Setting audio BG on 3&4

I am trying to set up BG audio on 3 or 3 &4 but can't seem to hear it. Can someone help.
Thanks
Jeff

Are the tracks "ON"? green button at the left side of the timeline next to the tracks in question.
Are you monitoring both left and right channels? If not, are the tracks in question panned to the middle or to the channel you are monitoring?
Have you tried increasing the track levels? (Select track, MODIFY > LEVELS)
rh

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setting audio record path?

Let me guess - your tempo's set to 120 and 4/4 time, right?
The limit is on the number of quarter notes Logic can handle (8550).
Set your song length to max (2158), reduce your tempo, and/or adjust your time signature. Example, temp 60, 4/8 time will give you 2hrs 22minutes of recording.
good luck

Setting audio level points?

I am trying to change the level of an audio in a sequence. I double clicked the audio clip to open the waveform in the Viewer. I know there is a way to select various spots on the audio waveform and mark them. Then you can drag the audio level line to a new setting to adjust that part of the waveform. But I don't know how to create the spots on the waveform. I can't find the answer in the Help manual.
Thanks for any info.

You need to select the Pen Tool which is to the right of the timeline.
You can also do it by pressing p
Then click the pink "level" line of the audio and a keyframe will be created. Click further along and you can drag the sound up or down etc.
Ian.

Set audio position microseconds/Player

Hay,
I've made this nifty little class for playing audio:
import java.io.*;
import javax.sound.sampled.*;
public class Player implements LineListener
     // Consts
     public static final float maxVol = 6.0206f;
     public static final float minVol = -80.0f;
     // Vars
     // The AudioInputStream to play
     private AudioInputStream toPlay = null;
     // If the playthread should pause
     private volatile boolean paused = false;
     // If the thingy is playing
     private volatile boolean playing = false;
     // The line
     private volatile SourceDataLine line = null;
     // The thread that will play the sound
     private Thread playThread = null;
     // Construtor
     // Body
     public void play(String toPlay)
          try {
               // If playing
               if (playing && paused && (!line.isRunning())) {
                    setPaused(false);
                    // The return so don't play again
                    return;
               play(new FileInputStream(toPlay));
          } catch (Exception ex) {ex.printStackTrace();}
     public void play(File toPlay) {
          try {
               // If playing
               if (playing && paused && (!line.isRunning())) {
                    setPaused(false);
                    // The return so don't play again
                    return;
               play(new FileInputStream(toPlay));
          } catch (Exception ex) {ex.printStackTrace();}
     public void play(InputStream toPlay) {
          try {
               // If playing
               if (playing && paused && (!line.isRunning())) {
                    setPaused(false);
                    // The return so don't play again
                    return;
               // Get AudioInputStream from given file.
               this.toPlay = AudioSystem.getAudioInputStream(toPlay);
          } catch (Exception ex) {ex.printStackTrace();}
          // Make the play thread
          playThread = new Thread(new PlayThread());
          // Start it
          playThread.start();
     public void stop()
          // Not playing
          playing = false;
          // If there is a line
          if (line != null)
               // Close the line
               line.close();
               line = null;
          // Distroy the play thread
          playThread = null;
     public void update(LineEvent e)
          // Stop
          if (e.getType()==LineEvent.Type.STOP)
               stop();
     // Getters
     public SourceDataLine getLine()
          if (line != null)
               return line;
          return null;
     public float getVolume()
          FloatControl con = (FloatControl) line
                    .getControl(FloatControl.Type.MASTER_GAIN);
          // Get value
          return con.getValue();
     public boolean getMute()
          BooleanControl con = (BooleanControl) line
                    .getControl(BooleanControl.Type.MUTE);
          // Get
          return con.getValue();
     // Setters
     public void setPaused(boolean inPaused)
          paused = inPaused;
          // If not null
          if (line != null)
               // If paused
               if (paused)
                    // Stop the line
                    line.stop();
               } // Resume
               else
                    // Start
                    line.start();
     // Setters
     public void setVolume(float vol)
          // TODO: Maybe test to see if within min an max
          FloatControl con = (FloatControl) line
                    .getControl(FloatControl.Type.MASTER_GAIN);
          // Set value
          con.setValue(vol);
     public void setMute(boolean mute)
          BooleanControl con = (BooleanControl) line
                    .getControl(BooleanControl.Type.MUTE);
          // Set
          con.setValue(mute);
     private class PlayThread implements Runnable
          public void run()
               try {
                    System.out.println("toPlay is null " + (toPlay == null));
                    // If not playing
                    if (!playing)
                         // Now playing
                         playing = true;
                         play();
                    System.out.println("Finished");
               } catch (Exception ex) {ex.printStackTrace();}
          private void play()
               try {
                    AudioInputStream din = null;
                    if (toPlay != null)
                         AudioFormat baseFormat = toPlay.getFormat();
                         AudioFormat decodedFormat = new AudioFormat(
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                                             .getSampleRate(), 16, baseFormat
                                             .getChannels(),
                                   baseFormat.getChannels() * 2, baseFormat
                                             .getSampleRate(), false);
                         // Get AudioInputStream that will be decoded by underlying
                         // VorbisSPI
                         din = AudioSystem
                                   .getAudioInputStream(decodedFormat, toPlay);
                         // Play now !
                         rawplay(decodedFormat, din);
                         toPlay.close();
               } catch (Exception e) {e.printStackTrace();}
          private void rawplay(AudioFormat targetFormat, AudioInputStream din)
                    throws IOException, LineUnavailableException
               byte[] data = new byte[4096];
               line = getLine(targetFormat);
               if (line != null)
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                    line.start();
                    int nBytesRead = 0, nBytesWritten = 0;
                    while (nBytesRead != -1)
                         // While still paused
                         while (paused) {}
                         nBytesRead = din.read(data, 0, data.length);
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                              nBytesWritten = line.write(data, 0, nBytesRead);
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                    line.drain();
                    line.stop();
                    line.close();
                    din.close();
          private SourceDataLine getLine(AudioFormat audioFormat)
                    throws LineUnavailableException
               SourceDataLine res = null;
               DataLine.Info info = new DataLine.Info(SourceDataLine.class,
                         audioFormat);
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               res.open(audioFormat);
               return res;
}I've got an app that uses JMF to play MP3's, but using JMF just to play MP3's is overkill, but I've found a SPI from javaZoom now, so that's okay.
Anyway, I want to set the position of the line in microseconds now, as well as get the duration.
Can anyone help with this?
By the way, if anyone can open .rar files (lots of complainants about that) then the, in much need of an update, app is here: http://acquiesce.awardspace.co.uk/Projects/Download/Music_Exploder.rar
I you do want to try it, you do need the JMF.
Why not pop into the forum as well:
http://acquiesce.informe.com/mucic-exploder-dc4.html
You know I need the comments.
Luke :D

okay, so u can do this in the Clip interface, but the Clip takes ages to open, at lest for mp3's.
I need it to open it right away
I understand a bit more now, the SourceDataLine plays each byte as it is writen which is why it plays it straight away, but the Clip loads it into memory, so therefore it knows how long, how many frames etc about the audio, but it means that it's not as fast as streaming.
I need the best of both worlds really, I've tried opening a Clip on another Thread as well as playing from a SourceDataLine, the idea was to have the Clip take over the playing once it has loaded, but the line doesn't play or read/write.
Is there someway to take advantage of both streaming an non-streaming in this way?
Or simply make the Clip.open_AudioImputStream) faster?
Thanks
Luke

How to set audio hardware input channels for narration & to what?

Searched forum - hope I didn't miss solution for the following:
Using PE3 in Windows XPSP3
Tried for the first time to add narration to a project. Was using a stereo/microphone headset. Got nothing but noise. Could record in Windows recorder and Audacity. However, the microphone as part of the set would not work in Windows Audio Hardware Test Wizard. The set was using a sound chip on the motherboard.
A friend of mine has the same OS, motherboard and is using PE3. He too had never tried adding narration - but tried with a USP connected set, and was successful.
I removed my stereo set, and connected a USB set. To make sure all settings were reset, I did a cold reboot. The USB set worked in Windows Audio Test Wizard. I opened my project again, and in Edit/Preferences/Audio Hardware, clicked AIIO Settings. Unchecked the old set (Realtek) in both the Output and Input Ports area, checked the USB Audio Device, moved to the top of both lists.
Then tried to add narration and got the error message "Your current audio hardware selection does not have any input channels. Please use Preferences>Audio Hardware to correct this problem.
Thinking that maybe this was a preset for the project and couldn't be changed, I started a new project, no video, just tried to add narration, and got the same message.
The entries in the columns in the ASIO Direct Sound for Full Duplex Setup menu for the USB Audio Device in the Input section are: 2048, 0, 1, 16, Yes
The only one of these that seems to be able to be adjusted is the second number - the Offset (Samples). The Audio Channels number is set to 1 and doesn't seem to be changeable. Suggestions?

When working with Audio in most NLE's, i.e. Premiere Elements, or Pro, there are basically three areas that can affect how the Audio is handled on one's system. First, there is the Sound settings in Windows (from Control Panel). Next, the Audio card/chip probably/possibly has a console to make settings at the card/chip level. Last, and these are really two locations, but I'm only counting them as one since they are both in the NLE - Edit>Preferences>Audio & Audio Hardware. This is where Steve is referring to. Check all settings, and I'd suggest doing so in the order that I listed them above: OS, Audio card, then Premiere. A Mute, or by-pass in the first two, can override the settings in Premiere, and you may have its setting correct, but still not be able to effectively use your hardware.
The exact settings chosen will depend on the exact hardware that is properly installed. Without knowing all of your hardware and other settings elsewhere, one cannot give you a definitive answer. Ideally, in the settings drop-downs, you *should* see your hardware listed. Look closely, as these can be a bit cryptic.
If you are still unable to get your hardware working properly, a fourth place to look would be Control Panel>System>Hardware>Device Manager for your Audio hardware. Check that Device Manager sees your hardware, and that it is working properly.
If all is set up properly, but still not working, report back with exact specs on your system, including all of your Audio hardware. There are possibly some more steps that can be taken. I will not mention them now, as they *should not* be needed. If you do need further assistance, please list every option in all of the drop-down lists from Premiere, so we can know what your choices are.
Good luck,
Hunt

Set audio on compressor using command line?

Anyone know how to supply the audio path using compressor on the command line?
Running compressor -help shows audio=/path/to/file but it's not too specific on how that looks in the actual command. I've tried every combination I can think of. Has anyone done this succussfully?
Thanks

When I try that format I don't get any response at the command line from compressor...it just exits. Here's my command:
compressor -clustername "Compressor" -batchname "test" -settingpath "/Volumes/theGrill/Resource_Library/Art_Anim/Presets/Compressor/Preview (H.264).setting" -asneededcopy -destinationpath "/Volumes/Scratch/temp/test/test.mov" -jobpath "/Volumes/Scratch/temp/test/Leadership_April_Update/" audio="/Volumes/Scratch/temp/test/Leadership_April_Update.wav"
The other formats I've tried include. Responses are what is returned in terminal.
-jobpath "/path/to/sourcevideo.mov",audio="/path/to/sourceaudio.aif"
     Response: Parameter error: Invalid Job path: /Volumes/Scratch/temp/test/Leadership_April_Update/,audio=/Volumes/Scratch/temp /test/Leadership_April_Update.wav
-jobpath "/path/to/sourcevideo.mov" -audio="/path/to/sourceaudio.aif"
     Response: None
-jobpath "/path/to/sourcevideo.mov" -audio "/path/to/sourceaudio.aif"
     Response: Invalid parameter: -audio
-jobpath "/path/to/sourcevideo.mov" -jobpath audio="/path/to/sourceaudio.aif"
     Response: Parameter error: Invalid Job path: audio=/Volumes/Scratch/temp/test/Leadership_April_Update.wav
-jobpath "file:///path/to/sourcevideo.mov?audio=file:///path/to/sourceaudio.aif"
     This submits, but the output doesn't have the audio.
Any other ideas?

Setting audio tone for exporting video (-12 or 0db?)

Hi,
I'm exporting a production as full uncompressed video to create a data DVD which will then be used to strike copies in different formats. I have 2 Q's.
1) I'm prepping bars & tone at the head, & I noticed that the default setting for FCP HD 4.5 is to have audio tones at -12db. I'm from the old school of audio where tone was always set to 0db. I'm guessing that the higher dynamic range of digital audio makes some arguement for the -12, but I want to be sure.
2) Also, its a music video. I'd heard the recommendation to have program audio levels peak at -6db to keep it from having unusually loud audio, but I'm wondering if that recommendation is for narratives & based towards dialog levels? Any issues with a music video having audio levels that peak at or close to 0db?
Thanks a lot.
Duane

Duane this is an example of where old school is completely trumped by new school.
In the analog world 0 dB was the point you wanted to be near at your loudest, and MAYBE peak over very quickly at the very loudest point of your track.
in our digital domain these days, 0dB is now the point of no return. Hit zero and your track is distortedwith no chance of saving it.
pretend that fcp's -12 equals 0 on an analog scale, and then imagine you've got up to -6 for headroom for the loudest things. NEVER go anywhere above -6 and you'll never have a problem.

Setting audio recording from mono to stereo

How do you set the audio recording from mono to sterio on windows vista?

Select them all the timeline and press Opt-L.

Setting audio levels

I must adjust the audio of several videos partly to make the clips in each video sound at the same sort of level, and to avoid distortion or startling viewers on loud bits. The video will be distributed using DVD Studio Pro 3. I have read:
http://discussions.apple.com/thread.jspa?messageID=7025739&#7025739
which recommends -12 db for the average level. In a typical example of the videos, if I reduce my levels to that average (-12 db) the repeated peaks (cattle mooing and banging and clanging into metal barriers at a cattle market) go to 0 db and the quiet parts go to -18 db.
Does that seem correct for a viewer watching the DVD on a TV set without his/her having to set the TV's volume control up or down to get a volume level normal for TV viewing?
Thanks.

It's not that big a difference. A live orchestra has a greater dynamic range. -12db is a good reference for the average level of your audio. If you're really really worried about this and you have FCE4 you could try applying the Normalize filter to your audio.

Setting audio volume

Hi,
I can't figure out how to set the volume level of the audio I have dragged into the opening screens. The volume is always way too loud when I put the DVD in. I want to be able to match the volume of the content to what is in the main pages and chapter screens.
Thanks,
Lori

You will need to change volumes in Soundtrack or other editor, DVD SP does not handle those changes...also if you encoded differently there can be volume shifts but it sounds like more than that...

Setting audio output levels automatically at login (even with command-line)

When I log in, the audio level seems to be at the level set by the last user. Is there a way to get this to automatically set to my preferred level when I log in?
A related question: Is there a command-line method of setting the audio level?
Thanks in advance,
Paul
Power Mac G5 Quad Mac OS X (10.4.8) 2GB ECC Memory

u could use scripting...
sudo osascript -e "set Volume 20"
using that at the command line will change the volume level to 20.
you could put that in a script or something and have it launch at login.
Beavis2084

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.

Set audio in/output with shell command

Hi.
I want to switch the systems audio input and output with a shell command. I've found a command that can set the volume. Maybe someone can tell me if this is possible and if so, which command i need.
osascript -e "set Volume 0"

I think there's a messy way of doing it in AppleScript by addressing the menus in the System Preferences. But if I were you I would have a look at the tool that someone has created here:
http://code.google.com/p/switchaudio-osx/
It addresses CoreAudio directly I think to change the input/output method:
will@Newton:~/Downloads/SwitchAudioSource-v1> ./SwitchAudioSource
Please specify audio device.
Usage: ./SwitchAudioSource [-a] [-c] [-t type] -s device_name
-a : shows all devices
-c : shows current device
-t type : device type (input/output/system). Defaults to output.
-s device_name : sets the audio device to the given device by name
will@Newton:~/Downloads/SwitchAudioSource-v1> ./SwitchAudioSource -a
Built-in Microphone (input)
Built-in Input (input)
Built-in Output (output)
will@Newton:~/Downloads/SwitchAudioSource-v1> ./SwitchAudioSource -c
Built-in Output
will@Newton:~/Downloads/SwitchAudioSource-v1> ./SwitchAudioSource -t input -s 'Built-in Input'
input audio device set to "Built-in Input"

Set audio channel controlled by multimedia buttons

In KDE, is there a way I can set the audio channel that is controlled by the multimedia buttons? I am having a problem where the raise and lower volume buttons are not controlling the correct (PCM) channel.
I have a Dell Inspiron 9300 with the Intel AC'97 audio controller. The computer has normal speakers, controlled by the Master channel and a subwoofer controlled by the Master Mono channel. Because of this, in kmix, I leave the two above channels set and use the PCM control to control overall volume, as adjusting the "Master" channel does not affect subwoofer volume. Now, when I use kmix to adjust volume, things are fine, however when I use the multimedia raise/lower volume buttons, only the Master channel gets adjusted. I assume there must be some place where the channel controlled by these buttons can be set, but I cannot find it. I believe I've read that kmilo handles this task, but I cannot find where I might configure that.
I would be willing to do some hacking of the appropriate code, if it is required. Someone would just have to point me to what library/executable contains the code that handles the mentioned multimedia buttons.
Thanks and regards,
jbro

Thanks for the suggestion, but that is actually one thing I tried with no luck. I could create a "Soft Master" channel and use it to control overall volume, the same as just selecting PCM as the master channel, but the multimedia keys still directly changed the "Master" channel. Note that I have no problem when changing the volume through kmix. It properly adjusts the PCM channel, which I have set as the master channel. Only when I press the multimedia keys do I have this problem, so it looks like kmilo isn't paying attention to the master channel and automatically adjusting the first channel. Unfortunately, according to the ALSA wiki, there is no way to override an existing channel.
OK, I took a look at the latest code (KDE 4.0) for the delli8k plugin for kmilo and it adjusts "masterVolume" via dcop. When I do the corresponding commands from a prompt everything works fine. This leads me to believe that KDE 3.5.7 doesn't have the code I'm looking at. This is odd, though, as the last change in the Dell kmilo plugin was in September of last year. Perhaps it has not been backported to KDE 3.5.7. When I get some time I will try to compile the new kmilo and plugins to see if that changes anything.
Thanks again for the suggestion.

Cannot set audio level to zero

Hi,
I know how to edit the audio levels and have no trouble setting it, from 0% to 150%.
With the video clips this works fine. I can go from complete silence to full volume.
When I try this with an audio only track, I can set the fader from 0% to 150%. The audio gets louder as I increase the percentage. When I set the percentage to zero however, I can still hear the audio. It's softer, but still there.
I've tried using the "fade" feature, dragging the line from starting level down to zero. The audio fades, but not to complete silence. I'd say the level is equivalent to the 10-15% level I get when fading video.
I have iMovie version 5.0.2.
Any and all suggestions appreciated!
iMac G5 2.1Ghz 20" Flat Panel Mac OS X (10.4.3) BondiMac G3, 233MHz, OS 9.2.2, iMac G4 Flat Panel, 800MHz, OS 10.2.8

Hi,
The audio clip contains only music. The video clip contains only spoken word. It was a good idea though!
Thanks for posting!
ap

Setting audio BG on 3&4

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