Audio stereo level adjustments

Hi...I'm working on an edit with just two audio tracks, L + R, audio and video loaded straight from the camera via Log and Transfer.
I have needed to adjust audio levels up and down to what's happening on the video and the only way I can do this is by adjusting the L & R tracks individually, I can't get both of the tracks up in the Viewer window simutaneously, like a stereo pair.
Video and audio tracks are linked.
There's obviously a setting somewhere I've missed because I do recall being able to adjust both tracks together at one time....
Can anyone tell me how to set this up please...
FCP Studio 7.0.3.
Thanks
PM

select the tracks, go to the modify menu and make sure the tracks are both linked and stereo pair.
Also, you can always copy and paste attributes if necessary. Copy the track with the correct audio level, select the clip you want to adjust and option-v paste attributes and make sure the correct items are checked.

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Why does compressor settings of aac audio, stereo, create 2 channel mono from 2 discrete audio channel movies?

Why does compressor settings of aac audio, stereo, create 2 channel mono from 2 discrete audio channel movies?

well depending on how your audio is brought into your system you can select it as two streams of mono or a stereo pair. In reality, unless it's mixed in stereo, that's just two streams of mono locked together. Other streams brought in via other means, can have a lot more audio channels, e.g. 8, 16, 24.
In your case, with what you're doing, two separate channels works best, so you can control the levels of two very different audio sources. If, for instance, you were taking the feed from the mixing board during a concert, you might want their stereo out. Otherwise you might be dealing with the 1.2 zillion channels that they're mixing live.
I like separating the tracks coming in. I work with people who prefer to lock them together. Locking means the tracks move together during editing. Ok, fine, but I'm usually dealing with different audio sources, like you are: a camera mic (safety on location) and a close mic (shotgun, lav). Or I have a shotgun source and a lav. I want to determine the best source audio for the shot. Sometimes it's a lav, sometimes a different feed from somewhere else, sometimes the shotgun, and sometimes a mix of two sources. But if they're locked together and pre-set to come out evenly from both speakers there no way for me to make a determination. The other guys like to cut and figure out the audio later.
To each his or her own. . .

STB QIP2500-3 audio output level

There is a noticeable difference between the audio output level (loudness) of the RF (ch3/4) and the baseband (L/R stereo) on the STB QIP2500-3. The RF is louder than the baseband. Is there any way to increase the output level of the baseband L/R stereo level, thanks.

Go to the Main Menu by pressing 'Menu' key on remote, Settings, Audio, Make sure the 'Set Top Box Volume' is at its max. Audio format can be put on 'Stereo'. Volume control can be put on 'Fixed'. Finally set the Dynamic Range to 'Heavy'. The following settings usually give the best audio output.
Javier
Verizon Telecom
Fiber Solution Center
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In adobe audition cc how do i edit my audio stereo in waveform

in adobe audition cc how do i edit my audio stereo in waveform i dont know where i tried everywhere please help

To expand on what ryclark says, you have to set the sample rate on your sound card or audio interface. If you're just using the onboard sound chip, that'll be on the Sound menu on the control panel. For that, go to the Playback Device menu, select your device, then properties, then Advanced. There'll be a drop down menu listing the available sample rates you can choose. You then have to do the same on the Recording Devices tab, at this has to match what you set on playback. Finally, you have to make sure that Audition (Edt/Preferences/Audio Hardware) is set to the same as your sound card.
If you have a specialist sound card, the setting will probably be in software you installed when you hooked that up.
Note that you're unlikely to find ANY sound card or audio interface that can use 20,000Hz as a sample rate though.
My normal advice if you need a low sample rate for some reason is to do your work at one of the standards (say 44,100 or 48,000) then convert the finished product. I know that some things like answering machines want lower sample rates. However, even this method won't give you 20,000. The presets in Audition jump from 16,000 to 22,050 which are standards in the industry.
Perhaps if you can explain why you want to use 20,000 Hz sampling?

Set audio input level manually on t5i

First post here. I've just started workig with my new t5i, which I'm using to work on my ongoing video tutorial series. In the manual I see how it is possible to set the audio input level maually for an external mic but I can't figure out how to navigate to the menu that allows me to do this. I have pored over the user manual a number of times and can't find this information. Can anyone help?
Solved!
Go to Solution.

Same camera in different shooting modes. The menus I described would be for shooting video in "P, Tv, Av, or M". Anything else will give you a smaller menu selection like you have already seen.

Quicktime Broadcaster AUDIO Input Level Setting

Greetings all,
I'm trying to record my band's rehearsal w/ QtB, but even when the Audio Gain level is on 0, but I still get distortion when playing back the recording.
Are there any ways to fix this?

Go to your system preference. Sound Prefs...Make sure your not taxing you sound. There is a control for input setting...

Group Level adjustments not required for consolidation report

HI Experts,
The requirement is as below:
*Requirement 1:*
My client want to post some adjustment entries at Group level after consolidation, but he need a report With and without the consolidation entries.
I.E. one balance sheet and PNL with the group level adjustments and another balance sheet and PNL without Group level adjustments.
Requirement: 2
Every period his manual entries are more than 240 and during year end it will go till 300.
As it is difficult to post these many entries manually, Client is asking for any upload functionality. Need some inputs on how to acheive this BCS.
Thanks in Advance,
Richard.

I agree with Eugene, you can't just create a user-defined posting level. The available posting levels are pre-defined by SAP in the UC00 type pool.
* posting level
begin of uc00_cs_plevel,
* cse posting level:
*   input_direct   type uc_plevel value is initial,
    input          type uc_plevel value '00', "data input
    input_corrctn type uc_plevel value '01', "correction of input
    input_pcc      type uc_plevel value '02', "PCC of input
*   input_itchange type uc_plevel value '08', "item change of input
    value_adj      type uc_plevel value '10', "value adjustment
    value_adj_pcc type uc_plevel value '12', "PCC of value adj
    pair           type uc_plevel value '20', "elim.of pairs
    pair_pcc       type uc_plevel value '22', "PCC of elim.of pairs
*   pair_mst       type uc_plevel value '23', "elim.of pairs mst.
*   pair_mst_pcc   type uc_plevel value '24', "PCC of elim/pairs mst.
    group          type uc_plevel value '30', "elim.within group
end of uc00_cs_plevel,
The virtual cube logic wouldn't know how to address a custom posting level without a number of core changes.

LIVE audio stereo stream

Hi there,
I'm looking for a way to offer a LIVE audio stereo stream to
the web client using Shockwave plug-in.
I was "banging my head off forums and search engines" (as
flashcomguru nice says) for some weeks
to find a way to stream LIVE audio stereo to Flash Media
Server but I couldn't find one viable direct
solution for this platform.
The client Flash Media Server app uses Microphone.get method
to capture sound from soundboard
which unfortunately is mono. it uses a speech Nellymoser
codec, at which music sounds pretty OK at 44Khz.
Possible solution: I can capture the LIVE line-in stereo
sound using two soundcards and thus beeing able to
create two sepparate mono audio streams one for left, one for
right using the only available Microphone.get method.
It will then maybe need some synchronization server scripts
later.
The trick is: In Director I use two .swf movies (action
script controlled) to connect to server
and play the streams but I couldn't find a way to direct the
streams to a specific channel
(one of the 8 available) from which point I can use the LINGO
Sound Channel PAN property, like this:
sound(1).pan = 100 for right
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From all that I read I came to the conclusion that you can
use the Sound Channel PAN property only to
.swa imported and prerecorded audio cast members, not to the
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in the main sound output channel to which I can't apply the
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The key is to route those two streams left and right in
Director.
Any hints?
Thanks for reading,
My deepest respect,
hOver

The microphone code is very similar to what you have posted. I can successfully use the enhanced microphone. When it is enabled, the issue I am having is exhibited.
A simple test I am using:
Use ffmpeg to stream a stereo mp3 file to the media server. I am using the following ffmpeg command line:
ffmpeg -re -i ~/alone.mp3 -vn -acodec copy -ac 2 -f flv rtmp://myserver.com:1935/video/3Daudio/alone
In this case the file is encoded at 44.1 kHz.
The client uses a netstream to play with bufferTime = 0
Without the microphone, the playback is as expected. With a normal microphone, not the enhanced microphone, the playback is as expected but there is little to no echo cancellation.
When the enhanced microphone is enabled, again using similar code to your post, the mp3 playback becomes severely distorted and is unacceptable.
In my opinion, this is an issue with the AEC algorithms of the enhancedMicrophone and stereo playback of a 'live' stream. If I modify the client playback code to bufferTime > 0, the mp3 playback is normal but there is no echo cancellation.
Thanks,
Sam

Why do voice-overs split up my audio stereo pairs?

I'm using FCP 5.1 in OS 10.5.8.
Sometimes after I do voice-overs, I discover that all the stereo pairs on the audio tracks are split apart, and often widely separated vertically. To get them back together on adjacent tracks, I have to un-pair them in the Modify menu, push them back together on the audio tracks, and then re-pair them.
This unwanted splitting-up of audio stereo-pairs is a nuisance. How can I make them stay together?
Tom

Thanks Meg! From now on, I'll do as you suggest and set the target audio track for voiceovers to be a blank one beneath all audio tracks that have anything on them. The project I'm working on has scads of stacked-up sound effects, so I have to aim voiceovers clear down to track A17, which is the first blank one I have! But that way they won't split up any stereopairs on the tracks above them.
I still don't see why the voiceover clips have to split up the stereo pairs to the right and left of them, when they're going onto a blank space on an audio track, but I guess that's just a peculiarity of the program. You've shown me that to prevent such trouble, a voiceover clip needs to appear on a completely blank track of its own, clear of all other audio clips. Then it can be moved up to wherever you want it to go on other tracks.
I was stymied at first when I couldn't get the A1 and A2 patches to appear in the patch panel, so that I could set the target track for voiceovers. They just weren't there. I finally figured out that to get the patches to appear, I have to double-click on any video clip in the Browser. That makes them show up.
Thanks again,
Tom

Changing audio monitoring levels?

Hey all,
I am working with 4 tracks of audio, two of which are a translation track of the recorded interviews. I would like to drop the levels of JUST the recorded audio, so I can hear the translated track above the interview audio. Is there a way I can lower two of the tracks for all the clips? If I change the levels for one of the clips using the audio mixer tool, it resets when I open a new clip, so unfortunately this is not working. Any suggestions?
Thanks in advance.
Gabe

Click the audio tracks you want to lower and once highlighted go to the viewer and click Stereo (a1a2) and just lower the red db level down 6 or so db and see if that does it for you. You will need to do this each time there is an edit in the audio tracks you want to lower.
Good luck,
corpguy

Audio:stereo to dualmono

Hi, i'm evaluating PP 5.5 after 8 eight of use of Final Cut.
About 80% of my clips have with a dual mono audio, ch1 is interview, ch2 is ambient audio.
When i import clips is adobe premiere, i get one stereo audio, instead of a dual mono.
If i edit to timeline, the clip goes to a stereo track, and if i assign two mono track, PP edit to timeline video but not audio.
I want the audio to go to two separate tracks instead of a single stereo one.
ps: if i use modify -> audio channels all options are disabled: i can.'t convert to mono, i can only swap the right and left channels.
Thank You
Alessandro

Copy the audio track to a new audio track, which will be stereo as well. Then on the first audio track apply Fill Left and on the second audio track apply Fill Right then you will end up with two different tracks, one with the interview and one with the ambient sound. You can apply these effects on the track level in the audio mixer, so you do not need to apply it to each and every clip.

Adjusting Audio Volume Levels

Often, one has Audio that is too loud, or too soft. Sometimes, they have added a music Track, that needs to drop down to clearly hear the dialog from another Clip. While there are Audio-editing programs, like Audition, SoundBooth, and even the free Audacity, PrE, or PrPro can handle such adjustment just fine. These adjustments can be made for the entire Clip, or can be Keyframed over time.
First method, and not one that I recommend, is to adjust the Audio Gain of the Audio Clip. This is done by either Rt-clicking on a separate Audio Clip, or by first Alt-clicking, and then Rt-clicking on a Muxed Audio Clip (Multiplexed with both Video and Audio in one file). Then, one would choose Audio Gain, and adjust that. This method will also allow one to then adjust the Volume by a greater amount, and has the bonus of increasing the Waveform Dislay vertically.
The next method is to look at the Audio Clip (or the Audio portion of a Muxed Clip), and click-drag on the yellow-orange “rubberband” for Volume, and move it up, or down. This can be Keyframed, but the next method is much better and more precise, in my estimation.
My first choice is to adjust the fixed Effect>Volume. This is done from the Effects Control Panel, and it is very easy to Keyframe these adjustments over time. This is much more accurate, and actually easier, than click-dragging the rubberbands in the Clip itself. Accessing the Effects Control Panel is slightly different, from PrE to PrPro. In PrE, one would Select the Audio Clip, then go to the Effects Tab, and choose Edit Effects, locating the Volume Effects and twirling it open to view the slider. In PrPro, if one does not already have the Effects Control Panel in a Pane in the GUI, it would be accessed from Window>Effects Control Panel. With the Audio Clip Selected, one would locate the fixed Volume Effect, twirl that open, and make the same adjustments, including Keyframes, if necessary. These will be Clip Keyframes, and will follow that Clip, where ever you move it. This is a difference from the next method.
My other choice would be to use the Audio Mixer (Window>Audio Mixer, if not open), and to automate the Volume settings. This is accomplished by playing the Timeline and adjusting the sliders for Volume for each Track. In PrPro, one would first set the Send to Write, or Touch, to make those adjustments, then switch back to Read, when done. There are no adjustable Sends in PrE, so one is always in Read, until adjustments are made, and then it’s set to Write, but you will not see this change. After you are done, you can playback the Timeline, and watch the slider(s) automatically move. These will be Track Keyframes, and will remain on the Track, regardless of what Clips you add/remove from that area of the Track. Unlike the Clip Keyframes, these will NOT follow the Clip.
One can toggle the Keyframe of an Audio Track between Clip, or Track. Also, the default frequency of Audio Track Keyframes in PrE & PrPro is a bit high, in my opinion. I drop these down a bit in Edit>Preferences.
For the Keyframing, one can alter the Velocity of the Keyframes to Bezier from Linear, and the adjustments will be a bit smoother. Besides the Bezier, one can use Ease-Out and Ease-In, which are Bezier, but basically in a Preset. In PrPro, one has several different Bezier options. In PrE, one is a bit more limited.
For setting the Volume, I recommend that one only glance at the Levels Meters (unless they are peaking in the red), and instead use a pair of high-quality headphones (noise canceling models are worth the money), or fully calibrated studio monitors. Let your ears make the decisions for you, as the dB settings should only be a starting point.
In PrPro, one can also Rt-click on the Audio Clip, and choose Edit in Audition (if you have it installed), or Edit n Soundbooth (if you have one of the suites).
Here is a look at Audio Keyframes in PrPro 2.0. Things are similar in PrPro later versions, and also in PrE, with the above noted differences:

Here are some AdobeTV tutorial links for Premiere Elements Audio:
http://tv.adobe.com/watch/learn-premiere-elements-11/adding-and-customizing-an-audio-effec t/
http://tv.adobe.com/watch/adobe-digital-school-collection/audio-mixing/
http://tv.adobe.com/watch/learn-premiere-elements-10/get-professionalquality-sound-through out-your-movie-/
http://tv.adobe.com/watch/premiere-elements-9-how-tos/enjoy-a-perfect-balance-of-audio/
Hunt

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);

Split Audio Broadcast Levels

Hey, guys. I'm cutting my first project for broadcast, and I have a question about split audio levels. My sequence is set to the usual -6dB ceiling, but when I convert from stereo to dual mono and split the music and dialogue tracks, the levels suddenly jump up 6 dB. Why is that? I've had to lower everything to -12dB in order to compensate, and I want to make sure that I'm doing it right.
Also, do I need to pan the bars and tone, or should I leave them in stereo?
Thanks, and happy holidays.

Well, it is true that every broadcaster has different specs for audio. Once had to do a show for Starz that also aired on MSNBC and local NBC... the specs for all were different. Video and audio...all different. Made it kinda expensive to deal with...almost restrictively so. We had to tell them that for this to be economical we needed a consensus... and they gave us some leeway.
Thank goodness.
Shane

Dialogue audio - stereo or mono?

I shot our short using a shotgun mic. When capturing the audio it brings in 2 audio tracks. Do i want to leave those as 2 mono tracks or 1 stereo track?
Thanks

Well - i'm still confused. I had one boom mic. Channel 1 was turned pretty high - and channel 2 was the same mic at a lower level in case of clipping - but there was none.
When i bring the track into the timeline, track 1 audio is the hotter one, track 2 is the lower one. I assume i don't want both.
Do i want to delete track 2, copy down track 1 and make them a stereo pair?
Or should i simply delete track 2, and make sure track 1 is panned to the center?
thanks!

Audio stereo level adjustments

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