Conflicts:

	src/audio/audiortp.cpp
	src/sipvoiplink.cpp

Instead of two!!

You can switch from ALSA to Pulseaudio from the config window

If pulseaudio is selected, the alsa configuration panel gets hidden

problem with capture: delay and audio quality
Dtmf, ringtones missing

No capture for now
No dtmf, no ringtones

Before loading the codecs the user set, check if this codec is loaded.
In fact, it could have been erased and causes gtk bug

At startup, the installation directory is scanned and the available shared library
(codecs) are loaded in memory.

The server sends the list of the sound card it founds on the system to the GTk without
checking it is a capture or a playback device
TODO: improve the list sent ( must add abstraction device plug, default, etc..)
TODO: improve sound quality for capture

Not implemented yet: ringtones, dtmf

#### Recommended play devices names (alsa-project.org)
#### http://www.alsa-project.org/main/index.php/DeviceNames
Device: #default --> capture : too much noise
		     playback : glitchs
		     dmix
	#plug:front  -->  playback almost perfect
		         capture :bad audio (noise noise noise !!!)
 		         dmix not enabled
 	#plug:rear  -->  doesn't exist on my sound card (Error: unknown PCM device)
	#plug:center_lfe --> idem
	#plug:side  --> idem

If there is multiple sound cards on your system, you can select the one you prefer like that:
  plug:front:0 select the card 0, 0 refers to the number in /proc/asound/cards.
  plug:front:Intel with the name of the card
My /proc/asound/cards:
0 [Intel          ]: HDA-Intel - HDA Intel
                      HDA Intel at 0xfdff4000 irq 23

##### Device names not recommmended (alsa-project.org)
These devices maps directly to the driver devices and doesn't handle an abstraction
	plughw -->  playback almost perfect
		    capture : noise plus weird behaviour
                    dmix not enabled
        hw  -->     playback : not audible ( maybe no resampling ?? no format conversion ??)
		    capture : XRUN mode
	            dmix not enabled

But size of the frames to work on

The codecs are loaded at the initialization of the rtp session and destroyed at the end.
GSM audio quality is far better now. Because the dynamic library had to load at every 20ms the libgsm, it introduced a delay at each frame

!!!!!!!!!!!  Must have libspeex1 and libspeex1-dev installed

The previous commit brought a segmentation fault when launching the client.
When the daemon is lauched separatly, it uses a wrapper script, but it doesn't when the client start the daemon.
SO the LD_LIBRARY_PATH was ignored and the daemon couldn't find the dynamic library for the codecs.

Codecs dynamic libraries are created at the compilation time if they don't exist and copied in /usr/lib/sflphone/codecs at the installation

Audio calls with ilbc codecs still not tested.

Note: Manually copy the .so library in /usr/lib , then ldconfig

Copier les librairies dynamiques (codec_*.so) dans le répertoire /usr/lib et upda
faire ldconfig