Peak Level Meter in

I'm using a SB Li've! Platinum 5. w/ Li've Dri've in my PC and I just upgraded to XP. When I installed the XP drivers for my card, I lost the peak level meter in my sound control window that I had back when I was in 98SE. I know there is 3rd party software that I can use to get a peak level meter, but are there ANY XP drivers for my card that have a built in peak level meter?
I ask, because an application I'm using (Sub Station Alpha) requires that I have a "built in" peak level meter in my drivers in order to use the audio triggering function.
Any help is greatly appreciated.
[EDIT]
If it helps at all, I am refferering to the meter you see on the left side next to the main audio slider. If you are runnin an SB card in XP and you have a peak level meter, please let me know what drivers you are using, thanks.
http://img368.imageshack.us/my.php?image=peaklevelmeter8yi.gif
Message Edited by Ruroni6 on 05-23-2006 09:4 AM

1) Bounce a file with the master at 0 DBfs, and bounce it with the master at -12 DBfs, and see if there is a difference.
2) The metering plugin is situated before the fader, that's why the metering isn't affected by fader movements. The same would go for any plugin inserted after the Meter: it will not influence the meter, even if it has a volume control that is fully closed (-∞). Signal flow is the keyword here.
regards, Erik.

Similar Messages

Confused with Logic level meter

Hi guys, could anyone help me with some piece of advice?I have a problem with track levels, my mixes are usually quiet, but my aim is to get them to the yet fatiguing but still standard very loud level. Also when mixing I can hear that the arrangement is groovy and punchy, but when I get it through Ozone or Logic mastering plug-ins it turns into a komplete mess without any definition, the waveform of the track is brushed heavily . One person said to get a loud nice master I need to watch the Logic Output fader meter for headroom about 6 or 3 dbs before zero and RMS about -9 db, the thing is I'm konfused now with the Level meter as RMS showing displays red klips , but the signal is actually lower, and I can see it, for ex. when klipping the signal is about - 15 dbs RMS, could anyone explain me the real way to get loud tracks, and what approximate settings should I stick to, as it starts getting on my nerves already cause I start to think I just can't do music..

First, try to separate the concepts of mixing and mastering. They're two entirely different processes. I'd suggest you first concentrated on getting your mix 'right' for you, and don't worry about the relative loudness until you've completed this first mixing stage. I wrote a brief blog post about this here which might help a little.
In general terms, you don't need to worry about occasional high levels on your individual tracks - only on mix bus on your stereo output, where a good peak level might be around -8 dbFS. That's the only place where you can't 'go into the red.'
Once you have your mix as good as you can get it, *regardless of volume,* then open that bounced file in a new Logic session. That's where you can experiment with Ozone, Adaptive Limiter, Multiband compression and other tools you can try.
You might want to do some further reading about mastering. Here are a few articles/forums which may help get you going:
http://www.soundonsound.com/sos/aug04/articles/computermastering.htm
http://community.sonikmatter.com/forums/index.php?/topic/4200-mastering-in-logic /
http://www.musicbizacademy.com/articles/mastering.htm
and there are dozens of posts in this forum - just search for "mastering"

JMF Challenge - Create audio volume level meter for rtp Audio Transmitter

Based on the following code at: http://java.sun.com/products/java-media/jmf/2.1.1/solutions/RTPConnector.html
The challenge is to build an Audio level meter, you know the lights that go up and down to your voice like on a stereo Hi-Fi system.
If we can start with ideas, give it time the challenge will be complete for all to use.

Heres the finished codec code, it just spits out System.out's with the peak level.
import javax.media.*;
import javax.media.protocol.*;
import javax.media.protocol.DataSource;
import javax.media.format.*;
import javax.media.control.TrackControl;
import javax.media.control.QualityControl;
import javax.media.control.SilenceSuppressionControl;//###### delete if no silence.
import javax.media.rtp.*;
import javax.media.rtp.rtcp.*;
import com.sun.media.rtp.*;
public class AudioLevelMeterCodec implements Effect {
    /** The effect name **/
    private static String EffectName="AudioLevelMeterCodec";
    /** chosen input Format **/
    protected AudioFormat inputFormat;
    /** chosen output Format **/
    protected AudioFormat outputFormat;
    /** supported input Formats **/
    protected Format[] supportedInputFormats=new Format[0];
    /** supported output Formats **/
    protected Format[] supportedOutputFormats=new Format[0];
    /** selected Gain **/
    protected float gain = 2.0F;
    /** initialize the formats **/
    public AudioLevelMeterCodec() {
        supportedInputFormats = new Format[] {
            new AudioFormat(
                    AudioFormat.LINEAR,
                    Format.NOT_SPECIFIED,
                    8,
                    Format.NOT_SPECIFIED,
                    AudioFormat.LITTLE_ENDIAN,
                    AudioFormat.SIGNED,
                    8,
                    Format.NOT_SPECIFIED,
                    Format.byteArray
        supportedOutputFormats = new Format[] {
            new AudioFormat(
                    AudioFormat.LINEAR,
                    Format.NOT_SPECIFIED,
                    8,
                    Format.NOT_SPECIFIED,
                    AudioFormat.LITTLE_ENDIAN,
                    AudioFormat.SIGNED,
                    8,
                    Format.NOT_SPECIFIED,
                    Format.byteArray
    /** get the resources needed by this effect **/
    public void open() throws ResourceUnavailableException {
    /** free the resources allocated by this codec **/
    public void close() {
    /** reset the codec **/
    public void reset() {
    /** no controls for this simple effect **/
    public Object[] getControls() {
        return (Object[]) new Control[0];
     * Return the control based on a control type for the effect.
    public Object getControl(String controlType) {
        try {
            Class cls = Class.forName(controlType);
            Object cs[] = getControls();
            for (int i = 0; i < cs.length; i++) {
                if (cls.isInstance(cs))
return cs[i];
return null;
} catch (Exception e) { // no such controlType or such control
return null;
/************** format methods *************/
/** set the input format **/
public Format setInputFormat(Format input) {
// the following code assumes valid Format
inputFormat = (AudioFormat)input;
return (Format)inputFormat;
/** set the output format **/
public Format setOutputFormat(Format output) {
// the following code assumes valid Format
outputFormat = (AudioFormat)output;
return (Format)outputFormat;
/** get the input format **/
protected Format getInputFormat() {
return inputFormat;
/** get the output format **/
protected Format getOutputFormat() {
return outputFormat;
/** supported input formats **/
public Format [] getSupportedInputFormats() {
return supportedInputFormats;
/** output Formats for the selected input format **/
public Format [] getSupportedOutputFormats(Format in) {
if (! (in instanceof AudioFormat) )
return new Format[0];
AudioFormat iaf=(AudioFormat) in;
if (!iaf.matches(supportedInputFormats[0]))
return new Format[0];
AudioFormat oaf= new AudioFormat(
AudioFormat.LINEAR,
iaf.getSampleRate(),
8,
iaf.getChannels(),
AudioFormat.LITTLE_ENDIAN,
AudioFormat.SIGNED,
8,
Format.NOT_SPECIFIED,
Format.byteArray
return new Format[] {oaf};
/** gain accessor method **/
public void setGain(float newGain){
gain=newGain;
/** return effect name **/
public String getName() {
return EffectName;
/** do the processing **/
public int process(Buffer inputBuffer, Buffer outputBuffer){
// == prolog
byte[] inData = (byte[])inputBuffer.getData();
int inLength = inputBuffer.getLength();
int inOffset = inputBuffer.getOffset();
byte[] outData = validateByteArraySize(outputBuffer, inLength);
int outOffset = outputBuffer.getOffset();
int samplesNumber = inLength / 2 ;
int valueMin = 255;
int valueMax = 0;
for (int i=0; i< samplesNumber;i++) {
int tempL = inData[inOffset ++];
int tempH = inData[inOffset ++];
int sample = tempH | (tempL & 255);
if (sample > valueMax) {
valueMax = sample;
if (sample < valueMin) {
valueMin = sample;
System.out.println((valueMax - valueMin) - 256);
System.arraycopy(inData,0,outData,0,inData.length);
// == epilog
updateOutput(outputBuffer,outputFormat, inData.length, 0);
return BUFFER_PROCESSED_OK;
* Utility: validate that the Buffer object's data size is at least
* newSize bytes.
* @return array with sufficient capacity
protected byte[] validateByteArraySize(Buffer buffer,int newSize) {
Object objectArray=buffer.getData();
byte[] typedArray;
if (objectArray instanceof byte[]) { // is correct type AND not null
typedArray=(byte[])objectArray;
if (typedArray.length >= newSize ) { // is sufficient capacity
return typedArray;
System.out.println(getClass().getName()+
" : allocating byte["+newSize+"] ");
typedArray = new byte[newSize];
buffer.setData(typedArray);
return typedArray;
/** utility: update the output buffer fields **/
protected void updateOutput(Buffer outputBuffer,
Format format,int length, int offset) {
outputBuffer.setFormat(format);
outputBuffer.setLength(length);
outputBuffer.setOffset(offset);

Is it possible to change the display mode of the dB Number on level meter?

the default dB number just shows up the maxi happened peak level and keeps
holding the level number until the next higher level. any one know if there is any other mode to have?

It's a headroom meter, and can be reset by clicking it (there is a key command for this also).
It's purpose is to let you know the maximum level your channel has reached, and therefore how much headroom you have (or not). I'm not sure what other "modes" you'd want - what did you have in mind?

Peak level + rtp server / client

Hi all JMF developper,
Here are my questions. I have to develop a simple application that capture voice coming from a microphone and then transmit it over a network (the Internet in general). I would like to develop a peak level as many audio tools provide (just to see if the recording level is good).
Then it will be very interesting for me to get materials, resources, codes or anything else about a simple JMF server / client. We plan to use the Darwin streaming server.
From the client side, I would like to know if there a buffer security mechanism built in or not (for instance, listener that warns that the buffer will become empty in order to stop anything else that is synchronized with the voice)
Kind regards,
St�phane

anybody does how I can create a Peak level in my GUI (a volume meter while recording)
Thanks

Signal Level Meter

Signal Level Meter
CATV Signal Level Meter GAO2002 is specially designed for CATV system maintenance. Signal Level Meter GAO2002 features small size (160mm x 130mm x 65mm) , light weight (368g including battery), long operating time (more than 6 hours), and well-built from appearance to architecture. Signal Level Meter GAO2002 combines the most practical functions and makes them easy to use. What is even better about this piece of equipment is that its added aseismatic design makes it more durable than you can expect. Tests show that it can continue to work well after being dropped from 5 meters high. It supports Level, V/A, Tilt and Trunk Level testing as well as testing two channels at the same time.
CATV Signal Level Meter GAO2002 Specifications:
Frequency
Frequency range: 46MHz ~ 860MHz
Frequency step: 50kHz, 1MHz, 10MHz and 100MHz
Level Measurement
Range: 20dBuV ~ 50dBuV
Accuracy: ±20dB (20ºC ± 5ºC)
Resolution: 0.5dB
Carrier to Noise (C/N)
Option Range: 20dBuV ~ 50dBuV
Input signal range: ≥ 85dB
Measure accuracy: ± 2.5dB (20ºC ± 5ºC) shut off the carrier
Resolution: 0.5dB
Voltage
Input range: 1~100V (AC/DC)
Measured accuracy: ±2V
Resolution: 1V
Miscellaneous
Size: 160mm x 130mm x 65mm
Weight: 368g (including battery)
Working: -10ºC ~ 40ºC
Audio: Built-in speaker (Auto on in single frequency mode)
Battery
Built-in battery: 3.6V/2.1AH Ni-MH
Working time: More than 6 hours (Shut off the audio and LCD backlight)
Charging time: 10 ~ 12 hours (power off)
Gao Tek Inc.

Do you have drivers for LabVIEW?
~~~~~~~~~~~~~~~~~~~~~~~~~~
"It’s the questions that drive us.”
~~~~~~~~~~~~~~~~~~~~~~~~~~

Sound Level Meter vi

It would be great if I could manage to have a Sound Level Meter vi. I do not have SVT.
Right now a .vi sharing would be best, but in my mind it is to start to make my own, this kind of tasks make you learn. The problem is time.
Any sharings? Suggestions? I found this in an old post, but, I get this error:
"Poly VI 'AI Acquire Waveform.vi': subVI is missing.
Attachments:
SPL.llb ‏88 KB

AI Acquire Waveform is part of the old traditional DAQ API. You can either replace the traditional DAQ VIs with DAQmx VIs or install traditional DAQ (provided you are on a supported, 32-bit OS). I would recommend the former. The sound level algorithms should still be good. Let us know if you run into issues.
This account is no longer active. Contact ShadesOfGray for current posts and information.

Level Meter Plug In -VS- Output 1/2 meter...Inconsistencies...

I'm mixing an 18 minute long recording for vinyl release. The vinyl pressing company has a minimum dbl on recordings they master and press. If the level is under, they charge hundreds of dollars more to re master and increase the volume via massive compression.
So to check and double check the overall volume in Logic Express 8, all I can find for a meter is the "Level Meter" in the visualization plug in toolbox. So I have the Level Meter "plugged" into the Output 1/2 track...The problem with the Level Meter is that it is showing levels around 8 dbl lower than the Output 1/2 meter. This could potentially be an expensive discrepancy not to mention may have a negative effect on the end product's sound quality.

Eriksimon wrote:
If you used a meter as an insert, it comes, like other effects plugins, before the fader. Sorry, I didn't read attentively the first time.
Still, the effect is the same as pre-fader metering. The signal passes the meter before it goes to the fader. If you turn pre-fader metering on you'll see that your faders don't affect the channel strip level meters anymore and behave like inserted level meters as well.
So if you lower the fader on channel 1&2 so you no longer see clipping, how does it work?
1. Does it lower the output simply telling you that you are not clipping going out from fader 1&2?
or
2. Does it lower the input from all contributing channels so that the source coming to fader 1&2 no longer clips?

Diagram or vi for Sound Pressure Level Meter (SPL)

Need help for implementing a Sound Pressure Level Meter (SPL) with LabVIEW 7.1 or 6.3....I must use a microfone and a laptop for my project.....Need diagram or ready vi ....!!!
How can I make an A,B,C Weighting Filter vi for my Sound Pressure Level Meter project....???
If there is one already made or you can give me any help please send it to me.......[email protected]
Thanks.....

Search the archives. A similar question about the weighting filters was posted within the past month or two, if I recall correctly. I am not aware of any ready-built VIs, but the filter specs are published (Google A-weighting).
Also be careful with the frequency response of inexpensive microphones. They can skew the results substantially if you do not have some way to measure and compenste for the response.
Lynn

Audio Level Meter

Does anyone know where I can find an audio level meter that is either written in Swing or can be integrated with my Swing app?
I have not found anything in my web searches that really makes sense.

I've just started looking for the exact same thing. I'm glad
you posted first, I think you've explained the need better than I
would have been able to. I was thinking I may have to use an
embeded SWF, but the only thing I've found was in AS3.
http://livedocs.adobe.com/flex/3/html/help.html?content=Working_with_Sound_14.html
I too will be checking out the asFFT Xtra.

Is there a simple sound level meter for iTunes for windows ?

I'm trying to check the sound level of my music in iTunes and was wondering if there is a simple sound level meter plug-in for iTunes out there.

HKO,
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Question about Level Meter Plug in

I know how to change the channel meter between pre/post fader metering is there a way to do this with Logic's Level Meter plug in and with Inspector?

Plugins are always pre-fader, as the audio path hits the plugins and then goes into the fader.
What you'd have to do is route your audio through a fader (eg aux channel) to control the level, and then leave the fader on your output set to 0dB and meter in your plugins on that.

Input level meter in Audition CS5.5

I can't find an input level meter in audition CS5.5. I have to go my windows 7 control panel find me device and set the levels. So, to set levels I have to have two windows open, one control panel audio adjustment window and one soundbooth window where I actually record and watch meter, then I erase and record for real.
I must be overlooking something -- every other audio program I have ever used lets you set input levels while watching the meter. I've been through the help, and Googled it and I can't figure it out. In Audition, how do I set input levels from within Audition CS5.5 (not in windows 7 control panel)?
Thanks!

Short answer, no!
Audition merely records EXACTLY what the input device is sending to it. In normal useage the "level" of the signal to be recorded would be set externally, e.g. by use of an audio mixer or a "gain" control on the input device. I assume you are recording by using a line or mic plugged straight to your computer? In which case, the Windows Audio Control Panel is the only way to set levels.
Jeff

After downloading and installing additional content, no sound will play. Logic has no connection to sound library. No level meter activity. Right before i installed additional content I was working on a project, and everything was OK.

After downloading and installing additional content, no sound will play. Logic has no connection to sound library. No level meter activity.
Right before i installed the additional content I was working on a project, and everything was OK. What is wrong. How do I make Logic reconnect to the instrument files?

Whitelab Records wrote:
So, 20 views and nobody's anybody the wiser?
A small update to this issue....
I have been trawling the forums to find an answer but to no avail...
And... I've tried exporting and bouncing individual stems and the full session, either by muting all bar the stem or soloing the stem.....and still nothing.
Same problems.....everything is audible in audition but nothing plays on playback.
Since the 'prelisten' channel strip is Solo safe this would indicate that you have a track or channel strip soloed somewhere else, out of sight. Are the Mute buttons on your channel strips blinking?
To view all objects in your mixer, click the button. You may have to scroll sideways to check the whole mixer.

Level meter stops randomly during input

using 64bit Logic 9- latest version. While using a channel with audio input with "I" selected in channel strip, the level meter will randomly go flat but sound is still passing. usually toggling the "R" box will get it working again, but repeatedly it will drop out- anyone experience this?? if so, any conclusions?
thanks!

I suspect a problem with your audio interface. The next time it happens, you could test that theory by trying this. Open Preferences > Audio and uncheck "Enabled" (under "Core Audio"). Press the button Apply Changes. Then check Enabled, and Apply Changes again. Does that fix it? Maybe your audio drivers need an update, or there is some other issue related to your audio interface.

Peak Level Meter in

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