Move src/ -> webrtc/

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d

9 files changed, 1590 insertions, 0 deletions

diff --git a/webrtc/modules/audio_coding/codecs/g722/Android.mk b/webrtc/modules/audio_coding/codecs/g722/Android.mk
new file mode 100644
index 0000000000..39dea9eb2d
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g722/Android.mk
@@ -0,0 +1,40 @@
+# 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.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_g722
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+    g722_interface.c \
+    g722_encode.c \
+    g722_decode.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+    $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+    $(LOCAL_PATH)/include \
+    $(LOCAL_PATH)/../../../..
+
+LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    libdl \
+    libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722.gypi b/webrtc/modules/audio_coding/codecs/g722/g722.gypi
new file mode 100644
index 0000000000..311b5a0fae
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g722/g722.gypi
@@ -0,0 +1,64 @@
+# 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.
+{
+  'targets': [
+    {
+      'target_name': 'G722',
+      'type': '<(library)',
+      'include_dirs': [
+        'include',
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          'include',
+        ],
+      },
+      'sources': [
+        'include/g722_interface.h',
+        'g722_interface.c',
+        'g722_encode.c',
+        'g722_decode.c',
+        'g722_enc_dec.h',
+      ],
+    },
+  ], # targets
+  'conditions': [
+    ['include_tests==1', {
+      'targets': [
+        {
+          'target_name': 'g722_unittests',
+          'type': 'executable',
+          'dependencies': [
+            'G722',
+            '<(webrtc_root)/test/test.gyp:test_support_main',
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+          ],
+          'sources': [
+            'g722_unittest.cc',
+          ],
+        },
+        {
+          'target_name': 'G722Test',
+          'type': 'executable',
+          'dependencies': [
+            'G722',
+          ],
+          'sources': [
+            'test/testG722.cc',
+          ],
+        },
+      ], # targets
+    }], # include_tests
+  ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_decode.c b/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
new file mode 100644
index 0000000000..499cc8fa30
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
@@ -0,0 +1,410 @@
+/*
+ * 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 <stdio.h>
+#include <memory.h>
+#include <stdlib.h>
+
+#include "typedefs.h"
+#include "g722_enc_dec.h"
+
+
+#if !defined(FALSE)
+#define FALSE 0
+#endif
+#if !defined(TRUE)
+#define TRUE (!FALSE)
+#endif
+
+static __inline WebRtc_Word16 saturate(WebRtc_Word32 amp)
+{
+    WebRtc_Word16 amp16;
+
+    /* Hopefully this is optimised for the common case - not clipping */
+    amp16 = (WebRtc_Word16) 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(g722_decode_state_t *s, int band, int d);
+
+static void block4(g722_decode_state_t *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 --------------------------------------------------------*/
+
+g722_decode_state_t *WebRtc_g722_decode_init(g722_decode_state_t *s,
+                                             int rate,
+                                             int options)
+{
+    if (s == NULL)
+    {
+        if ((s = (g722_decode_state_t *) 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(g722_decode_state_t *s)
+{
+    free(s);
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+int WebRtc_g722_decode(g722_decode_state_t *s, WebRtc_Word16 amp[],
+                       const WebRtc_UWord8 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++] = (WebRtc_Word16) (rlow << 1);
+            amp[outlen++] = (WebRtc_Word16) (rhigh << 1);
+        }
+        else
+        {
+            if (s->eight_k)
+            {
+                amp[outlen++] = (WebRtc_Word16) (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 ------------------------------------------------------------*/
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h b/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h
new file mode 100644
index 0000000000..d2d19b04b1
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h
@@ -0,0 +1,158 @@
+/*
+ * 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: g722.h,v 1.10 2006/06/16 12:45:53 steveu Exp $
+ *
+ * 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_
+
+/*! \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;
+} g722_encode_state_t;
+
+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;
+} g722_decode_state_t;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+g722_encode_state_t *WebRtc_g722_encode_init(g722_encode_state_t *s,
+                                             int rate,
+                                             int options);
+int WebRtc_g722_encode_release(g722_encode_state_t *s);
+int WebRtc_g722_encode(g722_encode_state_t *s,
+                       WebRtc_UWord8 g722_data[],
+                       const WebRtc_Word16 amp[],
+                       int len);
+
+g722_decode_state_t *WebRtc_g722_decode_init(g722_decode_state_t *s,
+                                             int rate,
+                                             int options);
+int WebRtc_g722_decode_release(g722_decode_state_t *s);
+int WebRtc_g722_decode(g722_decode_state_t *s,
+                       WebRtc_Word16 amp[],
+                       const WebRtc_UWord8 g722_data[],
+                       int len);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_encode.c b/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
new file mode 100644
index 0000000000..7487b64c7f
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
@@ -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 <stdio.h>
+#include <memory.h>
+#include <stdlib.h>
+
+#include "typedefs.h"
+#include "g722_enc_dec.h"
+
+#if !defined(FALSE)
+#define FALSE 0
+#endif
+#if !defined(TRUE)
+#define TRUE (!FALSE)
+#endif
+
+static __inline WebRtc_Word16 saturate(WebRtc_Word32 amp)
+{
+    WebRtc_Word16 amp16;
+
+    /* Hopefully this is optimised for the common case - not clipping */
+    amp16 = (WebRtc_Word16) 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(g722_encode_state_t *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 --------------------------------------------------------*/
+
+g722_encode_state_t *WebRtc_g722_encode_init(g722_encode_state_t *s,
+                                             int rate, int options)
+{
+    if (s == NULL)
+    {
+        if ((s = (g722_encode_state_t *) 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(g722_encode_state_t *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
+WebRtc_Word16 limitValues (WebRtc_Word16 rl)
+{
+
+    WebRtc_Word16 yl;
+
+    yl = (rl > 16383) ? 16383 : ((rl < -16384) ? -16384 : rl);
+
+    return (yl);
+}
+#endif
+
+int WebRtc_g722_encode(g722_encode_state_t *s, WebRtc_UWord8 g722_data[],
+                       const WebRtc_Word16 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++] = (WebRtc_UWord8) (s->out_buffer & 0xFF);
+                s->out_bits -= 8;
+                s->out_buffer >>= 8;
+            }
+        }
+        else
+        {
+            g722_data[g722_bytes++] = (WebRtc_UWord8) code;
+        }
+    }
+    return g722_bytes;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_interface.c b/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
new file mode 100644
index 0000000000..d559014225
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
@@ -0,0 +1,115 @@
+/*
+ *  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_interface.h"
+#include "g722_enc_dec.h"
+#include "typedefs.h"
+
+
+WebRtc_Word16 WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
+{
+    *G722enc_inst=(G722EncInst*)malloc(sizeof(g722_encode_state_t));
+    if (*G722enc_inst!=NULL) {
+      return(0);
+    } else {
+      return(-1);
+    }
+}
+
+WebRtc_Word16 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(
+        (g722_encode_state_t*) G722enc_inst, 64000, 2);
+    if (G722enc_inst == NULL) {
+        return -1;
+    } else {
+        return 0;
+    }
+}
+
+WebRtc_Word16 WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
+{
+    // Free encoder memory
+    return WebRtc_g722_encode_release((g722_encode_state_t*) G722enc_inst);
+}
+
+WebRtc_Word16 WebRtcG722_Encode(G722EncInst *G722enc_inst,
+                                WebRtc_Word16 *speechIn,
+                                WebRtc_Word16 len,
+                                WebRtc_Word16 *encoded)
+{
+    unsigned char *codechar = (unsigned char*) encoded;
+    // Encode the input speech vector
+    return WebRtc_g722_encode((g722_encode_state_t*) G722enc_inst,
+                       codechar, speechIn, len);
+}
+
+WebRtc_Word16 WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
+{
+    *G722dec_inst=(G722DecInst*)malloc(sizeof(g722_decode_state_t));
+    if (*G722dec_inst!=NULL) {
+      return(0);
+    } else {
+      return(-1);
+    }
+}
+
+WebRtc_Word16 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(
+        (g722_decode_state_t*) G722dec_inst, 64000, 2);
+    if (G722dec_inst == NULL) {
+        return -1;
+    } else {
+        return 0;
+    }
+}
+
+WebRtc_Word16 WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
+{
+    // Free encoder memory
+    return WebRtc_g722_decode_release((g722_decode_state_t*) G722dec_inst);
+}
+
+WebRtc_Word16 WebRtcG722_Decode(G722DecInst *G722dec_inst,
+                                WebRtc_Word16 *encoded,
+                                WebRtc_Word16 len,
+                                WebRtc_Word16 *decoded,
+                                WebRtc_Word16 *speechType)
+{
+    // Decode the G.722 encoder stream
+    *speechType=G722_WEBRTC_SPEECH;
+    return WebRtc_g722_decode((g722_decode_state_t*) G722dec_inst,
+                              decoded, (WebRtc_UWord8*) encoded, len);
+}
+
+WebRtc_Word16 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;
+    }
+}
+
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722_unittest.cc b/webrtc/modules/audio_coding/codecs/g722/g722_unittest.cc
new file mode 100644
index 0000000000..a828eddbe5
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_unittest.cc
@@ -0,0 +1,17 @@
+/*
+ *  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.
+ */
+
+/*
+ * Empty test just to get code coverage metrics for this dir.
+ */
+#include "g722_interface.h"
+#include "gtest/gtest.h"
+
+TEST(G722Test, EmptyTestToGetCodeCoverage) {}
diff --git a/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h b/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
new file mode 100644
index 0000000000..e50d66f56c
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
@@ -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
+ */
+WebRtc_Word16 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
+ */
+
+WebRtc_Word16 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
+ */
+WebRtc_Word16 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
+ */
+
+WebRtc_Word16 WebRtcG722_Encode(G722EncInst *G722enc_inst,
+                                WebRtc_Word16 *speechIn,
+                                WebRtc_Word16 len,
+                                WebRtc_Word16 *encoded);
+
+
+/****************************************************************************
+ * WebRtcG722_CreateDecoder(...)
+ *
+ * Create memory used for G722 encoder
+ *
+ * Input:
+ *     - G722dec_inst         : G722 instance for decoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+WebRtc_Word16 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
+ */
+
+WebRtc_Word16 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
+ */
+
+WebRtc_Word16 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
+ */
+
+WebRtc_Word16 WebRtcG722_Decode(G722DecInst *G722dec_inst,
+                                WebRtc_Word16 *encoded,
+                                WebRtc_Word16 len,
+                                WebRtc_Word16 *decoded,
+                                WebRtc_Word16 *speechType);
+
+/****************************************************************************
+ * WebRtcG722_Version(...)
+ *
+ * Get a string with the current version of the codec
+ */
+
+WebRtc_Word16 WebRtcG722_Version(char *versionStr, short len);
+
+
+#ifdef __cplusplus
+}
+#endif
+
+
+#endif /* MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_ */
diff --git a/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc b/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
new file mode 100644
index 0000000000..d2fdca3a85
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
@@ -0,0 +1,162 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+/*
+ * testG722.cpp : Defines the entry point for the console application.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include "typedefs.h"
+
+/* include API */
+#include "g722_interface.h"
+
+/* Runtime statistics */
+#include <time.h>
+#define CLOCKS_PER_SEC_G722  100000
+
+// Forward declaration
+typedef struct WebRtcG722EncInst    G722EncInst;
+typedef struct WebRtcG722DecInst    G722DecInst;
+
+/* function for reading audio data from PCM file */
+int readframe(WebRtc_Word16 *data, FILE *inp, int length)
+{
+    short k, rlen, status = 0;
+
+    rlen = (short)fread(data, sizeof(WebRtc_Word16), length, inp);
+    if (rlen < length) {
+        for (k = rlen; k < length; k++)
+            data[k] = 0;
+        status = 1;
+    }
+
+    return status;
+}
+
+int main(int argc, char* argv[])
+{
+    char inname[60], outbit[40], outname[40];
+    FILE *inp, *outbitp, *outp;
+
+    int framecnt, endfile;
+    WebRtc_Word16 framelength = 160;
+    G722EncInst *G722enc_inst;
+    G722DecInst *G722dec_inst;
+    int err;
+
+    /* Runtime statistics */
+    double starttime;
+    double runtime = 0;
+    double length_file;
+
+    WebRtc_Word16 stream_len = 0;
+    WebRtc_Word16 shortdata[960];
+    WebRtc_Word16 decoded[960];
+    WebRtc_Word16 streamdata[80*3];
+    WebRtc_Word16 speechType[1];
+
+    /* handling wrong input arguments in the command line */
+    if (argc!=5)  {
+        printf("\n\nWrong number of arguments or flag values.\n\n");
+
+        printf("\n");
+        printf("Usage:\n\n");
+        printf("./testG722.exe framelength infile outbitfile outspeechfile \n\n");
+        printf("with:\n");
+        printf("framelength  :    Framelength in samples.\n\n");
+        printf("infile       :    Normal speech input file\n\n");
+        printf("outbitfile   :    Bitstream output file\n\n");
+        printf("outspeechfile:    Speech output file\n\n");
+        exit(0);
+
+    }
+
+    /* Get frame length */
+    framelength = atoi(argv[1]);
+
+    /* Get Input and Output files */
+    sscanf(argv[2], "%s", inname);
+    sscanf(argv[3], "%s", outbit);
+    sscanf(argv[4], "%s", outname);
+
+    if ((inp = fopen(inname,"rb")) == NULL) {
+        printf("  G.722: Cannot read file %s.\n", inname);
+        exit(1);
+    }
+    if ((outbitp = fopen(outbit,"wb")) == NULL) {
+        printf("  G.722: Cannot write file %s.\n", outbit);
+        exit(1);
+    }
+    if ((outp = fopen(outname,"wb")) == NULL) {
+        printf("  G.722: Cannot write file %s.\n", outname);
+        exit(1);
+    }
+    printf("\nInput:%s\nOutput bitstream:%s\nOutput:%s\n", inname, outbit, outname);
+
+    /* Create and init */
+    WebRtcG722_CreateEncoder((G722EncInst **)&G722enc_inst);
+    WebRtcG722_CreateDecoder((G722DecInst **)&G722dec_inst);
+    WebRtcG722_EncoderInit((G722EncInst *)G722enc_inst);
+    WebRtcG722_DecoderInit((G722DecInst *)G722dec_inst);
+
+
+    /* Initialize encoder and decoder */
+    framecnt = 0;
+    endfile = 0;
+    while (endfile == 0) {
+        framecnt++;
+
+        /* Read speech block */
+        endfile = readframe(shortdata, inp, framelength);
+
+        /* Start clock before call to encoder and decoder */
+        starttime = clock()/(double)CLOCKS_PER_SEC_G722;
+
+        /* G.722 encoding + decoding */
+        stream_len = WebRtcG722_Encode((G722EncInst *)G722enc_inst, shortdata, framelength, streamdata);
+        err = WebRtcG722_Decode((G722DecInst *)G722dec_inst, streamdata, stream_len, decoded, speechType);
+
+        /* Stop clock after call to encoder and decoder */
+        runtime += (double)((clock()/(double)CLOCKS_PER_SEC_G722)-starttime);
+
+        if (stream_len < 0 || err < 0) {
+            /* exit if returned with error */
+            printf("Error in encoder/decoder\n");
+        } else {
+          /* Write coded bits to file */
+          if (fwrite(streamdata, sizeof(short), stream_len/2,
+                     outbitp) != static_cast<size_t>(stream_len/2)) {
+            return -1;
+          }
+          /* Write coded speech to file */
+          if (fwrite(decoded, sizeof(short), framelength,
+                     outp) != static_cast<size_t>(framelength)) {
+            return -1;
+          }
+        }
+    }
+
+    WebRtcG722_FreeEncoder((G722EncInst *)G722enc_inst);
+    WebRtcG722_FreeDecoder((G722DecInst *)G722dec_inst);
+
+    length_file = ((double)framecnt*(double)framelength/16000);
+    printf("\n\nLength of speech file: %.1f s\n", length_file);
+    printf("Time to run G.722:      %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
+    printf("---------------------END----------------------\n");
+
+    fclose(inp);
+    fclose(outbitp);
+    fclose(outp);
+
+    return 0;
+}