Audio Compression

What is the best audio compression for a music video? I want to keep audio as well as it being at standard levels.
for some reason i choosed dolby digital professioinal but the audio is low. . .
Thanks

Set Dialog Normalization to -31dbfs. Set Compression to NONE.
-DH

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Multi-clip Audio Compression

Is there a way to apply audio compression to multiple clips simultaneously? I had some audio difficulties and was using the camera's compression in early shots and not in later ones, so the balance, levels, and dynamic range are significantly different. The uncompressed sound has much wider dynamic range and clips in a few spots, so I can just pull the levels up.

Select a bunch of clips in the timeline.
Use Modify>Levels.
You use Absolute or Relative to make adjustments.
Relative may be what you are looking for.
Al

Correct audio compression for web playback.

I am making a webseries and I want to know what is the best audio compression for web playback.
I tried H.264, but the audio levels were much lower when I played it back outside of finalcut.
any advice?

this might help a little -
http://www.kenstone.net/fcphomepage/qt_movies_fromfcp.html
and
http://www.kenstone.net/fcphomepage/youtube_compressorgary.html

How do I stop beats audio compressing the sound up and down

How do I stop beats audio compressing the sound up and down.

Hi Strung,
I see that you are having some issues with compression and Beats Audio. I have done some research into your issue and will be happy to help. Here are a couple of links to HP Support Forum threads where they talk about the settings in the Beats Audio.
http://h30434.www3.hp.com/t5/Notebook-PC-Sound-and-Audio/DV7-w-Beats-sound-compression/td-p/396705/p...
http://h30434.www3.hp.com/t5/Notebook-PC-Sound-and-Audio/Beats-Audio-EQ-Setting-General-Sound-Advice...
Take a look and fiddle with your settings and let me know how it goes.
Thank you,
Please click “Accept as Solution ” if you feel my post solved your issue.
Click the “Kudos Thumbs Up" on the right to say “Thanks” for helping!
Thank you,
BHK6
I work on behalf of HP

Audio compression rate

Hi,
is there a way to change the audio compression rate on my Ipod? Bass and treble sound very distorted when connected to my home stereo amplifier.
Many thanks
acer aspire 3500 Windows XP

Distortion can be caused by several things -
1) Simply overloading the input to your hi-fi or by a high data compression rate or a combination of the two. Check the input sensitivity rating of the socket you are using for the iPod on your hi-fi is around the 1 volt level (RMS). If is much less than 1V (say 300mV or less), even a lowish volume setting on the iPod could cause distortion, at least in the bass. Input overload from high bass levels can cause clipping distortion that can blow speaker tweeters (not the woofers) and sound terrible. Also, file compression at less than 128kbps causes progressively more disagreeable distortion right across the audio bandwidth.
2) Downloaded files are sometimes highly compressed (less than 68kbps) and are not worth playing from any half-decent hi-fi. Also, avoid ripping CD's at less than 128kbps if you want to play back through your hi-fi. Preferably, use a bigger bit rate than 128kbps. iTunes material should be fine, at least for the non-audiophile, as it's AAC files are not too highly compressed.
3) If you use the Sound Enhancer (iTunes preferences) or an iPod Equalizer Preset, you could push the overall output toward overload for the hi-fi input.
Happy listening.
iPod 5G video 30GB

I can't live without this one ... Monkey's Audio Compression

I'm not sure how this package hasn't been created yet. I guess none of the ArchLinux users download music of the net ... ;-)
If you don't want to create the files I uploaded the data to the FTP incoming directory: ftp://ftp.archlinux.org/incoming/monkeys_audio.tar.gz
PKGBUILD
# Author: Casey McGinty
# Date: 9/10/04
pkgname=mac
pkgver=3.99
pkgrel=1
pkgdesc="Monkey's Audio Compression is a lossless audio encoder/decoder similar to FLAC"
url="http://www.monkeysaudio.com"
depends=()
makedepends=( 'nasm' )
source=( "http://mrttt.nm.ru/$pkgname-$pkgver-linux-1.tar.bz2" "mac.patch" )
md5sums=('c35c4fb9c8ffadd88918a376c6479040' '11a01eb44031a1e403fb03f71007a913')
build() {
echo
cd $startdir/src/$pkgname-$pkgver
patch -p1 -i ../$pkgname.patch
./configure --prefix=/usr
make || return 1
make prefix=$startdir/pkg/usr install
cd src/XMMS-Plugin
make build
install -D libmacinput.so $startdir/pkg/usr/lib/xmms/Input/libmacinput.so
# vim: ft=sh
mac.patch
diff -aur mac-3.99-old/src/MACLib/Assembly/Assembly.h mac-3.99/src/MACLib/Assembly/Assembly.h
--- mac-3.99-old/src/MACLib/Assembly/Assembly.h 2004-04-30 21:22:35.000000000 -0700
+++ mac-3.99/src/MACLib/Assembly/Assembly.h 2004-09-10 22:43:41.000000000 -0700
@@ -6,7 +6,7 @@
void Adapt(short * pM, const short * pAdapt, int nDirection, int nOrder);
int CalculateDotProduct(const short * pA, const short * pB, int nOrder);
BOOL GetMMXAvailable();
+}
#endif // #ifndef APE_ASSEMBLY_H
The build is output is nasty ... but it works.

cmcginty wrote:I'm not sure how this package hasn't been created yet. I guess none of the ArchLinux users download music of the net ... ;-)
no, it takes too long to download, but i copied all my cd's with abcde using ogg - is monkey better? (and still that small?)
i'll have a look at this pkg on Tue - after my first big exam this year

Audio compression issues in iBooks Author

Hi - I am having quality issues with audio compression. I have a voice only file. It is fine in the original recording. I have saved to aiff and m4a and dropped both into iBooks widget. The widget "optimizes" it and the resulting sound is full of echo and sounds like repeats of first and last consonants of words. I even thought that my stereo channels might be out of sync, so mixed down to mono. Same sound problems in iBooks.

Found the root of the problem and a fix
1. Your audio file is at 48hz
2. Convert to 44.1 hz in Logic or any other Digital Audio Workstations
3. Export to a labeled folder with settings AIFF or Wave 16bit
4. From the folder drag and drop audio files in iBooks Author
IBook Author auto 'opitimizing' can handle that with no glitches or dropouts
[email protected] Nevis WI

Best audio compression

Hi, trying to make a DVD of a music work, but I get a slight hiss when it is transcoded, I've noticed the audio bitrate, thing 192, but there are higher settings, up to 400... if I set the bitrate higher,w ill the audio compress better..thanks, I'm a bit clueless aobut all this...

The DVD Specifications state that Stereo Dolby Digital must be at least 192kbps. You can go up to 448kbps, and expect the disc to play on all players. Going any higher than this will make the disc out of spec if this is the sole stream available.
LPCM is uncompressed. This carries a data rate of 1536kbps, as no data is thrown away by the perceptual process. The downside is that you lose bitrate available for film, but if this is a music title, LPCM is your best bet.
Simply export from Premiere as Uncompressed 16 bit 48KHz.
You may - if you wish - elect to go with 24 bit 48KHz, at a data rate of 2336kbps or similar.

Airplay Audio Compression

When not using iTunes, is audio compressed when streamed over Airplay Display or when choosing Apple TV as the audio output device?

Ok, but this doeasn't work for all my streaming services, I'm streaming songs with spotify and sports with different streaming services. My other options is to use a looong hdmi cable, would be nice to do it wireless. And airplay works most of the time, even for hd streaming video.

Unwanted audio compression

Can anyone help?!!
I'm trying to output a movie to tape but it keeps compressing the audio. Louder passages keep being knocked down. I don't understand why as I have not added an audio compression filter. It's not compressed when I play it in the timeline. Any ideas anyone??
Mac Pro Mac OS X (10.4.9)

and a teensy bit more:
Your problem could have just as much to do with the "soundsticks". I just looked them up. They are small speakers intended to be used as computer speakers. They in no way represent or emulate a studio monitor that one might use to make audio decisions. Even if they sound great... in fact especially if they sound great. If they make "improvements" on your sound, then those are improvements your listeners aren't going to hear.
This should pretty much make the case: "SoundSticks II employs a new computer-optimized multi-band parametric equalization."
So the problems you're having could be originating with the prejudices of those epeakers affecting your audio processing decisions, which in turn affect what comes out the other end.
To have it come out sounding like "compression" is a little odd, but it may just have to do with your relative perception of the sound coming from these different speakers.
The best way to judge audio is in a clean soft neutral environment, with good nearfield monitors.In addition (and not simultaneously) you should be able to hear some representation of how your mix is going to sound on a little 3" speaker in an average TV set. One way to do that is to play it through your TV set.

Audio compression in FCE with AUPeakLimiter and AUDynamicsProcessor

I recently edited a video with poorly recorded audio, and learn some useful things about FCE that I want to share. Hope it will help someone else. I also posted this info, with pictures, here: http://members.dslextreme.com/users/craig.lawson/fce/audio_compression.html
Although the video was shot in a studio, the audio levels ranged from nearly 0 dB to down below 36 dB. Some of the talent were experienced and projected their voices, others were inexperienced, young kids, who did not. The audio system had intermittent ground-loop hum. And one of the XLR cables gave occasional bursts of static, perhaps when someone trod on it. And then there was the rare full-scale pop. A real mess.
The destination was broadcast (community) TV. The station's audio guidelines are, "at audible levels without distortion". Not much of a guideline, so I used the guidelines in the FCE manual and set as my goal -12 dB to -24 dB (I know, the recommendation is a 6 dB range, not 12 dB, but 6 dB sounded bad; maybe they use a compressor). In other words, I want to decrease my dynamic range by 24 dB (from 36 dB to 12 dB), with a maximum at 12 dB. How to achieve this in FCE?
Several people have written about Compressor/Limiter (and it's in the manual), but I was not satisfied with the result. Instead, I used AUPeakLimiter and AUDynamicsProcessor and it worked well. These are the settings I used and why.
*Component 1: Audio Filters > Apple > AUPeakLimiter*
A Peak Limiter prevents audio signals from clipping (in FCE, that means exceeding 0 dB). It does this by turning down the volume at the loud parts, and leaving it alone otherwise. The middle of this page has useful graphs which explain peak limiting: http://www.bluehaze.com.au/unix/level.html
Settings
*Attack: 0.01, units: seconds.* This tells AUPeakLimiter to respond to over-limit peaks within 10 msec. I chose the maximum setting for a more gradual volume changes (not sure if that is best - it may cause sudden peaks to go over limit).
*Release: 0.04, units: seconds.* When the peak has passed, return volume to normal within 40 msec. I chose the maximum setting for a more gradual volume change.
*Pre-gain: 20, units: dB.* Increase the input by 20 dB before doing the peak limiting. This makes the quiet sounds louder, and pushes the loud sounds will above the clipping limit (0 dB). But this is PRE-gain, and after applying this gain the peak limiter brings the signals above 0 dB back down. End result: my quiet sounds are louder, and my very loud sounds and perhaps only a little louder. And no clipping. This setting controls the lower value of the filter chain's dynamic range.
*Limiting amount.* FCE shows this as a setting, but in Garage Band it is a read-out. In Garage Band, it displays the amount of limiting currently being applied. I think that for FCE to display this as a setting is a program bug. Also, adjusting it doesn't do anything.
Result at this stage: all audio is loud, including the noise floor. In other words, it's like a compressor with a hard knee and low headroom.
*Component 2: Audio Filters > Apple > AUDynamicsProcessor*
A Dynamics Processor is a Compressor and Expander in one unit. The compressor decreases the dynamic range of the loud sounds, and the Expander increases the dynamic range of the quiet sounds, making them quieter. More on compression here: http://en.wikipedia.org/wiki/Dynamicrangecompression
*Compression threshold: -40, units: dB.* With AUDynamicsProcessor, the compression does not begin at the threshold, but instead begins about halfway to 0 dB. With this setting, the knee is around -20 dB. The Compressor appears to have a "soft knee" (see Wikipedia), but the softness is not adjustable.
*Headroom: 4, units: dB.* The headroom setting adjusts the amount of compression. Lower headroom values result in higher compression. Think of it as the amount of allowed dynamic range above the knee. This setting controls how much peaks may exceed the filter chain's dynamic range: set headroom low if you really don't want the output to exceed your target dynamic range, and set it higher if occasional peaks above your limit is OK.
*Expansion threshold: -60, units: dB.* Sounds below this threshold are expanded. Unlike the compression threshold setting, this sets the position of the knee directly. This setting, combined with expansion ratio, reduce background noise during quiet segments.
*Expansion ratio: 2.* How much expansion to apply to sounds below the threshold. Example: with threshold at -60 dB and ratio at 2, sounds levels at -70 dB (10 dB below the threshold), and pushed down to -80 dB (20 dB below the threshold).
*Attack time: 0.01, units: seconds.* Same as for AUPeakLimiter.
*Release time: 0.100, units: seconds.* Same as for AUPeakLimiter.
*Master gain: -12, units: dB.* This applies post-gain. This setting sets the upper value of the filter chain's dynamic range.
*Comp amount.* FCE shows this as a setting, but in Garage Band it is a read-out. In Garage Band, it displays the amount of compression currently being applied. I think that for FCE to display this as a setting is a program bug. Also, adjusting it doesn't do anything.
Summary
This filter chain effectively compressed my dynamic range, and pushed down my noise floor during quiet segments. Unfortunately, when the talent was talking quietly, the noise became apparent, but to address that requires an adaptive noise reduction filter (e.g. Sound Soap).
The dynamic range mapping created by this filter chain is not so great. Because pre-gain of AUPeakLimiter is so high, it makes the compression ratio infinite above the knee. Then AUDynamicsProcessor lowers the knee and compresses again. A better mapping would use something less than an infinite ratio, and that can be accomplished by putting AUDynamicsProcessor first in the chain. But the resulting filter chain would have more than 2 components, and FCE won't play it in real time (on my system).
If you want the effect I've described, the settings above will get you started. To tune your settings:
1. Disable AUDynamicsProcessor, turn your speaker volume way down, and then play with the AUPeakLimiter's pre-gain until your audio falls within your chosen minimum range on the audio meter. Pay attention to the quiet sounds, and AUPeakLimiter will take care of the loud ones.
2. When your lower range looks good, turn on AUDynamicsProcessor, set master gain, and then play with the compression threshold and headroom. Pay attention to the loud sounds.
3. Finally adjust the expander settings to quiet the noise floor.
Once you have filter settings you like, apply them to your top-most sequence.
Message was edited by: c555

Thanks Sasha for the reply, The Audio Rate on my files are 48KHz and Higher, when I mention the 16 bit I was talking about the Audio Format. The files I have are 16 bit integer where my sequence says it 32 bit floating point Audio Format (not really sure what that means). Would you know why the audio would be perfect on my laptop and not on the pc that is connected to the TV? Could it be because of the Audio Format? because I don't see what else it could be... :/
Message was edited by: Doodfsanfkldsjabfsda

Audio compress rate

Hello!
Does anyone know the tech-specs for airport express (latest ) with regard to at which rate audio files are compressed? What Im trying to establish is if the airport express is able to stream 24/192 Khz audiofiles or if it compresses audiofiles to some other bit-rate.
Thanks
Per

Hi sercher,
You'll need to record the tapes into MP3 format before you put them into your MP3 player. The way I've done this with my tapes in the past is to connect my old Walkman cassette player to my PC's sound card with the appropriate cable. Then I've used TotalRecorder or similar software to record it in MP3 format. It is all at li've speed: I hit "Play" on the tape player and "Record" in TotalRecorder, and come back to the PC 40 minutes later to stop things and flip the tape when it runs out.
It is a bit slow and tedious. I don't think there would be a faster way unless you could find some special hardware. A cassette player's going to play the cassette at regular speed, and that's the limiting factor.

Sorry in advance - Audio Compression

I am sure there is a posting somewhere. I think I am not putting the right keywords in.
I am looking for a good tutorial or some advice on peak limiting and compressing audio. Possible one that uses the filters that come with final cut or passing through Soundtrack. I have always had trouble getting the peeks right. Anybody have any advice? Thank you in advance

True story about everything in broadcast being compressed. At the very least there's a peak limiter at the end of the chain preventing overmodulation.
As far as compression in FCS, I always skip Final Cut Pro's Compressors as there is no metering. Also all the controls are linear rather than logarithmic. Not a big deal with the plain dynamics processor, but with the multiband version it is frustrating. Note to Apple: please fix this. I don't need to pick between 14,578Hz and 14,724Hz, but I do need to pick between 120Hz and 150Hz.
In Soundtrack I find the Compressor useful for vocals, the Multipressor useful for anything more complex (music, sfx), and if I'm really trying to hit it hard the Adaptive Limiter on subgroups or the main mix can get you right up to 0dB.
If you're really wanting to smash your signal be aware Final Cut Studio isn't the tool for the job. Most video applications don't call for the signal to be right up next to 0dB, but to have some headroom. I usually have my vocals center at about -12dB, and try to keep the peaks under -6dB. I'll even limit the vox at -3dB if it's an extremely dynamic person. If you're working on a CD project and you need to push it up to 0dB the Multipressor and Adaptive Limiter would be your best friends in FCS, but use software meant for audio rather than video if that's what is going on.

[SOLVED] Audio Compression Quality

I would be interested in views about which compression algorithm (MP3, OGG, AAC, etc) gives the best reproduction quality for a given bitrate (e.g 160kB/s).
Thanks in advance.
Last edited by myrlin (2013-10-14 19:48:58)

That one is easy: ogg vorbis is the best audio codec. Some might say opus is better. Well yes it is, but just kinda better. Why do I still prefer and recommend vorbis? Because opus only supports 48kHz. This results in an upsampling of most audio material, which usually comes in 44.1kHz. For lower bitrates (I'd say \leq 128kbps) opus of course beats vorbis, but with 160kbps (or -q 5 in oggenc) I'd say vorbis is the best for music.

Video/audio compression software

Hi,
I posted recently about audo compression software and
received 'Audacity' as a solution.
Now our needs have changed. We expect to be receiving videos
in a range of formats and from Macs and PCs.
Can anyone suggest suitable software to convert/compress both
audio and video for this task?
Thanks.

I've downloaded the SUPER product (after finding the proper
link on the
horrible website) but haven't played with it yet. This link
is to a
writeup on it instead of the product's home page, which is
horrible but
you should be able to eventually work your way through.
http://www.videohelp.com/tools/SUPER
HTH
Erik
ploughman1 wrote:
> Hi,
> I posted recently about audo compression software and
received 'Audacity' as
> a solution.
> Now our needs have changed. We expect to be receiving
videos in a range of
> formats and from Macs and PCs.
> Can anyone suggest suitable software to convert/compress
both audio and video
> for this task?
>
> Thanks.
>

Audio Compression

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