Best Codec?

[Yard]

Right now we are streaming with windows media player, 20kbps 16khz stereo

we want to move to a linux server, so need to use shoutcast, but I try the mp3 20kpbs 16khz mono (no stereo there) and the song is not very good... maybe I need to use another codec...

user connect with wmp9 and realmedia...
which codec it will be the best without people donwload one?

Yard

[jackherer]

Shoucast only broadcasts mp3. If you want to use something else you will need different server software. Real's Helix Universal Server will run on linux, but the free version only supports up to 1Mbps and/or 10 connections. You could also try icecast2 which supports ogg, but you can only get it in source code form so you have to know how to compile stuff. 20kbps 16khz is always going to sound pretty bad. Have you tried going up to 24kbps which you can do 22khz with.

[Yard]

I wanna transfert to shoutcast

for now we are streaming at 20 kbps 16 khz stereo with codec wma9, I try 20 kpbs 16 khz mono for mp3 and the sound is near the same... but in mono... wanna stream in stereo, we have too many people... more of 800, at 24 it take 0.5 kb/sec more for each people.. so many many bandwith per month over

Sorry for bad english ...

I'm using Sam encoder, and I don't have choice for 20kbs 16khz stereo... can I take other codec?

[jackherer]

I don't think MP3 supports 20kbps 16khz stereo with any encoder, i realy don't think the difference between stereo and mono matters at such a low frequency/bandwidth. Can you even hear the affect. The difference in quality is probably because Windows Media is a newer codec so is better at encoding at such low Kbps rather than the stereo thing. I say anything below about 80Kbps should be mono.

[Yard]

Hmm I will try, but maybe I can use something better of codec lame? I see somewhere about some other codev better of lame for the same quality?

[jackherer]

Lame is an encoder, mp3 is the codec. Everything i have read says that lame is the best MP3 encoder that you can get. If you want better quality at lower rates, don't use MP3, it is very old and Ogg is much better at lower bit rates.

[Yard]

but I don't think windows media player can read Ogg

[olden]

For decent bitrates and great quality, Ogg Vorbis rules.
Look no further. If you're new to this codec, have a quick glance at http://www.vorbis.com/faq.psp

About 2 years ago, I've tried virtually every codec you may think of. I was targeting a bitrate slightly higher than you, though, in the 64 to 80 kbps range for 32~44kHz stereo, and I tested mostly with rock/pop music and some voice.

The highest quality codecs IMHO were
- Sorenson QDesign Music 2
- AAC (the kind part of MPEG4-audio now)
- NTT/Yamaha's TwinVQ (aka VQF)
- Ogg Vorbis (in beta at the time)

But if you have some time, please don't take my word for it, try it too. A few readily encoded->decoded samples: http://www.xiph.org/ogg/vorbis/listen.html

However, unfortunately, "playability" with 3 out of the 4 aforementioned codecs was and still is problematic: only QuickTime reads QDesign format, AAC has many incompatible variants, only a few players have plugins for TwinVQ... remains Ogg Vorbis.

Ogg Vorbis support now comes standard with a bunch of players, including Winamp, MacAmp, RealPlayer etc... and plug-ins exist for most of the older ones, including WMP. More on http://www.vorbis.com/software.psp (non-exhaustive list)

RealNetworks' players, servers and other tools do or will support Ogg Vorbis as well.
If you're on a budget though, I'd say just stream with Icecast. Plus, you can get some fans to offload your server(s) with PeerCast –- everyone wins

[Peter Tate]

Guys,

I maybe going off half cocked here but!!!

I use the SAM encoder from spacial audio, specifically the Winamp plugin one.

This I find is the best and the only one that can encode mp3pro. The comments about shoutcast being crap at lower bit rates is not correct!

The big thing for me is the "wishy washy" sound when you get artifacts from mp3's as source files etc. The SAM encoder doesn't suffer this too much. For me an AM sounding stream with NO artifacts is better than anything TRYING to reproduce tops. My theory is don't encode it!!!! All audio processing here has this in mind.

Also use only the bit rate but 24Kbits @ 22Khz. This will keep modem people happy by not hogging all their bandwidth. 32/46/56 Kbits etc is a waste of time as the wishy washy comes back.

The next stop is the 44khz SAMPLING RATE----an encoder that will give you this down in modem area is not gonna happen. Although I've seen some working at 48/56, my research shows the best to be 64Kbits. Thus you are into people with broadband connections and good bye to modem listeners.

For me it would be the SAM encoder on two seperate computers one 24Kbits @22Khz----a second one runing 64Kbits @44Khz MONO (don't even think about stereo until 128Kbits).

The audio really needs to be processed with two schools of thought in mind hence two encoding machines. Both with nice Broadcast processsors in front of them one set for AM type audio characterisitcs the other FM but no pre-emph.

The isssue of Stereo vs mono is another issue (money mainly for the bandwidth)but for me top class HIFI can come at the 64Kbit level in mono!!!!

Hope this helps a bit---www.spacialaudio.com

An example of what I mean is my station at www.stationx.com.au

[jackherer]

Erm good advice, but if you read further up the thread he seems to have his heart set on 20Kbps and just wanted the best quality at that and no other bitrate.

[Vchat20]

i dont like to push people into things, but i would highly recommend he move up to the 24kbps, 22khz. ive messed with bitrates in shoutcast before, and 24kbps sounds much better than 20kbps if you ask me. if anything he should look into mp3pro if he wants that low of a bitrate, but again, mp3pro creates a lot of artifacts in the proccess.

my general rule for shoutcast broadcasting is as follows:
-------------
modem users: 24kbps, 22khz, mono -- best quality at this low of bitrate. and you dont chew up all the listeners bandwidth bt going at 32 or 46kbps.

broadband users: 128kbps, 44khz, stereo -- i have yet to see any broadband user that cant handle this bitrate. if anything, anyone who cant is obviously on ISDN which i still consider to be modem, if not dialup. and anyone who is should seriously upgrade to broadband. ISDN is way too expensive compared to broadband.

[jackherer]

Originally posted by Yard
I wanna transfert to shoutcast

for now we are streaming at 20 kbps 16 khz stereo with codec wma9, I try 20 kpbs 16 khz mono for mp3 and the sound is near the same... but in mono... wanna stream in stereo, we have too many people... more of 800, at 24 it take 0.5 kb/sec more for each people.. so many many bandwith per month over

Sorry for bad english ...

I'm using Sam encoder, and I don't have choice for 20kbs 16khz stereo... can I take other codec?

He can't afford the bandwidth, MP3 pro just sound like mp3 if you don't have the plug in.

[Vchat20]

im just wondering why he cant afford the bandwidth. call me crazy, but moving from 20kbps to 24kbps should make a very VERY small difference if none at all. id like to know how he's being hosted and cant afford a marginal bitrate change.

[jackherer]

dunno 4kbps * 800 listeners is 3200 Kbps, or 3.2 Mbps that is quite a lot realy.

[Vchat20]

not really when you look at it from how much the stream host has available as far as bandwidth is concerned. i mean look, stream hosts such as fast-serv (god help me. ), can manage to throw a mess of multi-hundred listener DNAS's on a single server. obviously there is some major bandwidth going on there. plus compare the 20kbps hes already using to that 4kbps difference. i mean, unless hes hosting this all by himself, i see no major financial or bandwidth issue when changing the bitrate. if he is hosting this by himself, still, i dont see where the problem is. if he can manage the massive pipe for his current stream, he should have no problem affording the extra 3.2Mbit

[jackherer]

Dunno mate, he says he can't afford the extra bandwidth no point arguing with me, the first thing i sugested was broadcasting at 24kbps and he said he didn't have the bandwidth. i'm borded of this now, take it up with him.

[Vchat20]

im not trying to argue, just trying to provide suggestions. obviously, if he wants more quality, maybe he should drop his listeneer count a bit to compensate for the higher bitrate?

[Yard]

Thank you very much for all answer _ I think I will keep the 20 kbps mono, it sound good with sam

Yard