BriansBrain

[quote author=”didac”]PD: Can you include the standard STL format (MPEG I Layer II)?[/quote]
My programs can only Stream what is available in your Send/Receive PC’s
Microsoft Audio Compression Manager

Open > Control Panel
Select > Sounds and Audio Devices
Then > Hardware
Look for Audio Codecs

In my systems it looks like this…

In my program you can Choose the Streaming format
like these three examples PCM MP3 OGG

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And here is the Receiver UDP and PCM over Lan at full whack 😛

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[quote author=”Modulator”]But can you adjust the buffering?[/quote]

In the receiver Max Buffer Size can be adjusted
100ms to 1000ms -or- 0.1 to > 1.0 second

When using UDP Protocol the delay over Lan is 0.1s even the though
the Min Available Buffer Size set to 0.5s.

The buffer is only needed and used for the TCP Protocol
when the internet connection is not very stable.

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[quote author=”BriansBrain”][quote author=”djscooby”]Is it possible to try this over lan & go better quality with pcm codec ?
[/quote]
I have fully tested this over my own Lan, sollid stable signal
and very low latency @ 200kbs using PCM and UDP protocol.
I will be posting the latency millisecond numbers soon 😛
[/quote]
How measured, seperate PC Audio recording in Cool Pro.
Left channel = original signal
Right channel = from the Receiver
Break the signal for 100ms, measure the delay difference of the recorded wav 😛

And here we go using UDP Protocol over Lan – streaming @
PCM 48000kHz, 16 Bit, Stereo – @ 200kbs = Latency 100 milliseconds Constant
Ogg Vorbis 48000 Hz, Stereo, About 450kbps – @ 60kbs = Latency 240 milliseconds Constant
MPEG Layer-3 320 kBit/s, 48000, Stereo – @ 41kbs = Latency 240 milliseconds Constant

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[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.

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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.

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