Intermittent Audio Compression

Shot with a Canon camera and converted the clips to Pro-res. The clips play fine on their own but when put into a project timeline there are intermittent audio issues throughout the clips - the closest I can describe it is heavy compression, eg. a CD is ripped at 192kbps and then suddenly the audio drops down to 24kbps. It continues on and off for the whole 72min. and then the last 10 minutes are perfectly fine. Any help would be really appreciated!!

Derdav - different cameras use different codecs so to begin with, please name the exact Canon camera, is it a DSLR or a camcorder or a Cinema camera such as a C300, etc. What codec?
Also, upon importing did you select to seperate mono, or combine stereo, etc.? When you have the clip in your storyline, if you select it and look in the inspector (command 4 to show inspector if it isn't showing) under the audio tab, what is it showing?
To take a shot at ideas, which is not the most efficient way to start, I'm wondering if you recorded a line feed from a source, like a mixer into one or both channels, or if you're talking about a camcorder and just on-board camera mic? Phasing issues can cause an effect like you're describing, or really I have no idea - need some help with more specific info.
Thanks for anything else you can provide about your setup.

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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.

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

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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?)
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Audio compression issues in iBooks Author

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Found the root of the problem and a fix
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I have noticed intermittant audio stream since I have had my computer. Whenever, I watch a video track or and audio track such as Apple Tips of the Week, the audio comes out broken and interrupted, not smooth audio or the voice output does not match up with the indivivdual speaking the words. I always thought this may had to do with the internal speed of the processor. I have been dealing with this so long that I finally decided to ask the Discussion Group about it.
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I recently started having intermittent audio cut off on my IMac G5 which is 6 months old. All audio to inclue iTunes stops for no apparent reason. This even happens when I am just listening to iTunes and performing no other tasks. Sometimes requires multiple reboots to fix. I checked audio output setting and it's 44100.0 Hz as recommended. Sometimes audio will restart without a reboot. Machine has 1.5G RAM and I do not use it for anything exotic or heavy duty.
Am using a USB hub with printer, scanner and external hard drive attached. Tried disconnecting all, but problem still recurs.
Thanks in advance for any help.
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If you plug in headphones into the headphones port do the audio problems persist? I'd like to know if the issue seems to be isolated to the speakers, or to the soundchip and/or sound software.
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http://discussions.apple.com/category.jspa?categoryID=134

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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.
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Can anyone help?!!
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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

TS3623 AppleTV 2nd gen intermittent audio content out.

My second generation AppleTV plays all content great to my Toshiba LED TV, but content out to the soundbar is intermittent. iTune movie "Let Me In" doesn't but "The Replacements" works perfect and so on. Same thing with Netflicks. But my Blueray player using netflicks works perfect on every tittle from netflicks. I have tested all four HDMI ports with the same result. The AppleTV is dissapointing at this point. Any Ideas? Oh, the sound bar is the Polk Audio 3000.
Dave

bothfly wrote:
I have two HDTV's bought in the last 2 years and see no option on either to upscale.
Your HDTVs panels will be fixed resolution - if the input signal differs the TVs have to up/downscale accordingly.
AppleTV 1 allows you to set resolution from 480p/576p through to 1080i/p under Settings.
If your TV is 1080p and you output 720p from AppleTV, the TV upscales automatically.
If the TV has a 720p panel, but accepts a 1080p input from an AppleTV 1 it will downscale.
For what it's worth I much prefer my AppleTV 1's. I too feel there's something subjectively worse about video quality on 2, menus in particular look rough edged/pixellated compared to 1- the output may just be sharper (or non anti-aliased menus) than gen 1.
AppleTV2 definitely buffers/navigates faster and with more responsiveness, but the interface is not as polished as ATV1 and some features are still missing compared to 1 - early days though for ATV2, so I hope software updates will remedy some of these. Having recently read something about possible ATV2 revision with better CPU however, I wonder if it will have the same lifespan software upgrade wise as gen 1 did, though being iOS based this may help apart from hardware dependent things.
AC
The point was that I found the ATV1 to be better than the ATV2, IMO of course.

Intermittent Audio Compression

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