[Iaxclient-cvs] iaxclient/lib TODO,NONE,1.1

[<st...@us...>]

Update of /cvsroot/iaxclient/iaxclient/lib
In directory sc8-pr-cvs1:/tmp/cvs-serv12012

Added Files:
	TODO 
Log Message:
some TODO thoughts and notes.


--- NEW FILE: TODO ---
TODO items:


1) Audio driver work:
	Properly abstract audio drivers (currently, we use only
		portaudio, but we may also want to support others.
		The most likely candidate here would be zaptel devices.

	Instead of the "switch" statements in the code, define an audio
	driver structure, with 
	
	- function pointers for actual driver entry points.	
		initialization: (scans available devices, sets up data
				structures)
		destruction: (stops everything, cleans up)
		"start":  starts audio for a particular call?
		"stop": stops audio for a particular call?
		"playsound": plays a particular sound: can be used for
			incoming call notification, ringback, dialtone etc?
		"select": select input and output devices to use?	
		[maybe extend this for zap devices to have "ring", etc
		functions?]

		
	- Common audio driver data members:
		a) perhaps an array of devices the driver has found,
			with for each device, a device name, an
			indication of whether this device is the default
			input or output, and whether this device
			supports input, output, or both.

	For portaudio, we probably want to switch to the "standard"
	portaudio callback interface, and away from pablio, which isn't
	really robust enough for our needs once we do this stuff.


2) Call handling:
	currently, the library really only supports one call, and not
	very well.  It should have a collection of calls (either an
	array, or a linked list), and keep track of the current state of
	each call.

	An array might be easiest to manage, and would map well to a
	softphone client.  We would then just refer to calls by their
	index, and a GUI client might present these like call
	appearances on their display.

	Incoming calls might come in on the first free call appearance,
	and outgoing calls by default would do the same.

	The state of each call might be similar to phonecore
	(incoming_incomplete, incoming, outgoing_incomplete, outgoing),
	but we'd also have to keep track of which call, if any, we
	currenly have "selected" -- i.e. which one we should connect to
	the audio system.

	We'd need to send events to the client whenever a call changed
	"state" in any way.

	We can make the number of calls in the array defined at runtime
	when the library is initialized.  A very simple client like
	testcall would just ask for a single call, so it wouldn't have
	to worry about a lot of this.

3) Codecs:  (I think that someone is working on this)

	Currently, the library assumes that all calls will be GSM only,
	and further assumes that all frames will be 20ms.  It can
	control the frame size (within reason) for frames it sends out,
	but should deal gracefully with incoming frames that aren't
	20ms.

	Codecs should probably be implemented via a similar set of
	structure abstractions as audio drivers, above.  They also need
	to handle incoming packets which may switch formats abruptly(?).

4) Events:
	We might want to consolidate the (currently three) callbacks
	that the library makes to clients, into a single callback, that
	passes back a structure with event info.  I was thinking of a
	structure with an event type, and then a union of different
	structures depending on the event type.

	The only thing is that we might want to decide whether or not,
	or how clients will "register" for different event types, even
	if they're handled through the same callback mechanism.

	Ideally, the library would handle all of the events itself, via
	some "default" handlers.  (I.e. for messages, it might just print
	them to stdout or stderr.  For incoming calls, it might accept
	them by default).

	So, the choices then are whether the client should register for
	individual events, or perhaps it can just decline events as they
	happen, and then the library could handle them.