I'm trying to accomplish a very simple tone:

The tone should:

-Start at a base frequency of 325hz

-Lower to 250hz

-Raise to 400hz

-Return to 325hz

-Accomplish this 1 time over a duration of 72 seconds

Can someone instruct me on accomplishing this in Audition 3.0.1?

Thanks!!!!

Hi,

3 years ago or something I had Windows Vista installed on my notebook and used Revalver to record my guitar tracks in audition. I haven't recorded anything now for 2 years and now I have Windows 7 ultimate on my notebook. But now I only get the clean signal and not the distorted signal from Revalver from my guitar when I try to use Revalver as vst and I don't know why or what I'm doing wrong. But when I click on play after recording, there's the normal distorted signal. And this is what I need while recording because I can't record metal songs when I just hear the clean signal from guitar...

If I remember right, all I did in the past was open audition, then I clicked on options>monitoring>extern.

Then I clicked on the record symbol in multi track session, clicked on the fx symbol> fx on>vst>Revalver

I think that was all. Do you know if I'm doing anything wrong?

I've been updating my studio, and have a new computer with more juice and a good 24-bit USB interface.  I want to record at 24-bit, but Audition has only 16 and 32-bit float.  I've read the manual and understand what that means.  I've also looked through the database, and find the past discussions on this somewhat confusing. 

What I really want to know is this: Will recording using the 32-bit float file setting work with a 24-bit interface?  Will anything be lost, distorted, etc.?  If not, where do the extra 8 bits of audio data come from when used?

Thanks for any thoughts.

Hey everybody,

I've got a huge problem with Audition 3.0: I can't record anything. In the Multitrack Session, the 'R'-Button is always grey so I can't prepare for the record (or whatever it is called in english). Also, when I check the Audio-Hardware-Settings (hope you know what I mean), there is no available input.

The problem has to be the program, because I can record with Audacity.

I'm using my onboard soundcard and a Shure livemic, so nothing special. System ist Win7.

Thanks, hope somebody can help me!

P.S.: Google is giving me many threads where people have the same problem, but no solutions.

hi

when I record in multi track - it records with pops and clicks.

I understand that in order to solve this problem I have to increase the buffer size.

where is that set?  under "Options" -> "Settings" -> "Multi -Track" -> "Play/Record" -> "Recording Buffer Size" ?

I was able to record when I had my Realtek driver selected. I subsequently downloaded and installed ASIO4ALL after reading notification that popped up on Audition recommending ASIO drivers, then selected the ASIO driver within Audition's 'Audio Hardware Setup' section. Since then, I have been unable to arm any track for recording.

Before downloading the ASIO4ALL driver, I tried to ascertain whether I already had an ASIO driver native to the Realtek soundcard, but could find nothing.

I have to admit, I have no idea about soundcards etc. Maybe it's not as simple as just downloading and installing the ASIO4ALL driver and assuming it will work. Maybe the ASIO4ALL driver is not compatible with my Realtek soundcard?? Maybe a lotta thangs...!

Anyway, if someone savvy could clue me into what might be going wrong, would be appreciative. I'm running Win 7 Home Premium (64bit).

Cheers
Ross

I have a track of two melody instruments playing in unison (they're playing virtually the same notes at the same time). Is there a way to eliminate or at least lower the volume of one of those instruments by zeroing in on its overtones/partials/frequencies and deleting them or lowering their amplitude?

I've been attempting this for two days and thought it's high time I ask for help because the process is insanely slow so far. I've tried using filters and EQ but they seem to be too inaccurate. Another problem is that I'm working with Adobe Audition 3.0, so it doesn't have as great of a notch filter as CS6 or whatever the latest is. I've tried going into the spectral waveform view and manually deleting overtones but it's completely time consuming so far. I don't mind spending a couple days doing this to a track, but after two days I've only really gotten rid of the offending instrument on one note. Too slow.

Any advice or thoughts?

Thanks!

I am a long time user of Audition from the CEP days.  Just upgraded from Au 1.5 to 3.0.  I use a MOTU 828mk2 interface.  For me, always a reliable pairing with CEP 2.1, Au 1.0 and 1.5.  With the upgrade, I'm finding that Au 3.0 keeps overriding the sample rate set in the MOTU driver applet, from 44.1khz to 48khz.  I've tried resetting the preference in the MOTU app, as well as setting within Audition, i.e., pulling up the hardware interface using the Control Panel button within Edit>>Audio Hardware Setup.  The selected driver is MOTU Audio ASIO.  In either case, whenever I go back to the Audio Hardware Setup, it has changed back to 48khz.  I see this change occur on the MOTU front panel sample rate indicator as well.  Anybody run into this with MOTU or other hardware?  I've also loaded the 3.0.1 patch, even though this problem isn't listed in the Bugfix.  Didn't solve it.  Advice appreciated.  Apologies if this has been covered.  I tried searching on 'hardware sample rate change' and similar permutations, but that kept bringing up basic info on setting hardware, sample rate defs, etc.  Thanks.

Hi,

Is there any solution for radio producers, to export from Audtion 3.0 separate files. For example i've got a few parts of radio program in one session - each one consists a HeadIn, Loop, voicetack and HeadOut (arrangedon differenttracks}. Now, I don't want to select one by one and export - I wan't to export everything in that session as mixed, separate files at one go.

Is it possible?

Thanks in advancefor your answer.

Bartek

I really like the way you did about frequency monitoring in AA 2 or 3. And, to be honest, I'mused to itpretty quickly.

But, for some reasons, I am moved to another DAW. I have tried a LOT spectrum analyzers, tried all the best vst in this direction, and all theydid not come close to subject.

So my predictable question to Adobe's developers: Can you make VST version of  frequency analysis, that contained in Adobe Audition 2/3? I'm really sure, that this idea will becomepopular and will get high demand. Thanks, I look forward toyour wisdom.

I'm running AA 3 (OK, I can't afford to upgrade to CS6, or whatever the current version is, so have mercy); and running on a Win7/64 bit machine.

I'm taking a local band's complete concert and trying to break up the individual songs to burn them to CD with track markers. AA 3 is also having an issue with the "write CD" function, where it merely closes the program. I've seen this issue discussed elsewhere, and my workaround has simply been to use another software program to burn CDs.

So I make markers between the songs (doesn't matter whether it's a cue marker, track marker, or what have you). If I save the selection as an mp3 (even a high sample rate version) then re-open that individual song, I see a slight gap (audio silence) at the head and tail of the song. This won't do when burning a CD as it will create a "blip" during playback. What I found is that if I save the selection as a .wav file instead, I do not get the gap at the head/tail of the file. I don't recall running into this before, but I had been using AA 3 on an older box with XP/32 bit; I don't know if that plays a part or not.

Thoughts/suggestions appreciated. In the meantime, I'll just make sure to save these selections as .wav files; these days there is enough hard drive space to support the larger files.

I was using Audition 2.0 just fine today and received a message that I need to reactivate my copy of Audition. I tried to activate online but kept receiving the Authentication Server unavailable (I know you recently shut it down)

Error Code: 93-9

After talking to an online chat person I was told they can't help with 2.0 on chat. I called the phone number given and the support person kept saying over and over he was able to activate my copy on his computer. But I can't get the message to away (server unavailable) and thus can't use the program putting all of my work to a stop.

I told him there is a phone activation option but the automated system kept saying the activation number was too short.

The third support tech told me to post a message on here and someone could help me through a live chat

At my job, I'm not able to use any optimization or effects on any of the audio we produce (long story) and I have to manually adjust the audio's peaks in volume, which is an incredibly long and painstaking process. Is there any process I can apply to the audio as a whole that will level the volume of the entire recording without compressing and leaving the audio sounding muddy and distorted?

I'm new to Audition - got 3.0 when Adobe made it available for free download. I have Cool Edit Pro and know my way around that at a basic level, and it seems Audition is similar. BUT, I can't get it to record.

I can arm the track and start recording, but no sound is being picked up from my mike.

I was able to record through this mike when I had my Realtek driver selected. I  subsequently downloaded and installed ASIO4ALL after reading notification that popped up on Audition recommending ASIO drivers, then  selected the ASIO driver within Audition's 'Audio Hardware Setup' section.

In the Control Panel my Realtek HD Audio soundcard does appear as the sole item in the WDM Device List, so I guess that means it is being associated with the ASIO driver.

I've checked the mike Properties (in Windows) and set the recording level pretty high (to 77). I just downloaded Audacity and my mike was picked up immediately when recording, no problem.

So, I'm stumped! Anyone have any idea as to why my mike is not being picked up when recording in Audition 3.0, when it was when I first downloaded Audition and had my Realtek soundcard selected.

I'm running Win 7 Home Premium (64bit).

Cheers all
Ross

Hello,

Sorry for my bad English! I am French!

I recently bought a Sony digital recorder (ICD-PX312) that records MP3 files.

I recorded my noisy neighbour, in continuous mode for three days. (I was not at home)

PX312 recorder has produced several files which maximum size is  511,845 kB corresponding to a lenght of 24H 15min.

Every file, checked with Mediainfo, have the same properties. (48kbps 44.1KHz)

I can read them with VLC media player without any problem, but I need to edit these files to keep only the interesting parts.

If I open (drag and drop or open file) these files with Audition 1.0 (I came from Cool-Edit), I found a very strange behavior.

The 24H 15min files are opened with a lenght of 1H and a half or so!.

I gave a try opening with Audition 3.0 and the result is strange too, and more "funny":

The lenght of one 24H15min file is 1H and a half

The lenght of another 24H 15min file is 21H or so.

In the Audition TMP directory, Audition has created several 4 GB temporay files that correspond to the limit of the WAV files.

I made a test with a 128kpbs, 44.1khz. This 511,845 kB file is  9H 5min long

Audition read it as 4H40min

The TMP file is 2,897,600 kB, far below the 4 GB limit

It seems Audition 1 and 3 (I read CS5 has the same problem too) does not share the WAV conversion very well.

Is it due to Audition itself or to the Fraunhofer MP3 codec?

Is there any workaround from Adobe?

As I am an AVS4YOU client, I tried AVS Audio Editor. It opens every file with the good lenght but does not have the needed editing functions. That demonstrates that it is possible to share very large files.

Many thanks in advance for any help or idea, because Audition is the editor I need!

Best Regards

Dear fellow users, thank you for reading my posts and the very helpful replies.  My earlier problem ergo 'missing presets' in the FAVORITES tab/dropdown menu (Edit window) is now solved thanks to Richard Yoeman-Clark who kindly sent me a copy of the AudCust.ini file.  I dropped that into my empty file folder.  Worked like a charm and I now have my presets back!  I've made a copy of that little file!  Heartfelt thanks to you Sir Richard!

Now I'd like to raise two other vexing questions:

1) I'm recording my audio tracks into separate stereo tracks in the main multi-track window directly in 'real time' from my stand alone Yamaha QY-700 sequencer.  For example, the current project (without any vocal tracks as yet) stands at 27 tracks of audio - I've split the drum parts up and every other instrument has its own stereo track in Au3's multi-track window.  These are all stereo because I'm taking a 'L' & 'R' audio output from the sequencer into my Edirol UA-25 inputs.  The USB connection is then carrying the digitised data directly into Au3 as a stereo input signal, not a mono source.

2) By lining up a programmed 'click' count-in at the start of every audio track, I'm then able to use a vertical marker and hone in on the onset of each click with sample accuracy at the beginning of each track and then line track 2 up with track 1 and so on... however, as the track progresses in time during playback things slowly drift out of time.  So I've been resorting to cutting and re-positioning segments of the audio to get them back into sync' with the preceding track.  Very laborious!  So why don't I use some sort of time code or master-slave midi arrangement from the sequencer to Au3 or the reverse?

Answer: It's not working!  I've connected 2 midi cables from the QY-700 to the rear of the UA-25 but when I press PLAY on the Yamaha no midi signal is being picked up by Au3 to initiate recording.  I then tried the reverse procedure, with Au3 in record standby mode, then pressed the record button but no midi-clock signal is being recognised by the QY-700 and it just sits there without advancing a single clock pulse. Any suggestions?

QUESTION 2: Playback across the multi-track is normal until I add EQ on any of the individual tracks and in particular if I add any compression/mastering effects on the master output channel strip.  Skips, drop outs, hiccups, juddering, dead stops - the same sort of sound you'd get if you knocked a record player stylus across your favorite album! - you know that screeching sound... So is this an audio buffer related problem or could it be something worse?  I did get a message on my screen saying 'Warning: Heavy CPU Load by Au3' so I'm wondering how to cure this.  My laptop isn't SO old!  Its an Acer 5720 with an Intel Core 2 Duo processor T7500.  Note: All session files are stored on the D drive and there's adequete memory - a healthy 58.7 Gigabytes.  The Au3 program is stored on C Drive - and there's 31 Gigabyte free on that drive.

Previously I was storing/reading back all my session files from an external Maxtor HD, but until I upgrade to a USB 3 connection from the laptop (later this month) I was advised that it was best to save all relevant session files internally on the D drive to speed up data flow.  I intend to save completed sessions to the external HD in any case, then delete the finished sessions/files from the D drive, then start a new project with a clean slate.  I also investigated the idea of buying a solid state hard drive but was advised that in practise I wouldn't get any noticeable improvement in CPU speed/Au3 performance.

I'm not a computer programmer but have tried to think about how to solve/improve performance issues like these.  It's great to be able to lay out big multi-track recordings, but if I can't utilise the EQ's, busses, reverbs or compressors on the master output strip without the above problems - then there's something seriously impeding that and I'm not sure what the problem can be. 

Re' the midi master/slave capability, if I could get that working it would solve the syncronisation headaches!

Thanks to those that have read this post and given it any consideration!  The midi settings available in the QY-700 are:

MID SYNC: Internal | Midi A | Midi B | MTC Midi - A | MTC Midi - B

MIDI CONTROL IN: OFF | IN - A | IN - B | IN - A, B

OUT: OFF | OUT - A | OUT - B | OUT - A, B

Does anyone know what I should be selecting in this chain and which corresponding commands in Au3 should be selected/de-selected to engage MASTER or SLAVE modes and vice versa with regard to the QY-700?

Au3 gives the following choices:  MIDI TRIGGER ENABLE | SMPTE/MTC SLAVE ENABLE | SMPTE/MTC MASTER ENABLE | SMPTE START OFFSET

Should I be selecting all of the above options? Or selecting only one?  In the 'Keyboard shortcuts and Midi triggers' pane, I've selected *any* midi channel.

If there are any another MIDI control panes, fields or switches I've not yet discovered them. - ( Andy posting as Day Of The Schmeckel )

I have a single-file recording of a vinyl record (containing multiple sequential tracks) that I want to export to multiple separate tracks. Is it possible to take thisfile of audio, add markers to it and export that file from multitrack view into separate tracks? I know its possible to do it using markers and to export a CD list and the burning to CD, but can you export to multiple gapless WAV files instead, with the markers as track/split points?

I am not burning to CD, just wanting to keep the tracks as files.

I've always recorded analogue audio with Audition 3.0, but am getting the bug to record midi and use some VSTi plugins.   According to this helpful doc from Adobe the process looks pretty straightforward: http://forums.adobe.com/message/1196432

Following this how-to to the letter, I have the Audition Polysynth showing in the Sequencer window.  The left and right arrows are currently set to <all>.   The problem is that I am hearing nothing when I click on the keys in the piano roll.   All other audio plays fine from the system. 

I double checked the Hardware Setup and am not seeing anything specific to MIDI.    In my Windows 7 audio setup, MIDI audio output is set to the same audio device as analogue output (which works fine). 

Right now I'm not worried about my MIDI keyboard/controller - I'm just trying to get sound from the piano roll in the Sequencer.           

I am using an older version of Audition - v1.5 to be exact - and am encountering problems with the audio on mix down. I am adding tags to show audio and then adjusting the length to fit the requirements of our broadcast clock. The time adjustments are no more than 2% and are done at "high precision". What I am experiencing with the resultant audio is wackiness with the audio quality. The best way to describe what I am hearing is it sounds like an audio recording on tape that has been "accordianed" or as fluttery audio. It does not happen throughout the entire file, only in certain places. It does not appear to have any rhyme or reason as to when it occurs in the file.

I have checked all other possiblities in the chain the audio passes through and have narrowed down to Audition. All processing is being done within Audition. All audio is the same bit depth and sample rate - 44.1K and 16 bit. Anyone have thoughts as to why this is happening and what I can do to remedy the situation?

Thanks in advance for any help you can offer.

Hello everybody! I'mlooking for suggestions aboutaproblem i have withAudition (3.0.1).

When i recordatrack whitthe Graphic Eq activated, i have a lot of latency in monitoring!

I noticed that theproblem occours only with this effect, and notwithreverbs, chorus,compressors, flangers, andevenwith otherequalizers(ex. "Parametric"). Only with the Graphic Equalizer.

I havealso triedrecording withASIO4ALLanda USB microphone (instead ofmy TascamUS 322)..

Really don't know why, andI don't believethatthe graphic equalizercanbe an effectso heavyfor the CPU.

Thanks in advance forany advice youcan give me!

Ciao,

Gian.