Audition

Hi people,So as you may be able to tell, I'm very new to adobe audition. I was messing around and trying to record myself rap when I noticed I kept being off beat. After doing a test, I noticed I was receiving latency on my recording.(NOTE: I am using a MXL990 Blizzard through a Focusrite Scarlet Solo Inerface for recording, and an Astro Mixamp plugged into the Optical port on my computer)When I checked my preferences, I noticed I could adjust the Latency down to 30ms, which is still noticeable for me.I then changed the device class from MME to ASIO (not sure what either of these are). Doing this disables all latency, and I have these options:ASIO4ALL v2: Does not register microphone, provides no playback device option and I get no sound.FL Studio ASIO: Registers the mic, is quiter and a slight delay, does not allow me to use my Blizzard/Focusrite interface.***Focusrite USB ASIO: Absolutely ZERO latency, registers the mic fine!HOWEVER, there is no audio playback via the Focusrite ASIO.

Hi there,I've made a selection:Then: Effects > Noise Reduction/Restoration > Sound Remover (process)And I get this:Why does it say Sound Model: Undefined?Your feedback will be kindly appreciated.

I can open up the video in After Effects and I can see and hear the video.I can open up the video in any media player, and I can see and hear the video.I drag/dropped the video into Adobe Audtion (and also tried Importing).The Files window shows both an .mp4 and .audio for the video.I double click the audio portion and press play, and no sound emits from the .audioYou can see in the Editor window that there is soundBut nothing is playing.Any advice?

Hi,I am working with a high-frequency recording underwater hydrophone that is not supplied with a frequency response curve. The sampling rate is 576 kHz, therefore the frequency range reaches up to 288 kHz. The hydrophone has a flat frequency response from 20 Hz - 150 kHz (+/- 3 dB) and then suffers a roll-off of 6 dB per octave above 150 kHz. The next octave would include all frequencies up to 300 kHz. I would like to know if I am applying the appropriate equalization settings in order to achieve a full normalized response over the range of the hydrophone. Ideally, I need to have a full flat response in order to calculate measurements of ultrasonic dolphin echolocation clicks in which some of the energy from each click extends well above 150 kHz.First, I calculated the roll-off correction values from 151 kHz - 300 kHz based on the 6 dB roll-off. The correction value at 288 kHz is 5.64 dB. However, I am not sure if I need to include an additional 3 dB from the flat response portion of

Am I missing something? Is there a way to view the processes that are queued up for Audition to complete because it seems at some point I always reach the stage where everything I try and do, including save, just pops up a box that says 'Waiting...' and the process stays at 0%. Obviously, as this includes saving, I end up losing my work as I can't actually complete the save, or any other process. Can I see what it's waiting for? And even more importantly can I kill a bad task in the queue that's causing my entire Audition to seize up?

I've got a 5.1 session in Au CS6 for a movie soundtrack with a lot of effects. I've done the usual of putting effects on tracks whenever possible (vs clips), and saving reverb for busses, but I've run up against my limit, and am getting artifacts now during playback. I know this won't affect render, but I wanted to know how to limit CPU time for effects that are on a track but where there is no audio file - if that's possible. For example, if I Power On the Track Effects Rack when there is no audio file, is there CPU usage for those effects? Would I benefit by Powering Off that Rack even when there are no audio files (data) being run through that Rack? Are there other methods, such as Mute? I'm just trying to determine how I might minimize CPU usage for effects that are on a track that aren't actually being used. It  would require a lot of automation, but worth it if I can continue to listen to my mix live without bouncing tracks.Thanks,Shane

I am trying to create speech test materials for audiology, and need to split recorded 5-word sentences into 5 separate clips and export those 5 clips as 5 WAV files. Those files will later be combined on the fly in other software to create new sentences. Using results from some pilot testing, the RMS amplitude of each file needs to be adjusted according to it's intelligibility. Obviously, for that to work and sound good (no clicks), there need to be ramps at each end of the file. I wish to have 5 ms ramps.All of this work is easily done in the Magix Sequoia DAW. Unfortunately, the one thing Sequoia doesn't have that I need is a way to get detailed amplitude statistics about a clip. I need to be able to see average RMS amplitude, which Audition can do. I had wanted to do everything in Audition, so if anyone can think of an alternative workflow to accomplish everything else that would be greatly appreciated.Maybe I'm having trouble because I'm used to working in multitrack mode in a very

Here is a sample of the sharp spiking effect. I attempted to use hard limiter (cs6 audition) and that did not improve the condition.What tool can I use to reduce, possible remove, that spiking? I have a 30 minutes video that I need to make corrections on.Thanks!ple

Hello,     I need to export a .wav file, so when the file is opened in a DAW, you are able to see the 2 different panned tracks, one L one R.  Any ideas, thanks!  (as of now I export as a mono and has just the 1 channel/track seen,  when i export as stereo it has what looks like the 2 channels but its the same audio obviously)  (I need it to look like the stereo with the 2 channels when the file is opened, but i need those to be the 2 different panned tracks from my multitrack session I exported as the .wav)any ideas??  Thanks so much!

I can't remember how to make the metronome display as a bus/track. I can only get it to toggle on or off, etc. I have this bus in some of my sessions, but I can't remember how I got it there for a new one I opened.

I have recently upgraded to Adobe CC. All of my files from previous versions will import into the multitrack at the cursor point, but the files produced with Adobe CC import to the beginning of the session on a new track. Is there a setting that causes this?

In editing an audio file from Premiere Pro, when I put it in Audition and check "noise reduction" it seems to do a good job of taking out a major hum.  However, when I 'save' or try to import back into Premiere Pro, the helpful effect is not saved.  What am I doing wrong?  Thanks for any help.

Hey! So I'm recording audio for a youtube channel, and we're using two exactly identical Blue Snowball mics, plugged in to USB. They each have their own direct 3.0 USB slots. We use an ASIO driver so that we record to one track, with one mic in the left channel, and one in the right. I'm using Adobe CC, on Windows 10.Here's the initial problem. Roughly every 5 minutes, one mic always falls just a LITTTTLE behind. I'm taking .5 seconds, enough to give an odd echo. So over a 30 minute period, it's even worse. Right now, I just pop into multitrack and agonizingly fix it, but we have 0 idea why this is happening. It normally occurs about an hour into recording, and then The second problem is that every now and then, that same mic that falls out, makes everything it records sound like it had a robotic filter put through it. It's very distorted, metallic, and rough. And it ramps up, it'll start with a small crackle, and eventually just builds into an unusable mess. I'm not SURE if this probl

I have a narration where the performer spoke a little too quickly. I need to insert a 'beat' in between one phrase and another. How can I do that?

Hello, on a song I've recently been recording, I'm having an issue where I will export to .mp3, play it on my computer, home stereo and car stereo. The song will sound fine on my computer (played at full volume) and my home stereo (haven't tested at full volume because, well, neighbors). However, when I play it back at full volume in my car, there is significant distortion on the vocals as if it is clipping. Now, I would be willing to accept that my car speakers aren't that great and move on but it doesn't happen on other music (even other stuff that I have recorded).I record using Audition (I guess that's a given) on a Macbook Pro via a M-Audio Fastrack Pro USB interface. The session itself is a stereo session with a bit depth of 32 (float) and 44.1kHz Sample Rate. The song has two vocal tracks (melody and harmony), both recorded using a Shure SM58 mic. Together (panned right 50 and left 50) they peak at about -3dB. I also double-tracked an acoustic guitar strumming chords (also recor

Hello, everyone,Audition CC user here. I am trying to figure out this pitch shift functionality and I can’t achieve what I need. I have multiple notes playing at the same time in the sound file, and I need to change pitch of only one of them. So my graph looks like this:If I make a selection of some range and apply pitch shift, it will shift both the lower, the medium, and the higher pitches, but I need to just shift this way:How can I achieve this? Would the SDK allow me to build a plugin that can help get this?

I often work with large multi track live recordings. The nature of the recordings mean a TON of volume automation is needed--But I only own an old Digi002 which won't work with Audition (not sure it even runs pro tools anymore). I do however have a midi keyboard that has a bunch of faders/knobs that are midi cc assignable. In presonus, it's relatively easy to assign those to do quick volume automation. Anyone know if you can perform similar operations in Audition? Or if any of the various ipad control surfaces will integrate with the software?Thanks!

hiI'm having trouble with audio jitter in my multitrack session, which is also playing dynamically linked video from Premiere Pro. It's a 17min film, and there are quite a number of tracks (maybe 50) although almost no real-time effects at all in the session. (And of those 50 tracks no more than 6 are playing at once)I'm using an imac on OSX 10.9.5, with plenty of free disc space, 3.4GHz quad-core i7 processor, AMD Radeon HD 6970M graphics, and audio through my MOTU 828mkii. I have 13Gb RAM assigned to Audition and Premiere Pro, Premiere is actually not even open, so basically 13Gb assigned to Audition.I've upped the I/O buffer size to the max available, and cleaned my media cache.So why is my audio jittering on playback? It's unfortunately impossible to edit like this.It seems to me I'm not asking Audition to do a great deal other than mix 3 or 4 tracks together at any one time, and simultaneously playback the dynamically linked video. In fact, when I work in Premiere I can have a hug

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I have a question which I would appreciate assistance on.When I record my voice over in Audition, after I do my noise removal when I listen back, there occurs a strange occasional 'bump'  noise at one or two random places in the recording. Has anyone else experienced this and can you offer an explanation?Les

So 90% of the time Audition works flawlessly. Great DAW and all. But one project causes massive latency/buffer issues. Slow/choppy/distorted sound. My equipment is below:Intel i7 3770s 3.9GHz24GB DDR3 RamPresonus FP10 Firewire interface Sonnet FireWire PCI cardWindows 10 ProI can't imagine with my specs computer performance is an issue. In fact, I've mixed larger sessions with the same setup before (I have a similar project in presonus one that is working just fine, even with a ton of processing heavy plugins).The session in question is a 30tk live recording, with a handful of plugins.The things I've tried:Changing every buffer/latency setting I can think ofTurning off effects for every single tracktrying different interfaces (M-Audio Fast Track Ultra, and an old tascam somethingsomething mkII)turning off all my network devicesclosing every other open applicationThings to note:Even though I've allocated up to 20GB for Adobe CC, my ram usage peaks about 8GB when playing back the session

Hi, I've gotten an omf from the recording studio and all the clips and tracks are there in order and labeled. but the source files have random names, is there a way to export it all or save it so that i just have the wavs cut like in the multitrack session and with the labels they have there too? I need to be able to import the files into Storyboard pro and not have them mixed down. I need to keep them seperate as i work out the timing for an animatic.So is there a way to save all the clips in my multitrack session as seperate wavs with their label as their filename?Thanks

Hi, I have an issue and I hope someone will help me, I'm sorry, but I don't have much time to dig into the old threads, there's too much. So,.I'm still using CS6 and I did create and use my own custom templates, where each track has it own set of plugins (chain)(pre-configurations of plugins are saved on my pc of course)every week I have to process a voice, always the same voice, so I use templates and custom plugins chains in the FX Rackbut when I load a session, CS6 never keep the Rack effects pré-configuration name I saved, instead, it always load the something else. (see pic1 bellow)so after loading my session, every week,  first, I always have to reload my plugin chain manually. even if it's not a big problem, but annoying as for example, in that session always I always use the same plugins chains. so, normaly, I should be able to load the session and see the name of my custom plug in chain Preconfiguration.. instead of "personnalisé" (as displayed in the pic bellow) , why do

I am using for recording sessions with identical track configurations the function "export session as template" successfully. However, when I completed mixing and arranging the 5.1 channels I would like to apply the mixing and arrangement properties to the subsequent sessions in order to make sure each of the other session mixes sound identical.I have not found a way how to accomplish this in AU CC 2017. Can anyone direct me to the right procedure/feature in AU?

I'm new to Audition. I have a terrible dialogue track, with pops, clicks, A/C and other noises, low input levels, just about every problem that can exist. I understand some of the tools, but in which order should they be applied? e.g. should I remove noise then increase the levels? when should limiters be used? where can I learn more in depth? I have watched videos but i find a lot of general overview type stuff and nothing in-depth. would like to find something in depth about cleaning up dialogue including EQ and ... well from A to Z. any advice is appreciated. thank you