Stretch/shrink audio samples to fit the beat?

Hey there, I'm new to GarageBand. I always used Sony Acid pro on a winbased pc because that piece of software is so fast to work with. The most convenient feature of Acid is the fact that it automatically stretches or shrinks your (wav/mp3) audio samples to fit in the the BPM you've set.
Is there a function in Garageband that can do something like this??

perhaps the ableton daw is a better choice if you are doing lots of beat matching and stretching, or want to go "live". pricey though.
http://www.ableton.com/
lots of us cheapskate garagebanders will use free audacity softo to manipulate sound outside of GB. http://audacity.sourceforge.net/
iBook G4, 1.33gHz, 768MB ram,SuperDrive Mac OS X (10.3.9)

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Is there a way to "shrink" or "stretch" an audio track to fit the video?

Hi all,
I've filmed a family event and decided to use an external audio recorder to pick up better detail of the participants further away from the cam's mike.
The length of the event is about 35 mins. and the audio (if sync'd at the start) will be out by a good 30 secs. by the end.
Can I "adjust" this from within iMovie or do I need an external app. to edit the .wav from the sound recorder?
I did search the forum, but some of the links were dead and some references are made to FCE or apps unknown to me.
Thanks for all your help!

Hi
As far as I know - This can not be done in any version of iMovie.
I would
Import material from Camera into iMovie (or other video editor eg iM HD6 or FCE/P)
Then export audio from this as .aiff via Export as QuickTime expert mode - 16-bit 48kHz so I can use this as a mould
Then I would use any Audio-Editor - Audacity (free on internet) and by using this import the other recorded audio
Now by having the Camera original audio on one set of tracks - I can put the recorded audio on-top of this and use Pitch shift to increase or de-crease it but without losing in frequence (Chipmunks or Giant - sounding)
Then when very exact match - Export out the audio from recorder and now as .aiff 16-bit 48kHz too
When imported this into iMovie and muted Camera audio - It should work.
There was a program "Amazing Slow-downer" which could do this too - but it was not free and You still had to do this outside iMovie.
Yours Bengt W

What is the best way to deal with different audio sample rates on the same timeline ?

what is the best way to deal with different audio sample rates on the same timeline ?

You don't have to do anything special. If possible, start your project with a clip that has the desired target frame rate and audio sample rate, and your project parameters will be set automatically. Other sample rates will be converted under the covers.
For example, if your video is shot at 48khz, you can add music files at 44.1khz with no problem.
If you are recording audio that you want to synch with video (multicam), you will get best results if everything is 48khz, but you can use 44.1 if that is all you have. Once I forgot to reset my Zoom to 48,000 and it still worked.

How do I change the audio sample rate from 48kHz to 44.1kHz for Mpeg2

Hey all,
I've been searching for a while but haven't found any direct answers in the forums or the user manual so here goes.
I have to dispatch a file to Bloomberg TV and the file specs they have given me are as follows:
Video Standard: MPEG-2, MP:ML, 4:2:0
Frame Rate: PAL 25fps
Video Size 720 x 405
Aspect Ratio 16:9
Audio Standard MPEG-1 Layer 2
MPEG-2 Program Stream Mux rate 6mbs per second
Bit Rate Type: CBR
Video Bit Rate 5.7mbs
Audio Bit Rate 192kbs
Audio Sample Rate 44.1Khz
Interlacing: Upper Field first (why they want interlaced for web streaming is beyond me)
GOP Structure: IBBP
I-Frame distance 12
Now everything above is fine except the audio encoding because even though I have set up a new setting from scratch I cant find anywhere to adjust the audio sample rate. The Inspector tells me the Audio encoder is set to:
Format: MPEG
Sample Rate: 48.000kHz
Channels: 2
Bits Per Sample: 16
Anyone Know how or even if I can change these audio settings? The only adjustments I can find are the filters or the transport/program stream option. I have it set to program as specified by Bloomberg.
Thanks in advance
J

The only setting that I could find in compressor that lets your change the bitrate to 44.1 is when you create a new dolby digital setting and then under the inspectors audio tab/Target System button, change the button to Generic AC-3. When done, you can change the Sample Rate to 44.1.
Hope this helps?

Project sample rate vs the audio interface clock

I used Audition for digitizing archival audio and I mostly work in 96 kHz because it's the current archival standard for analog audio, however I occasionally need to transfer DAT or ADAT tapes at 44.1 or 48 kHz. One thing I noticed when I went to change the project format was that my audio interface clock did not change and match the multitrack settings. I was able to change the clock manually but I had to go to the Audio Hardware preferences.
I find this behavior for a DAW very strange. If I have a multitrack open of one bit rate and my clock set to another, how does that work? How does Audition 6 play and record audio at differnt sample rates than the interface clock? When I open a multitrack that's 96 kHz and my clock is set to 48, what sample rate am I getting? The audio file being saved says 96 kHz but is it?
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loneraver1 wrote:
If true, this is indeed strange. I have used just about every DAW under the sun for the last 10 years based on what my employer purchased and I have never seen a DAW allow you to record sample rates where the project didn't match the incoming clock.
There should be an option that locks the audio interface internal clock to the opened project. It makes it annoying any time I have to switch between recording in 96 kHz to 48 kHz. That's annoying on it's own right, but what's really annoying is that it's an extra step that I have to train our archival technicians who don't have a techical background in audio.
I can definitely say it's true for my set-up!
Surely, the "extra step" you refer to is not necessary if AA is resampling the audio to match the desired sample rate?
As for the "option" to "lock the audio interface clock to the opened project", for those cards, like mine, which cannot have their sample rate changed by the audio software, how is that going to work? I suspect this was how AA CS5.5 and earlier worked: unless session/project and interface sample rate matched, no audio. I much prefer the current "resampling" method; no necessity to have to think about soundcard/interface settings every time a different sample rate is being used.
FWIW, if I want to record something at 44.1 I always ensure the card and AA settings are 44.1; but for "playback" (or video editing which I do quite a lot of and which requires 48 - my video editor software will not "play" unless the audio sampling rate of the card is 48 - ) 48 is my "default" setting.
JMO!

IMac hard drive making clicking sound (audio sample)

A couple of nights ago I realized my five year old iMac was making a sound I have never heard it make before. It is a scratching noise that I assume is coming from my internal hard drive (hard drive is stock, has never been replaced). It isn't a super loud noise, but it sounds like the noise some hard drives make when they are "searching" or "thinking". I'm wondering if my hard drive is beginning to fail, or if the problem could be something as simple as dust or dirt caught in the drive. An audio sample is in the link below, please help me diagnose my problem. Thanks!
Audio sample: http://cl.ly/073d3U1e1t280M2X0m3F

When you ran SMART Utility, were any of the sensors out of range or close to it, although not enough to trip the health measure? Just wondering. I once replaced an older hard drive that was showing error rates above normal, even when the drive was still 'verified'.
Other than that, you've run several tests that would say the hardware is OK. As GeekStacy said, new hard drive noises are typically not good. So I'd be sure you have a good backup regimen, and if the noises get yet louder or more noticeable, you can either run these various tests again or replace the hard drive.

Will iPhone 5 cases fit the iPhone 5C

Looking to purchase the iPhone 5C and am wondering if my all my iPhone 5 accessories including cases are compatible.

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Error message stops .wmv creation: "Inconsistent Audio Sample Rate..."

Just recently when attempting to export a video into the .wmv format, QuickTime Pro 7.2 gives me this alert message, "Inconsistent Audio Sample Rate. The media you are exporting contains audio with multiple sample rates." QuickTime will not compress (export) to .wmv. It will export to .mov. and .m4v.
This problem is new. For more than a year my weekly video exported to the .wmv format without fail.
I am working on a Mac G4 Dual Processor and iMovieHD.
Any help you can offer would be greatly appreciated.
Mac G4 Dual Processor Mac OS X (10.4.10)

Just in from flip4mac support:
The recent release of QuickTime 7.2 created an audio compression error for Flip4Mac WMV Studio, Studio Pro and Studio Pro HD customers. Specifically, when encoding WMV files with Flip4Mac WMV v.2.1.2 using uncompressed audio sources on QuickTime 7.2 systems, customers were receiving "Inconsistent Audio Sample Rate" error and the encode will fail. This problem has been resolved with the beta release of Flip4Mac WMV v2.1.33. If you are using QuickTime 7.2, please upgrade to this release at http://www.flip4mac.com/wmv_beta.htm

Why is FME audio sample rate restrictive?

I am unable to select higher than 22kHz audio sample rate on
the FME Control Panel - this results in very poor audio quality
delivered! The input device delivers up to 48kHz and is succesfully
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In case you are interested, I have listed the input device
spec below:
DVBETA (LairdTelemedia)
Output Ports available (I use FireWire)
SDI Digital Video: SMPTE 259, PAL or NTSC, on BNC a
connector
AES/EBU Digital Stereo Audio: 16 Bit @ 48 kHz, Embedded on
SDI connector
* IEEE-1394 (FireWire) x2: maximum 400 Mbps, PAL or NTSC, on
a 4-Pin and 6-Pin connectorsMachine Control, RS-422, on a 9-Pin
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I have the same issue when using my video capture card in FME
on Vista.
I am having a lot of issues actually.
1. CPU while encoding video & audio is only reaching
around 15% but I am dropping frames when I try and use high quality
encoding settings.
2. Audio and Video get out of sync very easily.
3. Audio sounds VERY odd, especially during music. It seems
like the speed of the audio is constantly shifting. Makes things
sound like oooOOOoooOOoo etc..
I hope Adobe is addressing these issues.
Thanks and take care.

Audio sample at variable frequency

I think this is the right forum for that question.<BR>
Do anyone know if it is possible to play an audio sample an modify the frequency while playing according to a variable value?
Any example?

write yourself a Mixer that can change the frequency of the data flowing through it.</p>
Mixer
<p>
Or find one on the internet that does it.
<p>
matfud

Can I move audio in logic by the sample in the sample editor to compensate

I wanted a final say on wether or not I can move audio with sample accurate percision by moving the anchor in the sample editor with update in arrange page turned off. i just need a verification of this.
Thanks guys

yes you can and should use the sample editor to make
such compensations. you do this by aligning the
anchor point to the sample or part of the wave form
that you want to correspond to a beat or subdivision
(ideally) though of course it can be any point you
want it to.
if you use 'pickup clock; move object or event to
SPL' KC the anchor point will jump to the SPL.
Philip K is right though; if your entire track is out
by the same amount that you can measure you could use
the sample delay (though of course that will only
delay it not move it forward in time). but you won't
see the waveforms align themselves to the beat
meaning you would have to be aware that that track
has been compensated for.
on balance i would edit with the sample editor.
I was thinking that you could delay the rest
of your tracks by however many samples thus
aligning by compensating everything else though
in real world situations I don't know how practical
this would be. I keep forgetting that you can measure
how many samples you select in the sample editor even
if you cannot see them individually. So I will defer
to the voice of experience on this. If Rohan would use
the Sample editor, then I would tend to agree

How to time stretch a 2 bar audio loop to fit logic tempo, please help

hi, i can match the logic tempo to fit the audio loop ok.
But say if i have a song which is 84.7153 tempo.
I have a 2 bar audio loop that i want to incorporate. The audio loop is a little faster than my logic tempo, so i just want it to fit exactly in two cycled logic bars.
Ive just managed to do it pretty close, in recycle, manually.
But can i do this in logic? Audio factory or something?
Ive been using logic for years, and ive never exlplored logics audio depths.
can someone help?
thankyou.

Use the Apple Loops Utility
http://www.digimixstudios.net/sd1al/
In my 3rd video here, I show how to use the Apple Loops Utility. Very simple.
The 4th video shows bringing the apple loop I just created, into a Logic 8 project.
Notice how I change the tempo and the apple loop stays within the 4 bar beat.

Sample rate off the audio input and out devices do not match - what to do?

This is fundamental, I know, but nevertheless I can't find my way around it. I get this error message when trying to recor:
Sample rate off the audio input and out devices do not match
and am asked to do this:
Use the appropriate operating system or audio device control panel to adjust the sample rates of the input and output devices to use the same settingt.
I have defined the sample rate to 44.1/16 bit in accordance with my inbuild soundcardt.
I am trying to record from LineIn.
When running on a M-Audio sound card I don't face any problems.
HP 8560W, sound card IDT/High definition audio Codec
Any suggestions?
Knud
Copenhagen

You're sure you have set BOTH the input and output settings to 44.1 16 bit?
Which version of Windows are you running? There are a number of posts on this forum about how to fully access both the Windows Mixer and the Mixer for your soundcard. Especially, you need to ensure that all "Windows Sounds" are turned OFF.

How can I scale just one dimension of a pic? I need to shrink the width only, to fit the frame.

HELP! How can I shink only one dimension? I need to shrink the width slightly to fit the frame; the program only allows scaling that is proportional, so when I shrink the width the heigh becomes too short. Father Mike

I'd recommend using the rectangular marquee and the crop command. This will crop the image to an exact size without any resampling or distortion. Cropping without constraint can cause distortion...like giving someone a cone head.
Check the size of the image first by either looking in the Image<Resize<Image Size. Make a notation of the height since you don't want to change that in this case. (While you are there, change the resolution to the desired output...make notation of height after this step if you changed the resolution.)
Select the Rectangular Marquee tool. Change the setting in the options bar to "Fixed Size" and input the desired size. (Along with the numbers, specify in this box in what units you desire...px, in, etc.
Drag the marquee to the desired location. (As long as the marquee tool is selected, this selection box can be moved around).
Go up to the menu bar and select Edit<Crop.
Reading material on Image<Resize<Image Size:
http://help.adobe.com/en_US/PhotoshopElements/7.0_Win/WSae2ea3b149d0c3591ae939f103860b3d59 -7ee8.html
Reading material on Crop Command:
http://help.adobe.com/en_US/PhotoshopElements/7.0_Win/WSae2ea3b149d0c3591ae939f103860b3d59 -7eed.html

Today is HP Expert Day - someone ask about the Beats audio problem!

Let's do it!

Hi
Try fn+b toggle just open more and more of the Beats.exe process. I too have had this issue. The fix is a simple one.
Make sure you have the SWSETUP folder from the factory install backed up. If not, you need to get a copy of the HPTONECONTROL HPQUICKLAUNCH, and the IDT DRIVER (Audio driver) from that folder.
I tried to get these drivers and programs from the HP support website for the Envy 17, fn+b with these just causes more and more of the Beats.exe process to open up. Even getting HPQUICKLAUNCH and the IDT DRIVER from the HP Support website for my Envy 17 did not give me a working Beats Audio.
Note that HPTONECONTROL is not listed on the HP Support website. You must get this from the SWSETUP folder.
I have tested different combinations of different versions of HPQUICKLAUNCH, HPTONECONTROL, and the IDT DRIVER. Only the factory default versions of these actually give a working Beats Audio and fn+b toggle.
Hope this helps and saves you the trouble. Now time to figure out how to get back my life.
Edit: This applies to the Envy 17 and Envy 14 since they both use the HM55 chipset.
Let us know how it goes!
"I work for HP."
****Click the (purple thumbs up icon in the lower right corner of a post) to say thanks****
****Please mark Accept As Solution if it solves your problem****
Regards
Manjunath

Stretch/shrink audio samples to fit the beat?

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