- take the RTP packet duration from the payload (Opus ptime > 20 ms)

AudioReceiver assumed every packet of a variable-size codec lasts
frameTime(), i.e. the local 20 ms ptime. A 40 ms Opus packet then looked
like 20 ms of audio followed by a 20 ms gap: decodePacketTo() emitted
silence before every packet and the jitter buffer dropped about half of
them (335 of 605 decoded on a 20/40 ms call), so vq-core scored clean
audio at MOS ~1.3. Codec::packetTime() now returns the duration of a
payload; Opus reads it with opus_packet_get_nb_samples(), fixed-frame
codecs keep the old formula. With it the same call decodes 603 of 605.

Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01LzU7iFc93X5jFATRKaoKiZ

src/engine/media/MT_AudioCodec.cpp

@@ -434,6 +434,13 @@ Codec::Info OpusCodec::info()
     };
 }
 
+int OpusCodec::packetTime(std::span<const uint8_t> payload)
+{
+    int samples = payload.empty() ? 0
+                                  : opus_packet_get_nb_samples(payload.data(), (opus_int32)payload.size(), mSamplerate);
+    return samples > 0 ? samples * 1000 / mSamplerate : frameTime();
+}
+
 Codec::EncodeResult OpusCodec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
     // opus_encode() takes the frame size in samples per channel and the output
@@ -1132,13 +1139,15 @@ G722Codec::~G722Codec()
 
 Codec::Info G722Codec::info()
 {
-    // ToDo: double check the G722 calls - remember RFC has bug about samplerate
+    // G.722 decodes to 16 kHz (640 bytes of PCM per 20 ms frame), but RFC 3551 keeps its RTP
+    // clock at 8 kHz for historical reasons: the timestamp unit is 1/8000, not 1/samplerate.
     return {.mName = G722_MIME_NAME,
-            .mSamplerate = 8000,
+            .mSamplerate = 16000,
             .mChannels = 1,
             .mPcmLength = 640,
             .mFrameTime = 20,
-            .mRtpLength = 160};
+            .mRtpLength = 160,
+            .mTimestampUnit = 1.0f / 8000};
 }
 
 Codec::EncodeResult G722Codec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
@@ -1186,7 +1195,8 @@ const char* G722Codec::G722Factory::name()
 
 int G722Codec::G722Factory::samplerate()
 {
-    // Although G722 uses 16000 as rate for timestamping RTP frames - in fact it is 8KHz codec. So return 8KHz here.
+    // The rate SDP advertises (rtpmap G722/8000) is the RTP clock, 8 kHz by RFC 3551, although the
+    // audio is 16 kHz; see G722Codec::info().
     return 8000;
 }
 

src/engine/media/MT_AudioCodec.h

@@ -69,6 +69,8 @@ protected:
     int          mDecoderChannels = 0;
 
 public:
+    int packetTime(std::span<const uint8_t> payload) override;
+
     struct Params
     {
         bool mUseDtx = false, mUseInbandFec = false, mStereo = false;

src/engine/media/MT_AudioReceiver.cpp

@@ -520,13 +520,11 @@ Codec* AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p)
 
             if (!codec)
                 time_length = 10;
-            else if (!codec->rtpLength())
-                time_length = codec->frameTime();
             else
-                time_length = lround(double(payloadLength) / codec->rtpLength() * codec->frameTime());
+                time_length = codec->packetTime(std::span{p->GetPayloadData(), (size_t)payloadLength});
 
             if (codec)
-                samplerate = codec->samplerate();
+                samplerate = codec->rtpClockRate(); // RTP timestamp units, not PCM rate (G.722)
         }
 
         // Process jitter anyway - can we decode payload or not
@@ -724,7 +722,7 @@ AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& out
     if (mLastPacketTimestamp && mLastPacketTimeLength && mCodec)
     {
         int units = rtp.GetTimestamp() - *mLastPacketTimestamp;
-        int milliseconds = units / (mCodec->samplerate() / 1000);
+        int milliseconds = units / (mCodec->rtpClockRate() / 1000);
         if (milliseconds > mLastPacketTimeLength)
         {
             auto silenceLength = std::chrono::milliseconds(milliseconds - mLastPacketTimeLength);
@@ -794,8 +792,11 @@ AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& out
                 mFrameCount = mCodec->rtpLength() ? rtp.GetPayloadLength() / mCodec->rtpLength() : 1;
                 int frameLength = mCodec->rtpLength() ? mCodec->rtpLength() : (int)rtp.GetPayloadLength();
 
-                // Save last packet time length
+                // Save last packet time length. Taken from the payload, not mFrameCount *
-                mLastPacketTimeLength = mFrameCount * mCodec->frameTime();
+                // frameTime(): an Opus packet of 40 or 60 ms is one "frame" here, and counting
+                // it as frameTime() (20 ms) made the gap check above emit silence before every
+                // packet and pushed the jitter buffer into discarding about half of them.
+                mLastPacketTimeLength = mCodec->packetTime(std::span{rtp.GetPayloadData(), payload_length});
 
                 // Decode
                 for (int i = 0; i < mFrameCount && !mCodecSettings.mSkipDecode; i++)
@@ -1157,12 +1158,17 @@ AudioReceiver::MediaInfo AudioReceiver::infoFor(jrtplib::RTPPacket& p)
 
     if (codec->rtpLength() != 0)
     {
-        int frameCount = static_cast<int>(p.GetPayloadLength() / codec->rtpLength());
+        // A G.729 (PT 18) SID frame is shorter than a voice frame but stands for one; other
-        if (p.GetPayloadType() == 9 /*G729A silence*/ && p.GetPayloadLength() % codec->rtpLength())
+        // codecs scale with the payload (G.722 at 10 ms ptime carries half a 160-byte frame).
-            frameCount++;
+        if (p.GetPayloadType() == 18 && p.GetPayloadLength() % codec->rtpLength())
-
+        {
+            int frameCount = static_cast<int>(p.GetPayloadLength() / codec->rtpLength()) + 1;
             packetTime = std::chrono::milliseconds(frameCount * codec->frameTime());
         }
+        else
+            packetTime = std::chrono::milliseconds(
+                lround(double(p.GetPayloadLength()) / codec->rtpLength() * codec->frameTime()));
+    }
     else if (typeid(*codec) == typeid(OpusCodec))
     {
         OpusCodec* oc = dynamic_cast<OpusCodec*>(codec);

src/engine/media/MT_AudioStream.cpp

@@ -114,7 +114,7 @@ void AudioStream::setTransmittingCodec(Codec::Factory& factory, int payloadType)
     mTransmittingCodec = factory.create();
     mTransmittingPayloadType = payloadType;
     if (mRtpSession.IsActive())
-        mRtpSession.SetTimestampUnit(1.0 / mTransmittingCodec->samplerate());
+        mRtpSession.SetTimestampUnit(mTransmittingCodec->timestampUnit());
 }
 
 PCodec AudioStream::transmittingCodec()
@@ -241,7 +241,7 @@ void AudioStream::addData(const void* buffer, int bytes)
                 ICELogMedia(<< "Sending RTP packet pt = " << mTransmittingPayloadType
                             << ", plength = " << (int)mEncodedAudio.size() << " to ");
                 mRtpSession.SendPacketEx(mEncodedAudio.data(), mEncodedAudio.size(), mTransmittingPayloadType, false,
-                                         packetTime * codec->samplerate() / 1000, 0, nullptr, 0);
+                                         packetTime * codec->rtpClockRate() / 1000, 0, nullptr, 0);
                 mEncodedAudio.clear();
                 encodedTime = 0;
             }

src/engine/media/MT_Codec.h

@@ -6,6 +6,7 @@
 #ifndef __MT_CODEC_H
 #define __MT_CODEC_H
 
+#include <cmath>
 #include <map>
 #include <span>
 
@@ -66,9 +67,24 @@ public:
     int          frameTime() { return info().mFrameTime; }
     std::string  name() { return info().mName; }
     float timestampUnit() { return info().mTimestampUnit == 0.0f ? 1.0f / info().mSamplerate : info().mTimestampUnit; }
+    // RTP clock rate in Hz, the unit of RTP timestamps. Equals samplerate() except for codecs
+    // whose RTP clock differs from the decoded audio: G.722 decodes to 16 kHz but its RTP clock
+    // is 8 kHz (RFC 3551). Use it for timestamp arithmetic and jitter, samplerate() for PCM.
+    int   rtpClockRate() { return static_cast<int>(1.0f / timestampUnit() + 0.5f); }
 
     Audio::Format getAudioFormat() { return Audio::Format(this->info().mSamplerate, this->info().mChannels); }
 
+    // Audio duration of one RTP payload, in milliseconds. The default suits fixed-size
+    // frames (rtpLength() bytes per frameTime()); a codec whose packets carry a variable
+    // amount of audio (Opus: 2.5-120 ms, whatever the sender's ptime) reads it from the
+    // payload instead, as frameTime() is only the local ptime.
+    virtual int packetTime(std::span<const uint8_t> payload)
+    {
+        if (!rtpLength())
+            return frameTime();
+        return static_cast<int>(std::lround(double(payload.size()) / rtpLength() * frameTime()));
+    }
+
     // Returns size of encoded data (RTP) in bytes
     struct EncodeResult
     {

src/libs/g722/CMakeLists.txt

@@ -1,16 +1,13 @@
 project (g722_codec)
 
-# Rely on C++ 11
-set (CMAKE_CXX_STANDARD 20)
-set (CMAKE_CXX_STANDARD_REQUIRED ON)
-
 set(CMAKE_POSITION_INDEPENDENT_CODE ON)
 
+# G.722 from Asterisk (codecs/g722): Steve Underwood's spandsp code, which he
+# placed in the public domain, on top of the CMU 1993 codec, whose use "for
+# any research or commercial purpose, is completely unrestricted". See README.md.
 set (G722_SOURCES
-    g722_bitstream.c
     g722_decode.c
     g722_encode.c
 )
 
 add_library(g722_codec ${G722_SOURCES})
-

src/libs/g722/README.md

@@ -0,0 +1,35 @@
+# G.722 codec
+
+Source: Asterisk, `codecs/g722/` (`g722.h`, `g722_encode.c`, `g722_decode.c`),
+https://github.com/asterisk/asterisk, taken 2026-09-21.
+
+## Licence
+
+No copyleft. The code has two layers:
+
+- **Steve Underwood, 2005** (spandsp): "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." (in each file's header)
+- **Carnegie Mellon University, 1993** (Chengxiang Lu and Alex Hauptmann, Speech
+  Group): "The Carnegie Mellon ADPCM program is Copyright (c) 1993 by Carnegie
+  Mellon University. Use of this program, for any research or commercial
+  purpose, is completely unrestricted. If you make use of or redistribute this
+  material, we would appreciate acknowlegement of its origin." (the terms as
+  reproduced in sippy's libg722, https://github.com/sippy/libg722, LICENSE)
+
+This replaces an older copy of the same spandsp code that was offered only
+under GPL-2 or LGPL-2.1.
+
+## Behaviour
+
+Asterisk's version maps 16-bit PCM to the codec's range: the encoder halves its
+input and the decoder doubles its output. Streams from Asterisk and ffmpeg
+therefore decode at their original level. The older copy did neither, so it
+decoded those streams 6 dB too quiet and clipped encoder input above -6 dBFS.
+
+Local changes:
+- the decoder saturates its output (`saturate()`) instead of casting. Asterisk's
+  cast wraps loud samples around int16: 259 samples in 11 s of speech peaking
+  at -0.2 dBFS came out as large values of the opposite sign;
+- `g722.h` includes `<stdint.h>` and maps `__inline__` to `__inline` for MSVC.

src/libs/g722/g722.h

@@ -1,5 +1,5 @@
 /*
- * VoIPcodecs - a series of DSP components for telephony
+ * SpanDSP - a series of DSP components for telephony
  *
  * g722.h - The ITU G.722 codec.
  *
@@ -7,21 +7,10 @@
  *
  * 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 -
- * This program is free software; you can redistribute it and/or modify
+ *  even the slimy ones who might try to proprietize my work and use it
- * it under the terms of the GNU General Public License version 2, or
+ *  to my detriment.
- * the Lesser GNU General Public License version 2.1, as published by
- * the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  *
  * Based on a single channel G.722 codec which is:
  *
@@ -29,17 +18,19 @@
  * Computer Science, Speech Group
  * Chengxiang Lu and Alex Hauptmann
  *
- * $Id: g722.h,v 1.17 2008/02/09 15:32:26 steveu Exp $
+ * $Id$
  */
 
 
 /*! \file */
 
-#if !defined(_SPANDSP_G722_H_)
+#if !defined(_G722_H_)
-#define _SPANDSP_G722_H_
+#define _G722_H_
-
-#include "inttypes.h"
 
+#include <stdint.h>   /* rtphone: the header uses int16_t/uint8_t itself */
+#if defined(_MSC_VER) && !defined(__inline__)
+#define __inline__ __inline   /* rtphone: MSVC has no __inline__ (the .c files use it) */
+#endif
 
 /*! \page g722_page G.722 encoding and decoding
 \section g722_page_sec_1 What does it do?
@@ -60,6 +51,13 @@ enum
     G722_PACKED = 0x0002
 };
 
+#ifndef INT16_MAX
+#define INT16_MAX       32767
+#endif
+#ifndef INT16_MIN
+#define INT16_MIN       (-32768)
+#endif
+
 typedef struct
 {
     /*! TRUE if the operating in the special ITU test mode, with the band split filters
@@ -136,48 +134,19 @@ typedef struct
     int out_bits;
 } g722_decode_state_t;
 
-#if defined(__cplusplus)
+#ifdef __cplusplus
-extern "C"
+extern "C" {
-{
 #endif
 
-/*! Initialise an G.722 encode context.
-    \param s The G.722 encode context.
-    \param rate The required bit rate for the G.722 data.
-           The valid rates are 64000, 56000 and 48000.
-    \param options
-    \return A pointer to the G.722 encode context, or NULL for error. */
 g722_encode_state_t *g722_encode_init(g722_encode_state_t *s, int rate, int options);
-
 int g722_encode_release(g722_encode_state_t *s);
-
-/*! Encode a buffer of linear PCM data to G.722
-    \param s The G.722 context.
-    \param g722_data The G.722 data produced.
-    \param amp The audio sample buffer.
-    \param len The number of samples in the buffer.
-    \return The number of bytes of G.722 data produced. */
 int g722_encode(g722_encode_state_t *s, uint8_t g722_data[], const int16_t amp[], int len);
 
-/*! Initialise an G.722 decode context.
-    \param s The G.722 decode context.
-    \param rate The bit rate of the G.722 data.
-           The valid rates are 64000, 56000 and 48000.
-    \param options
-    \return A pointer to the G.722 decode context, or NULL for error. */
 g722_decode_state_t *g722_decode_init(g722_decode_state_t *s, int rate, int options);
-
 int g722_decode_release(g722_decode_state_t *s);
-
-/*! Decode a buffer of G.722 data to linear PCM.
-    \param s The G.722 context.
-    \param amp The audio sample buffer.
-    \param g722_data
-    \param len
-    \return The number of samples returned. */
 int g722_decode(g722_decode_state_t *s, int16_t amp[], const uint8_t g722_data[], int len);
 
-#if defined(__cplusplus)
+#ifdef __cplusplus
 }
 #endif
 

src/libs/g722/g722_bitstream.c

@@ -1,137 +0,0 @@
-/*
- * VoIPcodecs - a series of DSP components for telephony
- *
- * bitstream.c - Bitstream composition and decomposition routines.
- *
- * Written by Steve Underwood <steveu@coppice.org>
- *
- * Copyright (C) 2006 Steve Underwood
- *
- * All rights reserved.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the Lesser Lesser GNU General Public License version 2.1.1, as
- * published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- * $Id: bitstream.c,v 1.8 2007/08/20 15:22:21 steveu Exp $
- */
-
-/*! \file */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#include "inttypes.h"
-#include <stdlib.h>
-#include <string.h>
-#include <assert.h>
-
-#include "g722_telephony.h"
-#include "g722_bitstream.h"
-
-void bitstream_put(bitstream_state_t *s, uint8_t **c, unsigned int value, int bits)
-{
-    value &= ((1 << bits) - 1);
-    if (s->residue + bits <= 32)
-    {
-        s->bitstream |= (value << s->residue);
-        s->residue += bits;
-    }
-    while (s->residue >= 8)
-    {
-        s->residue -= 8;
-        *(*c)++ = (uint8_t) (s->bitstream & 0xFF);
-        s->bitstream >>= 8;
-    }
-}
-/*- End of function --------------------------------------------------------*/
-
-void bitstream_put2(bitstream_state_t *s, uint8_t **c, unsigned int value, int bits)
-{
-    value &= ((1 << bits) - 1);
-    if (s->residue + bits <= 32)
-    {
-        s->bitstream = (s->bitstream << bits) | value;
-        s->residue += bits;
-    }
-    while (s->residue >= 8)
-    {
-        s->residue -= 8;
-        *(*c)++ = (uint8_t) ((s->bitstream >> s->residue) & 0xFF);
-    }
-}
-/*- End of function --------------------------------------------------------*/
-
-unsigned int bitstream_get(bitstream_state_t *s, const uint8_t **c, int bits)
-{
-    unsigned int x;
-
-    while (s->residue < (unsigned int) bits)
-    {
-        x = (unsigned int) *(*c)++;
-        s->bitstream |= (x << s->residue);
-        s->residue += 8;
-    }
-    s->residue -= bits;
-    x = s->bitstream & ((1 << bits) - 1);
-    s->bitstream >>= bits;
-    return x;
-}
-/*- End of function --------------------------------------------------------*/
-
-unsigned int bitstream_get2(bitstream_state_t *s, const uint8_t **c, int bits)
-{
-    unsigned int x;
-
-    while (s->residue < (unsigned int) bits)
-    {
-        x = (unsigned int) *(*c)++;
-        s->bitstream = (s->bitstream << 8) | x;
-        s->residue += 8;
-    }
-    s->residue -= bits;
-    x = (s->bitstream >> s->residue) & ((1 << bits) - 1);
-    return x;
-}
-/*- End of function --------------------------------------------------------*/
-
-void bitstream_flush(bitstream_state_t *s, uint8_t **c)
-{
-    if (s->residue > 0)
-    {
-        *(*c)++ = (uint8_t) ((s->bitstream << (8 - s->residue)) & 0xFF);
-        s->residue = 0;
-    }
-}
-/*- End of function --------------------------------------------------------*/
-
-void bitstream_flush2(bitstream_state_t *s, uint8_t **c)
-{
-    if (s->residue > 0)
-    {
-        *(*c)++ = (uint8_t) ((s->bitstream << (8 - s->residue)) & 0xFF);
-        s->residue = 0;
-    }
-}
-/*- End of function --------------------------------------------------------*/
-
-bitstream_state_t *bitstream_init(bitstream_state_t *s)
-{
-    if (s == NULL)
-        return NULL;
-    s->bitstream = 0;
-    s->residue = 0;
-    return s;
-}
-/*- End of function --------------------------------------------------------*/
-/*- End of file ------------------------------------------------------------*/

src/libs/g722/g722_bitstream.h

@@ -1,89 +0,0 @@
-/*
- * VoIPcodecs - a series of DSP components for telephony
- *
- * bitstream.h - Bitstream composition and decomposition routines.
- *
- * Written by Steve Underwood <steveu@coppice.org>
- *
- * Copyright (C) 2006 Steve Underwood
- *
- * All rights reserved.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the Lesser GNU General Public License version 2.1, as
- * published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- * $Id: bitstream.h,v 1.8 2007/12/13 11:31:32 steveu Exp $
- */
-
-/*! \file */
-
-#if !defined(_SPANDSP_BITSTREAM_H_)
-#define _SPANDSP_BITSTREAM_H_
-
-/*! \page bitstream_page Bitstream composition and decomposition
-\section bitstream_page_sec_1 What does it do?
-
-\section bitstream_page_sec_2 How does it work?
-*/
-
-/*! Bitstream handler state */
-typedef struct
-{
-    /*! The bit stream. */
-    unsigned int bitstream;
-    /*! The residual bits in bitstream. */
-    unsigned int residue;
-} bitstream_state_t;
-
-
-#if defined(__cplusplus)
-extern "C"
-{
-#endif
-
-/*! \brief Put a chunk of bits into the output buffer.
-    \param s A pointer to the bitstream context.
-    \param c A pointer to the bitstream output buffer.
-    \param value The value to be pushed into the output buffer.
-    \param bits The number of bits of value to be pushed. 1 to 25 bit is valid. */
-void bitstream_put(bitstream_state_t *s, uint8_t **c, unsigned int value, int bits);
-
-void bitstream_put2(bitstream_state_t *s, uint8_t **c, unsigned int value, int bits);
-
-/*! \brief Get a chunk of bits from the input buffer.
-    \param s A pointer to the bitstream context.
-    \param c A pointer to the bitstream input buffer.
-    \param bits The number of bits of value to be grabbed. 1 to 25 bit is valid.
-    \return The value retrieved from the input buffer. */
-unsigned int bitstream_get(bitstream_state_t *s, const uint8_t **c, int bits);
-
-unsigned int bitstream_get2(bitstream_state_t *s, const uint8_t **c, int bits);
-
-/*! \brief Flush any residual bit to the output buffer.
-    \param s A pointer to the bitstream context.
-    \param c A pointer to the bitstream output buffer. */
-void bitstream_flush(bitstream_state_t *s, uint8_t **c);
-
-void bitstream_flush2(bitstream_state_t *s, uint8_t **c);
-
-/*! \brief Initialise a bitstream context.
-    \param s A pointer to the bitstream context.
-    \return A pointer to the bitstream context. */
-bitstream_state_t *bitstream_init(bitstream_state_t *s);
-
-#if defined(__cplusplus)
-}
-#endif
-
-#endif
-/*- End of file ------------------------------------------------------------*/

src/libs/g722/g722_dc_restore.h

@@ -1,157 +0,0 @@
-/*
- * VoIPcodecs - a series of DSP components for telephony
- *
- * dc_restore.h - General telephony routines to restore the zero D.C.
- *                level to audio which has a D.C. bias.
- *
- * Written by Steve Underwood <steveu@coppice.org>
- *
- * Copyright (C) 2001 Steve Underwood
- *
- * All rights reserved.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2, as
- * published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- * $Id: dc_restore.h,v 1.18 2007/04/08 08:16:17 steveu Exp $
- */
-
-/*! \file */
-
-#if !defined(_SPANDSP_DC_RESTORE_H_)
-#define _SPANDSP_DC_RESTORE_H_
-
-/*! \page dc_restore_page Removing DC bias from a signal
-
-\section dc_restore_page_sec_1 What does it do?
-
-Telecoms signals often contain considerable DC, but DC upsets a lot of signal
-processing functions. Placing a zero DC restorer at the front of the processing
-chain can often simplify the downstream processing. 
-
-\section dc_restore_page_sec_2 How does it work?
-
-The DC restorer uses a leaky integrator to provide a long-ish term estimate of
-the DC bias in the signal. A 32 bit estimate is used for the 16 bit audio, so
-the noise introduced by the estimation can be keep in the lower bits, and the 16
-bit DC value, which is subtracted from the signal, is fairly clean. The
-following code fragment shows the algorithm used. dc_bias is a 32 bit integer,
-while the sample and the resulting clean_sample are 16 bit integers. 
-
-    dc_bias += ((((int32_t) sample << 15) - dc_bias) >> 14);
-    clean_sample = sample - (dc_bias >> 15); 
-*/
-
-/*!
-    Zero DC restoration descriptor. This defines the working state for a single
-    instance of DC content filter.
-*/
-typedef struct
-{
-    int32_t state;
-} dc_restore_state_t;
-
-#if defined(__cplusplus)
-extern "C"
-{
-#endif
-
-static __inline__ void dc_restore_init(dc_restore_state_t *dc)
-{
-    dc->state = 0;
-}
-/*- End of function --------------------------------------------------------*/
-
-static __inline__ int16_t dc_restore(dc_restore_state_t *dc, int16_t sample)
-{
-    dc->state += ((((int32_t) sample << 15) - dc->state) >> 14);
-    return (int16_t) (sample - (dc->state >> 15));
-}
-/*- End of function --------------------------------------------------------*/
-
-static __inline__ int16_t dc_restore_estimate(dc_restore_state_t *dc)
-{
-    return (int16_t) (dc->state >> 15);
-}
-/*- End of function --------------------------------------------------------*/
-
-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 > INT16_MAX)
-        return  INT16_MAX;
-    return  INT16_MIN;
-}
-/*- End of function --------------------------------------------------------*/
-
-/*#ifdef _MSC_VER
-__inline float rintf (float flt)
-{
-	_asm
-	{	fld flt
-		frndint
-	}
-}
-
-__inline double rint(double dbl)
-{
-    __asm 
-	{
-        fld dbl
-        frndint
-    }
-}
-
-__inline long lrintf (float flt)
-{
-	long retval;
-	_asm
-	{	fld flt
-		fistp retval
-	}
-	return retval;
-}
-#endif
-*/
-
-static __inline__ int16_t fsaturatef(float famp)
-{
-    if (famp > 32767.0)
-        return  INT16_MAX;
-    if (famp < -32768.0)
-        return  INT16_MIN;
-    return (int16_t) rintf(famp);
-}
-/*- End of function --------------------------------------------------------*/
-
-static __inline__ int16_t fsaturate(double damp)
-{
-    if (damp > 32767.0)
-        return  INT16_MAX;
-    if (damp < -32768.0)
-        return  INT16_MIN;
-    return (int16_t) rint(damp);
-}
-/*- End of function --------------------------------------------------------*/
-
-#if defined(__cplusplus)
-}
-#endif
-
-#endif
-/*- End of file ------------------------------------------------------------*/

src/libs/g722/g722_decode.c

@@ -1,5 +1,5 @@
 /*
- * VoIPcodecs - a series of DSP components for telephony
+ * SpanDSP - a series of DSP components for telephony
  *
  * g722_decode.c - The ITU G.722 codec, decode part.
  *
@@ -7,21 +7,10 @@
  *
  * 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 -
- * This program is free software; you can redistribute it and/or modify
+ *  even the slimy ones who might try to proprietize my work and use it
- * it under the terms of the GNU General Public License version 2, or
+ *  to my detriment.
- * the Lesser GNU General Public License version 2.1, as published by
- * the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  *
  * Based in part on a single channel G.722 codec which is:
  *
@@ -29,7 +18,11 @@
  * Computer Science, Speech Group
  * Chengxiang Lu and Alex Hauptmann
  *
- * $Id: g722_decode.c,v 1.20 2008/02/09 15:32:56 steveu Exp $
+ * $Id$
+ *
+ * rtphone: taken from Asterisk (codecs/g722). The only change is that the
+ * decoder saturates its output instead of letting a corrupt stream wrap
+ * around int16 (Asterisk casts (rlow << 1) and (xout >> 11) directly).
  */
 
 /*! \file */
@@ -38,18 +31,37 @@
 #include <config.h>
 #endif
 
-#include "g722_inttypes.h"
+#include <stdio.h>
+#include <inttypes.h>
 #include <memory.h>
 #include <stdlib.h>
-#if defined(HAVE_TGMATH_H)
+#if 0
 #include <tgmath.h>
 #endif
-#include <math.h>
 
-#include "g722_telephony.h"
-#include "g722_dc_restore.h"
 #include "g722.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 > INT16_MAX)
+        return  INT16_MAX;
+    return  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)
@@ -177,14 +189,8 @@ int g722_decode_release(g722_decode_state_t *s)
 
 int g722_decode(g722_decode_state_t *s, int16_t amp[], const uint8_t g722_data[], int len)
 {
-    static const int wl[8] =
+    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 };
-        -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,
@@ -193,18 +199,9 @@ int g722_decode(g722_decode_state_t *s, int16_t amp[], const uint8_t g722_data[]
         3228, 3298, 3371, 3444, 3520, 3597, 3676,
         3756, 3838, 3922, 4008
     };
-    static const int wh[3] =
+    static const int wh[3] = {0, -214, 798};
-    {
+    static const int rh2[4] = {2, 1, 2, 1};
-        0, -214, 798
+    static const int qm2[4] = {-7408, -1616,  7408,   1616};
-    };
-    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,
@@ -246,6 +243,7 @@ int g722_decode(g722_decode_state_t *s, int16_t amp[], const uint8_t g722_data[]
     {
            3,  -11,   12,   32, -210,  951, 3876, -805,  362, -156,   53,  -11,
     };
+
     int dlowt;
     int rlow;
     int ihigh;
@@ -369,19 +367,20 @@ int g722_decode(g722_decode_state_t *s, int16_t amp[], const uint8_t g722_data[]
 
         if (s->itu_test_mode)
         {
-            amp[outlen++] = (int16_t) (rlow << 1);
+            amp[outlen++] = saturate(rlow << 1);
             amp[outlen++] = (int16_t) (rhigh << 1);
         }
         else
         {
             if (s->eight_k)
             {
-                amp[outlen++] = (int16_t) rlow;
+                amp[outlen++] = saturate(rlow << 1);
             }
             else
             {
                 /* Apply the receive QMF */
-                memcpy(s->x, &s->x[2], 22*sizeof(s->x[0]));
+                for (i = 0;  i < 22;  i++)
+                    s->x[i] = s->x[i + 2];
                 s->x[22] = rlow + rhigh;
                 s->x[23] = rlow - rhigh;
 
@@ -392,8 +391,8 @@ int g722_decode(g722_decode_state_t *s, int16_t amp[], const uint8_t g722_data[]
                     xout2 += s->x[2*i]*qmf_coeffs[i];
                     xout1 += s->x[2*i + 1]*qmf_coeffs[11 - i];
                 }
-                amp[outlen++] = (int16_t) (xout1 >> 12);
+                amp[outlen++] = saturate(xout1 >> 11);
-                amp[outlen++] = (int16_t) (xout2 >> 12);
+                amp[outlen++] = saturate(xout2 >> 11);
             }
         }
     }

src/libs/g722/g722_encode.c

@@ -1,5 +1,5 @@
 /*
- * VoIPcodecs - a series of DSP components for telephony
+ * SpanDSP - a series of DSP components for telephony
  *
  * g722_encode.c - The ITU G.722 codec, encode part.
  *
@@ -9,19 +9,10 @@
  *
  * All rights reserved.
  *
- * This program is free software; you can redistribute it and/or modify
+ *  Despite my general liking of the GPL, I place my own contributions
- * it under the terms of the GNU General Public License version 2, or
+ *  to this code in the public domain for the benefit of all mankind -
- * the Lesser GNU General Public License version 2.1, as published by
+ *  even the slimy ones who might try to proprietize my work and use it
- * the Free Software Foundation.
+ *  to my detriment.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  *
  * Based on a single channel 64kbps only G.722 codec which is:
  *
@@ -29,7 +20,7 @@
  * Computer Science, Speech Group
  * Chengxiang Lu and Alex Hauptmann
  *
- * $Id: g722_encode.c,v 1.18 2008/02/09 15:32:56 steveu Exp $
+ * $Id$
  */
 
 /*! \file */
@@ -38,18 +29,37 @@
 #include <config.h>
 #endif
 
-#include "g722_inttypes.h"
+#include <stdio.h>
+#include <inttypes.h>
 #include <memory.h>
 #include <stdlib.h>
-#if defined(HAVE_TGMATH_H)
+#if 0
 #include <tgmath.h>
 #endif
-#include <math.h>
 
-#include "g722_telephony.h"
-#include "g722_dc_restore.h"
 #include "g722.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 > INT16_MAX)
+        return  INT16_MAX;
+    return  INT16_MIN;
+}
+/*- End of function --------------------------------------------------------*/
+
 static void block4(g722_encode_state_t *s, int band, int d)
 {
     int wd1;
@@ -269,13 +279,14 @@ int g722_encode(g722_encode_state_t *s, uint8_t g722_data[], const int16_t amp[]
         {
             if (s->eight_k)
             {
-                xlow = amp[j++];
+                xlow = amp[j++] >> 1;
             }
             else
             {
                 /* Apply the transmit QMF */
                 /* Shuffle the buffer down */
-                memcpy(s->x, &s->x[2], 22*sizeof(s->x[0]));
+                for (i = 0;  i < 22;  i++)
+                    s->x[i] = s->x[i + 2];
                 s->x[22] = amp[j++];
                 s->x[23] = amp[j++];
 
@@ -287,8 +298,8 @@ int g722_encode(g722_encode_state_t *s, uint8_t g722_data[], const int16_t amp[]
                     sumodd += s->x[2*i]*qmf_coeffs[i];
                     sumeven += s->x[2*i + 1]*qmf_coeffs[11 - i];
                 }
-                xlow = (sumeven + sumodd) >> 13;
+                xlow = (sumeven + sumodd) >> 14;
-                xhigh = (sumeven - sumodd) >> 13;
+                xhigh = (sumeven - sumodd) >> 14;
             }
         }
         /* Block 1L, SUBTRA */

src/libs/g722/g722_inttypes.h

@@ -1,103 +0,0 @@
-/*
- * SpanDSP - a series of DSP components for telephony
- *
- * inttypes.h - a fudge for MSVC, which lacks this header
- *
- * Written by Steve Underwood <steveu@coppice.org>
- *
- * Copyright (C) 2006 Michael Jerris
- *
- *
- * This file is released in the public domain.
- *
- */
-
-#if !defined(_INTTYPES_H_)
-#define _INTTYPES_H_
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-#ifdef _MSC_VER
-  #if (_MSC_VER >= 1400) // VC8+
-    #ifndef _CRT_SECURE_NO_DEPRECATE
-      #define _CRT_SECURE_NO_DEPRECATE
-    #endif
-    #ifndef _CRT_NONSTDC_NO_DEPRECATE
-      #define _CRT_NONSTDC_NO_DEPRECATE
-    #endif
-  #endif // VC8+
-
-  #include <windows.h>
-  typedef unsigned __int8		uint8_t;
-  typedef unsigned __int16	uint16_t;
-  typedef unsigned __int32	uint32_t;
-  typedef unsigned __int64    uint64_t;
-  typedef __int8		int8_t;
-  typedef __int16		int16_t;
-  typedef __int32		int32_t;
-  typedef __int64		int64_t;
-  #define inline __inline
-  #define __inline__ __inline
-  #define  INT16_MAX   0x7fff 
-  #define  INT16_MIN   (-INT16_MAX - 1) 
-  #define _MMX_H_
-
-  /* disable the following warnings 
-   * C4100: The formal parameter is not referenced in the body of the function. The unreferenced parameter is ignored. 
-   * C4200: Non standard extension C zero sized array
-   * C4706: assignment within conditional expression
-   * C4244: conversion from 'type1' to 'type2', possible loss of data
-   * C4295: array is too small to include a terminating null character
-   * C4125: decimal digit terminates octal escape sequence
-   */
-  #pragma warning(disable:4100 4200 4706 4295 4125)
-
-  //#pragma comment(lib, "ws2_32.lib")
-
-  #define strncasecmp _strnicmp
-  #define strcasecmp _stricmp
-  #define snprintf _snprintf
-
-  #if !defined(INFINITY)
-    #define INFINITY 0x7fffffff
-  #endif
-#else
-  #include <stdint.h>
-#endif
-
-#define PACKAGE "voipcodecs"
-#define VERSION "0.0.1andabit"
-
-#ifndef INT32_MAX
-  #define INT32_MAX	(2147483647)
-#endif
-#ifndef INT32_MIN
-  #define INT32_MIN	(-2147483647 - 1)
-#endif
-
-#define PRId8 "d"
-#define PRId16 "d"
-#ifndef PRId32
-# define PRId32 "ld"
-#endif
-
-#ifndef PRId64
-# define PRId64 "lld"
-#endif
-
-#define PRIu8 "u"
-#define PRIu16 "u"
-#ifndef PRIu32
-# define PRIu32 "lu"
-#endif
-#ifndef PRIu64
-# define PRIu64 "llu"
-#endif
-#ifdef __cplusplus
-}
-#endif
-
-#endif // _INTTYPES_H_
-

src/libs/g722/g722_telephony.h

@@ -1,74 +0,0 @@
-/*
- * VoIPcodecs - a series of DSP components for telephony
- *
- * telephony.h - some very basic telephony definitions
- *
- * Written by Steve Underwood <steveu@coppice.org>
- *
- * Copyright (C) 2003 Steve Underwood
- *
- * All rights reserved.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the Lesser GNU General Public License version 2.1, as
- * published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- * $Id: telephony.h,v 1.10 2007/04/05 19:20:50 steveu Exp $
- */
-
-#if !defined(_SPANDSP_TELEPHONY_H_)
-#define _SPANDSP_TELEPHONY_H_
-
-#define SAMPLE_RATE             8000
-
-/* This is based on A-law, but u-law is only 0.03dB different */
-#define DBM0_MAX_POWER          (3.14f + 3.02f)
-#define DBM0_MAX_SINE_POWER     (3.14f)
-/* This is based on the ITU definition of dbOv in G.100.1 */
-#define DBOV_MAX_POWER          (0.0f)
-#define DBOV_MAX_SINE_POWER     (-3.02f)
-
-/*! \brief A handler for pure receive. The buffer cannot be altered. */
-typedef int (span_rx_handler_t)(void *s, const int16_t amp[], int len);
-
-/*! \brief A handler for receive, where the buffer can be altered. */
-typedef int (span_mod_handler_t)(void *s, int16_t amp[], int len);
-
-/*! \brief A handler for transmit, where the buffer will be filled. */
-typedef int (span_tx_handler_t)(void *s, int16_t amp[], int max_len);
-
-#define ms_to_samples(t)    (((t)*SAMPLE_RATE)/1000)
-
-#if !defined(FALSE)
-#define FALSE 0
-#endif
-#if !defined(TRUE)
-#define TRUE (!FALSE)
-#endif
-
-#include <assert.h>
-#if (_MSC_VER >= 1400)			// VC8+
-#define vc_assert(expr) assert(expr);__analysis_assume( expr )
-#else
-#define vc_assert(expr) assert(expr)
-#endif
-
-#if defined(__cplusplus)
-/* C++ doesn't seem to have sane rounding functions/macros yet */
-#ifndef _MSC_VER
-#define lrint(x) ((long int) (x))
-#define lrintf(x) ((long int) (x))
-#endif
-#endif
-
-#endif
-/*- End of file ------------------------------------------------------------*/

src/libs/g722/g722codec.c

@@ -1,300 +0,0 @@
-/*
- * G.722 Plugin codec for OpenH323/OPAL
- *
- * Copyright (C) 2008 by Hermon Labs, All Rights Reserved
- *
- * The contents of this file are subject to the Mozilla Public License
- * Version 1.0 (the "License"); you may not use this file except in
- * compliance with the License. You may obtain a copy of the License at
- * http://www.mozilla.org/MPL/
- *
- * Software distributed under the License is distributed on an "AS IS"
- * basis, WITHOUT WARRANTY OF ANY KIND, either express or implied. See
- * the License for the specific language governing rights and limitations
- * under the License.
- *
- * The Original Code is Open Phone Abstraction Library.
- *
- * The Initial Developer of the Original Code is Eugene Mednikov
- *
- * Contributor(s): ______________________________________.
- *
- * $Revision: 23265 $
- * $Author: rjongbloed $
- * $Date: 2009-08-25 04:25:58 +0300 (вт, 25 серп 2009) $
- */
-
-#ifndef PLUGIN_CODEC_DLL_EXPORTS
-#include "plugin-config.h"
-#endif
-
-#include <codec/opalplugin.h>
-
-#include "VoIPCodecs/inttypes.h"
-#include "VoIPCodecs/g722.h"
-
-
-#define INCLUDE_SDP_16000_VERSION 0
-
-/* Due to a spec error, clock rate is 8kHz even though this is 16kHz codec,
-   see RFC3551/4.5.2. So some of these values have to be lied about so that
-   the OPAL system gets it right.
-  */
-#define CLOCK_RATE              8000  // Clock rate is not samples/second in this case!
-#define BITS_PER_SECOND         64000 // raw bits per second
-#define FRAME_TIME              1000  // Microseconds in a millisecond
-#define SAMPLES_PER_FRAME       16    // Samples in a millisecond
-#define	BYTES_PER_FRAME         8     // Bytes in a millisecond
-#define MAX_FRAMES_PER_PACKET   90    // 90 milliseconds, which means RTP packets smaller than 1500 bytes typical LAN maximum
-#define PREF_FRAMES_PER_PACKET  20    // 20 milliseconds
-
-static const char L16Desc[]  = "PCM-16-16kHz"; // Cannot use "L16" as usual, force 16kHz PCM
-static const char g722[]     = "G.722-64k";
-static const char sdpG722[]  = "G722";
-
-#if INCLUDE_SDP_16000_VERSION
-static const char g722_16[]  = "G.722-16kHz";
-#endif
-
-#define PAYLOAD_CODE 9
-
-
-/////////////////////////////////////////////////////////////////////////////
-
-static void * create_encoder(const struct PluginCodec_Definition * codec)
-{
-  return g722_encode_init(NULL, BITS_PER_SECOND, 0);
-}
-
-static void destroy_encoder(const struct PluginCodec_Definition * codec, void * context)
-{
-  g722_encode_release(context);
-}
-
-static int encode(const struct PluginCodec_Definition * codec,
-                                           void * context,
-                                     const void * from, 
-                                       unsigned * fromLen, 
-                                           void * to, 
-                                       unsigned * toLen, 
-                                   unsigned int * flag) 
-{ 
-  g722_encode_state_t * state = context;
-
-  if (*toLen < *fromLen / 4)
-    return 0; // Destination buffer not big enough
-
-  *toLen = g722_encode(state, to, from, *fromLen / 2);
-  return 1; 
-} 
-
-
-/////////////////////////////////////////////////////////////////////////////
-
-static void * create_decoder(const struct PluginCodec_Definition * codec)
-{
-  return g722_decode_init(NULL, BITS_PER_SECOND, 0);
-}
-
-static void destroy_decoder(const struct PluginCodec_Definition * codec, void * context)
-{
-  g722_decode_release(context);
-}
-
-
-static int decode(const struct PluginCodec_Definition * codec, 
-                                           void * _context, 
-                                     const void * from, 
-                                       unsigned * fromLen, 
-                                           void * to, 
-                                       unsigned * toLen, 
-                                   unsigned int * flag) 
-{
-  g722_decode_state_t * state = _context;
-
-  if (*toLen < *fromLen * 4)
-    return 0; // Destination buffer not big enough
-
-  *toLen = g722_decode(state, to, from, *fromLen) * 2; 
-  return 1;
-} 
-
-
-/////////////////////////////////////////////////////////////////////////////
-
-static struct PluginCodec_information licenseInfo = {
-  1084181196,                                   // timestamp = Mon 10 May 2004 09:26:36 AM UTC
-
-  "Eugene Mednikov",                            // source code author
-  "1.0",                                        // source code version
-  "em@hermonlabs.com",                          // source code email
-  "http://www.hermonlabs.com",                  // source code URL
-  "Copyright (C) 2008 by Hermon Labs, All Rights Reserved", // source code copyright
-  "MPL 1.0",                                    // source code license
-  PluginCodec_License_MPL,                      // source code license
-  
-  "ITU G.722",                                  // codec description
-  "Steve Underwood",                            // codec author
-  NULL,                                         // codec version
-  "steveu@coppice.org",                         // codec email
-  NULL,                                         // codec URL
-  NULL,                                         // codec copyright information
-  NULL,                                         // codec license
-  PluginCodec_License_LGPL                      // codec license code
-};
-
-/////////////////////////////////////////////////////////////////////////////
-
-static struct PluginCodec_Definition g722CodecDefn[] =
-{
-#if INCLUDE_SDP_16000_VERSION
-  // Include a version for SIP/SDP that indicates the more logical, though
-  // incorrect by RFC3551/4.5.2, 16000Hz version of G.722. We use dynamic
-  // payload types to avoid conflict with the compliant 8000Hz version.
-  { 
-    // encoder 
-    PLUGIN_CODEC_VERSION_WIDEBAND,        // codec API version
-    &licenseInfo,                         // license information
-
-    PluginCodec_MediaTypeAudio |          // audio codec
-    PluginCodec_InputTypeRaw |            // raw input data
-    PluginCodec_OutputTypeRaw |           // raw output data
-    PluginCodec_RTPTypeDynamic,           // specified RTP type
-
-    g722_16,                              // text decription
-    L16Desc,                              // source format
-    g722_16,                              // destination format
-
-    0,                                    // user data
-
-    16000,                                // samples per second
-    BITS_PER_SECOND,                      // raw bits per second
-    FRAME_TIME,                           // microseconds per frame
-    SAMPLES_PER_FRAME,                    // samples per frame
-    BYTES_PER_FRAME,                      // bytes per frame
-    PREF_FRAMES_PER_PACKET,               // recommended number of frames per packet
-    MAX_FRAMES_PER_PACKET,                // maximum number of frames per packe
-    PAYLOAD_CODE,                         // IANA RTP payload code
-    sdpG722,                              // RTP payload name
-
-    create_encoder,                       // create codec function
-    destroy_encoder,                      // destroy codec
-    encode,                               // encode/decode
-    NULL,                                 // codec controls
-
-    0,                                    // h323CapabilityType 
-    NULL                                  // h323CapabilityData
-  },
-
-  { 
-    // decoder 
-    PLUGIN_CODEC_VERSION_WIDEBAND,        // codec API version
-    &licenseInfo,                         // license information
-
-    PluginCodec_MediaTypeAudio |          // audio codec
-    PluginCodec_InputTypeRaw |            // raw input data
-    PluginCodec_OutputTypeRaw |           // raw output data
-    PluginCodec_RTPTypeDynamic,           // specified RTP type
-
-    g722_16,                              // text decription
-    g722_16,                              // source format
-    L16Desc,                              // destination format
-
-    0,                                    // user data
-
-    16000,                                // samples per second
-    BITS_PER_SECOND,                      // raw bits per second
-    FRAME_TIME,                           // microseconds per frame
-    SAMPLES_PER_FRAME,                    // samples per frame
-    BYTES_PER_FRAME,                      // bytes per frame
-    PREF_FRAMES_PER_PACKET,               // recommended number of frames per packet
-    MAX_FRAMES_PER_PACKET,                // maximum number of frames per packe
-    PAYLOAD_CODE,                         // IANA RTP payload code
-    sdpG722,                              // RTP payload name
-
-    create_decoder,                       // create codec function
-    destroy_decoder,                      // destroy codec
-    decode,                               // encode/decode
-    NULL,                                 // codec controls
-
-    0,                                    // h323CapabilityType 
-    NULL                                  // h323CapabilityData
-  },
-#endif
-
-  // Standards compliant version
-  { 
-    // encoder 
-    PLUGIN_CODEC_VERSION_WIDEBAND,        // codec API version
-    &licenseInfo,                         // license information
-
-    PluginCodec_MediaTypeAudio |          // audio codec
-    PluginCodec_InputTypeRaw |            // raw input data
-    PluginCodec_OutputTypeRaw |           // raw output data
-    PluginCodec_RTPTypeExplicit,          // specified RTP type
-
-    g722,                                 // text decription
-    L16Desc,                              // source format
-    g722,                                 // destination format
-
-    0,                                    // user data
-
-    CLOCK_RATE,                           // samples per second
-    BITS_PER_SECOND,                      // raw bits per second
-    FRAME_TIME,                           // microseconds per frame
-    SAMPLES_PER_FRAME,                    // samples per frame
-    BYTES_PER_FRAME,                      // bytes per frame
-    PREF_FRAMES_PER_PACKET,               // recommended number of frames per packet
-    MAX_FRAMES_PER_PACKET,                // maximum number of frames per packe
-    PAYLOAD_CODE,                         // IANA RTP payload code
-    sdpG722,                              // RTP payload name
-
-    create_encoder,                       // create codec function
-    destroy_encoder,                      // destroy codec
-    encode,                               // encode/decode
-    NULL,                                 // codec controls
-
-    PluginCodec_H323AudioCodec_g722_64k,  // h323CapabilityType 
-    NULL                                  // h323CapabilityData
-  },
-
-  { 
-    // decoder 
-    PLUGIN_CODEC_VERSION_WIDEBAND,        // codec API version
-    &licenseInfo,                         // license information
-
-    PluginCodec_MediaTypeAudio |          // audio codec
-    PluginCodec_InputTypeRaw |            // raw input data
-    PluginCodec_OutputTypeRaw |           // raw output data
-    PluginCodec_RTPTypeExplicit,          // specified RTP type
-
-    g722,                                 // text decription
-    g722,                                 // source format
-    L16Desc,                              // destination format
-
-    0,                                    // user data
-
-    CLOCK_RATE,                           // samples per second
-    BITS_PER_SECOND,                      // raw bits per second
-    FRAME_TIME,                           // microseconds per frame
-    SAMPLES_PER_FRAME,                    // samples per frame
-    BYTES_PER_FRAME,                      // bytes per frame
-    PREF_FRAMES_PER_PACKET,               // recommended number of frames per packet
-    MAX_FRAMES_PER_PACKET,                // maximum number of frames per packe
-    PAYLOAD_CODE,                         // IANA RTP payload code
-    sdpG722,                              // RTP payload name
-
-    create_decoder,                       // create codec function
-    destroy_decoder,                      // destroy codec
-    decode,                               // encode/decode
-    NULL,                                 // codec controls
-
-    PluginCodec_H323AudioCodec_g722_64k,  // h323CapabilityType 
-    NULL                                  // h323CapabilityData
-  }
-};
-
-
-PLUGIN_CODEC_IMPLEMENT_ALL(G722, g722CodecDefn, PLUGIN_CODEC_VERSION_WIDEBAND)
-
-/////////////////////////////////////////////////////////////////////////////