awah-sip / awah-sip_codec Goto Github PK

this function is not used anymore and can be removed

In GitLab by @andyweiss on Apr 16, 2021, 12:10

sound devices are only updated on application start up.

In GitLab by @LeeO86 on Apr 26, 2021, 22:24

Fctions:

• getGpioRoutes
• getGpioDevID
• addGpioDev(type, incount, outcout, name, uid = "")
• removeGpioDev
• getGpioPortsList
• connectGpioPort (schould be possible on the crosspoint)
• disconnectGpioPort
• changeGpioCrosspoint
• getGpioDevices
• setGPI
• setGPO
• getState

Signals:

• gpioDeviceChanged
• gpioRoutesChanged
• gpioRoutesTableChanged
• gpioChanged

In GitLab by @andyweiss on Apr 11, 2021, 09:33

Add a web GUI

value is inverted ... to check

In GitLab by @andyweiss on Apr 11, 2021, 09:33

Testen welche settings sich pro Account einstellen lassen. Z.B. Jitterbuffer wäre wichtig!

In GitLab by @andyweiss on Apr 11, 2021, 10:10

It should be possible to send and receive GPIO's

In GitLab by @andyweiss on Apr 11, 2021, 10:13

Ein TCP Socket Server integrieren um z.B. von einem ESP Routing Punkte setzen und Routing Tabelle auslesen zu können.

In GitLab by @andyweiss on Apr 11, 2021, 10:15

Add a VOX detection to set crosspoint on a certain audio level.

In GitLab by @andyweiss on Apr 11, 2021, 09:33

abchecken warum erst das zweite Signal Sip Status es bis ins GUI schafft

In GitLab by @andyweiss on Apr 11, 2021, 09:33

Issue seems to be in pjsua_aud.c

pjsua_aud.c ......Unable to open file for recording: No such file or directory [status=120002]

In GitLab by @andyweiss on Apr 11, 2021, 09:33

anstelle von mehreren calls pro account annehmen, ein option in den account hinzufügen, die den call bei besetzt an einen vordefinierten account weiterleitet.

So ist das handling und die Übersicht in dem router gewährleistet.

The user schould be able to set custom labels to the individual inputs and outputs of a sound device or a GPIO device.

In GitLab by @andyweiss on Apr 16, 2021, 12:46

Hier ein Auszug vom logverzeiniss und er aktuellen zeit

root@raspberrypi:/opt/AWAH-SIP_Codec/bin/AWAH-SIP_Codec/logs# ls
Log_2021_04_10__01_12.txt Log_2021_04_11__21_49.txt Log_2021_04_11__22_08.txt Log_2021_04_11__22_59.txt
Log_2021_04_10__08_51.txt Log_2021_04_11__21_59.txt Log_2021_04_11__22_20.txt Log_2021_04_11__23_01.txt
Log_2021_04_10__10_32.txt Log_2021_04_11__22_03.txt Log_2021_04_11__22_21.txt

root@raspberrypi:/opt/AWAH-SIP_Codec/bin/AWAH-SIP_Codec/logs# date
Fri 16 Apr 13:42:50 BST 2021

In GitLab by @andyweiss on Apr 11, 2021, 09:33

add buddy handling, inclusive save and load etc.

In GitLab by @andyweiss on Apr 11, 2021, 09:33

Assertion failed: (cport->tx_buf_count <= cport->tx_buf_cap), function write_port, file ../src/pjmedia/conference.c, line 1827.

die einkanalige version geht

In GitLab by @andyweiss on Apr 11, 2021, 09:34

mehrere callinfo Fenster sollen möglich sein

In GitLab by @andyweiss on Apr 11, 2021, 09:33

what are the optimal pool size parameters?

In GitLab by @andyweiss on Apr 11, 2021, 22:10

Tested with Octo sound card und a USB sound device. Both only show one channel.

Like this the spplittcomb crashes!

In GitLab by @andyweiss on Apr 11, 2021, 09:33

add a RX and TX levelmeter to the account. To show audio levels in the GUI

In GitLab by @andyweiss on Apr 11, 2021, 09:33

includes überprüfen und aufräumen

In GitLab by @andyweiss on Apr 11, 2021, 09:34

Fensterpositionen speichern

In GitLab by @andyweiss on Apr 11, 2021, 10:08

Einen GPIO Router hinzufügen. GPIO Provider sollen sich via ein socket registrieren können. In einer Matrix sollen diese geroutet werden können.

In GitLab by @andyweiss on Apr 16, 2021, 12:11

beim input nur devices welche auch inputs haben anzeigen

In GitLab by @andyweiss on Apr 11, 2021, 09:33

To be compatible with some legacy devices AAC support is needed

In GitLab by @andyweiss on Apr 11, 2021, 09:33

The ident Player plays Full-scale files with some distortion. Probably because a stereo file gets mononmixed.
A stereo ident Player would be nicer anyway.

In GitLab by @andyweiss on Apr 11, 2021, 09:33

codec parameter editierbar machen

In GitLab by @andyweiss on Apr 11, 2021, 09:33

Seriös testen ob wir sauberes audio übermitteln können.

vor allem auf Clock clicks, Quantisierungs Rauschen etc achten.

In GitLab by @andyweiss on Apr 11, 2021, 09:34

Im Feld zur Nummernwahl direkt immer den neuesten Callhistory Eintrag einfüllen. So kann man mit Ender direkt einen ReDial machen

In GitLab by @LeeO86 on Apr 26, 2021, 22:00

• Subscribe Events: each client has to subscribe to events such as:

  • logMessage
  • callInfo
  • gpIevents
  • gpOevents

• Add GPIO functions (set and reset GPI & O, set Routes, add and remove devices incl. Logical functions, see also GPIO-Issue)
  -rename audioRoutesChanged to routesChanged

if a call recorder and an announcement is configured the recording schould start after the announcement and not immediately

In GitLab by @andyweiss on Apr 11, 2021, 09:33

in configfile.cpp Line 216 is the port range set to 0 (with gives 10 addresses)
this is not EBU Tech 3329 compliant!!! schould we fix that?

In GitLab by @andyweiss on Apr 11, 2021, 09:34

Wenn eine Verbindung zu einem Nereus gemacht wird und im Router kein Audio auf den Account geroutet ist, lock der Decoder vom Nereus nicht und beendet den Call nach einer Weile

In GitLab by @andyweiss on Apr 16, 2021, 12:44

sollen wir alle Varianten auflisten? Sysdefault hw plughw dsnoop etc? Die Namen sind schon sehr kryptisch. Schön wäre hier einwenig aufzuräumen

In GitLab by @andyweiss on Apr 20, 2021, 09:56

wenn ein Soundevice nicht vorhanden ist, nicht gleich löschen, sondern user informieren und in der list behalten!
Benötigt eine Art notify user Funktion. -> oder kann einfach im Log vermerkt werden?

In GitLab by @andyweiss on Apr 11, 2021, 09:33

pjsua->sendPresenceStatus(busy); is commented out

todo: check Buddies presence status mechanism

In GitLab by @andyweiss on Apr 11, 2021, 09:33

Multichannel calls (greater than stereo) would be nice.

In GitLab by @andyweiss on Apr 11, 2021, 09:33

for each active call a inspection of the mediastream is needed to detect media loss

In GitLab by @andyweiss on Apr 11, 2021, 09:33

Add "first matching codec" to the codeclist. Like that we can offer multiple codecs to the remote client

In GitLab by @andyweiss on Apr 11, 2021, 10:17

Different systam states schould be available in the GPIO Router. E.g. SIP registered, Connection established....

In GitLab by @andyweiss on Apr 11, 2021, 09:34

siehe Ticket:

https://trac.pjsip.org/repos/ticket/2041

check if sound device already exists bevor adding it.

In GitLab by @andyweiss on Apr 11, 2021, 09:34

default volume im router abspeichern

In GitLab by @andyweiss on Apr 11, 2021, 09:33

alle settings validieren. Z.B Jitter Buffer einstellungen kann zu Assertion faults führen

In GitLab by @andyweiss on Apr 11, 2021, 09:33

The user schould be able to put custom labels to the codecs

call statistics schould no get the data from call dump. Use direct variables instead

In GitLab by @andyweiss on Apr 11, 2021, 09:33

there are still some structs copied!!!

In GitLab by @andyweiss on Apr 11, 2021, 09:33

Wie soll die API zur Aussenwelt aussehen und was für Schnittstellen gib es?

In GitLab by @andyweiss on Apr 11, 2021, 09:34

Jitterbuffer Max delay als define oder in den Settings nur im config File.
Die min und max werte beim eingeben checken bzw Ranges updaten.

In GitLab by @andyweiss on Apr 11, 2021, 09:33

test Transfer Call.