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[openh323.git] / include / lid.h

/*
 * lid.h
 *
 * Line Interface Device
 *
 * Open Phone Abstraction Library
 *
 * Copyright (c) 1999-2000 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 
 * Quicknet Technologies, Inc. http://www.quicknet.net.
 * 
 * Contributor(s): ______________________________________.
 *
 * $Log$
 * Revision 1.58  2005/11/30 13:05:01  csoutheren
 * Changed tags for Doxygen
 *
 * Revision 1.57  2005/07/03 13:59:43  shorne
 * Add LID type identifier (for Plugin LID support)
 *
 * Revision 1.56  2003/04/29 08:29:13  robertj
 * Cleaned up documentation for new wink duration functions.
 *
 * Revision 1.55  2003/04/28 01:47:52  dereks
 * Add ability to set/get wink duration for ixj device.
 *
 * Revision 1.54  2003/03/05 06:26:40  robertj
 * Added function to play a WAV file to LID, thanks Pietro Ravasio
 *
 * Revision 1.53  2002/11/05 04:26:48  robertj
 * Imported RingLine() by array from OPAL.
 *
 * Revision 1.52  2002/10/29 00:46:04  dereks
 * Fix typo in comments.
 *
 * Revision 1.51  2002/09/16 01:14:15  robertj
 * Added #define so can select if #pragma interface/implementation is used on
 *   platform basis (eg MacOS) rather than compiler, thanks Robert Monaghan.
 *
 * Revision 1.50  2002/09/03 06:19:37  robertj
 * Normalised the multi-include header prevention ifdef/define symbol.
 *
 * Revision 1.49  2002/08/05 10:03:47  robertj
 * Cosmetic changes to normalise the usage of pragma interface/implementation.
 *
 * Revision 1.48  2002/06/27 08:52:53  robertj
 * Fixed typo and naming convention for Cisco G.723.1 annex A capability.
 *
 * Revision 1.47  2002/06/26 05:45:41  robertj
 * Added capability for Cisco IOS non-standard name for G.723.1 Annex A so
 *   can now utilise SID frames with Cisco gateways.
 *
 * Revision 1.46  2002/06/25 08:30:08  robertj
 * Changes to differentiate between stright G.723.1 and G.723.1 Annex A using
 *   the OLC dataType silenceSuppression field so does not send SID frames
 *   to receiver codecs that do not understand them.
 *
 * Revision 1.45  2002/05/09 06:26:30  robertj
 * Added fuction to get the current audio enable state for line in device.
 * Changed IxJ EnableAudio() semantics so is exclusive, no direct switching
 *   from PSTN to POTS and vice versa without disabling the old one first.
 *
 * Revision 1.44  2002/01/23 01:58:25  robertj
 * Added function to determine if codecs raw data channel is native format.
 *
 * Revision 1.43  2001/07/19 05:54:27  robertj
 * Updated interface to xJACK drivers to utilise cadence and filter functions
 *   for dial tone, busy tone and ringback tone detection.
 *
 * Revision 1.42  2001/05/22 00:31:41  robertj
 * Changed to allow optional wink detection for line disconnect
 *
 * Revision 1.41  2001/03/29 23:42:34  robertj
 * Added ability to get average signal level for both receive and transmit.
 * Changed silence detection to use G.723.1 SID frames as indicator of
 *   silence instead of using the average energy and adaptive threshold.
 *
 * Revision 1.40  2001/02/13 05:02:39  craigs
 * Extended PlayDTMF to allow generation of additional simple tones
 *
 * Revision 1.39  2001/02/09 05:16:24  robertj
 * Added #pragma interface for GNU C++.
 *
 * Revision 1.38  2001/01/25 07:27:14  robertj
 * Major changes to add more flexible OpalMediaFormat class to normalise
 *   all information about media types, especially codecs.
 *
 * Revision 1.37  2001/01/24 05:34:49  robertj
 * Altered volume control range to be percentage, ie 100 is max volume.
 *
 * Revision 1.36  2001/01/11 05:39:44  robertj
 * Fixed usage of G.723.1 CNG 1 byte frames.
 *
 * Revision 1.35  2000/12/19 06:38:57  robertj
 * Fixed missing virtual on IsTonePlaying() function.
 *
 * Revision 1.34  2000/12/17 22:08:20  craigs
 * Changed GetCountryCodeList to return PStringList
 *
 * Revision 1.33  2000/12/11 01:23:32  craigs
 * Added extra routines to allow country string manipulation
 *
 * Revision 1.32  2000/11/30 08:48:35  robertj
 * Added functions to enable/disable Voice Activity Detection in LID's
 *
 * Revision 1.31  2000/11/26 23:12:18  craigs
 * Added hook flash detection API
 *
 * Revision 1.30  2000/11/24 10:46:12  robertj
 * Added a raw PCM dta mode for generating/detecting standard tones.
 * Modified the ReadFrame/WriteFrame functions to allow for variable length codecs.
 *
 * Revision 1.29  2000/11/20 03:15:13  craigs
 * Changed tone detection API slightly to allow detection of multiple
 * simultaneous tones
 * Added fax CNG tone to tone list
 *
 * Revision 1.28  2000/11/06 02:00:18  eokerson
 * Added support for AGC on ixj devices under Linux.
 *
 * Revision 1.27  2000/11/03 06:22:48  robertj
 * Added flag to IsLinePresent() to force slow test, guarenteeing correct value.
 *
 * Revision 1.26  2000/10/13 02:24:06  robertj
 * Moved frame reblocking code from LID channel to LID itself and added
 *    ReadBlock/WriteBlock functions to allow non integral frame sizes.
 *
 * Revision 1.25  2000/09/25 22:31:19  craigs
 * Added G.723.1 frame erasure capability
 *
 * Revision 1.24  2000/09/23 07:20:28  robertj
 * Fixed problem with being able to distinguish between sw and hw codecs in LID channel.
 *
 * Revision 1.23  2000/09/22 01:35:03  robertj
 * Added support for handling LID's that only do symmetric codecs.
 *
 * Revision 1.22  2000/08/31 13:14:40  craigs
 * Added functions to LID
 * More bulletproofing to Linux driver
 *
 * Revision 1.21  2000/08/30 23:24:24  robertj
 * Renamed string version of SetCountrCode() to SetCountryCodeName() to avoid
 *    C++ masking ancestor overloaded function when overriding numeric version.
 *
 * Revision 1.20  2000/08/18 04:17:48  robertj
 * Added Auto setting to AEC enum.
 *
 * Revision 1.19  2000/06/19 00:32:13  robertj
 * Changed functionf or adding all lid capabilities to not assume it is to an endpoint.
 *
 * Revision 1.18  2000/06/01 07:52:19  robertj
 * Changed some LID capability code back again so does not unneedfully break existing API.
 *
 * Revision 1.17  2000/05/30 10:19:17  robertj
 * Added function to add capabilities given a LID.
 * Improved LID capabilities so cannot create one that is not explicitly supported.
 *
 * Revision 1.16  2000/05/24 06:42:18  craigs
 * Added calls to get volume settings
 *
 * Revision 1.15  2000/05/10 04:05:26  robertj
 * Changed capabilities so has a function to get name of codec, instead of relying on PrintOn.
 *
 * Revision 1.14  2000/05/02 04:32:24  robertj
 * Fixed copyright notice comment.
 *
 * Revision 1.13  2000/04/14 17:18:25  robertj
 * Fixed problem with error reporting from LID hardware.
 *
 * Revision 1.12  2000/04/10 17:44:52  robertj
 * Added higher level "DialOut" function for PSTN lines.
 * Added hook flash function.
 *
 * Revision 1.11  2000/04/05 18:04:12  robertj
 * Changed caller ID code for better portability.
 *
 * Revision 1.10  2000/03/30 23:10:50  robertj
 * Fixed error in comments regarding GetFramerate() function.
 *
 * Revision 1.9  2000/03/29 20:54:19  robertj
 * Added function on LID to get available codecs.
 * Changed codec to use number of frames rather than number of bytes.
 *
 * Revision 1.8  2000/03/23 23:36:48  robertj
 * Added more calling tone detection functionality.
 *
 * Revision 1.7  2000/03/23 02:43:15  robertj
 * Fixed default DTMF timing.
 *
 * Revision 1.6  2000/03/22 17:18:49  robertj
 * Changed default DTMF tone string times.
 *
 * Revision 1.5  2000/03/21 03:06:48  robertj
 * Changes to make RTP TX of exact numbers of frames in some codecs.
 *
 * Revision 1.4  2000/01/13 12:39:29  robertj
 * Added string based country codes to LID.
 *
 * Revision 1.3  2000/01/13 04:03:45  robertj
 * Added video transmission
 *
 * Revision 1.2  2000/01/07 08:28:09  robertj
 * Additions and changes to line interface device base class.
 *
 * Revision 1.1  1999/12/23 23:02:35  robertj
 * File reorganision for separating RTP from H.323 and creation of LID for VPB support.
 *
 */

#ifndef __OPAL_LID_H
#define __OPAL_LID_H

#ifdef P_USE_PRAGMA
#pragma interface
#endif


#include "codecs.h"
#include "h323caps.h"


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

/** Line Interface Device abstraction.
    Note all functions in this device abstraction are assumed to be thread atomic.
 */
class OpalLineInterfaceDevice : public PObject
{
  PCLASSINFO(OpalLineInterfaceDevice, PObject);

  public:
    /**Construct a new line interface device.
      */
    OpalLineInterfaceDevice();

    /**Open the line interface device.
      */
    virtual BOOL Open(
      const PString & device      ///<  Device identifier name.
    ) = 0;

    /**Determine if the line interface device is open.
      */
    virtual BOOL IsOpen() const;

    /**Close the line interface device.
      */
    virtual BOOL Close();

    /**Determine the type of line interface device.
       This is a string indication of the card type for user interface
       display purposes or device specific control. The device should be
       as detailed as possible eg "Quicknet LineJACK".
      */
    virtual PString GetName() const = 0;

    /**Get the total number of lines supported by this device.
      */
    virtual unsigned GetLineCount() = 0;

    /**Get the type of the line.
       A "terminal" line is one where a call may terminate. For example a POTS
       line with a standard telephone handset on it would be a terminal line.
       The alternative is a "network" line, that is one connected to switched
       network eg the standard PSTN.
      */
    virtual BOOL IsLineTerminal(
      unsigned line   ///<  Number of line
    );


    /**Determine if a physical line is present on the logical line.
      */
    virtual BOOL IsLinePresent(
      unsigned line,      ///<  Number of line
      BOOL force = FALSE  ///<  Force test, do not optimise
    );


    /**Determine if line is currently off hook.
       This function implies that the state is debounced and that a return
       value of TRUE indicates that the phone is really off hook. That is
       hook flashes and winks are masked out.
      */
    virtual BOOL IsLineOffHook(
      unsigned line   ///<  Number of line
    ) = 0;

    /**Set the hook state of the line.
       Note that not be possible on a given line, for example a POTS line with
       a standard telephone handset. The hook state is determined by external
       hardware and cannot be changed by the software.
      */
    virtual BOOL SetLineOffHook(
      unsigned line,        ///<  Number of line
      BOOL newState = TRUE  ///<  New state to set
    ) = 0;

    /**Set the hook state of the line.
       This is the complement of SetLineOffHook().
      */
    virtual BOOL SetLineOnHook(
      unsigned line        ///<  Number of line
    ) { return SetLineOffHook(line, FALSE); }

    /**Set the hook state off then straight back on again.
       This will only operate if the line is currently off hook.
      */
    virtual BOOL HookFlash(
      unsigned line,              ///<  Number of line
      unsigned flashTime = 200    ///<  Time for hook flash in milliseconds
    );

    /**Return TRUE if a hook flash has been detected
      */
    virtual BOOL HasHookFlash(unsigned line);


    /**Determine if line is ringing.
       This function implies that the state is "debounced" and that a return
       value of TRUE indicates that the phone is still ringing and it is not
       simply a pause in the ring cadence.

       If cadence is not NULL then it is set with the bit pattern for the
       incoming ringing. Note that in this case the funtion may take a full
       sequence to return. If it is NULL it can be assumed that the function
       will return quickly.
      */
    virtual BOOL IsLineRinging(
      unsigned line,          ///<  Number of line
      DWORD * cadence = NULL  ///<  Cadence of incoming ring
    );

    /**Begin ringing local phone set with specified cadence.
       If cadence is zero then stops ringing.

       Note that not be possible on a given line, for example on a PSTN line
       the ring state is determined by external hardware and cannot be
       changed by the software.

       Also note that the cadence may be ignored by particular hardware driver
       so that only the zero or non-zero values are significant.
      */
    virtual BOOL RingLine(
      unsigned line,    ///<  Number of line
      DWORD cadence     ///<  Cadence bit map for ring pattern
    );

    /**Begin ringing local phone set with specified cadence.
       If nCadence is zero then stops ringing.

       Note that not be possible on a given line, for example on a PSTN line
       the ring state is determined by external hardware and cannot be
       changed by the software.

       Also note that the cadence may be ignored by particular hardware driver
       so that only the zero or non-zero values are significant.

       The ring pattern is an array of millisecond times for on and off parts
       of the cadence. Thus the Australian ring cadence would be represented
       by the array   unsigned AusRing[] = { 400, 200, 400, 2000 }
      */
    virtual BOOL RingLine(
      unsigned line,     ///<  Number of line
      PINDEX nCadence,   ///<  Number of entries in cadence array
      unsigned * pattern ///<  Ring pattern times
    );


    /**Determine if line has been disconnected from a call.
       This uses the hardware (and country) dependent means for determining
      */
    virtual BOOL IsLineDisconnected(
      unsigned line,   ///<  Number of line
      BOOL checkForWink = TRUE
    );


    /**Directly connect the two lines.
      */
    virtual BOOL SetLineToLineDirect(
      unsigned line1,   ///<  Number of first line
      unsigned line2,   ///<  Number of second line
      BOOL connect      ///<  Flag for connect/disconnect
    );

    /**Determine if the two lines are directly connected.
      */
    virtual BOOL IsLineToLineDirect(
      unsigned line1,   ///<  Number of first line
      unsigned line2    ///<  Number of second line
    );


    /**Get the media formats this device is capable of using.
      */
    virtual OpalMediaFormat::List GetMediaFormats() const = 0;

    /**Set the media format (codec) for reading on the specified line.
      */
    virtual BOOL SetReadFormat(
      unsigned line,    ///<  Number of line
      const OpalMediaFormat & mediaFormat   ///<  Codec type
    ) = 0;

    /**Set the media format (codec) for writing on the specified line.
      */
    virtual BOOL SetWriteFormat(
      unsigned line,    ///<  Number of line
      const OpalMediaFormat & mediaFormat   ///<  Codec type
    ) = 0;

    /**Get the media format (codec) for reading on the specified line.
      */
    virtual OpalMediaFormat GetReadFormat(
      unsigned line    ///<  Number of line
    ) = 0;

    /**Get the media format (codec) for writing on the specified line.
      */
    virtual OpalMediaFormat GetWriteFormat(
      unsigned line    ///<  Number of line
    ) = 0;

    /**Set the line codec for reading.
       Note this function is now deprecated as it could not distinguish between
       some codec subtypes (eg G.729 and G.729B) or non standard formats. Use
       the SetReadFormat function instead.

       The default behaviour now finds the first OpalMediaFormat with the
       payload type and uses that.
      */
    virtual BOOL SetReadCodec(
      unsigned line,    ///<  Number of line
      RTP_DataFrame::PayloadTypes codec   ///<  Codec type
    );

    /**Set the line codec for writing.
       Note this function is now deprecated as it could not distinguish between
       some codec subtypes (eg G.729 and G.729B) or non standard formats. Use
       the SetReadFormat function instead.
      */
    virtual BOOL SetWriteCodec(
      unsigned line,    ///<  Number of line
      RTP_DataFrame::PayloadTypes codec   ///<  Codec type
    );

    /**Set the line codec for reading/writing raw PCM data.
       A descendent may use this to do anything special to the device before
       beginning special PCM output. For example disabling AEC and set
       volume levels to standard values. This can then be used for generating
       standard tones using PCM if the driver is not capable of generating or
       detecting them directly.

       The default behaviour simply does a SetReadCodec and SetWriteCodec for
       PCM data.
      */
    virtual BOOL SetRawCodec(
      unsigned line    ///<  Number of line
    );

    /**Stop the read codec.
      */
    virtual BOOL StopReadCodec(
      unsigned line   ///<  Number of line
    );

    /**Stop the write codec.
      */
    virtual BOOL StopWriteCodec(
      unsigned line   ///<  Number of line
    );

    /**Stop the raw PCM mode codec.
      */
    virtual BOOL StopRawCodec(
      unsigned line   ///<  Number of line
    );

    /**Set the read frame size in bytes.
       Note that a LID may ignore this value so always use GetReadFrameSize()
       for I/O.
      */
    virtual BOOL SetReadFrameSize(
      unsigned line,    ///<  Number of line
      PINDEX frameSize  ///<  New frame size
    );

    /**Set the write frame size in bytes.
       Note that a LID may ignore this value so always use GetReadFrameSize()
       for I/O.
      */
    virtual BOOL SetWriteFrameSize(
      unsigned line,    ///<  Number of line
      PINDEX frameSize  ///<  New frame size
    );

    /**Get the read frame size in bytes.
       All calls to ReadFrame() will return this number of bytes.
      */
    virtual PINDEX GetReadFrameSize(
      unsigned line   ///<  Number of line
    ) = 0;

    /**Get the write frame size in bytes.
       All calls to WriteFrame() must be this number of bytes.
      */
    virtual PINDEX GetWriteFrameSize(
      unsigned line   ///<  Number of line
    ) = 0;

    /**Low level read of a frame from the device.
     */
    virtual BOOL ReadFrame(
      unsigned line,    ///<  Number of line
      void * buf,       ///<  Pointer to a block of memory to receive data.
      PINDEX & count    ///<  Number of bytes read, <= GetReadFrameSize()
    ) = 0;

    /**Low level write frame to the device.
     */
    virtual BOOL WriteFrame(
      unsigned line,    ///<  Number of line
      const void * buf, ///<  Pointer to a block of memory to write.
      PINDEX count,     ///<  Number of bytes to write, <= GetWriteFrameSize()
      PINDEX & written  ///<  Number of bytes written, <= GetWriteFrameSize()
    ) = 0;

    /**High level read of audio data from the device.
       This version will allow non-integral number of frames to be read.
     */
    virtual BOOL ReadBlock(
      unsigned line,    ///<  Number of line
      void * buf,   ///<  Pointer to a block of memory to receive the read bytes.
      PINDEX count  ///<  Count of bytes to read.
    );

    /**High level write audio data to the device.
     */
    virtual BOOL WriteBlock(
      unsigned line,    ///<  Number of line
      const void * buf, ///<  Pointer to a block of memory to write.
      PINDEX count      ///<  Count of bytes to write.
    );


    /**Get average signal level in last frame.
      */
    virtual unsigned GetAverageSignalLevel(
      unsigned line,  ///<  Number of line
      BOOL playback   ///<  Get average playback or record level.
    );


    /**Enable audio for the line.
      */
    virtual BOOL EnableAudio(
      unsigned line,      ///<  Number of line
      BOOL enable = TRUE
    );

    /**Disable audio for the line.
      */
    BOOL DisableAudio(
      unsigned line   ///<  Number of line
    ) { return EnableAudio(line, FALSE); }

    /**Determine if audio for the line is enabled.
      */
    virtual BOOL IsAudioEnabled(
      unsigned line      ///<  Number of line
    );


    enum {
      MaxVolume = 100
    };

    /**Set volume level for recording.
       A value of 100 is the maximum volume possible for the hardware.
       A value of 0 is the minimum volume possible for the hardware.
      */
    virtual BOOL SetRecordVolume(
      unsigned line,    ///<  Number of line
      unsigned volume   ///<  Volume level from 0 to 100%
    );

    /**Set volume level for playing.
       A value of 100 is the maximum volume possible for the hardware.
       A value of 0 is the minimum volume possible for the hardware.
      */
    virtual BOOL SetPlayVolume(
      unsigned line,    ///<  Number of line
      unsigned volume   ///<  Volume level from 0 to 100%
    );

    /**Get volume level for recording.
       A value of 100 is the maximum volume possible for the hardware.
       A value of 0 is the minimum volume possible for the hardware.
      */
    virtual BOOL GetRecordVolume(
      unsigned line,      ///<  Number of line
      unsigned & volume   ///<  Volume level from 0 to 100%
    );

    /**Set volume level for playing.
       A value of 100 is the maximum volume possible for the hardware.
       A value of 0 is the minimum volume possible for the hardware.
      */
    virtual BOOL GetPlayVolume(
      unsigned line,      ///<  Number of line
      unsigned & volume   ///<  Volume level from 0 to 100%
    );


    enum AECLevels {
      AECOff,
      AECLow,
      AECMedium,
      AECHigh,
      AECAuto,
      AECAGC,
      AECError
    };

    /**Get acoustic echo cancellation.
       Note, not all devices may support this function.
      */
    virtual AECLevels GetAEC(
      unsigned line    ///<  Number of line
    );

    /**Set acoustic echo cancellation.
       Note, not all devices may support this function.
      */
    virtual BOOL SetAEC(
      unsigned line,    ///<  Number of line
      AECLevels level   ///<  AEC level
    );

    /**Get wink detect minimum duration.
       This is the signal used by telcos to end PSTN call.
      */
    virtual unsigned GetWinkDuration(
      unsigned line    ///<  Number of line
    );

    /**Set wink detect minimum duration.
       This is the signal used by telcos to end PSTN call.
      */
    virtual BOOL SetWinkDuration(
      unsigned line,        ///<  Number of line
      unsigned winkDuration ///<  New minimum duration
    );

    /**Get voice activity detection.
       Note, not all devices, or selected codecs, may support this function.
      */
    virtual BOOL GetVAD(
      unsigned line    ///<  Number of line
    );

    /**Set voice activity detection.
       Note, not all devices, or selected codecs, may support this function.
      */
    virtual BOOL SetVAD(
      unsigned line,    ///<  Number of line
      BOOL enable       ///<  Flag for enabling VAD
    );


    /**Get Caller ID from the last incoming ring.
       The idString parameter is either simply the "number" field of the caller
       ID data, or if full is TRUE, all of the fields in the caller ID data.

       The full data of the caller ID string consists of the number field, the
       time/date and the name field separated by tabs ('\t').
      */
    virtual BOOL GetCallerID(
      unsigned line,      ///<  Number of line
      PString & idString, ///<  ID string returned
      BOOL full = FALSE   ///<  Get full information in idString
    );

    /**Set Caller ID for use in next RingLine() call.
       The full data of the caller ID string consists of the number field, the
       time/date and the name field separated by tabs ('\t').

       If the date field is missing (two consecutive tabs) then the current
       time and date is used. Using an empty string will clear the caller ID
       so that no caller ID is sent on the next RingLine() call.
      */
    virtual BOOL SetCallerID(
      unsigned line,            ///<  Number of line
      const PString & idString  ///<  ID string to use
    );

    /**Send a Caller ID on call waiting command
       The full data of the caller ID string consists of the number field, the
       time/date and the name field separated by tabs ('\t').

       If the date field is missing (two consecutive tabs) then the current
       time and date is used. Using an empty string will clear the caller ID
       so that no caller ID is sent on the next RingLine() call.
      */
    virtual BOOL SendCallerIDOnCallWaiting(
      unsigned line,            ///<  Number of line
      const PString & idString  ///<  ID string to use
    );

    /**Send a Visual Message Waiting Indicator
      */
    virtual BOOL SendVisualMessageWaitingIndicator(
      unsigned line,            ///<  Number of line
      BOOL on
    );


    enum {
      DefaultDTMFOnTime = 180,
      DefaultDTMFOffTime = 120
    };

    /**Play a DTMF digit.
       Any characters that are not in the set 0-9, A-D, * or # will be ignored.
      */
    virtual BOOL PlayDTMF(
      unsigned line,            ///<  Number of line
      const char * digits,      ///<  DTMF digits to be played
      DWORD onTime = DefaultDTMFOnTime,  ///<  Number of milliseconds to play each DTMF digit
      DWORD offTime = DefaultDTMFOffTime ///<  Number of milliseconds between digits
    );

    /**Read a DTMF digit detected.
       This may be characters from the set 0-9, A-D, * or #. A null ('\0')
       character indicates that there are no tones in the queue.
       Characters E through P indicate the following tones:

         E = 800   F = 1000  G = 1250  H = 950   I = 1100  J = 1400
         K = 1500  L = 1600  M = 1800  N = 2100  O = 1300  P = 2450

      */
    virtual char ReadDTMF(
      unsigned line   ///<  Number of line
    );

    /**Get DTMF removal mode.
       When set in this mode the DTMF tones detected are removed from the
       encoded data stream as returned by ReadFrame().
      */
    virtual BOOL GetRemoveDTMF(
      unsigned line   ///<  Number of line
    );

    /**Set DTMF removal mode.
       When set in this mode the DTMF tones detected are removed from the
       encoded data stream as returned by ReadFrame().
      */
    virtual BOOL SetRemoveDTMF(
      unsigned line,     ///<  Number of line
      BOOL removeTones   ///<  Flag for removing DTMF tones.
    );


    enum CallProgressTones {
      NoTone    = 0x00,   // indicates no tones
      DialTone  = 0x01,   // Dial tone
      RingTone  = 0x02,   // Ring indication tone
      BusyTone  = 0x04,   // Line engaged tone
      ClearTone = 0x08,   // Call failed/cleared tone (often same as busy tone)
      CNGTone   = 0x10,   // Fax CNG tone
      AllTones  = 0x1f
    };

    /**See if any tone is detected.
      */
    virtual unsigned IsToneDetected(
      unsigned line   ///<  Number of line
    );

    /**See if any tone is detected.
      */
    virtual unsigned WaitForToneDetect(
      unsigned line,          ///<  Number of line
      unsigned timeout = 3000 ///<  Milliseconds to wait for
    );

    /**See if a specific tone is detected.
      */
    virtual BOOL WaitForTone(
      unsigned line,          ///<  Number of line
      CallProgressTones tone, ///<  Tone to wait for
      unsigned timeout = 3000 ///<  Milliseconds to wait for
    );

    /**Set a tones filter information.
       The description string is of the form
          frequence ':' cadence
      where frequency is either
          frequency
          low '-' high
      and cadence is
          mintime
          ontime '-' offtime
          ontime '-' offtime '-' ontime '-' offtime
      examples:
          300:0.25      300Hz for minimum 250ms
          1100:0.4-0.4  1100Hz with cadence 400ms on, 400ms off
          900-1300:1.5  900Hz to 1300Hz for minimum of 1.5 seconds
          425:0.4-0.2-0.4-2    425Hz with cadence
                                400ms on, 200ms off, 400ms on, 2 seconds off
      */
    virtual BOOL SetToneFilter(
      unsigned line,              ///<  Number of line
      CallProgressTones tone,     ///<  Tone filter to change
      const PString & description ///<  Description of filter parameters
    );

    /**Set a tones filter information.
      */
    virtual BOOL SetToneFilterParameters(
      unsigned line,            ///<  Number of line
      CallProgressTones tone,   ///<  Tone filter to change
      unsigned lowFrequency,    ///<  Low frequency
      unsigned highFrequency,   /// High frequency
      PINDEX numCadences,       /// Number of cadence times
      const unsigned * onTimes, /// Cadence ON times
      const unsigned * offTimes /// Cadence OFF times
    );

    /**Play a tone.
      */
    virtual BOOL PlayTone(
      unsigned line,          /// Number of line
      CallProgressTones tone  /// Tone to be played
    );

    /**Determine if a tone is still playing
      */
    virtual BOOL IsTonePlaying(
      unsigned line   /// Number of line
    );

    /**Stop playing a tone.
      */
    virtual BOOL StopTone(
      unsigned line   /// Number of line
    );

    /**Play a wav file
      */
    virtual BOOL PlayAudio(
      unsigned line,            /// Number of line
      const PString & filename  /// File Name
    );
    
    /**Stop playing the Wave File
      */
    virtual BOOL StopAudio(
      unsigned line   ///Number of line
    );


    /**Dial a number on network line.
       The takes the line off hook, waits for dial tone, and transmits the
       specified number as DTMF tones.

       If the requireTones flag is TRUE the call is aborted of the call
       progress tones are not detected. Otherwise the call proceeds with short
       delays while it tries to detect the call progress tones.

       The return code indicates the following:
          DialTone  No dial tone detected
          RingTone  Dial was successful
          BusyTone  The remote phone was busy
          ClearTone Dial failed (usually means rang out)
          NoTone    There was an internal error making the call
      */
    virtual CallProgressTones DialOut(
      unsigned line,                /// Number of line
      const PString & number,       /// Number to dial
      BOOL requireTones = FALSE     /// Require dial/ring tone to be detected
    );


    enum T35CountryCodes {
      Japan, Albania, Algeria, AmericanSamoa, Germany, Anguilla, AntiguaAndBarbuda,
      Argentina, Ascension, Australia, Austria, Bahamas, Bahrain, Bangladesh,
      Barbados, Belgium, Belize, Benin, Bermudas, Bhutan, Bolivia, Botswana,
      Brazil, BritishAntarcticTerritory, BritishIndianOceanTerritory, 
      BritishVirginIslands, BruneiDarussalam, Bulgaria, Myanmar, Burundi,
      Byelorussia, Cameroon, Canada, CapeVerde, CaymanIslands,
      CentralAfricanRepublic, Chad, Chile, China, Colombia, Comoros, Congo,
      CookIslands, CostaRica, Cuba, Cyprus, Czechoslovakia, Cambodia,
      DemocraticPeoplesRepublicOfKorea, Denmark, Djibouti, DominicanRepublic,
      Dominica, Ecuador, Egypt, ElSalvador, EquatorialGuinea, Ethiopia,
      FalklandIslands, Fiji, Finland, France, FrenchPolynesia,
      FrenchSouthernAndAntarcticLands, Gabon, Gambia, Germany2, Angola, Ghana,
      Gibraltar, Greece, Grenada, Guam, Guatemala, Guernsey, Guinea, GuineaBissau,
      Guayana, Haiti, Honduras, Hongkong, Hungary, Iceland, India, Indonesia,
      Iran, Iraq, Ireland, Israel, Italy, CotedIvoire, Jamaica, Afghanistan,
      Jersey, Jordan, Kenya, Kiribati, KoreaRepublic, Kuwait, Lao, Lebanon,
      Lesotho, Liberia, Libya, Liechtenstein, Luxemborg, Macao, Madagascar,
      Malaysia, Malawi, Maldives, Mali, Malta, Mauritania, Mauritius, Mexico,
      Monaco, Mongolia, Montserrat, Morocco, Mozambique, Nauru, Nepal,
      Netherlands, NetherlandsAntilles, NewCaledonia, NewZealand, Nicaragua,
      Niger, Nigeria, Norway, Oman, Pakistan, Panama, PapuaNewGuinea, Paraguay,
      Peru, Philippines, Poland, Portugal, PuertoRico, Qatar, Romania, Rwanda,
      SaintKittsAndNevis, SaintCroix, SaintHelenaAndAscension, SaintLucia,
      SanMarino, SaintThomas, SaoTomeAndPrincipe, SaintVicentAndTheGrenadines,
      SaudiArabia, Senegal, Seychelles, SierraLeone, Singapore, SolomonIslands,
      Somalia, SouthAfrica, Spain, SriLanka, Sudan, Suriname, Swaziland, Sweden,
      Switzerland, Syria, Tanzania, Thailand, Togo, Tonga, TrinidadAndTobago,
      Tunisia, Turkey, TurksAndCaicosIslands, Tuvalu, Uganda, Ukraine,
      UnitedArabEmirates, UnitedKingdom, UnitedStates, BurkinaFaso, Uruguay,
      USSR, Vanuatu, VaticanCityState, Venezuela, VietNam, WallisAndFutuna,
      WesternSamoa, Yemen, Yemen2, Yugoslavia, Zaire, Zambia, Zimbabwe,
      NumCountryCodes,
      UnknownCountry = -1
    };

    /**Get the country code set for the device.
      */
    T35CountryCodes GetCountryCode() const { return countryCode; }

    /**Get the country code set for the device as a string.
      */
    PString GetCountryCodeName() const;

    /**Get the country code set for the device as a string.
      */
    static PString GetCountryCodeName(T35CountryCodes code);
    static T35CountryCodes GetCountryCode(const PString & name);

    /**Set the country code set for the device.
       This may change the line analogue coefficients, ring detect, call
       disconnect detect and call progress tones to fit the countries
       telephone network.
      */
    virtual BOOL SetCountryCode(
      T35CountryCodes country   /// COuntry code for device
    );

    /**Set the country code set for the device.
      */
    virtual BOOL SetCountryCodeName(
      const PString & countryName   /// COuntry code for device
    );

    /**Get the list of countries actually supported by the device
     */
    virtual PStringList GetCountryCodeNameList() const;


    /**Return number for last error.
      */
    int GetErrorNumber() const { return osError; }

    /**Return text for last error.
      */
    PString GetErrorText() const;

    virtual void PrintOn(
      ostream & strm
    ) const;

        /** LID Device Type  
         */
        enum DeviceType{
                POTSLine,          // Behaves as a normal Phone 
                Gateway,                   // Behaves as a gateway
                CellEmulate                // Behaves as like a CellPhone
                
        };

        /** Get the LID Device Type.
         */
        virtual OpalLineInterfaceDevice::DeviceType GetDeviceType() { return POTSLine; };

  protected:
    int             os_handle;
    int             osError;
    T35CountryCodes countryCode;
    PBYTEArray      readDeblockingBuffer, writeDeblockingBuffer;
    PINDEX          readDeblockingOffset, writeDeblockingOffset;

#if PTRACING
    friend ostream & operator<<(ostream & o, CallProgressTones t);
#endif
};


/**This class describes an I/O channel that uses a Line Interface Device.
 */
class OpalLineChannel : public PChannel
{
  PCLASSINFO(OpalLineChannel, PChannel);

  public:
    /**Create a new channel to a line interface device.
      */
    OpalLineChannel(
      OpalLineInterfaceDevice & device, ///<  Device for channel I/O
      unsigned line,                    ///<  Number of line
      const H323AudioCodec & codec      ///<  Codec that uses the channel
    );
    ~OpalLineChannel();

    /**Get the device name.
      */
    PString GetName() const;

    /**Close the channel.
      */
    BOOL Close();

    /** Low level read from the channel. This function may block until the
       requested number of characters were read or the read timeout was
       reached. The GetLastReadCount() function returns the actual number
       of bytes read.

       The GetErrorCode() function should be consulted after Read() returns
       FALSE to determine what caused the failure.

       @return
       TRUE indicates that at least one character was read from the channel.
       FALSE means no bytes were read due to timeout or some other I/O error.
     */
    virtual BOOL Read(
      void * buf,   ///<  Pointer to a block of memory to receive the read bytes.
      PINDEX len    ///<  Maximum number of bytes to read into the buffer.
    );

    /** Low level write to the channel. This function will block until the
       requested number of characters are written or the write timeout is
       reached. The GetLastWriteCount() function returns the actual number
       of bytes written.

       The GetErrorCode() function should be consulted after Write() returns
       FALSE to determine what caused the failure.

       @return
       TRUE if at least len bytes were written to the channel.
     */
    virtual BOOL Write(
      const void * buf, ///<  Pointer to a block of memory to write.
      PINDEX len        ///<  Number of bytes to write.
    );


    /**Get the device the channel is writing to.
      */
    OpalLineInterfaceDevice & GetDevice() const { return device; }

    /**Get the line number on device the channel is writing to.
      */
    unsigned GetLineNumber() const { return lineNumber; }


  protected:
    OpalLineInterfaceDevice &   device;
    unsigned                    lineNumber;
    BOOL                        reading;
    BOOL                        useDeblocking;
};





/**This class describes the LID based codec capability.
 */
class H323_LIDCapability : public H323AudioCapability
{
  PCLASSINFO(H323_LIDCapability, H323AudioCapability);

  public:
  /**@name Construction */
  //@{
    /**Create a new capability.
     */
    H323_LIDCapability(
      const OpalMediaFormat & mediaFormat   ///<  RTP codec type for capability
    );
  //@}

  /**@name Overrides from class PObject */
  //@{
    /**Compare the object with another of the same class.
      */
    Comparison Compare(const PObject & obj) const;

    /**Create a copy of the object.
      */
    virtual PObject * Clone() const;

    /**Get the name of the media data format this class represents.
     */
    virtual PString GetFormatName() const;
  //@}

  /**@name Identification functions */
  //@{
    /**Get the sub-type of the capability. This is a code dependent on the
       main type of the capability.

       This returns H245_AudioCapability::e_nonStandard.
     */
    virtual unsigned GetSubType() const;
  //@}

  /**@name Operations */
  //@{
    /**Create the codec instance, allocating resources as required.
     */
    virtual H323Codec * CreateCodec(
      H323Codec::Direction direction  ///<  Direction in which this instance runs
    ) const;

    /**Determine if the codec specified for the object is valid.
      */
    BOOL IsValid() const;

    /**Add all of the capabilities that a LID is capable of.
     */
    static void AddAllCapabilities(
      const OpalLineInterfaceDevice & device, ///<  Device to get codecs from
      H323Capabilities & capabilities,        ///<  Table to add capabilities to
      PINDEX descriptorNum,   ///<  The member of the capabilityDescriptor to add
      PINDEX simultaneous     ///<  The member of the SimultaneousCapabilitySet to add
    );
  //@}

  /**@name Protocol manipulation */
  //@{
    /**This function is called whenever and outgoing TerminalCapabilitySet
       or OpenLogicalChannel PDU is being constructed for the control channel.
       It allows the capability to set the PDU fields from information in
       members specific to the class.

       The default behaviour sets the data rate field in the PDU.
     */
    virtual BOOL OnSendingPDU(
      H245_AudioCapability & pdu,  ///<  PDU to set information on
      unsigned packetSize          ///<  Packet size to use in capability
    ) const;

    /**This function is called whenever and incoming TerminalCapabilitySet
       or OpenLogicalChannel PDU has been used to construct the control
       channel. It allows the capability to set from the PDU fields,
       information in members specific to the class.

       The default behaviour gets the data rate field from the PDU.
     */
    virtual BOOL OnReceivedPDU(
      const H245_AudioCapability & pdu,  ///<  PDU to get information from
      unsigned & packetSize              ///<  Packet size to use in capability
    );
  //@}

  protected:
    OpalMediaFormat mediaFormat;
    PINDEX codecTableIndex;
};


/**This class describes the Cisco non-standard G.723.1 Annex A codec capability.
 */
class H323_CiscoG7231aLIDCapability : public H323NonStandardAudioCapability
{
  PCLASSINFO(H323_CiscoG7231aLIDCapability, H323NonStandardAudioCapability);

  public:
  /**@name Construction */
  //@{
    /**Create a new Cisco non-standard G.723.1 Annex A capability.
     */
    H323_CiscoG7231aLIDCapability();
  //@}

  /**@name Overrides from class PObject */
  //@{
    /**Create a copy of the object.
      */
    virtual PObject * Clone() const;
  //@}

  /**@name Operations */
  //@{
    /**Create the codec instance, allocating resources as required.
     */
    virtual H323Codec * CreateCodec(
      H323Codec::Direction direction  ///<  Direction in which this instance runs
    ) const;
  //@}

  /**@name Identification functions */
  //@{
    /**Get the name of the media data format this class represents.
     */
    virtual PString GetFormatName() const;
  //@}
};


/**This class is a LID codec that only converts data for playback on some
   other audio device.
 */
class H323_LIDCodec : public H323AudioCodec
{
  PCLASSINFO(H323_LIDCodec, H323AudioCodec);

  public:
    /**Create a new LID based codec.
     */
    H323_LIDCodec(
      const char * mediaFormat,   ///<  Media format for codec
      Direction direction,        ///<  Direction in which this instance runs
      unsigned numFrames,         ///<  Number of frames to transmit in packet
      PINDEX codecTableIndex      ///<  Type of codec to use
    );

    /**Encode the data from the appropriate device.
       This will encode a frame of data for transmission. The exact size and
       description of the data placed in the buffer is codec dependent but
       should be less than H323Capability::GetTxFramesInPacket() *
       OpalMediaFormat::GetFrameSize() in length.

       The length parameter is filled with the actual length of the encoded
       data, often this will be the same as the size parameter.

       This function is called every GetFrameRate() timestamp units, so MUST
       take less than (or equal to) that amount of time to complete!

       Note that a returned length of zero indicates that time has passed but
       there is no data encoded. This is typically used for silence detection
       in an audio codec.
     */
    virtual BOOL Read(
      BYTE * buffer,            ///<  Buffer of encoded data
      unsigned & length,        ///<  Actual length of encoded data buffer
      RTP_DataFrame & rtpFrame  ///<  RTP data frame
    );

    /**Decode the data and output it to appropriate device.
       This will decode a single frame of received data. The exact size and
       description of the data required in the buffer is codec dependent but
       should be less than H323Capability::GetRxFramesInPacket() *
       OpalMediaFormat::GetFrameSize()  in length.

       It is expected this function anunciates the data. That is, for example
       with audio data, the sound is output on a speaker.

       This function is called every GetFrameRate() timestamp units, so MUST
       take less than that amount of time to complete!
     */
    virtual BOOL Write(
      const BYTE * buffer,          ///<  Buffer of encoded data
      unsigned length,              ///<  Length of encoded data buffer
      const RTP_DataFrame & frame,  ///<  Entire RTP frame
      unsigned & written            ///<  Number of bytes used from data buffer
    );

    /**Return flag indicating raw channel is native.
       For audio codecs, FALSE typically means that the format is PCM-16.
       For video codecs, FALSE typically means that the format is YUV411P.

       The default behaviour returns TRUE.
      */
    virtual BOOL IsRawDataChannelNative() const;

    /**Check frame for a talk burst.
       This does the deadband calculations on the average signal levels
       returned by the GetAverageSignalLevel() function and based on the
       levelThreshold, signalDeadbandFrames and silenceDeadbandFrames
       member variables.
      */
    virtual BOOL DetectSilence();

    /**Calculate the energy in the audio stream.
       This is called from within DetectSilence() to calculate the average
       energy since the last call to DetectSilence().
      */
    virtual unsigned GetAverageSignalLevel();


    PINDEX GetCodecTableIndex() const { return codecTableIndex; }

  protected:
    PINDEX codecTableIndex;
    unsigned packetSize;
    unsigned missedCount;
    BYTE     lastSID[4];
    BOOL     lastFrameWasSignal;
};


#endif // __OPAL_LID_H


// End of File ///////////////////////////////////////////////////////////////