ageet Forums

Hello Michael,
congratulation for your project.
Yes, great product !

Would be REALLY GREAT if speakerswitch* will work with ALMOST every mobile, including ARTEMIS P3300... (now this is possible with troca2.exe if the app is launched with a mortscript, before dialing/answer in Agephone).

Maybe not the best approach, but the problem can be solved with RIL API: see RIL_Initialize, RIL_SetAudioDevices, RIL_Deinitialize.
http://msdn.microsoft.com/en-us/library/aa923031.aspx
http://wiki.xda-developers.com/index.ph ... rilgsm.dll
and "play sound througth the earpiece"
http://www.eggheadcafe.com/software/asp ... the-e.aspx

Also, integrating an acoustic echo and noise canceller (like the attached one - used by Fring) can be a ++ plus....

Best regards,
Dan

The info for Artemis:

Hardware Information:

OEM Info: ARTE10000
OEM ID: 5
Number of processors: 1
Processor type: 2577

System Information:

Platform Type: PocketPC
Platform Name: Artemis
Platform Manufacturer: HTC
Build Number: 00005294
Major Version: 00000005
Minor Version: 00000002
Platform ID: 00000003
Local IP:
Global IP:
HostIndex: -1
NIC: 192.168.123.1, fe80:..:1,

Driver Enumeration:

WAV1: \windows\wavedev.dll (92704)
WAV8: \windows\bta2dp.dll (55636)

Hi Michael,

How is going?

I suppose you are not responding to every desperate guy hwo asking on some help to route the voice trough the earpiece.
As an xdadev member you seems to be a little bit selfish: you don't mind to help in any way this comunity, but are expecting the feedback...

Ok, check this:
http://forum.xda-developers.com/showpos ... count=1444

Now is your turn to give us a feedback.

Regards,
x

Hi dan_u_be and welcome to the forums! Please excuse my selfishness that has been putting too much focus on the XDA forums lately and neglected the home ground unduly First off, thank you very much for pointing out Fring's echo cancellation metod! We can't just implement it without negotiating for a license from the people behind it, but "Canec" sounds definitely like a great way to do the job and I am already in touch with DSP Algorithms. As for your other suggestion, I have already pointed our developer at it and will let you know what he had to say. It may take a while, but rest assured that we are just as interested as you are in this matter and look forward to the day when we really support all the devices out there. In the meantime: Did you already check out AM 2.75? We enabled some more devices and it might just work for your Artemis!

I have just gotten word from our programmers and they are positive about giving the RIL approach a shot. We usually operate on a deeper level to reroute the audio, but this approach can have its advantages. You'll hear from me by mail shortly.

Dear sir,
we just buy a mobile version to run on HTC T3232, the outgoing voice is choppy with G.711 ulaw

Do we have any way to get voice quality improvement?

best regards
kaiser

AGEPhone 2.75
Windows Mobile 6.1

The Touch 3G has no issues with sound quality that we know of and there are a row of settings that might improve the situation. First off, I hope that you don't mind me asking a little question though: What kind of internet connection and VoIP provider are you using? Please note that 2G and EDGE connections are not recommended and all other connections have to provide at least a constant 100 kbit/s upstream to be usable with the G.711 codec (which uses 64 kbit/s + 20 kbit/s overhead in each direction). If your connection provides enough bandwidth you may want to try

1. Changing the Packet Size (page 3 provider settings) to 40 ms if you are using a mobile broadband connection.
2. Raising the Record Buffer to 80 - 120 ms. We don't recommend going much higher than that as the added delay makes proper conversations impossible from some point on.

Please let me know if this helped you to improve the situation - I am looking forward to hearing from you again!

we set 120 ms for recording buffer and get a better result.
thanks..

For bandwidth issue,

How do we select prefered CODEC for the call?

best regards
kaiser

That's great to hear and as for selecting the audio codec, there is currently no entry in the GUI to do so. Please close down AGEphone Mobile and go to "My Documents / My Phone Booth" on your device and open "sipd.conf" with a text editor. Under [General] you will find the string

MediaTypes=3 0 8 101

where 3 stands for GSM-FR, 0 for G.711u and 8 for G.711a. You can rearrange and delete the different options at will. Removing "3" for example will force G.711 to be used. The 101 is for DTMF events and should never be deleted or rearranged. I hope this helps you to set the right codec and we will definitely add an option to do this without having to change the "sipd.conf".