc],[Enable G.722 webrtc codec (default: yes)]),want_g722_webrtc=$enableval,want_g722_webrtc=yes)
if [[ "x$want_g722_webrtc" = "xyes" ]]; then
AC_MSG_CHECKING([for G.722 miniwebrtc in libs])
basedir=`cd "$srcdir" && pwd`
incg722="libs/miniwebrtc/audio/coding_g722"
if [[ -f "$basedir/$incg722/g722_interface.h" ]]; then
HAVE_G722WR=yes
basedir=`echo "$basedir" | sed 's/\([[^\]]\)\([[[:space:]\$\"'\'']]\)/\1\\\2/g'`
G722WR_INC="-I$basedir/$incg722"
fi
AC_MSG_RESULT([$HAVE_G722WR])
fi
AC_SUBST(HAVE_G722WR)
AC_SUBST(G722WR_INC)
HAVE_SPEEX=no
SPEEX_INC=""
SPEEX_LIB="-lspeex"

3
libs/miniwebrtc/Makefile.in
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@ -58,6 +58,9 @@ OBJS := audio/common/resampler/resampler.o \
audio/common/vad/vad_gmm.o \
audio/common/vad/vad_sp.o \
audio/common/vad/webrtc_vad.o \
audio/coding_g722/g722_decode.o \
audio/coding_g722/g722_encode.o \
audio/coding_g722/g722_interface.o \
audio/coding_isac/fix/arith_routines.o \
audio/coding_isac/fix/arith_routines_hist.o \
audio/coding_isac/fix/arith_routines_logist.o \

408
libs/miniwebrtc/audio/coding_g722/g722_decode.c Normal file
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@ -0,0 +1,408 @@
/*
* SpanDSP - a series of DSP components for telephony
*
* g722_decode.c - The ITU G.722 codec, decode part.
*
* Written by Steve Underwood <steveu@coppice.org>
*
* Copyright (C) 2005 Steve Underwood
*
* Despite my general liking of the GPL, I place my own contributions
* to this code in the public domain for the benefit of all mankind -
* even the slimy ones who might try to proprietize my work and use it
* to my detriment.
*
* Based in part on a single channel G.722 codec which is:
*
* Copyright (c) CMU 1993
* Computer Science, Speech Group
* Chengxiang Lu and Alex Hauptmann
*
* $Id: g722_decode.c,v 1.15 2006/07/07 16:37:49 steveu Exp $
*
* Modifications for WebRtc, 2011/04/28, by tlegrand:
* -Removed usage of inttypes.h and tgmath.h
* -Changed to use WebRtc types
* -Changed __inline__ to __inline
* -Added saturation check on output
*/
/*! \file */
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <memory.h>
#include <stdio.h>
#include <stdlib.h>
#include "g722_enc_dec.h"
#include "typedefs.h"
#if !defined(FALSE)
#define FALSE 0
#endif
#if !defined(TRUE)
#define TRUE (!FALSE)
#endif
static __inline int16_t saturate(int32_t amp)
{
int16_t amp16;
/* Hopefully this is optimised for the common case - not clipping */
amp16 = (int16_t) amp;
if (amp == amp16)
return amp16;
if (amp > WEBRTC_INT16_MAX)
return WEBRTC_INT16_MAX;
return WEBRTC_INT16_MIN;
}
/*- End of function --------------------------------------------------------*/
static void block4(G722DecoderState *s, int band, int d);
static void block4(G722DecoderState *s, int band, int d)
{
int wd1;
int wd2;
int wd3;
int i;
/* Block 4, RECONS */
s->band[band].d[0] = d;
s->band[band].r[0] = saturate(s->band[band].s + d);
/* Block 4, PARREC */
s->band[band].p[0] = saturate(s->band[band].sz + d);
/* Block 4, UPPOL2 */
for (i = 0; i < 3; i++)
s->band[band].sg[i] = s->band[band].p[i] >> 15;
wd1 = saturate(s->band[band].a[1] << 2);
wd2 = (s->band[band].sg[0] == s->band[band].sg[1]) ? -wd1 : wd1;
if (wd2 > 32767)
wd2 = 32767;
wd3 = (s->band[band].sg[0] == s->band[band].sg[2]) ? 128 : -128;
wd3 += (wd2 >> 7);
wd3 += (s->band[band].a[2]*32512) >> 15;
if (wd3 > 12288)
wd3 = 12288;
else if (wd3 < -12288)
wd3 = -12288;
s->band[band].ap[2] = wd3;
/* Block 4, UPPOL1 */
s->band[band].sg[0] = s->band[band].p[0] >> 15;
s->band[band].sg[1] = s->band[band].p[1] >> 15;
wd1 = (s->band[band].sg[0] == s->band[band].sg[1]) ? 192 : -192;
wd2 = (s->band[band].a[1]*32640) >> 15;
s->band[band].ap[1] = saturate(wd1 + wd2);
wd3 = saturate(15360 - s->band[band].ap[2]);
if (s->band[band].ap[1] > wd3)
s->band[band].ap[1] = wd3;
else if (s->band[band].ap[1] < -wd3)
s->band[band].ap[1] = -wd3;
/* Block 4, UPZERO */
wd1 = (d == 0) ? 0 : 128;
s->band[band].sg[0] = d >> 15;
for (i = 1; i < 7; i++)
{
s->band[band].sg[i] = s->band[band].d[i] >> 15;
wd2 = (s->band[band].sg[i] == s->band[band].sg[0]) ? wd1 : -wd1;
wd3 = (s->band[band].b[i]*32640) >> 15;
s->band[band].bp[i] = saturate(wd2 + wd3);
}
/* Block 4, DELAYA */
for (i = 6; i > 0; i--)
{
s->band[band].d[i] = s->band[band].d[i - 1];
s->band[band].b[i] = s->band[band].bp[i];
}
for (i = 2; i > 0; i--)
{
s->band[band].r[i] = s->band[band].r[i - 1];
s->band[band].p[i] = s->band[band].p[i - 1];
s->band[band].a[i] = s->band[band].ap[i];
}
/* Block 4, FILTEP */
wd1 = saturate(s->band[band].r[1] + s->band[band].r[1]);
wd1 = (s->band[band].a[1]*wd1) >> 15;
wd2 = saturate(s->band[band].r[2] + s->band[band].r[2]);
wd2 = (s->band[band].a[2]*wd2) >> 15;
s->band[band].sp = saturate(wd1 + wd2);
/* Block 4, FILTEZ */
s->band[band].sz = 0;
for (i = 6; i > 0; i--)
{
wd1 = saturate(s->band[band].d[i] + s->band[band].d[i]);
s->band[band].sz += (s->band[band].b[i]*wd1) >> 15;
}
s->band[band].sz = saturate(s->band[band].sz);
/* Block 4, PREDIC */
s->band[band].s = saturate(s->band[band].sp + s->band[band].sz);
}
/*- End of function --------------------------------------------------------*/
G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
int rate,
int options) {
if (s == NULL)
{
if ((s = (G722DecoderState *) malloc(sizeof(*s))) == NULL)
return NULL;
}
memset(s, 0, sizeof(*s));
if (rate == 48000)
s->bits_per_sample = 6;
else if (rate == 56000)
s->bits_per_sample = 7;
else
s->bits_per_sample = 8;
if ((options & G722_SAMPLE_RATE_8000))
s->eight_k = TRUE;
if ((options & G722_PACKED) && s->bits_per_sample != 8)
s->packed = TRUE;
else
s->packed = FALSE;
s->band[0].det = 32;
s->band[1].det = 8;
return s;
}
/*- End of function --------------------------------------------------------*/
int WebRtc_g722_decode_release(G722DecoderState *s)
{
free(s);
return 0;
}
/*- End of function --------------------------------------------------------*/
int WebRtc_g722_decode(G722DecoderState *s, int16_t amp[],
const uint8_t g722_data[], int len)
{
static const int wl[8] = {-60, -30, 58, 172, 334, 538, 1198, 3042 };
static const int rl42[16] = {0, 7, 6, 5, 4, 3, 2, 1,
7, 6, 5, 4, 3, 2, 1, 0 };
static const int ilb[32] =
{
2048, 2093, 2139, 2186, 2233, 2282, 2332,
2383, 2435, 2489, 2543, 2599, 2656, 2714,
2774, 2834, 2896, 2960, 3025, 3091, 3158,
3228, 3298, 3371, 3444, 3520, 3597, 3676,
3756, 3838, 3922, 4008
};
static const int wh[3] = {0, -214, 798};
static const int rh2[4] = {2, 1, 2, 1};
static const int qm2[4] = {-7408, -1616, 7408, 1616};
static const int qm4[16] =
{
0, -20456, -12896, -8968,
-6288, -4240, -2584, -1200,
20456, 12896, 8968, 6288,
4240, 2584, 1200, 0
};
static const int qm5[32] =
{
-280, -280, -23352, -17560,
-14120, -11664, -9752, -8184,
-6864, -5712, -4696, -3784,
-2960, -2208, -1520, -880,
23352, 17560, 14120, 11664,
9752, 8184, 6864, 5712,
4696, 3784, 2960, 2208,
1520, 880, 280, -280
};
static const int qm6[64] =
{
-136, -136, -136, -136,
-24808, -21904, -19008, -16704,
-14984, -13512, -12280, -11192,
-10232, -9360, -8576, -7856,
-7192, -6576, -6000, -5456,
-4944, -4464, -4008, -3576,
-3168, -2776, -2400, -2032,
-1688, -1360, -1040, -728,
24808, 21904, 19008, 16704,
14984, 13512, 12280, 11192,
10232, 9360, 8576, 7856,
7192, 6576, 6000, 5456,
4944, 4464, 4008, 3576,
3168, 2776, 2400, 2032,
1688, 1360, 1040, 728,
432, 136, -432, -136
};
static const int qmf_coeffs[12] =
{
3, -11, 12, 32, -210, 951, 3876, -805, 362, -156, 53, -11,
};
int dlowt;
int rlow;
int ihigh;
int dhigh;
int rhigh;
int xout1;
int xout2;
int wd1;
int wd2;
int wd3;
int code;
int outlen;
int i;
int j;
outlen = 0;
rhigh = 0;
for (j = 0; j < len; )
{
if (s->packed)
{
/* Unpack the code bits */
if (s->in_bits < s->bits_per_sample)
{
s->in_buffer |= (g722_data[j++] << s->in_bits);
s->in_bits += 8;
}
code = s->in_buffer & ((1 << s->bits_per_sample) - 1);
s->in_buffer >>= s->bits_per_sample;
s->in_bits -= s->bits_per_sample;
}
else
{
code = g722_data[j++];
}
switch (s->bits_per_sample)
{
default:
case 8:
wd1 = code & 0x3F;
ihigh = (code >> 6) & 0x03;
wd2 = qm6[wd1];
wd1 >>= 2;
break;
case 7:
wd1 = code & 0x1F;
ihigh = (code >> 5) & 0x03;
wd2 = qm5[wd1];
wd1 >>= 1;
break;
case 6:
wd1 = code & 0x0F;
ihigh = (code >> 4) & 0x03;
wd2 = qm4[wd1];
break;
}
/* Block 5L, LOW BAND INVQBL */
wd2 = (s->band[0].det*wd2) >> 15;
/* Block 5L, RECONS */
rlow = s->band[0].s + wd2;
/* Block 6L, LIMIT */
if (rlow > 16383)
rlow = 16383;
else if (rlow < -16384)
rlow = -16384;
/* Block 2L, INVQAL */
wd2 = qm4[wd1];
dlowt = (s->band[0].det*wd2) >> 15;
/* Block 3L, LOGSCL */
wd2 = rl42[wd1];
wd1 = (s->band[0].nb*127) >> 7;
wd1 += wl[wd2];
if (wd1 < 0)
wd1 = 0;
else if (wd1 > 18432)
wd1 = 18432;
s->band[0].nb = wd1;
/* Block 3L, SCALEL */
wd1 = (s->band[0].nb >> 6) & 31;
wd2 = 8 - (s->band[0].nb >> 11);
wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
s->band[0].det = wd3 << 2;
block4(s, 0, dlowt);
if (!s->eight_k)
{
/* Block 2H, INVQAH */
wd2 = qm2[ihigh];
dhigh = (s->band[1].det*wd2) >> 15;
/* Block 5H, RECONS */
rhigh = dhigh + s->band[1].s;
/* Block 6H, LIMIT */
if (rhigh > 16383)
rhigh = 16383;
else if (rhigh < -16384)
rhigh = -16384;
/* Block 2H, INVQAH */
wd2 = rh2[ihigh];
wd1 = (s->band[1].nb*127) >> 7;
wd1 += wh[wd2];
if (wd1 < 0)
wd1 = 0;
else if (wd1 > 22528)
wd1 = 22528;
s->band[1].nb = wd1;
/* Block 3H, SCALEH */
wd1 = (s->band[1].nb >> 6) & 31;
wd2 = 10 - (s->band[1].nb >> 11);
wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
s->band[1].det = wd3 << 2;
block4(s, 1, dhigh);
}
if (s->itu_test_mode)
{
amp[outlen++] = (int16_t) (rlow << 1);
amp[outlen++] = (int16_t) (rhigh << 1);
}
else
{
if (s->eight_k)
{
amp[outlen++] = (int16_t) (rlow << 1);
}
else
{
/* Apply the receive QMF */
for (i = 0; i < 22; i++)
s->x[i] = s->x[i + 2];
s->x[22] = rlow + rhigh;
s->x[23] = rlow - rhigh;
xout1 = 0;
xout2 = 0;
for (i = 0; i < 12; i++)
{
xout2 += s->x[2*i]*qmf_coeffs[i];
xout1 += s->x[2*i + 1]*qmf_coeffs[11 - i];
}
/* We shift by 12 to allow for the QMF filters (DC gain = 4096), less 1
to allow for the 15 bit input to the G.722 algorithm. */
/* WebRtc, tlegrand: added saturation */
amp[outlen++] = saturate(xout1 >> 11);
amp[outlen++] = saturate(xout2 >> 11);
}
}
}
return outlen;
}
/*- End of function --------------------------------------------------------*/
/*- End of file ------------------------------------------------------------*/

160
libs/miniwebrtc/audio/coding_g722/g722_enc_dec.h Normal file
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@ -0,0 +1,160 @@
/*
* SpanDSP - a series of DSP components for telephony
*
* g722.h - The ITU G.722 codec.
*
* Written by Steve Underwood <steveu@coppice.org>
*
* Copyright (C) 2005 Steve Underwood
*
* Despite my general liking of the GPL, I place my own contributions
* to this code in the public domain for the benefit of all mankind -
* even the slimy ones who might try to proprietize my work and use it
* to my detriment.
*
* Based on a single channel G.722 codec which is:
*
***** Copyright (c) CMU 1993 *****
* Computer Science, Speech Group
* Chengxiang Lu and Alex Hauptmann
*
* $Id$
*
* Modifications for WebRtc, 2011/04/28, by tlegrand:
* -Changed to use WebRtc types
* -Added new defines for minimum and maximum values of short int
*/
/*! \file */
#if !defined(_G722_ENC_DEC_H_)
#define _G722_ENC_DEC_H_
#include "typedefs.h"
/*! \page g722_page G.722 encoding and decoding
\section g722_page_sec_1 What does it do?
The G.722 module is a bit exact implementation of the ITU G.722 specification for all three
specified bit rates - 64000bps, 56000bps and 48000bps. It passes the ITU tests.
To allow fast and flexible interworking with narrow band telephony, the encoder and decoder
support an option for the linear audio to be an 8k samples/second stream. In this mode the
codec is considerably faster, and still fully compatible with wideband terminals using G.722.
\section g722_page_sec_2 How does it work?
???.
*/
#define WEBRTC_INT16_MAX 32767
#define WEBRTC_INT16_MIN -32768
enum
{
G722_SAMPLE_RATE_8000 = 0x0001,
G722_PACKED = 0x0002
};
typedef struct
{
/*! TRUE if the operating in the special ITU test mode, with the band split filters
disabled. */
int itu_test_mode;
/*! TRUE if the G.722 data is packed */
int packed;
/*! TRUE if encode from 8k samples/second */
int eight_k;
/*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
int bits_per_sample;
/*! Signal history for the QMF */
int x[24];
struct
{
int s;
int sp;
int sz;
int r[3];
int a[3];
int ap[3];
int p[3];
int d[7];
int b[7];
int bp[7];
int sg[7];
int nb;
int det;
} band[2];
unsigned int in_buffer;
int in_bits;
unsigned int out_buffer;
int out_bits;
} G722EncoderState;
typedef struct
{
/*! TRUE if the operating in the special ITU test mode, with the band split filters
disabled. */
int itu_test_mode;
/*! TRUE if the G.722 data is packed */
int packed;
/*! TRUE if decode to 8k samples/second */
int eight_k;
/*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
int bits_per_sample;
/*! Signal history for the QMF */
int x[24];
struct
{
int s;
int sp;
int sz;
int r[3];
int a[3];
int ap[3];
int p[3];
int d[7];
int b[7];
int bp[7];
int sg[7];
int nb;
int det;
} band[2];
unsigned int in_buffer;
int in_bits;
unsigned int out_buffer;
int out_bits;
} G722DecoderState;
#ifdef __cplusplus
extern "C" {
#endif
G722EncoderState* WebRtc_g722_encode_init(G722EncoderState* s,
int rate,
int options);
int WebRtc_g722_encode_release(G722EncoderState *s);
int WebRtc_g722_encode(G722EncoderState *s,
uint8_t g722_data[],
const int16_t amp[],
int len);
G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
int rate,
int options);
int WebRtc_g722_decode_release(G722DecoderState *s);
int WebRtc_g722_decode(G722DecoderState *s,
int16_t amp[],
const uint8_t g722_data[],
int len);
#ifdef __cplusplus
}
#endif
#endif

434
libs/miniwebrtc/audio/coding_g722/g722_encode.c Normal file
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@ -0,0 +1,434 @@
/*
* SpanDSP - a series of DSP components for telephony
*
* g722_encode.c - The ITU G.722 codec, encode part.
*
* Written by Steve Underwood <steveu@coppice.org>
*
* Copyright (C) 2005 Steve Underwood
*
* All rights reserved.
*
* Despite my general liking of the GPL, I place my own contributions
* to this code in the public domain for the benefit of all mankind -
* even the slimy ones who might try to proprietize my work and use it
* to my detriment.
*
* Based on a single channel 64kbps only G.722 codec which is:
*
***** Copyright (c) CMU 1993 *****
* Computer Science, Speech Group
* Chengxiang Lu and Alex Hauptmann
*
* $Id: g722_encode.c,v 1.14 2006/07/07 16:37:49 steveu Exp $
*
* Modifications for WebRtc, 2011/04/28, by tlegrand:
* -Removed usage of inttypes.h and tgmath.h
* -Changed to use WebRtc types
* -Added option to run encoder bitexact with ITU-T reference implementation
*/
/*! \file */
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <memory.h>
#include <stdio.h>
#include <stdlib.h>
#include "g722_enc_dec.h"
#include "typedefs.h"
#if !defined(FALSE)
#define FALSE 0
#endif
#if !defined(TRUE)
#define TRUE (!FALSE)
#endif
static __inline int16_t saturate(int32_t amp)
{
int16_t amp16;
/* Hopefully this is optimised for the common case - not clipping */
amp16 = (int16_t) amp;
if (amp == amp16)
return amp16;
if (amp > WEBRTC_INT16_MAX)
return WEBRTC_INT16_MAX;
return WEBRTC_INT16_MIN;
}
/*- End of function --------------------------------------------------------*/
static void block4(G722EncoderState *s, int band, int d)
{
int wd1;
int wd2;
int wd3;
int i;
/* Block 4, RECONS */
s->band[band].d[0] = d;
s->band[band].r[0] = saturate(s->band[band].s + d);
/* Block 4, PARREC */
s->band[band].p[0] = saturate(s->band[band].sz + d);
/* Block 4, UPPOL2 */
for (i = 0; i < 3; i++)
s->band[band].sg[i] = s->band[band].p[i] >> 15;
wd1 = saturate(s->band[band].a[1] << 2);
wd2 = (s->band[band].sg[0] == s->band[band].sg[1]) ? -wd1 : wd1;
if (wd2 > 32767)
wd2 = 32767;
wd3 = (wd2 >> 7) + ((s->band[band].sg[0] == s->band[band].sg[2]) ? 128 : -128);
wd3 += (s->band[band].a[2]*32512) >> 15;
if (wd3 > 12288)
wd3 = 12288;
else if (wd3 < -12288)
wd3 = -12288;
s->band[band].ap[2] = wd3;
/* Block 4, UPPOL1 */
s->band[band].sg[0] = s->band[band].p[0] >> 15;
s->band[band].sg[1] = s->band[band].p[1] >> 15;
wd1 = (s->band[band].sg[0] == s->band[band].sg[1]) ? 192 : -192;
wd2 = (s->band[band].a[1]*32640) >> 15;
s->band[band].ap[1] = saturate(wd1 + wd2);
wd3 = saturate(15360 - s->band[band].ap[2]);
if (s->band[band].ap[1] > wd3)
s->band[band].ap[1] = wd3;
else if (s->band[band].ap[1] < -wd3)
s->band[band].ap[1] = -wd3;
/* Block 4, UPZERO */
wd1 = (d == 0) ? 0 : 128;
s->band[band].sg[0] = d >> 15;
for (i = 1; i < 7; i++)
{
s->band[band].sg[i] = s->band[band].d[i] >> 15;
wd2 = (s->band[band].sg[i] == s->band[band].sg[0]) ? wd1 : -wd1;
wd3 = (s->band[band].b[i]*32640) >> 15;
s->band[band].bp[i] = saturate(wd2 + wd3);
}
/* Block 4, DELAYA */
for (i = 6; i > 0; i--)
{
s->band[band].d[i] = s->band[band].d[i - 1];
s->band[band].b[i] = s->band[band].bp[i];
}
for (i = 2; i > 0; i--)
{
s->band[band].r[i] = s->band[band].r[i - 1];
s->band[band].p[i] = s->band[band].p[i - 1];
s->band[band].a[i] = s->band[band].ap[i];
}
/* Block 4, FILTEP */
wd1 = saturate(s->band[band].r[1] + s->band[band].r[1]);
wd1 = (s->band[band].a[1]*wd1) >> 15;
wd2 = saturate(s->band[band].r[2] + s->band[band].r[2]);
wd2 = (s->band[band].a[2]*wd2) >> 15;
s->band[band].sp = saturate(wd1 + wd2);
/* Block 4, FILTEZ */
s->band[band].sz = 0;
for (i = 6; i > 0; i--)
{
wd1 = saturate(s->band[band].d[i] + s->band[band].d[i]);
s->band[band].sz += (s->band[band].b[i]*wd1) >> 15;
}
s->band[band].sz = saturate(s->band[band].sz);
/* Block 4, PREDIC */
s->band[band].s = saturate(s->band[band].sp + s->band[band].sz);
}
/*- End of function --------------------------------------------------------*/
G722EncoderState* WebRtc_g722_encode_init(G722EncoderState* s,
int rate,
int options) {
if (s == NULL)
{
if ((s = (G722EncoderState *) malloc(sizeof(*s))) == NULL)
return NULL;
}
memset(s, 0, sizeof(*s));
if (rate == 48000)
s->bits_per_sample = 6;
else if (rate == 56000)
s->bits_per_sample = 7;
else
s->bits_per_sample = 8;
if ((options & G722_SAMPLE_RATE_8000))
s->eight_k = TRUE;
if ((options & G722_PACKED) && s->bits_per_sample != 8)
s->packed = TRUE;
else
s->packed = FALSE;
s->band[0].det = 32;
s->band[1].det = 8;
return s;
}
/*- End of function --------------------------------------------------------*/
int WebRtc_g722_encode_release(G722EncoderState *s)
{
free(s);
return 0;
}
/*- End of function --------------------------------------------------------*/
/* WebRtc, tlegrand:
* Only define the following if bit-exactness with reference implementation
* is needed. Will only have any effect if input signal is saturated.
*/
//#define RUN_LIKE_REFERENCE_G722
#ifdef RUN_LIKE_REFERENCE_G722
int16_t limitValues (int16_t rl)
{
int16_t yl;
yl = (rl > 16383) ? 16383 : ((rl < -16384) ? -16384 : rl);
return (yl);
}
#endif
int WebRtc_g722_encode(G722EncoderState *s, uint8_t g722_data[],
const int16_t amp[], int len)
{
static const int q6[32] =
{
0, 35, 72, 110, 150, 190, 233, 276,
323, 370, 422, 473, 530, 587, 650, 714,
786, 858, 940, 1023, 1121, 1219, 1339, 1458,
1612, 1765, 1980, 2195, 2557, 2919, 0, 0
};
static const int iln[32] =
{
0, 63, 62, 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, 0
};
static const int ilp[32] =
{
0, 61, 60, 59, 58, 57, 56, 55,
54, 53, 52, 51, 50, 49, 48, 47,
46, 45, 44, 43, 42, 41, 40, 39,
38, 37, 36, 35, 34, 33, 32, 0
};
static const int wl[8] =
{
-60, -30, 58, 172, 334, 538, 1198, 3042
};
static const int rl42[16] =
{
0, 7, 6, 5, 4, 3, 2, 1, 7, 6, 5, 4, 3, 2, 1, 0
};
static const int ilb[32] =
{
2048, 2093, 2139, 2186, 2233, 2282, 2332,
2383, 2435, 2489, 2543, 2599, 2656, 2714,
2774, 2834, 2896, 2960, 3025, 3091, 3158,
3228, 3298, 3371, 3444, 3520, 3597, 3676,
3756, 3838, 3922, 4008
};
static const int qm4[16] =
{
0, -20456, -12896, -8968,
-6288, -4240, -2584, -1200,
20456, 12896, 8968, 6288,
4240, 2584, 1200, 0
};
static const int qm2[4] =
{
-7408, -1616, 7408, 1616
};
static const int qmf_coeffs[12] =
{
3, -11, 12, 32, -210, 951, 3876, -805, 362, -156, 53, -11,
};
static const int ihn[3] = {0, 1, 0};
static const int ihp[3] = {0, 3, 2};
static const int wh[3] = {0, -214, 798};
static const int rh2[4] = {2, 1, 2, 1};
int dlow;
int dhigh;
int el;
int wd;
int wd1;
int ril;
int wd2;
int il4;
int ih2;
int wd3;
int eh;
int mih;
int i;
int j;
/* Low and high band PCM from the QMF */
int xlow;
int xhigh;
int g722_bytes;
/* Even and odd tap accumulators */
int sumeven;
int sumodd;
int ihigh;
int ilow;
int code;
g722_bytes = 0;
xhigh = 0;
for (j = 0; j < len; )
{
if (s->itu_test_mode)
{
xlow =
xhigh = amp[j++] >> 1;
}
else
{
if (s->eight_k)
{
/* We shift by 1 to allow for the 15 bit input to the G.722 algorithm. */
xlow = amp[j++] >> 1;
}
else
{
/* Apply the transmit QMF */
/* Shuffle the buffer down */
for (i = 0; i < 22; i++)
s->x[i] = s->x[i + 2];
s->x[22] = amp[j++];
s->x[23] = amp[j++];
/* Discard every other QMF output */
sumeven = 0;
sumodd = 0;
for (i = 0; i < 12; i++)
{
sumodd += s->x[2*i]*qmf_coeffs[i];
sumeven += s->x[2*i + 1]*qmf_coeffs[11 - i];
}
/* We shift by 12 to allow for the QMF filters (DC gain = 4096), plus 1
to allow for us summing two filters, plus 1 to allow for the 15 bit
input to the G.722 algorithm. */
xlow = (sumeven + sumodd) >> 14;
xhigh = (sumeven - sumodd) >> 14;
#ifdef RUN_LIKE_REFERENCE_G722
/* The following lines are only used to verify bit-exactness
* with reference implementation of G.722. Higher precision
* is achieved without limiting the values.
*/
xlow = limitValues(xlow);
xhigh = limitValues(xhigh);
#endif
}
}
/* Block 1L, SUBTRA */
el = saturate(xlow - s->band[0].s);
/* Block 1L, QUANTL */
wd = (el >= 0) ? el : -(el + 1);
for (i = 1; i < 30; i++)
{
wd1 = (q6[i]*s->band[0].det) >> 12;
if (wd < wd1)
break;
}
ilow = (el < 0) ? iln[i] : ilp[i];
/* Block 2L, INVQAL */
ril = ilow >> 2;
wd2 = qm4[ril];
dlow = (s->band[0].det*wd2) >> 15;
/* Block 3L, LOGSCL */
il4 = rl42[ril];
wd = (s->band[0].nb*127) >> 7;
s->band[0].nb = wd + wl[il4];
if (s->band[0].nb < 0)
s->band[0].nb = 0;
else if (s->band[0].nb > 18432)
s->band[0].nb = 18432;
/* Block 3L, SCALEL */
wd1 = (s->band[0].nb >> 6) & 31;
wd2 = 8 - (s->band[0].nb >> 11);
wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
s->band[0].det = wd3 << 2;
block4(s, 0, dlow);
if (s->eight_k)
{
/* Just leave the high bits as zero */
code = (0xC0 | ilow) >> (8 - s->bits_per_sample);
}
else
{
/* Block 1H, SUBTRA */
eh = saturate(xhigh - s->band[1].s);
/* Block 1H, QUANTH */
wd = (eh >= 0) ? eh : -(eh + 1);
wd1 = (564*s->band[1].det) >> 12;
mih = (wd >= wd1) ? 2 : 1;
ihigh = (eh < 0) ? ihn[mih] : ihp[mih];
/* Block 2H, INVQAH */
wd2 = qm2[ihigh];
dhigh = (s->band[1].det*wd2) >> 15;
/* Block 3H, LOGSCH */
ih2 = rh2[ihigh];
wd = (s->band[1].nb*127) >> 7;
s->band[1].nb = wd + wh[ih2];
if (s->band[1].nb < 0)
s->band[1].nb = 0;
else if (s->band[1].nb > 22528)
s->band[1].nb = 22528;
/* Block 3H, SCALEH */
wd1 = (s->band[1].nb >> 6) & 31;
wd2 = 10 - (s->band[1].nb >> 11);
wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
s->band[1].det = wd3 << 2;
block4(s, 1, dhigh);
code = ((ihigh << 6) | ilow) >> (8 - s->bits_per_sample);
}
if (s->packed)
{
/* Pack the code bits */
s->out_buffer |= (code << s->out_bits);
s->out_bits += s->bits_per_sample;
if (s->out_bits >= 8)
{
g722_data[g722_bytes++] = (uint8_t) (s->out_buffer & 0xFF);
s->out_bits -= 8;
s->out_buffer >>= 8;
}
}
else
{
g722_data[g722_bytes++] = (uint8_t) code;
}
}
return g722_bytes;
}
/*- End of function --------------------------------------------------------*/
/*- End of file ------------------------------------------------------------*/

114
libs/miniwebrtc/audio/coding_g722/g722_interface.c Normal file
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@ -0,0 +1,114 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <stdlib.h>
#include <string.h>
#include "g722_enc_dec.h"
#include "g722_interface.h"
#include "typedefs.h"
int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
{
*G722enc_inst=(G722EncInst*)malloc(sizeof(G722EncoderState));
if (*G722enc_inst!=NULL) {
return(0);
} else {
return(-1);
}
}
int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
{
// Create and/or reset the G.722 encoder
// Bitrate 64 kbps and wideband mode (2)
G722enc_inst = (G722EncInst *) WebRtc_g722_encode_init(
(G722EncoderState*) G722enc_inst, 64000, 2);
if (G722enc_inst == NULL) {
return -1;
} else {
return 0;
}
}
int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
{
// Free encoder memory
return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
}
int16_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
const int16_t* speechIn,
int16_t len,
uint8_t* encoded)
{
unsigned char *codechar = (unsigned char*) encoded;
// Encode the input speech vector
return WebRtc_g722_encode((G722EncoderState*) G722enc_inst,
codechar, speechIn, len);
}
int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
{
*G722dec_inst=(G722DecInst*)malloc(sizeof(G722DecoderState));
if (*G722dec_inst!=NULL) {
return(0);
} else {
return(-1);
}
}
int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
{
// Create and/or reset the G.722 decoder
// Bitrate 64 kbps and wideband mode (2)
G722dec_inst = (G722DecInst *) WebRtc_g722_decode_init(
(G722DecoderState*) G722dec_inst, 64000, 2);
if (G722dec_inst == NULL) {
return -1;
} else {
return 0;
}
}
int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
{
// Free encoder memory
return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);
}
int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
int16_t *encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType)
{
// Decode the G.722 encoder stream
*speechType=G722_WEBRTC_SPEECH;
return WebRtc_g722_decode((G722DecoderState*) G722dec_inst,
decoded, (uint8_t*) encoded, len);
}
int16_t WebRtcG722_Version(char *versionStr, short len)
{
// Get version string
char version[30] = "2.0.0\n";
if (strlen(version) < (unsigned int)len)
{
strcpy(versionStr, version);
return 0;
}
else
{
return -1;
}
}

190
libs/miniwebrtc/audio/coding_g722/g722_interface.h Normal file
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@ -0,0 +1,190 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_
#define MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_
#include "typedefs.h"
/*
* Solution to support multiple instances
*/
typedef struct WebRtcG722EncInst G722EncInst;
typedef struct WebRtcG722DecInst G722DecInst;
/*
* Comfort noise constants
*/
#define G722_WEBRTC_SPEECH 1
#define G722_WEBRTC_CNG 2
#ifdef __cplusplus
extern "C" {
#endif
/****************************************************************************
* WebRtcG722_CreateEncoder(...)
*
* Create memory used for G722 encoder
*
* Input:
* - G722enc_inst : G722 instance for encoder
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst);
/****************************************************************************
* WebRtcG722_EncoderInit(...)
*
* This function initializes a G722 instance
*
* Input:
* - G722enc_inst : G722 instance, i.e. the user that should receive
* be initialized
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
/****************************************************************************
* WebRtcG722_FreeEncoder(...)
*
* Free the memory used for G722 encoder
*
* Input:
* - G722enc_inst : G722 instance for encoder
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
/****************************************************************************
* WebRtcG722_Encode(...)
*
* This function encodes G722 encoded data.
*
* Input:
* - G722enc_inst : G722 instance, i.e. the user that should encode
* a packet
* - speechIn : Input speech vector
* - len : Samples in speechIn
*
* Output:
* - encoded : The encoded data vector
*
* Return value : >0 - Length (in bytes) of coded data
* -1 - Error
*/
int16_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
const int16_t* speechIn,
int16_t len,
uint8_t* encoded);
/****************************************************************************
* WebRtcG722_CreateDecoder(...)
*
* Create memory used for G722 encoder
*
* Input:
* - G722dec_inst : G722 instance for decoder
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
/****************************************************************************
* WebRtcG722_DecoderInit(...)
*
* This function initializes a G729 instance
*
* Input:
* - G729_decinst_t : G729 instance, i.e. the user that should receive
* be initialized
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
/****************************************************************************
* WebRtcG722_FreeDecoder(...)
*
* Free the memory used for G722 decoder
*
* Input:
* - G722dec_inst : G722 instance for decoder
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
/****************************************************************************
* WebRtcG722_Decode(...)
*
* This function decodes a packet with G729 frame(s). Output speech length
* will be a multiple of 80 samples (80*frames/packet).
*
* Input:
* - G722dec_inst : G722 instance, i.e. the user that should decode
* a packet
* - encoded : Encoded G722 frame(s)
* - len : Bytes in encoded vector
*
* Output:
* - decoded : The decoded vector
* - speechType : 1 normal, 2 CNG (Since G722 does not have its own
* DTX/CNG scheme it should always return 1)
*
* Return value : >0 - Samples in decoded vector
* -1 - Error
*/
int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
int16_t *encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType);
/****************************************************************************
* WebRtcG722_Version(...)
*
* Get a string with the current version of the codec
*/
int16_t WebRtcG722_Version(char *versionStr, short len);
#ifdef __cplusplus
}
#endif
#endif /* MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_ */

1
libs/ysdp/parser.cpp
View File

@ -34,6 +34,7 @@ const TokenDict SDPParser::s_payloads[] = {
{ "lpc10", 7 },
{ "slin", 11 },
{ "g726", 2 },
{ "g722/16000", 9 },
{ "g722", 9 },
{ "g723", 4 },
{ "g728", 15 },

8
modules/Makefile.in
View File

@ -174,6 +174,10 @@ ifneq (@HAVE_ISAC@,no)
PROGS := $(PROGS) isaccodec.yate
endif
ifneq (@HAVE_G722WR@,no)
PROGS := $(PROGS) g722webrtc.yate
endif
ifneq (@HAVE_SPEEX@,no)
PROGS := $(PROGS) speexcodec.yate
endif
@ -384,6 +388,10 @@ isaccodec.yate: ../libs/miniwebrtc/libminiwebrtc.a
isaccodec.yate: LOCALFLAGS = @ISAC_INC@ -I@top_srcdir@/libs/miniwebrtc/audio/common/processing -I@top_srcdir@/libs/miniwebrtc
isaccodec.yate: LOCALLIBS = -L../libs/miniwebrtc -lminiwebrtc
g722webrtc.yate: ../libs/miniwebrtc/libminiwebrtc.a
g722webrtc.yate: LOCALFLAGS = @G722WR_INC@ -I@top_srcdir@/libs/miniwebrtc/audio/common/processing -I@top_srcdir@/libs/miniwebrtc
g722webrtc.yate: LOCALLIBS = -L../libs/miniwebrtc -lminiwebrtc
gsmcodec.yate: EXTERNFLAGS = @GSM_INC@
gsmcodec.yate: EXTERNLIBS = @GSM_LIB@

292
modules/g722webrtc.cpp Normal file
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@ -0,0 +1,292 @@
/**
* g722webrtc.cpp
* This file is part of the YATE Project http://YATE.null.ro
*
* G.722 codec using library based on WebRTC project.
*
* Yet Another Telephony Engine - a fully featured software PBX and IVR
* Copyright (C) 2019 Null Team
*
* This software is distributed under multiple licenses;
* see the COPYING file in the main directory for licensing
* information for this specific distribution.
*
* This use of this software may be subject to additional restrictions.
* See the LEGAL file in the main directory for details.
*
* 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.
*
*
* Copyright (c) 2011, The WebRTC project authors. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are
* met:
*
* * Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* * Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in
* the documentation and/or other materials provided with the
* distribution.
*
* * Neither the name of Google nor the names of its contributors may
* be used to endorse or promote products derived from this software
* without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
* HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
* LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
* DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
* THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include <yatephone.h>
extern "C" {
#include "signal_processing_library.h"
#include "g722_interface.h"
}
// G.722 was erroneously declared as 8000 samples/s but it's really 16000
// Minimum frame is 10ms (80 octets) but we will make all calculations for 20
#define G722_SAMPL 320
#define G722_SAMP8 160
#define G722_BLOCK 640
#define G722_FRAME 160
using namespace TelEngine;
namespace { // anonymous
class G722Codec : public DataTranslator
{
public:
G722Codec(const char* sFormat, const char* dFormat, bool encoding);
~G722Codec();
virtual bool valid() const
{ return m_enc || m_dec; }
virtual unsigned long Consume(const DataBlock& data, unsigned long tStamp,
unsigned long flags);
private:
bool m_encoding; // Encoder/decoder flag
G722EncInst* m_enc; // Encoder instance
G722DecInst* m_dec; // Decoder instance
DataBlock m_data; // Incomplete input data
DataBlock m_outdata; // Codec output
};
class G722Factory : public TranslatorFactory
{
public:
G722Factory(const TranslatorCaps* caps);
virtual const TranslatorCaps* getCapabilities() const
{ return m_caps; }
virtual DataTranslator* create(const DataFormat& sFormat, const DataFormat& dFormat);
private:
const TranslatorCaps* m_caps;
};
class G722Module : public Module
{
public:
G722Module();
~G722Module();
inline void incCount() {
Lock mylock(this);
m_count++;
}
inline void decCount() {
Lock mylock(this);
m_count--;
}
virtual void initialize();
virtual bool isBusy() const
{ return (m_count != 0); }
protected:
virtual void statusParams(String& str);
private:
int m_count; // Current number of codecs
G722Factory* m_g722; // Factory used to create codecs
};
INIT_PLUGIN(G722Module);
static TranslatorCaps s_caps[] = {
{ 0, 0 },
{ 0, 0 },
{ 0, 0 }
};
UNLOAD_PLUGIN(unloadNow)
{
if (unloadNow)
return !__plugin.isBusy();
return true;
}
/*
* G722Codec
*/
G722Codec::G722Codec(const char* sFormat, const char* dFormat, bool encoding)
: DataTranslator(sFormat,dFormat), m_encoding(encoding),
m_enc(0), m_dec(0)
{
Debug(&__plugin,DebugAll,"G722Codec(\"%s\",\"%s\",%scoding) [%p]",
sFormat,dFormat,m_encoding ? "en" : "de",this);
__plugin.incCount();
if (encoding) {
::WebRtcG722_CreateEncoder(&m_enc);
::WebRtcG722_EncoderInit(m_enc);
}
else {
::WebRtcG722_CreateDecoder(&m_dec);
::WebRtcG722_DecoderInit(m_dec);
}
}
G722Codec::~G722Codec()
{
if (m_enc)
::WebRtcG722_FreeEncoder(m_enc);
if (m_dec)
::WebRtcG722_FreeDecoder(m_dec);
Debug(&__plugin,DebugAll,
"G722Codec(%scoding) destroyed [%p]",m_encoding ? "en" : "de",this);
__plugin.decCount();
}
unsigned long G722Codec::Consume(const DataBlock& data, unsigned long tStamp, unsigned long flags)
{
RefPointer<DataSource> src = getTransSource();
if (!(src && valid()))
return 0;
if (data.null() && (flags & DataSilent))
return src->Forward(data,tStamp,flags);
ref();
if (m_encoding && (tStamp != invalidStamp()) && !m_data.null())
tStamp -= (m_data.length() / 2);
m_data += data;
int frames,consumed;
// G.722 declared rate and timestamps are for 8000 samples/s so it needs tweaking
if (m_encoding) {
tStamp /= 2;
frames = m_data.length() / G722_BLOCK;
consumed = frames * G722_BLOCK;
if (frames) {
m_outdata.resize(frames * G722_FRAME);
uint8_t* d = (uint8_t*)m_outdata.data();
const int16_t* s = (const int16_t*)m_data.data();
for (int i=0; i<frames; i++) {
::WebRtcG722_Encode(m_enc,s,G722_SAMPL,d);
s += G722_SAMPL;
d += G722_FRAME;
}
}
}
else {
tStamp *= 2;
frames = m_data.length() / G722_FRAME;
consumed = frames * G722_FRAME;
if (frames) {
m_outdata.resize(frames * G722_BLOCK);
int16_t* d = (int16_t*)m_outdata.data();
const uint8_t* s = (const uint8_t*)m_data.data();
int16_t t = G722_WEBRTC_SPEECH;
for (int i=0; i<frames; i++) {
// encoded data gets casted to uint8_t and then const uint8_t
::WebRtcG722_Decode(m_dec,(int16_t*)s,G722_FRAME,d,&t);
s += G722_FRAME;
d += G722_SAMPL;
}
}
}
if (!tStamp)
tStamp = timeStamp() + (frames * G722_SAMP8);
XDebug("G722Codec",DebugAll,"%scoding %d frames of %d input bytes (consumed %d) in %d output bytes",
m_encoding ? "en" : "de",frames,m_data.length(),consumed,m_outdata.length());
unsigned long len = 0;
if (frames) {
m_data.cut(-consumed);
len = getTransSource()->Forward(m_outdata,tStamp,flags);
}
deref();
return len;
}
/*
* G722Factory
*/
G722Factory::G722Factory(const TranslatorCaps* caps)
: TranslatorFactory("g722"),
m_caps(caps)
{
}
DataTranslator* G722Factory::create(const DataFormat& sFormat, const DataFormat& dFormat)
{
if ((sFormat == "slin/16000") && (dFormat == "g722/16000"))
return new G722Codec(sFormat,dFormat,true);
else if ((sFormat == "g722/16000") && (dFormat == "slin/16000"))
return new G722Codec(sFormat,dFormat,false);
return 0;
}
/*
* G722Module
*/
G722Module::G722Module()
: Module("g722webrtc","misc"),
m_count(0), m_g722(0)
{
char ver[24] = {0};
::WebRtcG722_Version(ver,sizeof(ver));
Output("Loaded module G722 - based on WebRTC G.722 library version %s",ver);
const FormatInfo* f = FormatRepository::addFormat("g722/16000",160,20000,"audio",16000);
s_caps[0].src = s_caps[1].dest = f;
s_caps[0].dest = s_caps[1].src = FormatRepository::getFormat("slin/16000");
// FIXME: put proper conversion costs
s_caps[0].cost = s_caps[1].cost = 5;
m_g722 = new G722Factory(s_caps);
}
G722Module::~G722Module()
{
Output("Unloading module G722 with %d codecs still in use",m_count);
TelEngine::destruct(m_g722);
}
void G722Module::initialize()
{
static bool s_first = true;
Output("Initializing module G722");
if (s_first) {
installRelay(Level);
installRelay(Status);
installRelay(Command);
s_first = false;
}
}
void G722Module::statusParams(String& str)
{
str << "codecs=" << m_count;
}
}; // anonymous namespace
/* vi: set ts=8 sw=4 sts=4 noet: */

2
modules/yiaxchan.cpp
View File

@ -46,7 +46,7 @@ static const TokenDict dict_payloads[] = {
{"adpcm", IAXFormat::ADPCM},
{"g726", IAXFormat::G726},
{"slin", IAXFormat::SLIN},
{"g722", IAXFormat::G722},
{"g722/16000", IAXFormat::G722},
{"amr", IAXFormat::AMR},
{"gsmhr", IAXFormat::GSM_HR},
{0, 0}

2
modules/yjinglechan.cpp
View File

@ -3851,7 +3851,7 @@ void YJGDriver::initialize()
s_knownCodecs.add("4", "G723", "8000", "g723");
s_knownCodecs.add("7", "LPC", "8000", "lpc10");
s_knownCodecs.add("8", "PCMA", "8000", "alaw");
s_knownCodecs.add("9", "G722", "8000", "g722");
s_knownCodecs.add("9", "G722", "8000", "g722/16000");
s_knownCodecs.add("11", "L16", "8000", "slin");
s_knownCodecs.add("15", "G728", "8000", "g728");
s_knownCodecs.add("18", "G729", "8000", "g729");

2
modules/yrtpchan.cpp
View File

@ -42,7 +42,7 @@ static TokenDict dict_payloads[] = {
{ "lpc10", 7 },
{ "slin", 11 },
{ "g726", 2 },
{ "g722", 9 },
{ "g722/16000", 9 },
{ "g723", 4 },
{ "g728", 15 },
{ "g729", 18 },

1
packing/rpm/yate.spec.in
View File

@ -154,6 +154,7 @@ for small to large scale projects.
%{_libdir}/yate/ilbccodec.yate
%{_libdir}/yate/ilbcwebrtc.yate
%{_libdir}/yate/isaccodec.yate
%{_libdir}/yate/g722webrtc.yate
%{_libdir}/yate/gvoice.yate
%{_libdir}/yate/server/dbwave.yate
%{_libdir}/yate/server/dbpbx.yate