Audio 96 000 Khz to 44 100 .

Hello ;
i have record some audio parts in 96 000 Khz , i need to save the full mastering in 44 100 khz , but i don't find any thing in options to change , or to record in 44100 ... i can't change it when i want to export the record ... how can i do ?
(if i try to record , it makes an empty track)
Thanks

You need to use Convert Sample Type from the Edit menu or shortcut F11.

Similar Messages

Can I make my input/output audio 96.000 kHz & 32bit as default?

Just what the title said. I turned on Soundflower, and changed audio input and output to 96.000kHz and 32bit float. But will those settings remain after I close Soundflower? Will they stay the same after I turn off the machine?

leroydouglas wrote:
Your change should stick
~/Applications/Utilities/AudioMIDI setup.app
So, if I restart/turn off and on the laptop, the changes will still be there?

Horrible audio capturing 2nd stereo channel 12 bit 32,000 kHz

Hello,
I have DV tapes with 2 stereo channels 12 bit 32,000 kHz. I added a Capture Preset Editor with 16 bit 32,000 kHz 2nd 2 channels.
I captured only the sound of the second 2 channels.
The sound is horrible. At some places there is no sound, at other places noise.
How can I capture this audio with the same quality as when I play the DV tapes on my TV ?
Thanks,
Hubert

Use a preset that is 12bit.
Or make one, if one doesn't exist.
CaptM

Why can't I record in 96.000 khz sample rate ?

I am running Logic Pro 9.1.7 (1700.57) (32-bit). My computer is Mac OS 10.6.8. I am using a Duet Apogee
I have always recorded in 44.100 khz, but now someone has asked me to record in 96.000 khz. Everything is fine until I change the sample rate in Logic. Then I get a tinny, distorted sound, and this is for any audio I play through my computer, not just the track that's in Logoc. Is there a way out of this ?
Thanks

You own the hardware and don't know how to use it?
Didn't you install some software for the Duet? Somewhere on your system will be a software applet that let's you change settings on the Duet.
I just took a quick look online.. the "Maestro software Mixer Applet". Try looking in Applications or Utilities.
Need to read up about what you're using. Do not change Duet settings with Logic running.
I don't own a Duet but all of this stuff works basically the same.

Sample Rate 48.001 not 48.000 kHz

Hi,
Sometimes when I capture clips the resulting sample rate of the audio is 48.001. Then, in FCE HD, when playing the clip from the sequence there is a beeping over the normal audio, indicating I need to render the audio.
Does anyone have any advice to get my audio to import at 48.000 and not 48.001 kHz sample rate?
I'm using a Sony TRV310 D8 camera. Computer is G5 Dual 2.3, QT7.0.2, FCE HD 3.0, Mac OS X 10.4.2. Capturing to stock 250GB hard drive.
Thanks,
Chad

Yes, easy setup is set to 48 kHz and my camera is set to 16 bit.
I have the feeling it has something to do with the start/stop points of the footage screwing things up somehow. The footage in question is of a soccer game, about 45 minutes in length and with about 20 start/stops.
FCE HD was set to "stop capture with dropped frames" and the capture (using "clip mode") was done with no perceivable problems.
For now I ended up doing the following workaround on the sound track portion (sorry, I can't remember who originally posted this):
Open your iMovie .dv file in Quicktime Pro.
Extract the audio track
Export the audio track as a Quicktime Movie, Linear PCM format, Channels Stereo (L R), 48kHz, 16bit sample size
Delete the audio track from the .dv file
Open the exported audio file, select all and copy.
Select All in the .dv file
Add the exported Audio file using Add to Movie
Export/Save As the .dv file as a self contained Quicktime Movie
Use this movie in FCE
Trouble is, the sound is slightly out of sync now (I suppose the sound file shrunk by a wee bit going from 48.001 to 48.000 kHz). For this footage the sync is not critical.
I'm still curious as to what exactly is going on with my captures.
Thanks,
Mace

'XiFL', Stereo, 48.000 kHz --- NO SOUND

I was trying to open a .mkv file with the audio format
'XiFL', Stereo, 48.000 kHz
with quicktime 7
I can still see the picture, but there's just no sound.
Then I tried using VLC, iSubtitle, the results are the same - no sound.
I've got Perian DivX Flip4Mac installed. And all the other playbacks are just fine.
But this one with audio format 'XiFL', Stereo, 48.000 kHz just wouldn't make a sound...
I tried converting it to .mov and used Quicktime X to open it, it says that it can not play such a file even if it is a .mov file due to certain missing components. But previewing the .mov file is okay, with full picture, but still no sound.
I'm guessing it's some audio encoding component that I'm missing?
What's this 'XiFL' format anyway...
Please help me =. =

It could be a 'difficult' form of Matroska file. FYI Matroska is just a container like MOV or AVI, and some mkv have the video encoded with Real Video Codec (RV9 RV10). Perian+QT can't decode this, but VLC will play .mkv files just fine on OS X, as long as they don't use H.264!
It's a bit of a lottery until Matroska decide to support the Apple platform. I have found this:
http://mkv4mac.free.fr/index.php?lang=en&p=lec
which may help.
But there may be another workaround: 1. Get Info the mkv file
2. Change to extension to wmv. Save as wmv and that could work better.
Or this link may be helpful:
http://www.squidoo.com/mkv-converter-mac
Recently, a poster (ZachP14) suggested this:
"There's a program called Plex that runs .mkv files awesome on my 2.0 ghz macbook."
http://www.plexapp.com/
Maybe worth a try?

Recording external audio at 48 khz?

If I record all my audio externally, should I be doing so at 48khz? Or does IM5 automatically adjust. The reason I ask is I usually record music and I don't want it to be in the wrong key.
PowerBook 1.67 Mac OS X (10.4.8)

Not "key".
It's frequency and iMovie uses an oversampling of your original source to make a 48kHz audio file.
Larger sample rates (like those found in apps like Garageband) will be reduced to 48kHz.
An audio CD is recorded at 44.1kHz in AIFF format. If you exceed those frequencies your audio source file will be reduced in file size but you will not be able to hear any difference.
Use the Audio MIDI Setup app (Utilities folder) to make sure you're recording at the correct sample rate and frequency. Higher settings are OK but know that iMovie can't make a file above 48kHz.

Using iTunes vs. FCP to convert audio to 48 kHz AIFF?

Are there any real benefits of using one program over the other to do this?
Thanks.

iTunes has a menu option to do this...select the clip, convert. FCP doesn't. You'll have to load the clip into the viewer, FILE>EXPORT>QUicktime COnversion, then do a lot of sub menu maneuvering.
iTunes is faster, and is a good way to keep things organized.
Shane

Audio Rendering Problems

Hello all,
I searched through previous discussions about this, but couldn't find a solution that worked or that I hadn't already tried.
I'm currently working on a class project to make a movie intro, and I want to put it to music from the film I am doing it on. When I RAM preview the video, I have no problems at all. However, when I render it, the audio does not play at all. I know about going into the output module to check the box for Audio Output. However, despite clicking it, the audio does not work at all when I render it, whether it be a quicktime or flv file.
In my frustration to get this working, I've created a new project file with only the audio in it. I've tried turning up the decibals and turning up the volume on my system and speakers as high as they can go, still there's no audio.
I'm still a failry new user to After affects, and this issue is driving me nuts. Would anyone have an idea of what the issue could be? Thanks.

I am running on Mac OSX. I just tried rendering in audio only as a WAV file, but despite multiple attempts the WAV file will does not get created. I cleared my desktop of everything else and set the Output to there, but no WAV file gets created. There are no audio effects being used with this project. I re-imported my Audio into after affects, and the waveforms are there. I also imported the file the into Adobe Soundbooth, the waveforms showed, and I was able to save a Quicktime movie of it with working audio. When you ask "MP use perhaps?" I assume that means media player? I've been using quicktime thus far, and as said before the WAV files aren't even showing up, so I cannot say if that would be a problem for those. I tried closing all programs aside from the Media player for my rendered files, and still no sound. So it seems there isn't anything interfering with it.
Posting some render logs with the settings I used to see if that helps.
Render Log for when I try to render as a Quicktime:
Adobe After Effects v9.0.2 (Macintosh Intel) render log for project “AE Project”
Rendering started on 3/29/10 at 1:17:18 PM
3/29/10 1:17:18 PM: Starting composition “15 Jamukha Is Following 2”.
Render Settings: Best Settings
Quality: Best
Resolution: Full
Size: 720 x 480
Proxy Use: Use No Proxies
Effects: Current Settings
Disk Cache: Read Only
Color Depth: Current Settings
Frame Blending: On for Checked Layers
Field Render: Off
Pulldown: Off
Motion Blur: On for Checked Layers
Use OpenGL: Off
Solos: Current Settings
Time Span: Work Area
Start: 0;00;00;00
End: 0;01;29;22
Duration: 0;01;29;23
Frame Rate: 29.97 (comp)
Guide Layers: All Off
Storage Overflow: On
Skip Existing Files: Off
Output Module: Lossless
Output To: Macintosh HD:Users:Sam:Desktop:Test.mov
Format: QuickTime Movie
Output Info: Animation
Output Info: Spatial Quality = Most (100)
Include: Source XMP Metadata
Output Audio: 48.000 kHz / 16 bit / Stereo
Channels: RGB
Depth: Millions of Colors
Color: Premultiplied
Stretch: -
Crop: -
Final Size: 720 x 480
Profile: -
Embed Profile:
Post-Render Action: None
3/29/10 1:18:01 PM: Finished composition “15 Jamukha Is Following 2”.
Total Time Elapsed: 43 Seconds
Log closed on 3/29/10 at 1:18:02 PM
Render Log for when I try to render as a WAV file:
Adobe After Effects v9.0.2 (Macintosh Intel) render log for project “AE Project”
Rendering started on 3/29/10 at 1:23:26 PM
3/29/10 1:23:26 PM: Starting composition “15 Jamukha Is Following 2”.
Render Settings: Best Settings
Quality: Best
Resolution: Full
Size: 720 x 480
Proxy Use: Use No Proxies
Effects: Current Settings
Disk Cache: Read Only
Color Depth: Current Settings
Frame Blending: On for Checked Layers
Field Render: Off
Pulldown: Off
Motion Blur: On for Checked Layers
Use OpenGL: Off
Solos: Current Settings
Time Span: Work Area
Start: 0;00;00;00
End: 0;01;29;22
Duration: 0;01;29;23
Frame Rate: 29.97 (comp)
Guide Layers: All Off
Storage Overflow: On
Skip Existing Files: Off
Output Module: Lossless
Output To: Macintosh HD:Users:Sam:Desktop:Testing.wav
Format: WAV
Output Info: -
Output Audio: 48.000 kHz / 16 bit / Stereo
Channels: -
Depth: -
Color: -
Stretch: -
Crop: -
Final Size: -
Profile: -
Embed Profile:
Post-Render Action: None
3/29/10 1:23:28 PM: Finished composition “15 Jamukha Is Following 2”.
Total Time Elapsed: 2 Seconds
Log closed on 3/29/10 at 1:23:29 PM
My audio hardware for my MacBook Pro:
Intel High Definition Audio:
Device ID:    0x106B00A4
Audio ID:    58
Available Devices:
Speaker:
Connection:    Internal
Headphone:
Connection:    Combo
Internal Microphone:
Connection:    Internal
Line In:
Connection:    Combo
S/P-DIF Out:
Connection:    Combo
S/P-DIF In:
Connection:    Combo

2 Different Bit Depths- Which is the real one?

I apologize, this gets confusing. Reply and I'll try to clarify the best I can. I'll even submit the example files I've been experimenting with.
In iTunes, I have been converting WAV files into Apple Lossless .m4a files (ALAC). Recently, some of the ones I have converted have had some inconsistent bit depth values. Since I work in Adobe Audition CS6, I can output various formats: format settings of 32/64-bit (Integer OR Floating Point IEEE), sometimes 24-bit Integer, and sample types of 192.000 kHz or 44.100 kHz and up to 32-bit (float).
Contradictory? YES. It sounds weird to have format settings of up to 64-bit Integer/Float, but sample types of various sample rates, and almost always 32-bit (float). Is this a possible reason?
Another problem: iTunes has the ability to convert files to ALAC, among other formats. As it turns out, iTunes has the ability to convert to "32-bit," but only under specific circumstances. Working in Adobe Audition CS6, I have two options for "32-bit:" 32-bit Integer, and 32-bit Floating Point (IEEE). In order for iTunes to be able to maintain that bit depth, I need to select 32-bit INTEGER under Format Settings, and 32-bit under Sample Type. When the file has been exported, I reopen it in Adobe Audition where Bit Depth (in the "Files" box in the top right-hand corner) is listed as "32 (float)" all of a sudden. All I did was export it, I didnt convert it, but despite specifiaclly choosing "Integer," I have 1 32-bit Float audio file. In iTunes, it can open just fine. iTunes calls it "32-bit" as well, and it converts to other formats in iTunes as 32-bit (proabably float) just fine. However, if I export an audio file with 32-bit Floating Point (IEEE) Format Settings from Audition, iTunes fails to convert it to 32-bit, but instead converts it to 16-bit.
Here's what really odd: lately, every single ALAC file opens in Adobe Audition as having a Bit Depth of "32 (float)."
So... Where are the flaws? What are the real bit depths? ALAC files with the same sample rate and different bit depths (according to iTunes) still show up as being "32 (float)" in Adobe Audition. Is iTunes converting them all to 32-bits, and it just doesn't know it?
Thoughts?

32 bit Floating Point is Audition's default file operating mode. It offers considerable advantages over any fixed bit depth integer format .wav files.
See your other threads elsewhere on this forum.
http://forums.adobe.com/thread/1291155?tstart=30

Multiple sample rates!!!

I am doing something I have done a million times but this time I am unable to do so! Here it is.
Pull a clip on to the timeline viewer, extract the audio, split the video clip in half, trash one of the halves, insert other clips to fill in the video, turn the audio down to 0% on the new clips. When I try to export I get a failure that says I am trying to export multiple audio sample rates. I am doing nothing different than earlier today, but now it won't let me. I have tried several variations of this work to fool the computer but nothing works. I am moving on to another movie but if I run into this problem again it's off to Final Cut Express with no regrets.
Any ideas?
I thank you in advance.
Steve

I couldn't reproduce your error although I tried to follow your workflow (I had to use PAL, though).
Anyway, there is an audio sample rate bug in iMovie 6 because it extracts audio at 44.100 kHz instead of 48.000 kHz:
DV works best with 48.000 kHz, 16-bit audio (32.000 kHz, 12-bit audio can produce problems). If the DV audio is 48.000 kHz (as it should), and you use the Advanced/Extract Audio -command, the extracted audio is 44.100 kHz. (You can find the extracted *.aiff audio files at iMovie project package's /Media -folder).
http://www.sjoki.uta.fi/~shmhav/iMovieHD_6_bugs.html#extractaudio
FWIW, iMovie 4 also had an audio sample rate bug:
Imported audio in .mov and MooV files is converted to 32 kHz:
http://www.sjoki.uta.fi/~shmhav/iMovie4bugs.html#32khz

Composition switches to 'not real time'; incomplete render

Hi there,
I'm a film student at Ryerson University in Toronto, new to AE CS3, and having a problem.
I have a 30-second composition utilizing keyframes, and all of the groundwork is done, but in the preview (both regular and RAM) it is not moving in realtime (29.97 fps) and the composition stops 5 seconds early in the RAM preview. When it renders, the image is out of synch with the music.
Even my professor is stuck with it. I've tried disabling the layer cache and set a purge for every frame, but nothing's working.
Any help?
computer is a dell inspiron 1420 with 2GB RAM and plenty of hard drive space.

Thanks Jonas and Mylenium.
Followed every direction and insight, and none fix the problem.
The audio file was created using Audacity; as far as i can tell it was a perfect capture; imported from .mp3 and converted to .wav with no problems on playback, both in WMP and AE during the ram preview.
Once a .mov file, the synching comes and goes. The first 7 seconds are fine, then it slows down and goes out of synch and stays that way.
Render report is as follows:
Adobe After Effects v8.0x247 (Windows) render log for project VS assignment 1 (food).aep
Rendering started on 13/11/2008 at 5:07:22 PM
13/11/2008 5:07:22 PM: Starting composition Comp 1.
Render Settings: DV Settings
Quality: Best
Resolution: Full
Size: 720 x 480
Proxy Use: Use No Proxies
Effects: Current Settings
Disk Cache: Read Only
Color Depth: Current Settings
Frame Blending: On For Checked Layers
Field Render: Lower Field First
Pulldown: Off
Motion Blur: On For Checked Layers
Use OpenGL: Off
Solos: Current Settings
Time Span: Work Area
Start: 0;00;00;00
End: 0;00;29;29
Duration: 0;00;30;00
Frame Rate: 29.97 (comp)
Guide Layers: All Off
Storage Overflow: On
Skip Existing Files: Off
Output Module: Microsoft DV NTSC 48kHz
Output To: C:\Users\Paul\visual studies assignment 2 (food)\Comp 1.avi
Format: Video for Windows Movie
Output Info: Microsoft DV
Spatial Quality = 100
Key frame every 1 frames
Embedding: Project Link
Output Audio: 48.000 kHz / 16 bit / Stereo
Channels: RGB
Depth: Millions of Colors
Color: Premultiplied
Stretch: -
Crop: -
Final Size: 720 x 480
Profile: -
Embed Profile:
Post-Render Action: None
couple more questions:
- Why is exporting a bad idea? The result is the same in both the render queue and export as QT movie.
- how / where do I post the project? You guys sound like pros, that might be the best idea. I could email the files to you (less than 30mb) and if you open it in cs3, it might be easier for you guys to take a look.
Thanks again!
Paul.

Running Lightroom 6 on Windows machine (after having upgraded from LR5... While attempting to add or change music in the Slideshow module, LR unexpectedly quits. After several attempts at this, LR6 won't even launch properly...shows the splash screen but

Running Lightroom 6 on Windows machine (after having upgraded from LR5... While attempting to add or change music in the Slideshow module, LR unexpectedly quits. After several attempts at this, LR6 won't even launch properly...shows the splash screen but that's it. The program is buggy...need an update/revision quickly!!

If you're on Windows, check out a free program called Audacity--it's fantastic. With Audacity the process of converting a 48.000 kHz to 44.100 kHz is as simply as opening, changing the sample rate, and then exporting again, as an MP3.

After Effect error: Rendering error while writing to file

Greetings,
I'm a beginner and am starting to learn most of the basics but every time I try to render a specific type of composition, I get this massage:
After Effect error: Rendering error while writing to file "C:Users\Admin\Videos\720p.mp4".Unknown Error! (-1610153453)
RENDER SETTINGS:
LENGTH: 8min. and 30 sec
FORMAT:H.264
CODEC: mp4
RESOLUTION: 1280 x 720
AUDIO: 48.000 kHz, Stereo
OUTPUT LOCATION: C DRIVE (22GB FREE)
I got Windows Ultimate 64-bit and Adobe After Effect CS5.5 64-bit (both on C drive) installed in my laptop.
When I first installed after effect, I rendered a part of the whole composition (AVI, 720p) and it worked but after that when I rendered with
format H.264 it showed this massage. Then I tried to change it back to AVI. and rendered. again the same massage.
My C drive isn't full, I have 22GB SPACE AVAILABLE.
And when I try to save it on D (114GB free) or E drive (33GB free), It shows me the SAME MASSAGE.
PLEASE HELP
Thanks in advance.

Hi HR and everyone that have the problem error (-1610153453)
In my case I solved the problem changing the destination folder, I was rendering in a folder with a non english character! in my spanish language we have the "Ñ" ('n' with a wave on top ) after effects mp4 codec NO Ñ or cyrillic, etc...
I finally solve it thanks the russian friends: (http://nix-studio-edition.ru/tutorials/faq/1083-ae-1610153453.html) and Google translator
Hope it helps to desperate workers
Thanks

Rendering composition ends up huge

Hello,
I'm very new to AfterEffect so I appologize if this sounds obvious to other people. But I wonder why my output file is so big while the input files I've used to create the composition were not.
My input files are videos shot with a Canon 5D MkII and they are respectively 98Mb, 136Mb and 36Mb. I created a composition and added a warp stabilizer effect, that's it.
Then I render using the following ouput settings:
H.264 with bitrate encoding set to CBR 190 Mbps
Audio 48,000 Khz Stéréo.
This is as close as I could get to the original videos in term of picture quality, yet it's not quite as good even with these settings. And the problem is that I end up with a 1Gb mp4 file !
I don't understand how I can keep the hight quality of the input videos without increasing the file size....
Any help would be appreciated !
Thank you

I don't understand how I can keep the hight quality of the input videos without increasing the file size....
It's impossible. You're working off an already compressed format and add a second pass of compression on top of it. Image quality will always be worse than the original, especialyl since you're also applying a destructive process that distorts the source footage, changing any existing pixel patterns considerably so even if AE did some sort of pass-through encoding, it would still do a full re-encode since there is no original data left that would allow things like a fast transcode. Things are just as they are. If you want to retaain better image quality, you wil lhave to use intermediat workflwos based on image sequences or things like Quicktime files with Animation compression. That would also allow using tools like AME to achieve better compression results externally. Still, the original points stick: Compression on top of compression will never look like the original.
Mylenium

Audio 96 000 Khz to 44 100 .

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