View Full Version : encoder settings
18th January 2002, 11:42
ok dont call me ghey or nothing but for some reson i can only get a max of 56kbps 22.050kHz
and the other people who are playing on the same shout server as me have 64kbps avalible
why? and how?
i have a sblive and win2k
is it my mic?
some 411 help :)
18th January 2002, 11:45
Piece-o'-crap Microsoft MP3 ACM codec to blame.
Get Radium (http://www.riphelp.com/downloads/radium_codec.html)
18th January 2002, 11:47
any better places to d/l them from the yahoo thing is overused :/
18th January 2002, 11:57
ok i have it now how do it get it working?
i have installed it but i dont see anything inside winamp to select that
18th January 2002, 19:42
I'm going to assume that you are not a hardware techie.
Open Contol Panel --> Sounds and Multimedia. Pick the Hareware tab. In the list box select Audio Codecs. Click the Properties button.
You should see a list box with all the audio codecs installed on the machine. You should see 2 MP3 codecs listed, the "Advanced" and the "Professional"
Select the "Advanced" one first and click the Propeties button. Move its priority to 1. Click ok to save the settings.
Now select th "Professional" one and do the same thing to it.
Close all th windows and reboot.
This setup should alow you to have as many as 2 encoders running. The first one that starts will be able to use the new codec and the second one will be limited to the old maximum bit rate.
I hope this helps
19th January 2002, 02:01
thanks i will just reboot now :)
when i am on the shout cast server every one says the music is very quiet how do i fix that?
19th January 2002, 02:51
Well my friend this is another can of worms. A number 10 sized can.
For this to be handled you first need most of you MP3 files to have ben encoded with much the same level, it does not have to be right on but not all over the place. If you are ripping all of your own music you cna do a lot by setting some "normalization" settings in your ripper. You want the peaks to be in a range of about 15db.
Next you need to set the levels on your player Winamp, I like to keep mine at about 60% but you should set yours while listening to another computer playing your stream back to you.
To get that fat in your face sound that people associate with FM Rock & Roll, you will need to do some compression and maybe some limiting. I'm now using Audio Stocker for compresion and RockSteady for limiting. You will find that the real difference between compressing and limiting is the amount of gain reduction and the speeds with which the units or DSPs work. High compression ratios and fast acting units produce what we all limiting.
If you are using any anlalouge sources or going out and back to a mixing console you will have another headake to worry about: The A->D convertion. The imput to most comsumer an prosumer sound cards is an active gain stage. All level controls available on the system are after this gain stage. If you are using a pro or prosumer audio mixer the line level output is usualy in the range of +4dbm, nominal. The input to the sound cards we are talking about is -10dbv. To make matters worse many of the consoles that in a resonable price/performance range have a great deal of headroom and don't have a strong indication of when you are going above the nominal output level.
We are using console from Mackey that guarantees no distortion out to +28dbm and does not show yellow until +8dbm. This level will cause the input of the sound card to distort and no amount a level setting on the computer will help.
You might decide to not run the console at full level but that would move the noise floor in the console up into the level region that the music and voices ocupy. An I have yet to meet a DJ who is willing and able to keep the level down that low into the "green"
My solution to that part of this problem is to install 30db loss pads between the console and the input to the sound card.
The second part of the A->D problem is the amount of high frequency energy in the sound you are handling. Rock & Roll has a very high concetration of high order homonics and sharp attcack transients. A->D converters have very rigid frequency cielings determined by the sampling rate. The highest frequency the converter can handle must be less tha on half the sampling frquecy.
In all A->D converters you will find a set of low pass filters to prevent the signal from containing frequecies that are to high for the sample rate. In all but a very, very few cases the filters have very sharp skirt slopes. this is to allow as much information through with out going over the limit. The nature of filters or wqualizers, which is what they are, is that when the are driven with to much level they will ring and the level at which they will ring decresses with the increse in the frequecy they are fed.
Most of the exprimental data that is used to design these "antiailiasing filters" was based on the spectral content of Classical Music, which has much less high energy high frequency content than that of Rock & Roll. So when you raise the average level of your signal these filters will tend to ring on cymbols, snare drums and loud high notes from guitars.
If you are doing a stream with a bit rate below say about 56Kb you may want to employ an outboard equalizer to contol the highfrequency energy.
I hope this helps
19th January 2002, 03:02
im going 2 have to print this out :) looks liek you know your stuff
but lets say i was just playing from winamp the mp3's
how do i boost the volume?
19th January 2002, 03:36
If you are encoding in the Wianamp you are using as a player then only the first four paragraphs apply to getting your level up. The rest would apply to the sound you are injecting into your voice over music funtion in the encoder or a plugin.
The first thing to do is to get you MP3s into the ball park levle wise. Do that while ripping them or load each one into an editor such as Sound Forge and set the audio levels of each file there Then save them back, and be sure to make sure you are putting the ID3 tags back in.
Next use another computer to listen to anothe station that sounds the way you want your stream to sound. With out changing anthing on this monitoring computer listen to your stream with it.
Now WHILE LISTENING to your stream with the SECOND computer set the level o the Winamp that is playing back and encoding your stream. The only way to know how your stream really sounds is to listen to it coming back to you over the Net.
I hope this helps.
19th January 2002, 03:39
i get ya
so i burn then re rip them??
cous there is no way i can find a cd with the fraggle rock theam song on it
19th January 2002, 03:49
For things that you do not have the source CD to reripp, you will have to use a sound editor. I like Sound Fordge but Cool Edit Pro works fine too.
19th January 2002, 03:59
do ithere of them just alter the mp3 with out having 2 reburn?
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