Topic: VST plugins & mixing/mastering workflow in your DAWs

[phil_er_up]

Would the auto-align2 fix the combing effect with 2 sets of cardioid  mics from the same mic stand...like one set of cards with Supercardioid  or wide card?
Also, like the idea of using it with SBD and aud recordings.

Yes. That's actually my use case. I mainly record 4 or 5 audience channels + a stereo soundboard feed and matrix them in post.
Cf. the screenshot below. The darker areas in the Correlation diagram show the corrections.

In your screenshot of your DAW you have a "tape" plugin at the end of bus one and bus two plugin stack.
What is that "tape" plugin?

[morst]

I don't mean to cast shade on these amazing tools, and enjoy thinking through and discussing the implications.  As always, what sounds right is right.  You needn't think too deeply about all this.  Just listen carefully when making the decisions to be sure of doing more good than harm.  Please let me know if/when you come across outlier situations when using these tools. Thanks.

No matter what they say, no matter what they do,
Gonna end up in the middle of that rock & roll stew

~Rock & Roll Stew [Traffic]

[Gutbucket]

 

[if_then_else]

Would the auto-align2 fix the combing effect with 2 sets of cardioid  mics from the same mic stand...like one set of cards with Supercardioid  or wide card?
Also, like the idea of using it with SBD and aud recordings.

Yes. That's actually my use case. I mainly record 4 or 5 audience channels + a stereo soundboard feed and matrix them in post.
Cf. the screenshot below. The darker areas in the Correlation diagram show the corrections.

In your screenshot of your DAW you have a "tape" plugin at the end of bus one and bus two plugin stack.
What is that "tape" plugin?

Those "tape plugins" can help "glue" audio tracks together and add a polished, cohesive sound to the mix.
Think of these like one type of (gentle) saturation plugins...

Tape plugins emulate the (subtle) harmonic distortion of specific analog tape machines:

• They add warmth and depth to digital recordings
• They smooth out harsh transients, creating a more cohesive sound
• They can be used on individual tracks, busses, or the master channel

I think there was a thread somewhere on here where people used real tape machines to add that kind of "warmth". High-end tape machines have become really expensive these days. These are the software equivalents. I don't know if they sound exactly like the specific types of hardware tape machines but at least they lend the same type of "warmth" to a mix/master.

There are even a few decent free tape plugins. Proper reply after work.

[if_then_else]

Re. tape plugins: There are some decent free ones. So, try them out first:

• Chow Tape Model: https://chowdsp.com/products.html
• FerricTDS mkII (version 2): https://varietyofsound.wordpress.com/2009/10/29/ferrictds-released-today-withi-the-kvr-dc09/ (This is was originally created for 32-bit DAWs. In the meantime Variety Of Sound also created a 64-bit version which is available from the website's Downloads section.)

Among the commercial plug-ins, which are often designed to emulate specific hardware, there are some that are more suitable for the mix bus and others that are more suitable for the master bus. Some sound “dirtier” (suitable for heavy metal or noise rock) and others sound cleaner.

An example of tape plug-ins for the mix bus would be the Studer A800 tape recorder emulation from UAD. I personally don't like it at all in the master bus, but it sounds good for electric guitars in the mix bus. For the master, for example, UAD's Ampex ATR-102 would be a good choice. The tape plug-ins from u-he (“Satin”) or Softube (“Tape”) would be suitable for both the mix bus and the master bus. In general: If you decide on one of these commercial plug-ins, only buy them on sale. Some of them are reduced from over a hundred dollars to 30 or 40 dollars. I wouldn't spend more than that under any circumstances.

As for Gullfoss: As already mentioned, this is more of a plugin that makes small cosmetic improvements. Don't push it to its limits. Gullfoss comes in three versions: one for individual instruments, one with an optimized footprint for real-time applications (e.g. for live mixes) and the master version. I have only ever used the master version. The parameters are explained quite well here: https://www.soundonsound.com/reviews/soundtheory-gullfoss.

The two main parameters are Recover and Tame. Recover accentuates those elements that are in danger of getting masked by other sounds, while Tame suppresses those over-dominant sounds responsible for the masking. Both have their own amount percentage controls, and a third control called Bias determines how Gullfoss reacts in situations where it is less clear whether Recover or Tame would be most appropriate. Positive Bias values favour the Recovery mode at lower signal levels, while negative values lean towards the Tame mode. While Gullfoss matches the subjective level of the output to the input, it is designed not to adversely impact the dynamics of the treated signal — indeed, in the case of an over-compressed or over-limited signal, a useful degree of dynamic improvement can be made to transients. This was certainly borne out in my tests, where some delicate percussion that was in danger of getting buried in the mix came across much more cleanly.

The next control, Brighten, adjusts the overall tonality of the processed signal; as its name suggests, it can be used to lift the top end, but it can also be set to a negative value to reduce the highs if the source is already too bright. Finally, there’s a Boost setting, but this isn’t just a level control; it also affects the overall tonal balance in a way that reflects the way human hearing perception changes with level. Its action is to some extent based on the familiar Fletcher-Munson equal loudness curves, but there’s more to it than that, as it also has a dynamic impact on the EQ rather than applying a static curve — increasing Boost emphasises bass frequencies while cutting back the mids. Actual output level can be adjusted using the vertical slider at the right of the plug-in window.

[phil_er_up]

Thanks If_then_else for the info, pics and links.

Do have 2 tape plugins currently - Ozone vintage tape and PSP vintage warmer 2.

Early on in taping career used the PSP vintage warmer 2 though not sure if I was improving the sound or not. Never used the Ozone vintage tape until yesterday after reading this thread. Never thought about using it at the end of the bus stack. So this thread to me is about the VST plugin though also where and what ones to use in the bus 1, bus 2, master bus etc. This info could help newbies think about which plugins to use and where in your bus stacks. Good stuff.

You are very helpful and really thank you for that.

[if_then_else]

Thanks. I'm not going to pollute this thread further with my specific preferences in terms of plugins or workflows. This was just intended as an overview of some plugins I found useful. I understand that people's priorities and preferences in terms of post-processing are highly subjective and the kind of music I'm recording as well as matrices/multi-track recordings might only appeal to a certain subgroup of people here. That said, I've got two shows coming up next month where I've got open taping (+ filming) permission. I'll try to document my post-processing workflow in a separate thread while I'm on it. Just in case that it might be helpful to others.

[Gutbucket]

Thanks for the detailed info if_then_else. It's great to get this kind of insight into what fellow TS'ers are currently using, how they are using it and for what. Particularly so due to this area constantly evolving with the introduction of new tools.

[opsopcopolis]

Ran auto-align 2 and Mauto-align. Thought Mauto-align sounded good till I heard the auto-align 2. Auto-align 2 file sounded better IMO.
The low end of the file from the auto-align 2 processed file was there and sounded more natural the Mauto-align file sounded more mid-range and high end not very much low end for the file I was running. YMMV.

Also used TBProAudio DSEQ3 and it is a very good plugin and I like it and would use this one. Is easy to use and sound output was musical.

Tried Fabfilter Pro C2 and Fabfilter Pro L2 and both are really good though both are expensive.
Really liked the Pro C2 compressor and would see if I use the Pro L2 limiter that much though both seemed to do a really good job and sound output was musical.

Have anyone tried the fabfilter plugins and is it worth the $179 for each?

Saw online people raved about the Fabfilter Pro Q4 though have not tried that one yet.

=================================================================

Could not get Kush Omega and Sonible smartLimit to work. So can not comment on those.

The Fabfilter stuff is absolutely worth the money IMO. Especially the EQ, Comp, and Limiter. Pro Q4 is a game changer for audience tapes imo. Allows you to perform wide q boosts, while also dynamically reducing resonances

[Billy Mumphrey]

Thanks. I'm not going to pollute this thread further with my specific preferences in terms of plugins or workflows. This was just intended as an overview of some plugins I found useful. I understand that people's priorities and preferences in terms of post-processing are highly subjective and the kind of music I'm recording as well as matrices/multi-track recordings might only appeal to a certain subgroup of people here. That said, I've got two shows coming up next month where I've got open taping (+ filming) permission. I'll try to document my post-processing workflow in a separate thread while I'm on it. Just in case that it might be helpful to others.

Feel free to share any and all preferences, in this thread or another as I've been (slowly) following along and learning A LOT. The pace and progress of these digital tools is scary, in (mostly) good and bad ways.

The alignment tools are of particular interest, as they do seem to contradict what we try to do with stereo imaging: introduce time arrival differences for a fuller, more complete sound. I do believe in going with what sounds best, and look forward to hopefully trying these out myself.

[if_then_else]

Speaking of (multi-)track alignment (not phase-alignment): There seem to be some freeware/open source programs that are supposed to handle this. Among others, two stand-alone projects on github and a Nyquist plugin for Audacity. I haven't tried any of them yet, but I'll definitely have a look at the software.

https://audionyq.com/aligning-multiple-tracks/
https://github.com/protyposis/AudioAlign
https://github.com/benfmiller/audalign/

[morst]

Speaking of (multi-)track alignment (not phase-alignment): There seem to be some freeware/open source programs that are supposed to handle this. Among others, two stand-alone projects on github and a Nyquist plugin for Audacity. I haven't tried any of them yet, but I'll definitely have a look at the software.

https://audionyq.com/aligning-multiple-tracks/
https://github.com/protyposis/AudioAlign
https://github.com/benfmiller/audalign/

Looks like a good start.




Potential workflow:

• [size=78%]Align near start of show using plugin[/size]
• [size=78%]Mark start or end points of every track[/size]
• [size=78%]Align near end of show using plugin[/size]
• [size=78%]Feed offsets into my spreadsheet[/size]

https://docs.google.com/spreadsheets/d/1pQGfYwPgBFFzcY5m6aRj-Zbu9HsRumLy-tJB1d8Eufg/edit?gid=583050244#gid=583050244

• [size=78%]Sync[/size]
• [size=78%]Save work[/size]
• [size=78%]Listen and see if everything is correct...[/size]

I really hate the way this website changed at some point...

[Gutbucket]

More on channel alignment (please ignore this post if not your bag)-

A very interesting way to gain additional insight into some of the peculiarities of channel alignment is to mess around with applying a very slight delay to the Side (difference) channel verses the Mid (Sum) channel and listen for what that does.  The recording need not have been be made with a Mid/Side or X/Y microphone setup.  You can use any stereo recording by simply converting it to Mid/Side, applying the delay and converting back to L/R again.  That's most easily done in a single time delay plugin which includes Mid/Side routing options as well as Left/Right.  Here's a free one that can do that, Voxengo Sound Delay- https://www.voxengo.com/product/sounddelay/

Tiny delays of a fraction of a millisecond do interesting things in changing the perception of depth in the recording, particularly over headphones. Too much screws up the left right imaging while creating an overemphasised sense of font-back depth.  Way too much will create an echo effect. But a just touch can help bring the imaging of headphone listening "out of head" and increase the sense of 3d dimension without altering the Left/Right imaging significantly.  Whether one wants to do this to their recordings or not isn't really the point, it's more about hearing what it does to the perceived stereo imaging to gain insight into the nature of time alignment and how the best sounding alignment may not be one that is perfectly in-phase (which for a spaced pair of microphones can only be achieved along a single axis of arrival, as discussed in previous posts above), and how the choice alignment can legitimately vary over some range.

This all came to mind again today upon coming across this Youtube video- Headphones Are Not Stereo (mid side phase trickery) https://youtu.be/uZ9WQDojQt8?si=2UuKHi4Ecbq_z4CZ
The most relevant part in that video about the effect of delayed time/phase alignment starts here- https://youtu.be/uZ9WQDojQt8?si=jYi74B0U_i3VlMmi&t=442
If you need a primer on how sum/difference(Mid/Side) stereo works, just back up a bit along the timeline.

[Gutbucket]

^
This is also related to how alignment is intentionally varied in some microphone arrays by positioning the center microphone or center stereo pair slightly forward of a flanking pair, as we've discussed in other threads.  A wave-front arriving from directly ahead will reach the center microphone or pair a fraction of a millisecond before the flanking mics. When mixed down to stereo, that center mic or pair provides the majority of the Mid (sum) stereo content while the flanking mics provide most of the Side (difference) stereo content.  Positioning the center mic position slightly forward of the flanking pair effectively delays arrival at the flanking pair that is providing the majority of the Side (difference) content by some fraction of a millisecond.

I don't think I made this particular association clear in my posts when we were discussing center mic position in those other treads, although we did certainly talk around it. It similarly came up when discussing the extension of Improved PAS technique from 2 channels to additionally include PAS solutions for 3-microphone positions for the use of 3 or 4 microphones.  In that case the microphone angle/spacing/position solutions from the Michael Williams multi-mic Stereo Zoom data, the Sengpielaudio Visualizer and the Schoeps Image Assistant all call for pushing the center microphone position forward of the flanking mics by something like ~8-20 cm or so, as do most surround recording microphone configurations.  That effectively ends up sort of creating two axes of in-phase alignment angled somewhat right and left of center, rather than a single axis for sound arriving from directly ahead. (More technically correct - one axis angled right of center where the right spaced mic and center mic (or pair) is in full alignment, and another angled left of center where the left mic and center mic (or pair) are in alignment.)

Here are a few of the threads in which we've discussed center microphone (or pair) positioning:
> Relationships between spaced omnis and center mic- https://taperssection.com/index.php?topic=182579.0
> 3 mic (LCR) distance- https://taperssection.com/index.php?topic=200878.0
> Re: Improved PAS technique- now as setup diagrams & extended to 3-4 microphones- https://taperssection.com/index.php?topic=167549.msg2411846#msg2411846
(fair warning, some of the above threads get wordy!)

[if_then_else]

I've just tried my luck with Audalign (https://github.com/benfmiller/audalign/). There's good news and bad news. The good news is that it works perfectly. The bad news: it is not trivial to install.

The installation instructions on the project page should be taken with a grain of salt. If you install the whole thing with a current Python version (e.g. under a current Anaconda on Windows or Python 3.12 or 3.13 on Linux), you will encounter errors because some versions of the libraries used are no longer compatible with each other.

The last update of Audalign released a year ago is specific to Python 3.11.

The solution is to create a virtual environment with Python 3.11 and install the modules. I did this on Linux, it should work similarly on Windows:

Code: [Select]

pyenv install 3.11
/home/markus/.pyenv/versions/3.11.10/bin/python3.11 -m pip install --upgrade pip
/home/markus/.pyenv/versions/3.11.10/bin/pip3 install audalign
/home/markus/.pyenv/versions/3.11.10/bin/pip3 install audalign[visrecognize]

Do not install audalign[noisereduce]. It will pull some *big* Nvidia specific modules that would probably require a Nvidia graphics card and drivers.

This is a small sample Python script to align two files that are located in the folder "bak" and places the aligned files in a directory called "destination".

Code: [Select]

import audalign as ad

# Create a recognizer (in this case, using the FingerprintRecognizer)
fingerprint_rec = ad.FingerprintRecognizer()

# Optionally, configure the recognizer
fingerprint_rec.config.set_accuracy(3)

# Align files in a folder
results = ad.align("target/folder/", recognizer=fingerprint_rec)

# Or align specific files
results = ad.align_files(
    "bak/20160117_CHVE_44.1Khz-16bit.wav",
    "bak/AUD.wav",
    destination_path="destination/",
    recognizer=fingerprint_rec
)

# For fine-tuning the alignment
fine_results = ad.fine_align(
    results,
    recognizer=ad.CorrelationSpectrogramRecognizer()
)

Call the script with Python 3.11 from your virtualenv environment

Code: [Select]

/home/markus/.pyenv/versions/3.11.10/bin/python ./align_files.py

The sample output would look similar to the following.

Code: [Select]

Directory contains 0 files or could not be found
No matches detected
0 out of 0 found and aligned
Fingerprinting 20160117_CHVE_44.1Khz-16bit.wav
Fingerprinting AUD.wav
Finished fingerprinting 20160117_CHVE_44.1Khz-16bit.wav
Finished fingerprinting AUD.wav
20160117_CHVE_44.1Khz-16bit.wav: Finding Matches...  Aligning matches
AUD.wav: Finding Matches...  Aligning matches
Writing destination/20160117_CHVE_44.1Khz-16bit.wav
Writing destination/AUD.wav
Writing destination/total.wav
2 out of 2 found and aligned

Total fingerprints: 10395127
Fine Aligning...
100%|█████████████████████████████████████████████████████████████████████████████████████████████████████████████████████████████████████████████| 2/2 [00:00<00:00, 3480.75it/s]
Comparing AUD.wav against 20160117_CHVE_44.1Khz-16bit.wav...
Comparing 20160117_CHVE_44.1Khz-16bit.wav against AUD.wav...
Calculating correlation... Finding Local Maximums... done
Calculating correlation... Finding Local Maximums... done
2 out of 2 found and aligned

I've loaded the original files in Reaper. You can see that the original files (track 1 and 2) are unaligned.
The two tracks taken from the "destination" directory are aligned though. One of the files (track 2) has been padded with silence at the beginning.

Again, this is for "track align" purposes. For phase alignment, you might want to use one of the phase alignment plugins that have been discussed earlier in this thread.

Oh, and unless I've done anything wrong, the output in destination appears to be in mono. (Which would be backed by the animated gif / demo on the project website. I'm going to contact the author about it anyways.) You might want to use mono stems as input. This could also be scripted:

Code: [Select]

ffmpeg -i input.wav -filter_complex "[0:a]channelsplit=channel_layout=stereo[left][right]" -map "[left]" left.wav -map "[right]" right.wav
To join two mono channels back to stereo:

Code: [Select]

ffmpeg -i left.wav -i right.wav -filter_complex "[0:a][1:a]amerge=inputs=2[a]" -map "[a]" output_stereo.wav
Alternatively, there is an internal workaround with Audalign:
https://github.com/benfmiller/audalign/issues/67

I'll try to come up with a Python script tailored to "our" workflow when I have time.