I just bought Logic Studio so expect more posts on this in the future. Upon trying to install the application, I had the problem many people have been experiencing - Compressor is grayed out and refuses to install. I also noticed that upon completing the installation, Soundtrack Pro would crash when launching, citing a missing framework (ProFX.framework) as the cause. I don't know what causes this as I don't have Final Cut Studio installed on my Logic machine to conflict with it, and there were no receipts related to Compressor on my system. However, there is a simple workaround.
1. Install all the other components and let the installer skip Compressor for the time being.
2. On the main Logic installer disc, ctrl-click on Install Logic Studio
and select Show Original
3. Click on the Packages
4. Double-click Compressor.mpkg
and complete the installation process.
5. If you also got the same error as me about the missing ProFX framework when launching Soundtrack Pro, double-click the file PluginManager.pkg
and follow the installer.
6. I was then able to launch Compressor but upon performing a test render, I noticed that Batch Monitor was missing (why, I don't know). So if you get that problem too, install Qmaster.mpkg
This finally solved all of my problems and I was able to operate the Studio fully.
P.S. In case you were wondering, the difference between .pkg and .mpkg is that .mpkg (meta-packages) are containers for sub-packages (.pkg). The .pkg files only install one set of files whereas the .mpkg files can install a group of related packages.
Posted by Jon Chappell on Sunday October 19 2008 5:10 PM to Apple, Software, Sound
This is the second part of the Final Cut Pro Audio Filter Guide. Part 1 covered the Audio Units that ship with Mac OS X and is available here
. This part will cover the audio filters that ship as part of Final Cut Pro.
This guide is intended to inform Final Cut Pro users of the options available to them for fixing bad sound. It is worth mentioning at this point that this guide is intended for users of Final Cut Pro 6.
Now, one thing you may have noticed is that there are often duplicate filters such as AUBandpass (an Audio Unit) and the built-in Band Pass Filter. So which should you use? Well, for a start, bear in mind that some of those filters have the same function but different parameters. I would generally advise you to pick the built-in filters over the Audio Units whenever possible, as they tend to use up fewer system resources.
3 Band Equalizer
This allows you to take three separate bands (low, med and high) and adjust the gain up and down individually. I personally prefer AUFilter (in the Apple folder) because it gives you a bit more control and allows you to adjust 5 bands.
This is useful when you need to boost the bass of a voice or improve a flat-sounding voice.
Band Pass Filter
This will allow a range of frequencies on either side of the center frequency to pass through and reduce (attenuate) frequencies outside this range.
This is identical to the AUBandpass filter (in the Apple folder) except that instead of the bandwidth parameter, it has a mysterious one called Q. Q stands for Quality Factor and is a different way of representing the bandwidth. There are numerous articles about it on the internet that get quite technical but all you need to know is that it's the relationship between the center frequency and the bandwidth (f/b) so Q is inversely proportional to the bandwidth (i.e. when the bandwidth goes up, Q goes down by a proportional amount). It is worth noting that Q is not the bandwidth itself but it is related to it. If you want to find out the bandwidth, just divide the center frequency by Q.
If you don't understand the above explanation just play it by ear or use the AUBandpass filter.
The Compressor / Limiter reduces the volume of sounds above the threshold amplitude. This is a useful way of minimizing the difference between two subjects talking at different volumes or making sure that the audio fits within the limits of the playback device.
Obviously this reduces the volume of the overall audio so Preserve Volume rectifies this (although I find it is often then too loud). Attack time refers to the time it takes for the filter to decrease the volume once it has detected a frequency with an amplitude above its threshold. Release time refers to the time taken for the filter to increase the volume again once the high amplitude frequency has finished playing. Higher values allow for a smoother and less noticeable response but set them too high and the compressor won't respond quickly enough. This is something inherited from live audio mixing where you don't know what's coming next. Setting the threshold to just under your preferred limit allows time for the compressor to lower the volume in anticipation for a louder sound once the threshold is reached.
Ratio tells the compressor by how much it should reduce the volume when a sound exceeds the threshold. If the ratio is set to 2 (2:1), then a 10 dB increase in volume above the threshold will be halved to a 5 dB increase. It is worth noting that the compressor lowers the volume of sounds above the threshold but does not necessarily reduce them to a value at or below the threshold. Be aware that very loud sounds could still theoretically peak.
Finally, it is also worth mentioning that the compressor reduces the volume difference between the subject and any background noise, so background noise will be more noticeable upon boosting the audio after applying the compressor.
Sometimes you may experience a DC current leakage through the mic, causing noise in the recorded audio. A DC Notch filter will remove the DC offset component which you probably need a degree in audio engineering to fully understand. It has no parameters.
This is one of the lesser-used filters FCP provides.
This one obviously adds an echo. Effect Mix allows you to mix the echoing audio with the original in order to better blend it in. Effect Level controls the volume level of the echoes (but not the original audio). Brightness controls the degree that the echoes will overlap. Feedback controls how long each echo will last and delay controls the spacing between each repeat.
It is worth noting that the echoes will be abruptly cut off unless you lengthen the audio clip or fade it out.
Expander / Noise Gate
The opposite of a compressor. A compressor takes high amplitude sounds and lowers them whereas an expander takes low amplitude sounds (sounds below the threshold) and lowers them. I know it probably seems strange to lower sounds that are already quieter than the rest of the audio but this is designed to increase the dynamic range of the clip.
Threshold is the amplitude below which the expander will kick in. Ratio controls the proportion of expansion - for example with a ratio of 2 (2:1), a 3 dB fall below the threshold will be adjusted to a 6 dB fall. Attack time is the time for the volume change to be applied once it falls below the threshold. Release time is the time for the volume to return to normal once the signal goes above the threshold again.
A noise gate is a more extreme expander that will completely eliminate frequencies below the amplitude threshold. This can be achieved with high ratios (e.g. 10).
This is a simple filter that raises or lowers the volume of the audio. It is added automatically to clips when you use the Modify > Audio > Apply Normalization Gain command.
High Pass Filter
This will attenuate (reduce) low frequencies below the threshold. This is the same as a Low Shelf Filter. Use the Q slider to modify the bandwidth (width of the frequency range).
Useful for cutting out low frequency noise such as the rumbling of traffic or very low notes in a deep voice.
High Shelf Filter
This will attenuate (reduce) high frequencies above the threshold. This is the same as a Low Pass Filter. Gain allows you to adjust the volume of audio that passes through the filter.
Useful for filtering out high frequency noise such as buzzing on the soundtrack.
This is similar to the Shelf / Pass filters but it has several extra parameters. Frequency, Q and Gain have been explained many times above. In order to explain what the harmonic check boxes mean, we need to delve into a little audio theory.
The fundamental frequency is the lowest frequency in a harmonic series (the Frequency value in this case). Harmonics are integer multiples of the fundamental frequency - e.g. if f=60, 2f=120, 3f=180, etc. These play at the same time as the fundamental frequency and contribute to the tone of a sound. The Hum Remover is more powerful than a Shelf / Pass filter because it allows you to remove these specific frequencies without removing any frequencies in between these. For example, if f=60 and you wanted all harmonics up to 5f removed, a Low Shelf or High Pass filter would remove ALL frequencies up to 300 Hz, potentially affecting the quality of your sound. The Hum Remover would not do this.
Despite being called Hum Remover, the use of harmonics makes it useful for other purposes such as enhancing or reducing a musical instrument on a soundtrack.
Low Pass Filter
The Low Pass Filter has the same effect as the High Shelf Filter - it will attenuate (reduce) frequencies higher than the specified frequency range, keeping lower ones intact.
Frequency refers to the center frequency and Q is a way of representing the bandwidth (the width of the frequency range).
Useful for reducing high frequency noise such as buzzing.
Low Shelf Filter
The Low Shelf Filter has the same effect as the High Pass Filter - it will attenuate (reduce) frequencies lower than the specified frequency range, keeping higher ones intact.
Useful for reducing low frequency noise such as air conditioner hums.
The opposite of a Band Pass filter. Instead of only allowing frequencies within a certain range, this cuts out frequencies within a certain range.
Frequency refers to the center frequency and Q is a way of representing the bandwidth (the width of the frequency range).
Useful if you have noise of a constant frequency on your soundtrack (such as a buzzing sound).
This is a combination of Band Pass, Notch and Shelf filters combined into in a single filter. Frequency is the center frequency, Q is related to the bandwidth (see the explanation above) and gain allows you to boost or cut the frequencies passing through the filter.
This is similar to the Echo filter but considerably more sophisticated. Rather than simply repeating sounds with a delay, it allows you to mimic the characteristics of echoes within various locations. This is incredibly useful when performing ADR (Automated Dialogue Replacement) because it is highly likely that the sound booth you record the ADR in will sound nothing like the original location. This allows you to mimic the effect of sound waves bouncing off walls, with some canceling each other out and some increasing in intensity. It can also be used sparingly to improve a flat-sounding voice.
Subtlety is often the key with this filter and Effect Mix allows you to mix the reverb with the original sound to help blend it in. Effect Level controls the intensity, Brightness controls the degree that the echoes overlap and Type allows you to specify various preset locations.
It is worth mentioning that the reverberation will end abruptly unless you extend your audio clip or fade it out at the end.
This helps to reduce the intensity of "s" sounds, most noticeable if the actor has a lisp. The controls are similar to a compressor but it is optimized for reducing sibilant ("s") sounds. Ratio controls the amount of reduction - e.g. if the "s" sound is 6 dB and the ratio is 2, it will be reduced to 2 dB.
Emphasis controls the sensitivity of the filter and Broad Band Mode widens the bandwidth so that more frequencies around the center frequency are affected.
Sometimes if a microphone is directly in front of an actor's mouth they will accidentally breathe into it while speaking, causing a wind-type noise to be generated. This filter aims to minimize these.
The parameters are similar to a compressor - ratio controls the ratio of reduction proportional to the intensity of the sound. Broad Band Mode widens the bandwidth so that more frequencies around the center frequency are affected.
One Final Note
These filters do a decent job of repairing troublesome audio but they are not miracle cures. Sometimes (and this is never popular with producers) it just isn't cheaper to fix it in post. Even big Hollywood movies with access to multi-million dollar sound studios re-record a lot of their audio. It is important to always think realistically.
However, I hope this two-part guide has been useful in showing you just what can be improved. I noticed a significant improvement in the sound quality of my projects once I understood more about the audio filters available to me, and I hope you'll be able to say the same.
Posted by Jon Chappell on Sunday October 5 2008 10:12 AM to Final Cut Studio, Sound, Analysis
In my conversations with other editors it became clear that for a lot of them (myself included), audio was a major area of weakness. They could handle editing, color correction, graphics, etc, but only knew a few ways of improving bad audio. This inspired me to create this guide to show people exactly what Final Cut Pro offers out of the box.
Filters in FCP are split into two categories - Apple Audio Units (AU) which ship with the operating system and are available to any application, and FCP-specific ones. Today I'll be giving details on the Audio Units with the FCP-specific ones coming in Part 2.
The bandpass filter allows a certain range of frequencies to pass through and rejects any frequencies that are outside of that range. So if you have a person talking with rumbling traffic and high-pitched TV static in the background, you could use the bandpass filter to isolate the center frequencies of the person's voice.
This filter allows you to adjust the dynamic range of the audio (the difference between the loudest and softest sounds). Compression lowers the volume when it exceeds a certain threshold (making loud sounds quieter and therefore decreasing dynamic range). Expansion lowers the volume when it is lower than the threshold (thus making quiet sounds quieter and therefore increasing dynamic range).
Using a compressor and expander at the same time allows you to avoid overly-loud peaks while still maintaining dynamic range. The compressor will reduce the louder portions of the audio in proportion to their distance from the threshold (i.e. louder sounds will be reduced by a much larger amount), helping to even out amplitude variations. The expander will then take the quieter sounds and reduce their volume (again, proportionally), thus evening out the amplitude variation of the quieter sounds and increasing the dynamic range.
Imagine some music where you have loud crashing drums, a relatively quiet triangle and a trumpet somewhere in between. The drums are far too loud so you apply a compressor to bring them down. The drums are now a lot quieter but you've noticed that the subtlety of the triangle has been lost - it is now far too prominent. So you apply an expander to lower the volume of the triangle to an acceptable level. The AUDynamicsProcessor allows you to do this with one filter instead of two, which is a much more efficient use of system resources.
Attack time refers to the amount of time it takes for the compressor to implement a change in volume and release time is the time taken to reduce the compressor back to its original level afterwards.
There's a bit of debate over what "headroom" actually means as it's not a standard term. To the best of my knowledge, it refers to the number of dB the signal is permitted to exceed the threshold of the compressor/limiter.
The Dynamics Processor is a great way of compressing and/or expanding your audio to make sure that it fits within the audible range of your listening equipment.
This repeats parts of your audio. Dry/wet mix controls whether or not the repeated sounds overlap each other. 100% dry means no overlapping and 100% wet means full overlapping. Delay time is the delay before the repeat starts. Feedback controls the number of repetitions. Lowpass cutoff frequency stops certain low sounds from passing through the filter.
This filter is normally used for atmospheric effects. For more environmentally realistic results (at the cost of performance), I recommend the Reverberation filter that comes with Final Cut Pro.
AUDistortion (Leopard only)
This is a very comprehensive distortion filter offering a lot of control. I can't give much advice on the various parameters as I don't actually know what they mean and there's no documentation available. I would suggest adjusting by ear (it is probably worthwhile lowering the render quality while adjusting, and then raising it again once you're happy with the results).
I use this one a lot. It allows you to take 5 bands (5 ranges of frequencies) and adjust the gain up or down for each one. If a person's voice is muffled, increasing the gain on the higher frequencies can often improve this. I also sometimes use it to add bass to make a person's voice sound richer.
The bandwidth slider allows you to adjust the size of each frequency range, with the frequency sliders referring to the center of the range.
AUGraphicEQ(image truncated for space reasons)
This is similar to the AUFilter above but the size of the frequency ranges cannot be adjusted and you can have up to 31 bands instead of just 5. This site
provides a guide to common frequency ranges.
I've used this a couple of times to boost bass in a person's voice but I generally prefer AUFilter for this. It's good for removing or reducing a specific frequency range.
This cuts off frequencies above the cutoff frequency, allowing lower sounds to pass through. The gain slider controls the amount that the frequencies passing through will be boosted.
Good for reducing high-frequency noise.
This reduces lower frequencies and allows higher ones to pass through. Resonance controls the intensification of the higher frequencies that pass through.
Good for removing low rumbling sounds such as traffic or for reducing very deep voices (I have had to do this at times).
Reduces (attenuates) higher frequencies and lets lower ones through, controlled by the cutoff frequency. Resonance controls the intensification of the frequencies that pass through.
Good for removing high-pitched noise.
Reduces frequencies lower than the cutoff frequency and lets higher ones pass through. This is similar to the AUHighPass filter above but this offers the ability to adjust the gain instead of resonance.
Good for removing low hums.
This allows you to compress multiple bands (frequency ranges) individually for more control than a traditional compressor.
Pre-gain boosts the signal before it is processed which is useful if the signal is too low to be processed adequately by the compressor, and post-gain reduces the gain back to a normal level afterwards.
Crossovers 1, 2 and 3 define the point at which the previous band ends and the next one begins. Threshold refers to the amplitude level at which the compressor kicks in. Headroom, as stated above, probably gives you extra leeway above the threshold. Eq allows you to boost or lower each band.
Possible uses for the Multi Band Compressor include lowering the dynamic range of bass sounds without interfering with higher frequency sounds. This is different from the High Shelf Filter which indiscriminately cuts all bass below a certain frequency.
AUNetSend is one of the most interesting filters in that it's not actually a filter. It does absolutely nothing to affect the way your audio sounds but what it does do is allow you to send audio across a network.
You need to have an application that implements audio generators. Audio editing applications such as Logic implement these. Alternatively, there is a tool called AU Lab in /Developer/Applications/Audio if you have the developer tools installed. Soundtrack Pro does not support this. Add an AUNetReceive generator to the track in your audio application, add an AUNetSend filter to your FCP audio clip and hit play in FCP. You should see "AUNetSend" pop up in the AUNetReceive configuration dialog. Select it and your track should be receiving the audio from Final Cut Pro. Note that it only appears while the timeline is playing in FCP.
In my testing, the Status parameter appeared to do absolutely nothing.
This allows you to boost or lower the amplitude of a signal within a certain range. This is useful if for example you have a high-pitched buzzing noise in the background and you only want to eliminate that particular frequency and keep your higher sounds intact.
The Peak Limiter differs from a compressor in that a compressor reduces the volume of an entire track when a frequency reaches a certain level whereas the Peak Limiter reduces just that frequency. This is particularly useful if there is background noise on the track that would produce a noticeable fluctuation if the entire track were to receive a volume adjustment.
Attack and release time, as mentioned above, control the amount of time it takes for the filter to implement a change in amplitude, with longer times allowing a smoother transition. Pre-gain allows you to boost the volume before it reaches the filter in order to change the number of frequencies being affected. The limiting amount allows you to limit the amount that the filter will reduce the amplitude.
Adjusts the pitch of your audio, obviously. There are a LOT of controls though, and I have to admit that I don't have a clue what a Glb Trigger Thresh or a Loud Aggr K is. Effect blend blends the pitch-shifted audio with the original and is sometimes necessary to make voices sound natural. It is worth mentioning that I've gotten perfectly acceptable results by adjusting the first two parameters and not bothering with any of the others. I'd imagine that most people wouldn't have to adjust more than these.
I tried a Google search on some of these parameters but they only turned up forum threads where people were asking what on Earth they meant and no-one was able to solve their problem. If anyone does know, I'd be interested to hear from them.
This can be useful for making a male actor's voice more masculine (yes, I have had to do this!).
AURogerBeep (Leopard only)
This emulates the "roger beep" sound when someone lets go of the button on a walkie talkie. It will automatically play the sound when the audio level drops below a certain threshold for a certain amount of time (as if the person stopped speaking).
In gate threshold and in gate time refer to the time that the threshold amplitude must be maintained before the sound kicks in. Out gate threshold and out gate time refer to the amount of time that the threshold must be maintained before it ends the roger beep (adjust this if there is background noise). Roger level is the volume of the roger beep, and I think sensitivity and roger type are self-explanatory.
This is an effects filter with quite a narrow purpose so it's not something that gets used very often.
Similar to AUDelay except that the delay time is set as a number of audio samples instead of a number of seconds.
Two of the above filters are Leopard only. If you use those filters and then transfer your project to a Tiger machine, you will receive an error message and will be unable to use those filters within the project.Part 2
covers the filters built into FCP.
Posted by Jon Chappell on Saturday September 27 2008 2:31 PM to Apple, Final Cut Studio, Sound