Normalize level of 48 KHz audio

I have a video project with 48 KHz audio from the camera, and also a separate audio recording at 48 KHz, with too-low levels. I want to try normalizing its levels to see if it sounds better than the camera audio. It may avoid some echo, because of better microphone placement.
Garageband will normalize levels (in fact does it on output by default) but renders the waveform to 41 KHz on input. I'll use that version if I have to, but I would prefer to stay at 48 KHz.
Surely there is a way to normalize the level in Soundtrack 1.5 (which came with Final Cut Express 3.5). Or a free or cheap application that will do this simple job? I'd rather do it all in Soundtrack because that gives me easier access to other tools to clean up the audio.

Recently dowloaded a copy of the free audio editor Audacity. It's sometimes unstable but very useful, and it does let me normalize audio levels without changing the sampling rate.

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No 24-bit 96 kHz audio in media pla

I have a vista premium HTPC and a Audigy 2 NX USB sound card connected to a NAD analogue stereo used mainly for TV and music listening with a 2. focal system. Vista is fully updated. I checked 24 bit 96-kHz audio in Vista when i installed it in march -06 and had no audio problems. Some time ago i experienced audio cutting of suddenly while watching or recording TV. I checked the audio settings and found that all ripped audio (wma/mp3) in media player was dead silent without error messages. CD audio works but no ripped material. 24-bit 48 kHz worked, but not 96kHz. Other media players like winamp or foobar2000 plays in 96 kHz, but not media player/media center, is the codec broken or what?. How is the bit depth and frequency you set in creative media console related to the settings in vista? Does it matter what signal the nx is getting in the USB cable? What is the optimal setting in vista and creative audio console?

Doen't matter which faders you use. The final output is all that matters.
For digital broadcasting we use -20dB as the standard tone level. Our max peak (max) runs around -6 dB. Most of the music is much lower because it's RMS value is so much higher that the music is a lot fuller in volume. It goes over poorly when they flip on the video and get blasted out of their chairs.

Help! Still desperate because of this 32 - 48 kHz audio issue...

Hello fellow FCP-users,
I posted a few days ago asking what it means to have -along with the DVcamvideo- both 32 kHz and 48 kHz audio footage for one project.
After getting some very useful hints and suggestions I thought it would be no big problem, but even after carefully studying old postings about the same issue I'm still puzzled and indecisive. It seems the more I read, teh more confusing it gets.
But let me start with the first BIG problem: the capturing. Reading all replies so far, I understand that this should be done with a 32 kHz capturing preset. The only thing that I should do then is duplicate an existing setting (f.i. DvCam 48 kHz) to 32 kHz. Ok.
But I haven't found HOW I should do this simple duplicating action. So far I only found a DV 48 kHz-preset, by going to audio/video-settings, choosing the capture preset tab and then duplicate the setting, but at that point I can't find any option to change the audio capture to 32 kHz whatsoever. So there is where it basically ends, not having found any other option anywhere else to duplicate a capture setting.
Then, suppose the capture 32-question above will be answered, what will then be the best workaround?
I already tested this possibility: that is to log all my clips first, with only the video -no audio!- to be batch-captured. Then, as soon as I had all my logged files together, I duplicated them and gave new names to the these duplicates. Then I selected the first set of duplicates and batch-captured resulting in clips with onbly video. But when I tried to batch capture the newly named duplicates -only asking for the audio to be captured- it was again only the video that I got.
The idea behind this method was to separate video and audio, giving me the possibility to export the captured 32 kHz audio, thus getting the right 48kHz audio and then to delete all the 32 kHz audio in order to keep my project clean.
Would this separate capturing-method just be impossible or did I do something wrong here?
Or -if this was not a clever idea after all- should I just capture both video and 32 kHz audio, then create new 48 kHz audio and leave the mediafiles -both video and 32 khz audio- as I first captured them intact?
I hope one of you out there might give an answer to these questions...Especially the first one, because so far I can't even start with my project.
Thank you all for your attention. All help is more than appreciated.
Best regards,
Robbie V., Amsterdam, the Netherlands

...going to audio/video-settings, choosing the capture preset tab and then duplicate the setting, but at that point I can't find any option to change the audio capture to 32 kHz whatsoever.
After you've duplicated the preset select the duplicated preset and press the Edit button which will bring up a screen that looks like this:
When I work on a 32 project all the audio is 32 so there's really no need to transcode although your project demands may differ in which case you could export the audio by selecting File/Export/Audio to AIFF(s) and reimport if necessary. Capturing seperate audio and video is not necessary for this workflow.

Adjusting audio levels on an entire audio track

it may sound like a rookie question, but how do i change the level on an entire audio track, start to end. i am not interested in a "nest items" command, but in a situation that lowers every audio clip on the track by x and allows me still to see all the clips in the sequence and do further adjusting on a single clip if needed.
any ideas?
many thanks,
KN

To elaborate on what Max advised (also available through Modify->Levels), you have two choices for a gain level changes: "absolute" and "relative."
A "relative" adjustment will change the levels relative to where they are now. For example, the current gain level is at 6db then keyframed to 9db over the course of a clip. Making a relative adjustment of -4db will result in the gain levels being set to 2db at the first keyframe and 5db at the second keyframe.
An "absolute" adjustment will set the gain level to whatever value you enter, regardless of what its setting is prior to the change. Using the above example, making an absolute change of -4db to the same clip would result in the entire clip set to -4db.
-DH

Audio levels on the timeline & Audio Mixer

I'm confused why when you change the audio levels on a clip using the Audio Mixer it changes the levels for everything on the track?
I'm from an FCP & Avid background where you can adjust each clip using the Audio Mixer, is there a way I can do this? I find using the yellow sliders in each clip is very unaccurate and having to open each audio clip in the viewer is such a time waster.
Thanks, Andy

Audio in Premiere Pro is different--but quite a lot, in many circumstances. I'd recommend going through the help file's audio section to get a better idea of what some of the main controls are: Adobe Premiere Pro CS5 & CS5.5 * Editing Audio
But to answer your question more directly: the Audio Mixer controls track volume, not clip volume. By default, the Audio Mixer isn't set to drop keyframes, so it will affect the level of a track as a whole; check out Recording audio mixes to get more background on setting up the Audio Mixer for automation, so that you can record level changes.
The rubberbands (the yellow lines you're referring to) can be more finitely controlled by holding down the Ctrl/Cmd key as you drag; the values will move in smaller increments, then.
You don't have to load a clip into the Source Monitor to adjust the levels, if you want to adjust by the numbers. Just single click a clip in the sequence, and if the Effects Controls panel is forward, you'll be able to adjust the volume with the standard effects sliders/dials. Note that keyframing is enabled by default for the Level property, so if all you want to do is adjust the total level of a clip, it might be faster to do it in the sequence.
Audio mixing in Premiere Pro is pretty capable and the feature set is pretty deep, but I'll grant you it's a little overwhelming coming from a different NLE. I don't know if any of the previous helps you, but post back with any specific questions and we'll try to figure out a way to approach this in a fashion that makes sense to you.

Is it possible to trim a video layer according to sound levels found in its audio spectrum?

Hi There,
I'm interested to know if there is any way via a script or an existing plugin that would allow me to trim down a video layer based on the sound levels in it audio track? For example, if I wanted to set the in point for a source footage element to where sound started. Is that possible with scripting? If not, is there a plugin that analyzes the audio spectrum so that a script could possibly determine that point?
Thanks for your time and help!

This example uses the "Convert Audio to Keyframes" menu command to find the first frame where the selected layer's audio level is above the value specified in variable "threshold" (0.1 in this example). It then sets the selected layer's in point to that frame.
    var threshold = .1;
    function main(){
        var myComp = app.project.activeItem;
        if (myComp == null || ! (myComp instanceof CompItem)){
            alert("No comp active.");
            return;
        if (myComp.selectedLayers.length == 0){
            alert("No layer selected.");
            return;
        if (myComp.selectedLayers.length > 1){
            alert("More that one layer selected.");
            return;
        var myLayer = myComp.selectedLayers[0];
        if (! myLayer.hasAudio){
            alert("Selected layer doesn't have audio.");
            return;
        app.executeCommand(app.findMenuCommandId("Convert Audio to Keyframes"));
        var myAudioLayer = myComp.layer(1);
        var myProp = myAudioLayer.effect("Both Channels")("Slider")
        audioFound = false;
        for (var i = 1; i <= myProp.numKeys; i++){
            if (myProp.keyValue(i) > threshold){
                audioFound = true;
                myLayer.inPoint = myProp.keyTime(i);
                break;
        myAudioLayer.remove();
        if (! audioFound){
            alert("No audio above threshold found.");
            return;
    main();
Dan

Can I set the normalization level in the Bounce dialog?

I want to normalize my tracks to, say, -1db instead of 0db. Is there any way to set the normalize feature in the bounce dialog to do this?

Hi
The Art Of Sound wrote:
Which in turn reduces the dynamic range......
Louder parts get quieter... and quieter parts get louder....
No
Turning the gain up or down to ensure that the peak levels ( or RMS levels) merely turns the whole signal level up or down: no compression or expansion
http://en.wikipedia.org/wiki/Audio_normalization
CCT

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);

Level change when adding audio keyframes

Hi all,
I've been in Avid Land for the last few months and just came back and noticed something that was getting on my nerves when I left (I don't recall seeing this before FCP 6). When I add keyframes to my audio levels with the pen tool, I get a keyframe, but at the same time, it lowers my level as if I had added a keyframe and then dragged it down. As far as I can tell it does this every time.
Does anyone know what this is about?
Thanks,
Matt

I was having this problem, so just before I composed this reply I did some tests. It could well be 'sloppy mouse work' but I don't think so. This time the audio level went up by 1 db but before it was dropping by 1 db when adding a keyframe. During the test, no matter how accurate I was the audio level changed when adding a keyframe. The track I was working on was enlarge somewhat and the problem didn't happen on standard sized tracks. I feel that this has to be some sort of bug.
I have always found FCP a little bit on the 'accurate' side. You have to be very precise when clicking on items otherwise you will miss. Some buttons are way too small and I think Apple could improve the FCP workflow by enlarging some buttons and allowing more space around clickable items.
A good example of this is the 3 way colour corrector. The left and right buttons for blacks/mids/whites are very hard to hit and I quite often hit the maximum by mistake.
I have worked on many non-linear systems and FCP is the only one where mouse accuracy is so important.
I am, to some extent used to this now and I suppose it is a minor inconvenience in comparison to the benefits of using FCP but, in the interests of improvement and ease of use it would be good if Apple could take this onboard.
Weenie.

Level Meters on "Record Audio" have disappeared

I'm using 10.0.8 with OS 10.8.3 and yesterday connected a Blue Microphone Yeti as the Sound Input Device. When clicking Window>Record Audio, the window to select the Destination, Input Device, Gain and so on appears but the level meters at the top of the window are unresponsive. That is, the sound level scale is present but no green bars show input level. I can still record a voice-over fine, the clips appear in the timeline and the sound meters on playback display properly but yesterday the "Record Audio" meters worked and today they're gone. Any ideas on what I could have done to make them disappear?

So it doesn't sound like it's a hardware or drivers problem.
I updated to 10.0.8 and I cannot find a way to reproduce this. (But I'm not running 10.8.)
Try the trashing your preferences with Preference Manager.
Good luck.
Russ

Bug with 22 KHz audio

When I load an MP3 using new Sound(new URLRequest("path"));,
and then subsequently play it with an argument passed to the play
method indicating how many miliseconds into the file Flash should
begin playing the audio... it always starts twice as far into the
file as it's supposed to if the file is encoded at 22 KHz instead
of 44 KHz.

Thank you Steve. I do have Liz Castro's books, they are very good. Creating the audio content is not the problem, have done it many times. The problem is I believe a bug in iBooks that incorrectly places the audio controls when the section of the eBook set as linear="no" in the <spine>. This setting takes the section out of the linear reading flow of the book and opens it up in a new page, outside of the book. It's very useful for large tables, etc. Liz has examples of this on her site also. It works very well, except for the audio controls. While the function, it seems impossible to control their location on the page.

Does Apple Tv 3 support 24 bit/192 khz audio Data?

I Tryed to Google this but answers are not clear. Can anybody help?

I just bought a Pioneer N-50 in order to play 24bit audio files from my iMac by Airplay. I can't see if it does and can't find any documentation about it. I read somewhere (can't find it anymore) that the Pioneer plays 24bit files just fine from iTunes over Airplay. What's the secret?

Weird 44.1 kHz to 48 kHz audio sampling

Can anyone please advise...
After many years of editing with Premiere, I have not noticed until today a problem dropping a 44.1kHz mp3 track into a 48kHz timeline.
The music track plays correctly when previewed in the source monitor, but lowers in pitch when played from the timeline.
I resampled the file in Soundbooth by saving as a WAV at 48kHz, but, once back in the PP timeline, the pitch appears to be even lower than when played at a 44.1kHz file. Naturally, the piece plays correctly when dropped into a 44.1 sequence but I can't edit at 44.1 becase of all my other material is recorded at 48.
In all these years, I have never had to resample music to play in Premeire. All that's changed is that I have installed CS5 and (but removed) the K-Lite codec pack.
Can anyone help please. Is it something that has always been there, and I've never noticed. I really can't believe that to be the case!

Hi Bill
I saved them back both as 16 and 24 bit PCM/WAVs. I finally had to ask the client to resubmit them as 48kHz WAV files which, naturally, solved the issue.
I thought that Premiere made an attempt to resample audio files which differ from the targeted frequency of the project.
As I said in my original post, in my 12 or so years editing with Premiere, I have not been aware of encountering this before!
All very strange.

44.1 kHz audio recording

I read part of a link that mentioned new iPod will record audio at 44.1kHz. unfortunately the link was broken. The bit I read says at the bottom of the tech specs says this. can anyone confirm this or provide a link to a tech sheet page?
thanks

Click here to see the iPod technical specifications page.
(9246)

48 Khz Audio Rate with 12-bit Format

I'm capturing some old DV footage shot in 12bit format. However, when I capture, it is showing as 48KHz audio rate. As I understand it, 12-bit format should equal 32KHz rate.
I've got a Canon HV30
I've tried NTSC Firewire 32KHz in easy setup and also NTSC Firewire Basic 32KHz setup.
But the captured video Item Properties is still showing 48KHz.
Is this a problem?
Thanks,
Billy

Ian,
I think I figured out what happened. If you could verify this, it would be helpful.
As I went back through the video with the audio out of sync, I realized that it hit a spot in the tape where the timecode got out of sorts because I had a block that had no recorded info on it. In other words, I did not do as I should have and pre-recorded the tape to get the consistent timecode. The out of sync problem started happening immediately after that point where the "gap" occurred.
For some reason, iMovie seems to handle this problem better than FCE by essentially breaking the clip off at that point and starting a new clip when the new timecode comes into effect. FCE seems to "choke" on it a bit and then the audio goes out of sync thereafter.
That is my best guess as to what happened from what I'm seeing. Could what I describe cause the out of sync problem?
Thanks,
Billy

Normalize level of 48 KHz audio

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