Is Chris 1.2.6 still the best?

Executive Summary:
I’m hoping you processing experts can provide some timely advise.
Is Chris’s (sp?) Compressor 1.2.6 still the best among the options?
Are there any other plugins that I should explore for my purposes?

I’m processing lots of spoken word audio for audiobooks. I use the 1.2.6 version almost daily, and overall I’m happy with the results (although I have to chop off the beginning/end where that compressor adds some noise.)

When researching compression and tools, that product is the most recommended I’ve seen among the Audacity crowd.

I also read Chris passed a few years ago, and I found the site where it now lives (including the 1.2.7 beta version, which I do NOT use).

I found a de-Esser I like better than Spitfish (which is the most recommended.)

Because software tends to improve regularly, I’m wondering if anybody else has found an upgrade or something better than Chris’s. It’s unusual something that old hasn’t been upgraded by someone. I’m also not using the stock EQ options, as I like the KarmaFX version better. (I’m always looking for better tools, so if you know of something better, I’m all ears.)

Do you still feel that is the best compressor among the options? (assuming my application: spoken word, no music.)
Any advantages to the built-in version?

I realize I could play around with some of them, but wondering if anybody has any thoughts.

My needs are simple: Making quality audio books and compression is standard among the processing for that audio.

All feedback welcomed!

PS: I’ve been blown away at how helpful this community is.

Chris’s 1.2.6 dynamic compressor is still very good for spoken word and preferred by many to other compressors for this job.
The 1.2.7 version was experimental and has some unresolved issues.

I don’t agree that “new” is always “better” (I know you didn’t actually say that it was :wink:) The “Amplify” effect in Audacity is virtually unchanged from the version in Audacity 1.0.0 because it does exactly what it should do and there is no need to change it. When something works really well, “New” and “Improved” is often no more than a marketing ploy (or a cheaper to make version).

It would certainly be worth considering if Chris’ Compressor couldn’t be improved such that the beginnings and endings do not stand out the way they do now.
It may be sufficient to add some silence and to trim the sound after processing.

I appreciate your input! (Nice to have power users like you provide some perspective.)

I agree new isn’t always better, but as a former software developer myself, I find over time most software can be improved IF someone has time/budget and desire. Sometimes it’s making it faster, sometimes because the hardware is so much stronger these days, some options are available to solve complex issues that wouldn’t have been realistic when this software was created.

Compression with look-ahead seems like a very complex issue to me, so it won’t surprise me if someone eventually finds a few tweaks.

I want to be sure someone hadn’t found another alternative they like better. I’m far from an authority on what is available as plugins these days.

That said, the current one has the minor flaw that it adds noise to the beginning/end of my tracks, and I just snip it out, no big deal. In theory someone may have taken his code base and figured out a way to remove that one minor issue OR found another add-in that works as well, but eliminates that final couple steps.

It’s not that I think this is bad or missing much, but I don’t want to assume since I haven’t done much research on this topic. (If I could cut a step or two, it’s a win for me… I’m processing lots of audio these days, with more arriving regularly.)

Nice to know that experienced users like you still prefer that compressor.

If anybody has other input (or other options they like as well or better,) I’d be interested in hearing about it.

A workaround is to add a few seconds of some typical audio (copy n paste) to the beginning and end of your recording , then apply Chris’s Compressor, then remove the bookends. Alternatively use the envelope tool to reshape the ends to reduce the hiss/loudness at the beginning and end which can occur with Chris’s compressor.

Yes, I tend to add a few seconds to the beginning/end then cut them out. But that does get old as I already have a set of steps and I process lots of individual files. It’s not something I can automate with ‘Chains’ from what I can tell (I can’t see how to have a Chain select things, either select all, or select X seconds, etc. BUT that should be a different thread.)

My gut tells me the issue is NOT an easy fix, or Chris would have done it as it seems to be a widely known issue. (Alternately, if it was easy, I’m guessing someone else would have figured it out in the few years it’s been out and he’s been gone.)

That said, someday I’m hoping to find a plugin that works as well, but doesn’t add an extra set of steps. I used to write code for a living, but it’s been too many years and I know just enough to be dangerous in terms of the mathematics related to sound.

I also suspect some of the commercial plug-ins (paid) can match the Chris version, BUT totally a WAG on my part. That’s part of why I’m asking here, if someone else has compared I’d love to hear their experiences. (If I find something I’ll report back too…)

I am VERY impressed with this community. I do appreciate your input and suggestions!


I don’t have Chris’s compressor installed at the moment, but are you sure this is not a user error?

Chris’s dynamic compressor is intended to even out the “dynamics” (loud and soft), so that loud sounds become relatively quieter and quiet sounds become relatively louder.
Your description sounds like you have very quiet audio at the start of the tracks, which becomes louder when you apply the dynamic compressor, but that is exactly what a dynamic compressor is supposed to do. In order to prevent “silence” (background noise level) from being made louder you need to set the “Noise floor” to a higher level than the background “silence” level. Try raising the noise floor control and see if that fixes the problem (I’m not sure without looking if that will be moving the slider to the left or right, but if in doubt try both).

Well it’s it’s user error, I’d love to know the secret to eliminating it. All input welcomed.

My gut says a compressor shouldn’t bring up anything that is below its noise floor. I could be missing something. Chris’s doesn’t have the same behavior on all the other “silence” in the file. It handles everything very well, except the silence at the beginning/end of the file, which it ALWAYS adds some sound which is NOT wanted. In my case, when the compressor is applied, those beginning/end sections are down between -50 and -60 dB (room tone), sometimes lower.

If I have 3 seconds of room tone, it doesn’t bring it all up, it brings up about a half second of noise, in a repeatable pattern, just before my audio (voice) starts. Reminds me of an an ocean wave, starts low, gets louder, fades out. It repeats this behavior after the last spoken word in the file. IF there is a 6 second buffer (room tone) after the last word, the ocean wave sound happens in the first couple seconds of the silence after the last word, then the last half of the buffer section is silent again. The compressor is inconsistent within that last X seconds of “silence” and brings up only a portion of it. I’ve verified BEFORE applying the compressor that there is nothing in the before/after room tone that is above the noise floor.

I also figured out the same workaround someone else posted. If I extend my beginning/end silence with extra, then it’s easy to snip out the extra noise after the compressor is applied.

If you listen at a low volume you may not hear it. If you listen moderately loud, it’s obvious. To see it visually if I zoom in with one left mouse click on the scale.

And again, I have to suspect after all this time there is a compressor that works as well as Chris’s, but doesn’t have this one feature. I could be wrong, and overall I am very happy with the way it works within the audio.

I’ll upload samples this weekends highlighting this. From what I can tell, it’s a bug. Not a feature. I hope someone shows me I’m just doing it wrong. That would make my day since it’s extra steps I don’t need. Happy to be wrong! Please make my day. :slight_smile:

So did you try raising the noise floor setting to see what happens?

No, but I’m laughing at myself because that’s worth trying and I didn’t do it.

I think I didn’t try that because it says “raise the floor to keep the quiet parts quiet” and I really didn’t want to do that with spoken voice. I want the quiet parts to be closer to the louder parts.

Chris defaults the “Floor” to -32 and my standard is currently -28.
I settled on that after testing a few dozen audio files a couple months ago (while also tweaking the other settings.) In other words, I have raised it by 4 dB already compared to the default, but I’ll try more just to see what happens.
If I understand it right: If I raise the floor high enough, it should practically leave the file alone (not bring up any of the soft parts.)

In case it make any difference, here are my other settings.
(Maybe something else I have set is effecting the output too.)

Compress ratio: .8
Comp Hardness: .35
Floor: -28
… gate falloff: 2.0
Max amplitude: .85 (depending on the file, sometimes .70)

Of course I’m open to trying it, but if I understand it, getting the floor higher means I get less compressor effect on the soft parts of my audio.

I’ll let you know what I find, and I appreciate the suggestions.


Audacity 2.0.6
Windows 8 (64 bit)

Noise floor at -28, always hear it, see it front and end of the file (comparing to the noise floor before compression is added.) No visible or audible sound before compression, obvious in both cases after.

As I approach -20, it is almost totally gone on the start of the sound (around -75 dB, close enough for me), but the tail end still barely has it. I can see/hear that end. I can see the front end noise if I zoom in enough, but for practical purposes it’s silent. (Someone here on the board posted an “Amplitude…” plugin and I use that to check that I didn’t have noise I couldn’t see, plus I zoomed way in before applying the compressor.) Note that it’s adding something, but for practical purposes it’s not enough to matter.

I did lots of testing. It’s inconsistent with itself between different files. The AMOUNT of the added noise is longer/shorter depending on how the audio starts/ends. Not by much, but there is some variation. That implies to me if that you have music, it will depend on how it starts and ends, what it’s level is. I also didn’t test this audio with it EQed, it’s just raw voice with the noise removed via the noise tool.

It’s still adding the sound at the beginning no matter what, but for practical purposes I have test cases where that is low enough to be practical IF I raise the floor to -20 (or higher), but I’m not clear that works for me. IF I get to -15 or so then I can’t hear it or see it.

I was paying no attention to the quality of the compression with that floor on the voice, just checking out the noise based on this thread. I’ll next compare some audio with the floor at -28 vs. -20 and see what that’s like in terms of compression. I’m out of time for testing today (actually need to process some audio), but if the floor is high enough, the extra noise goes down.

BTW - Anybody else doing audiobook/podcast or other naked voice type processing: What settings are you using for Chris/compression? Anybody else using a noise floor that high? Maybe I’m missing something so please let me know what you are doing.

I have said for a while now that properly applied Chris Compressor can solve all your Spoken Word Standards problems in one swoop. I have never noticed the beginning and end problem, but I’m trying to remember if I cut both of those off anyway each time I use it. I know it was a very serious issue with an early version and I thought 1.2.6 was designed to solve that… It’s been a while.

Chris designed this compressor so he could listen to opera in the car. He said so. He designed it to be musical and still carefully remove all the enormous volume changes in each dramatic scene. That’s why it doesn’t work like other compressors. He started with Verdi and Puccini and worked up instead of starting with attack and release differential algorithms and working down.

I posted three waveforms. An original “podcast” performance, and two Chris variations: Standard Settings and Compression changed to 0.77 from 0.50.

The 0.77 version is indistinguishable from the same show broadcast on the local FM station using broadcast compressors, and in fact, I can ride around in my truck/lorry without having to constantly adjust the music system volume.

It would not shock me that one of the two settings resulted in a show that conforms to Spoken Word sound standards, since they and FM Radio have similar requirements.


You gave me some idea, couldn’t Chris’ Compressor be used for exactly the opposite, namely to bring broadcasted music back to its original dynamic range?
You could experiment to figure out which (negative) factor should be entered to just achieve this.
There’s a floor when used as expander though, the gain is not adapted accordingly. Certainly a point for improvement, along with some input correction (some values produce division by zero).

As I mentioned, I’m fine with the sound quality of the compressor but I certainly am not going to assume someone couldn’t create something stronger someday. I have no idea, but it works great for me except the extra noise it adds. And I’ve found workarounds, but still would love to find a way to eliminate them.

Next question:

Please cure my ignorance:
When you say “Standard Settings” I’m guessing/assuming that means “Defaults”? And if you changed Compression you mean “Compression Ratio”, and you’re leaving all the other settings per the default?

What settings do you like for spoken word?

I hope you’re right and it’s the best choice, since it’s what I’m using regularly.

As a side bar: My wife and I are both musicians (she has a voice degree from University of Michigan) so I heard tons of operas as a young man along with paying some dues working in a recording studio when I was young too many decades ago.

Opera is not exactly what I would consider close to spoken word, and I’m not a huge fan of what FM stations do to most music, in terms of compressing. I understand why they do it, and it sucks the life out of great dynamic music, but that’s a totally different discussion. I’m not saying a product designed for opera would be unsuitable for an audiobook, but in my mind they are in different leagues in terms of dynamic range and what is needed as an end product. So far Chris’s IS the best I’ve found, but still I expect over time someone will find ways to make it even better or a different way of doing it. If I’m wrong that’s fine, I’m already a huge fan of it other than the one issues I’ve outlined in previous posts.

I love hearing your insights into the original design. Great to have some additional depth. Thanks for sharing with those of us who are newer.

The Dolby Corporation made its name by designing a compressor/expander system that you couldn’t hear working. It was exactly symmetrical and in its “A” version made tape overdubbing possible because the 10dB noise boost that resulted was precisely the 10dB suppression that Dolby offered. Whole worlds opened up. The first layer of Analog Stereo Television was Dolby processed to get rid of the FM Stereo Noise Penalty. Analog TV sound is (was) FM. It worked. TV Stereo appeared in living rooms without fffffffffff in the background.

You only get that kind of transparency if you can control both ends. Nobody has managed to “de-Dolby” an audio performance except Dolby (several posts on the forum) and I suspect a look-ahead compressor with semi-unknown audio characteristics is unlikely to uncompress an existing performance. There’s no way to know what the trigger points are without knowing what the original compressor was thinking about. Even worse is the assumption of only one compressor or processor. FM Broadcast systems have two. One overall level setter and a second peak grabber – and later ones were even worse. I don’t know any way to reverse one of those.

What were you going to use as a test clip?


The particular application I have is a spoken word podcast/broadcast (plus telephone voices and music) by two people who have an enormous range between “mumbling in one’s beer” and a thermo-nuclear laugh. Whereas the squashed radio broadcast is perfectly enjoyable in the car, the podcast is not. The podcast has all the common complaints of people who fail to conform to Spoken Word requirements. The sound peaks are all over the map, the RMS value is too low, the voice volume varies too much, etc. etc. etc. This on top of room noise and echo problems common to people trying to record in their kitchen.

So yes, I do have Spoken Word performance examples where Chris worked very well and continues to. That waveform example is this particular show which returns from a musical bumper (the level segment) into voices. Note the variation in the upper one is all but completely eliminated in the bottom (Compression Ratio 0.77), plus the overall volume increases without clipping or overload damage. Further, because of look-ahead, the sound neither pumps nor breathes.

It will be very difficult to beat that.


And yes, Look-Ahead is the problem at the top and bottom of the clip. At the beginning, Look Ahead presets the compression compared to a clip that isn’t there yet, and at the end the gain used to blast off because the show sound would “mysteriously” drop to zero. I don’t remember any more than that, other than a fuzzy idea that this is the solution. Legacy versions were much worse.


When starting a recording it should be:
“Record,… 3…2…1…go…”
rather than

The very start of a recording can be a bit iffy for many reasons. Allowing a few seconds at each end of the recording allows space to tidy up when editing. Some recording programs even have a time line that starts at -2 seconds so as to encourage a 2 second lead in.

Quite so. Compressors can use “lookahead” or not.
Compressors work by changing the gain (amplifying up/down) according to the input level. If the gain changes too rapidly then it is very noticeable (commonly described as “pumping”).

Without lookahead, the gain will change only in response to the input level. If the input level goes up suddenly then it takes a while (“attack time”) for the compressor to catch up and reduce the gain. Real time compressors tend to do this because they don’t know what is coming until it happens. In order to prevent sudden peaks from going too high, a “limiter” may be used after the compressor. A limiter is similar to a compressor but has a very fast “attack” so that it can limit sudden peaks.

Compressors that are used for processing recordings may use “lookahead”. The compressor looks at what the level will be doing next. If a sudden peak is approaching then the compressor will start reducing the gain a short time (“lookahead time”) before the peak arrives. This prevents “overshoot” of the output level, but if there are a lot of sudden peaks (transients) then it will tend to push the level low and sound too quiet. For recordings of percussive sounds it is often better to use a compressor without lookahead (or a short lookahead time) and follow it with a peak limiter.

It is also important that after a high peak has passed that returning to normal gain does not occur too quickly. This is the “release time” (the Audacity compressor calls this “decay”). If this is too fast then you can hear “someone has turned the volume up”. If it is too slow then the sound after the peak is likely to be too quiet until the compressor has “released” the attenuation.

As Koz wrote, there is a problem at the start/end of the recording because the best that the compressor can do is to guess what the signal level might have been before the start and after the end of the recording. If the compressor guesses that the signal level was 0 dB before the start then the gain should start at minimum and then “release” to the current level, which will tend to make the start too quiet. If the compressor guesses that the signal level was silence before the start, then the gain should start at unity (1:1) which may allow the input to overshoot until the compressor has caught up. Similarly at the end of the recording. Probably the “best guess” is that the level before the start is the same as the level at the start, and the level after the end is the same as the level at the end, but even this will not always provide the desired result.
The only solution guaranteed to work is if there is sufficient lead in / out to allow the compressor the time that it needs to change the gain to the correct level.

I just processed a Saturday radio show and I didn’t hear anything wrong with the beginning or end.

You should post the damage you’re hearing on the forum. You may have to do it in multiple passes due to forum restrictions.


I’ll upload some samples with the issues ASAP. I used the compressor last night on a 30 min clip and had the issues front and back. (To be clear: This is minor, and not exactly a show stopper…)

I’ll probably just run that same clip, see if I can duplicate it with 10-30 seconds of audio. My gut says it has to do with how much room tone/silence is at the front/back of the cuts. We ask our narrators to leave at least 5 seconds up front and at the end. I’ve duplicated the issue 20 times, so no problem uploading samples.

(I have a project due by tomorrow, so you’ll see the upload within a day or two.)

I appreciate all your insights!