Audition

When I record singing, or any audio, it changes the pitch and its lower than recorded.How do I fix this?Thankyou!

no longer recording direct pickup@ sound from speaker only

nun ist endlich die Benutzeroberfläche so kaputt, dass man nur noch Gemüse sieht. Selbs die Buttons in der Multitrack Ansicht sind nur noch Buchstaben - mit Hintergrund wenn aktiviert.Wann hört dieses unsägliche glatt hobeln von Oberflächen endlich auf?! Bei Audition jedenfalls ist nichts mehr von einer einst sehr schönen Benutzer Oberfläche übrig. Ich hoffe, irgendein Product  Manager liest diesen Beitrag...

Hi!So I have noticed the following odd thing.When I move the slider across the timeline then the levels top right and the audio that is played back are somehow "ahead" of the waveform.As you can see in this screenshot, the levels peak, while at that point in the waveform there should be 0db.Anyone an idea what is going on here?

Hi! I recently purchased Audition because I have an old Microcassette tape that needs to be restored. So, yes, I am new to Audition. My problem is the voice on the tape is hard to hear and hard make out what she is saying clearly. I have removed background noises using the spectral frequency filter (it works very good) and then amplified the voice to make it easier to hear. I was successful at removing the background interference but It is almost like the person is talking too fast. Is there a way to slow the voice down slightly? Or any ideas or tips or steps etc on how I should approach this restoration. I only have a week or so to figure it out. All help is greatly appreciated! Thanks!

Hello everyone!!So, I was recording an interview for my school project, and one of the clips turned out to have some kind of "noise" or "eco" in the background, unfortunatly I cannot record it again, because the interviewee is alredy out of the country, I tried to fix it, however unsuccefully.I wonder if any of you audition masters can fix it ??Thank you very much!!DiogoPS: Apparently I cannot upload the audio file, so if you think you might be able to help, please add me on skype: manosayon.❤️

I am using an audient ID14 mixer with audition. I set it up in audio hardware as Audient Audio USB ASIO Driver. I can see my levels when I speak into the microphone but when I press record it is not recording anything. Is there something else I need to set up too?

Could someone explain to me the proper settings for processing audio for broadcast on FM radio, Internet, & Archive (podcast)? I have bee using Adobe Audition for a while, but this software has so many bells & whistles that I am overwhelmed. I know how to remove dead air, splice files, & use normalization, but I'm a bit cloudy as far as what processing settings I should use before saving to .WAV or .MP3. I'm most concerned with keeping the level even, as I have to occasionally deal with 78RPM records that were recorded before modern audio processing was invented. Perhaps someone can help me with this, as I didn't have the luxury of going to radio school.

I am trying to download CS6.After I download the 2 files, I open the smaller file, and it goes thru part of setup, then stops and I have nothing.

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?