My Audio Over IP, STL , PCM, QDesign MP2, Ogg Vorbis, mp3

[quote author=”bennylein1985″]hrhr…..

try using ct´s he-aac encoder@48khz 32kpbs an PS with edcast and a icecast2 server.[/quote]

What about the delay 😕

[quote author=”Modulator”]Are delays lower than 500ms possible? Something like 50ms would be cool. FLAC![/quote]

Even on tests streaming one PC to another @ PCM 44100, 16kHz, Stereo
just throuth a router (no internet) I can’t get better than 100ms 😕

It’s because it is Streaming 🙂
So the server makes the packets up first before it sends them,
then the Client has to get all the packet before it can make it
back into the wav it started life as.

8)

[quote author=”yorkie98″]If you need someone to assist with sending audio out to test the software, PM me, I can spare a couple of Mbps without a problem.[/quote]

Thanks a lot Yorkie,
I might take you up on your offer 😛

We have 10Mbps down in Gran Canaria but only 0.82Mbps up MAX if we are lucky 😕
Infact it’s more like 0.3Mbps most of the time 🙁

So I can’t give it a full test.

BB 8)

Codec Formats Tested

Streaming Bandwidth Needed (Upload) @

Microsoft PCM
From 8000kHz, 8 Bit, Mono – @ 10kbs
To 48000kHz, 16 Bit, Stereo – @ 200kbs << Better than CD Quality but 200kbs upload needed
MPEG Layer-3
From 18 kBit/s, 12000, Stereo – @ 3kbs
To 320 kBit/s, 48000, Stereo – @ 41kbs
Microsoft ADPCM
From 8000kHz, 4 Bit, Mono – @ 5kbs
To 44000kHz, 4 Bit, Stereo – @ 45kbs
GSM 6.1
From 8000kHz, Mono – @ 2kbs
To 44000kHz, Mono – @ 10kbs
CCIT A-Law or u-Law
From 8000kHz, 8 Bit, Mono – @ 10kbs
To 44000kHz, 8 Bit, Stereo – @ 90kbs

And now with Ogg Vorbis Codecs
Using Mode1 the Original Stream Compatible.
Format Tag: 26447 (mode1) = VBR
Format Tag: 26479 (mode1+) = CBR

Streaming Bandwidth Needed (Upload) @ min > max
Only tested the top end 128kbps upwards with Music & Speech

In Mono
48000 Hz, Mono, About 128kbps (Q:0.7) – @ 16kbs > 20kbs
48000 Hz, Mono, About 144kbps (Q:0.8 ) – @ 18kbs > 22kbs
48000 Hz, Mono, About 192kbps (Q:0.9) – @ 23kbs > 27kbs
48000 Hz, Mono, About 256kbps (Q:1.0) – @ 28kbs > 34kbs

In Stereo
48000 Hz, Stereo, About 128kbps (Q:0.4) – @ 17kbs > 24kbs
48000 Hz, Stereo, About 160kbps (Q:0.5) – @ 20kbs > 30kbs
48000 Hz, Stereo, About 192kbps (Q:0.6) – @ 25kbs > 34kbs
48000 Hz, Stereo, About 240kbps (Q:0.7) – @ 28kbs > 37kbs
48000 Hz, Stereo, About 256kbps (Q:0.8 ) – @ 30kbs > 40kbs
48000 Hz, Stereo, About 350kbps (Q:0.9) – @ 40kbs > 48kbs
48000 Hz, Stereo, About 450kbps (Q:1.0) – @ 48kbs > 60kbs <<< This was the one I was after 😛

And…
QDesign MPEG I Layer II 48000 Hz, 256 kBits/s – @ 32kbs

Using the UDP protocol over the Internet gives the lowest possible delay (500ms).
But, if the internet connection is not very stable with UDP you will loose data.

TCP protocol has guaranteed delivery of packets, you will not hear disturbances.
One penalty with TCP is a delay of a few seconds in the audio signal.

I will be looking for testers if anyone is interested (Yorkie has already offered)
No special requirements are needed, Windows 32 bit system, Pentium 3 minimum.
For the Sender my program and the ACM codecs.
For the Receiver, a Static Internet IP Number is requiered, my program,
the ACM codecs and a Port has to be opend up in NAT settings in your router.

BB 8)

[quote author=”Boki”]I am already using 48kHz 16bit PCM (~1.6Mbits bandwidth) with around 100-200ms with Wifi Link…

But always want to check new things. 🙂[/quote]

100-200ms is very good, you must be using UDP not TCP ?
I will have to do a speed test of mine over a private network.

What sort of WiFi distance are you using ?

[quote author=”Boki”]It’s always UDP. Do not ever think about TCP for low-latency audio over IP. Also when you think about Low-Latency on PC+Windows, don’t forget ASIO.
Distance is about 5.5km with 90cm dish antennas and Bullet5 on both sides. Soon will update to Bullet5MP.[/quote]

Sounds a great link you have.
Do you have to have Line of Sight for the antennas ?

I have made my STL with low speed (Upload) internet in mind, like mine.
My ADSL line specs = 10Mbps Download and 0.82Mbps Upload.
Infact it’s more like +/- 0.3Mbps unstable Upload most of the time.

So 25kbs > 35kbs upload would be safe.
So I have to go Codec.
Ogg Vorbis @ 48000 Hz, Stereo, 240kbps – @ 28kbs > 37kbs

Because the Codec Encode + Decode itself produces delay, plus the Internet delay
it is Impossible to get low-latency audio over IP in this situation.

So using UDP to obtain the lowest possible delay is pointless because of the erratic
low Upload speed and the possibility of loosing data, also I’m forgetting ASIO.

That’s why I’m going with TCP protocol, guaranteed delivery of packets,
penalty the delay of an extra second in the audio signal.

BB 8)