FeenPhone

No-cost software for real-time high-quality remote podcasting, radio and voiceover
 

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Features

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Check mark check3means it has been implemented already.

 

Do next. More complex to implement, but are the most important for making FeenPhone a “must have”, and for making it future proof:

–BETTER ERROR CORRECTION to deal with non-perfect networks.

–IPv6 support.
–WASAPI on audio in.
–Automatic port forwarding.

–Ability to raise volume of input to 150%
–More robust password authentication.
–Recording (more info HERE)

–Hot Swapping mics and headphones

–a couple small issues: cursor should go straight into text fields to enter Server. And highlighting “Change me” should delete that whole placeholder, not just part of it. Change BTC address to website with updatable address.

–ANDROID SUPPORT for easily doing remotes.

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PROOF OF CONCEPT:

P2P HQ mono true-duplex audio direct from one computer to another via Opus codec.check3

No noise reduction or echo cancellation, by choice. check3

Solid networking built from the ground up. check3

Extensible for other protocols.check3

Drop-down menus to pick audio input, audio output, buffer settings and codec settings.check3

Text chat for troubleshooting.check3

Saves settings (including remote IP) on exit.check3

Buffer meter check3

 

BASIC REQUIREMENTS:

Low-latencycheck3

No creep in latency. check3

Roll separate server module into program.check3

Multi-protocol support (TCP, UDP, Telnet)check3

Master volume VU metercheck3

Additional audio format / codec: PCM, G.722. (were later removed from interface, didn’t sound as good as Opus) check3

Minimal clicking and drop-outs. check3

Doesn’t crash.check3

Works almost flawlessly for two hours of solid two-way talk. check3

Ability to add third person to conversation. check3

More variations in Opus settings. check3

Default to best buffer settings on all three sliders.check3

Input VU meter for each hostcheck3

Volume sliderscheck3

Default to Opus settings: Sample rate 24 kHz, Bit rate: 32 Kb/s, Super Wide Band (12 kHz) but variable.check3

Remove codec settings that don’t test well or aren’t needed.check3

Decide proper license.check3

Make buffer reset button a little bigger, and easier to see.check3

Label buffer reset “Buff Dump” (radio broadcasters will confuse simply “Dump” with a cuss-dump button).check3

Headphone icon on Audio Out sectioncheck3

Hide little-used options, make accessible with Advanced Options buttoncheck3

About Tab with: Readme, manual, credits, License, donate link, BTC address and web links.check3

Basic icons and minimalist branding.check3

 

ADDITIONAL GOALS FOR 1.0 RELEASE TO PUBLIC:

Extensive testing completed, bugs worked out.check3

Works flawlessly for two hours of solid two-way talk.check3

Well-written, easy to understand manual.check3

 

GOAL LIST FOR FULL SECOND ROUND OF FUNDING:

Create installer, package for public use.check3

Code added on GitHubcheck3

Make volume slider on Audio Out work more smoothly.check3

Switchable auto-answer function (with password protection).check3

Input / Output level sliders.check3

–Change “Run Server” button to a tick box, or make it change color when pressed.check3
–Add a “Phone book” to save different IP connections and add name or label.check3
–Cough button check3
–Audio alerts on incoming call (DJ’s “FeenPhone, FeenPhone” sample.) check3
–Add call clock (for each caller, in case the third person comes in after the second person.) check3
–Hide WASAPI option for Audio In drop down.check3
–Passwords not obscured in interface.check3

–WASAPI support for Audio In / Lower latency in program.check3

–Fix issue with passwords, not logging on perfect every time.check3

–One-direction only option.check3
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.OTHER FEATURES TO CONSIDER ADDING

SIP support

Port randomizing option (within a range).

–Ability to mute and block persons in conversation.

Ability to block IPs.

–Advanced Option: Toggable gentle EQ boost at 12hz. And/or maybe a 4-band EQ with sliders.

IP address display, and ability to copy IP address to the clipboard.

See what else we can move to advanced options.
–Auto-reconnect if the connection drops.

Add ability to save different configurations, and name them so you call them up easily.

Option to auto-adjust settings for best sound and lowest latency depending on network.

IP ACL

RADIUS support

Ability to use stereo transmission.

Ability to roll over to a second pre-set IP if first connection is lost.

Mix Minus (this would be a holy grail, it would make this a killer app by replacing audio hardware).

Automation interface with relay closures

PAN left and right: for recording two-host shows for later mixing.

General overall improvements in all functionalities.

WebRTC support (for callers calling in to a radio show or podcast).

Pro interface and website.

No clicking or drop outs.

Windows Phone version.

 

WISH LIST OF FEATURES ADDED BY OTHERS, VIA OPEN-SOURCING:

Linux version.

Mac version.

Android version.
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