gsmcodec.cxx

mgcp协议源代码。支持多种编码：g711

/*
 * gsmcodec.cxx
 *
 * H.323 protocol handler
 *
 * Open H323 Library
 *
 * Copyright (c) 1998-2000 Equivalence Pty. Ltd.
 *
 * 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 H323 Library.
 *
 * The Initial Developer of the Original Code is Equivalence Pty. Ltd.
 *
 * Portions of this code were written with the assisance of funding from
 * Vovida Networks, Inc. http://www.vovida.com.
 *
 * Contributor(s): ______________________________________.
 *
 * $Log: gsmcodec.cxx,v $
 * Revision 1.14  2000/07/13 17:24:33  robertj
 * Fixed format name to be consistent will all others.
 *
 * Revision 1.13  2000/07/12 10:25:37  robertj
 * Renamed all codecs so obvious whether software or hardware.
 *
 * Revision 1.12  2000/07/09 14:55:15  robertj
 * Bullet proofed packet count so incorrect capabilities does not crash us.
 *
 * Revision 1.11  2000/05/10 04:05:33  robertj
 * Changed capabilities so has a function to get name of codec, instead of relying on PrintOn.
 *
 * Revision 1.10  2000/05/02 04:32:26  robertj
 * Fixed copyright notice comment.
 *
 * Revision 1.9  2000/03/21 03:06:49  robertj
 * Changes to make RTP TX of exact numbers of frames in some codecs.
 *
 * Revision 1.8  1999/12/31 00:05:36  robertj
 * Added Microsoft ACM G.723.1 codec capability.
 *
 * Revision 1.7  1999/11/20 00:53:47  robertj
 * Fixed ability to have variable sized frames in single RTP packet under G.723.1
 *
 * Revision 1.6  1999/10/08 09:59:03  robertj
 * Rewrite of capability for sending multiple audio frames
 *
 * Revision 1.5  1999/10/08 08:30:45  robertj
 * Fixed maximum packet size, must be less than 256
 *
 * Revision 1.4  1999/10/08 04:58:38  robertj
 * Added capability for sending multiple audio frames in single RTP packet
 *
 * Revision 1.3  1999/09/27 01:13:09  robertj
 * Fixed old GNU compiler support
 *
 * Revision 1.2  1999/09/23 07:25:12  robertj
 * Added open audio and video function to connection and started multi-frame codec send functionality.
 *
 * Revision 1.1  1999/09/08 04:05:49  robertj
 * Added support for video capabilities & codec, still needs the actual codec itself!
 *
 */

#include <ptlib.h>
#include "gsmcodec.h"
#include "h245.h"
#include "rtp.h"


#define new PNEW


H323_GSM0610Capability::H323_GSM0610Capability()
  : H323AudioCapability(7, 4)
{
  maxFrameSize = H323_GSM0610Codec::BytesPerFrame;
}


PObject * H323_GSM0610Capability::Clone() const
{
  return new H323_GSM0610Capability(*this);
}


PString H323_GSM0610Capability::GetFormatName() const
{
  return "GSM-06.10{sw}";
}


unsigned H323_GSM0610Capability::GetSubType() const
{
  return H245_AudioCapability::e_gsmFullRate;
}


BOOL H323_GSM0610Capability::OnSendingPDU(H245_AudioCapability & cap,
                                          unsigned packetSize) const
{
  cap.SetTag(H245_AudioCapability::e_gsmFullRate);

  H245_GSMAudioCapability & gsm = cap;
  gsm.m_audioUnitSize = packetSize*H323_GSM0610Codec::BytesPerFrame;
  return TRUE;
}


BOOL H323_GSM0610Capability::OnReceivedPDU(const H245_AudioCapability & cap,
                                           unsigned & packetSize)
{
  if (cap.GetTag() != H245_AudioCapability::e_gsmFullRate)
    return FALSE;

  const H245_GSMAudioCapability & gsm = cap;
  packetSize = gsm.m_audioUnitSize / H323_GSM0610Codec::BytesPerFrame;
  if (packetSize == 0)
    packetSize = 1;
  return TRUE;
}


H323Codec * H323_GSM0610Capability::CreateCodec(H323Codec::Direction direction) const
{
  return new H323_GSM0610Codec(direction);
}


/////////////////////////////////////////////////////////////////////////////

extern "C" gsm_state * gsm_create();
//gsm_state * gsm_create() { return NULL; }

H323_GSM0610Codec::H323_GSM0610Codec(Direction dir)
  : H323FramedAudioCodec(dir, SamplesPerFrame, BytesPerFrame)
{
  rtpPayloadType = RTP_DataFrame::GSM;
  gsm = gsm_create();
  PTRACE(3, "Codec\tGSM " << (dir == Encoder ? "en" : "de")
         << "coder created");
}


extern "C" void gsm_destroy(gsm_state *);
//void gsm_destroy(gsm_state *) { }

H323_GSM0610Codec::~H323_GSM0610Codec()
{
  gsm_destroy(gsm);
}


extern "C" void gsm_encode(gsm_state *, const short *, BYTE *);
//void gsm_encode(gsm_state *, const short *, BYTE *) { }

BOOL H323_GSM0610Codec::EncodeFrame(BYTE * buffer, unsigned &)
{
  gsm_encode(gsm, sampleBuffer, buffer);
  return TRUE;
}


extern "C" void gsm_decode(gsm_state *, const BYTE *, short *);
//void gsm_decode(gsm_state *, const BYTE *, short *) { }

BOOL H323_GSM0610Codec::DecodeFrame(const BYTE * buffer, unsigned length, unsigned &)
{
  if (length < BytesPerFrame)
    return FALSE;

  gsm_decode(gsm, buffer, sampleBuffer.GetPointer());
  return TRUE;
}


/////////////////////////////////////////////////////////////////////////////