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Hello folks,
Thank you very much indeed for all the help and advice you have given me on this forum and on my latest thread on mastering.
I would be grateful if you would give me some pointers on clipping when working with audio.
Can I assume the following is correct?
1 Clipping (above 0db) should not be tolerated in any part of the audio process from start to finished mixdown or master? This includes clipping in the input audio on all tracks and on all plug-ins on the track and on the master stereo out as well?
2 How do you personally deal with a short burst of clipping on a track? Do you lower the channel fader for the whole duration of the song, or if it’s not too noticeable just lower it for the duration of the clipping part?
3 When dealing with the Ozone mastering plug-in (placed in the master stereo out channel) I find that it may well be showing as clipping in the Ozone plug-in itself, but not as clipping in the Daw master stereo out channel, is this a fairly normal occurrence?
Thanks very much again for all your helpful advice.
Musiclover
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For me, I just isolate the portion that clips and use a gain change to drop it 1 db at a time until it stops tripping the clip-o-meter. I know that sometimes there is a random measure somewhere that is at a higher level of attenuation than the rest of the track.
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Your mix should not be more than -6 before mastering, that way you the have headroom to master.
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Yes, yes and yes.
To avoid clipping, I record in 24-bit for more headroom.
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Thank you very much for replies and advice.
If its ok I would like to seek your advice on another problem though really I should probably start a new thread, but I don't want to have a lot of different threads going as it may affect other posters.
I don't have a studio and just use my main computer as my DAW, I have 2 different pairs of speakers hooked up, one logitech 5.1 (though I do have it switched to 2 speakers) for listening to multimedia from the internet, playing cd's etc.
The other pair are an old goodman set I had that I hooked up to my m-audio usb interface and I use those for DAW work etc.(they are sort of behind the monitor due to lack of space, but that is not my main problem.
The problem is how do I calibrate them correctly for playback? I spent hours mixing and then exporting the mixdown but when I opened it in windows media player which plays it thoguh the logitech it was really really bassy, the goodman speakers have 3 controls volume, bass and treble and I have bass and treble set at approx 1 o' clock. Commerical recordings play fine thoguh the logitech, so I take it its the goodmans that are not setup properly, if indeed that is possible with them.
Do I just adjust the treble and bass more on the goodmans so that a commerical recording sounds sort of similar on both sets of speakers?
Do proper monitor speakers have the above controls volume bass and treble and would I get anything for less than say $200 if I do decide to buy?
Thanks very much again for your patience with me.
ps Just an afterhtought on this, since its even possible to edit the eq in the mixer in the computer itself for the logitech speakers this will probably even make them poor references.
Musiclover
Last edited by musiclover; 06/05/12 01:27 PM.
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Without proper acoustic treatment and a sound level meter, you probably will not be able to 'calibrate your speakers', in the sense you presented it. Mixing involves learning about the sound of your monitors and what the results are of mixing on them, then making the necessary adjustments.
Two quick mixing tricks:
1. Test your tentative mixes on everything you can find, including car speakers, boom boxes, cheap computer speakers, home stereo etc.
2. Try, when mixing, reducing the volume to just barely audible. Does any instrument stand out? That may be a problem (even for vocals). Can you hear every instrument? You should be able to.
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Quote:
2. Try, when mixing, reducing the volume to just barely audible. Does any instrument stand out? That may be a problem (even for vocals). Can you hear every instrument? You should be able to.
I agree with the first part of this but not the second, Matt. I was just discussing this with someone over the weekend and we went for a drive in my car to test it. You do not hear every instrument at very low volumes. My Benz is a nice quiet car with the Bose sound system but it's still a noisy environment compared to a studio. We spent a good half hour driving around and listening to everything from country to jazz to heavy metal to classic hiphop. When we turned the radio down to where we could barely hear anything what's left is always the lead, either vocals or instrument, and drums. That's it 95% of the time (there's always a few exceptions) and of those two vocals or the lead instrument like your sax are always on top of everything. I've used this as a rough test of the mix for years. Of course this is just rough, it has little to do with panning, EQ, compression, all of that stuff. It's just that when I listen to songs people post here and elsewhere most of the time the lead is buried. The basic mix is way off and my quick down and dirty test will at least put someone on the right track.
My guess is amateurs have no confidence in their singing or playing so they don't mix themselves properly. My feeling is if you're going to do it, then do it warts and all. If you don't then the whole production sounds like amateur hour not just the singing. I'm occasionally guilty of that too btw. I have some live recordings where my piano solo had a few too many clams in it so I buried it a little. One I was able to cover by pasting in a short phrase I did earlier and that little repeat sounded much better than the totally awful chord I played. When I played it back you would never tell I did that, it just sounds like I repeated a lick. Whatever, at least I try to do it right and also improve my playing.
Bob
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Quote:
To avoid clipping, I record in 24-bit for more headroom.
I'm not sure that 24 bit versus 16 bit recording will give you any more 'headroom.' What it WILL give you is more dynamic range. From Rane's Professional Reference Manual
Quote:
headroom A term related to dynamic range, used to express in dB, the level between the typical operating level and the maximum operating level (onset of clipping). For example, a nominal +4 dBu system that clips at +20 dBu has 16 dB of headroom. Because it is a pure ratio, there are no units or reference-level associated with headroom -- just "dB." Therefore (and a point of confusion for many) headroom expressed in dB accurately refers to both voltage and power. Which means our example has 16 dB of voltage headroom, as well as 16 dB of power headroom. It's not obvious, but it's true. (The math is left to the reader.)
If my understanding of this is correct (and I've been wrong before) if you record at -12dBr (referenced to 0dBFS), then you should have 12dB of headroom. If you're using 16 bits, then you have a dynamic range of 96dBr versus 24 bits which gives you 144dBr of dynamic range. However, if you're still recording at -12dBR, you still only have 12dB of headroom, 16 or 24 bit.
If anyone disagrees, please discuss civilly.
Gary
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In response to the tweaking speakers, I don't know if this will work or not, but it's worth trying.
Create a series of individual test tones, between 20Hz and 20kHz. You should be make them all the same level. Position a GOOD microphone at your mixing position, and then record each tone via the microphone to an individual track. Make sure the speakers are set to neutral settings, i.e. no bass or treble gain or cut, and are correctly positioned for mixing.
Use a Real Time Analyzer plug in to play back each tone recording. Ideally, the RTA should indicate what the original test tone was sent out. If they aren't, then you can attempt to adjust the speakers bass and treble, although with only two controls this will be difficult.
Finally, mix the signals into a single file and then play it back. Record another track of all the signals being played back through your now tweaked speakers. You should be able to see a broad spectrum analysis of what your speakers are sending out.
From there, you should be able to use either a 10 band or parametric EQ to get closer to a flat curve. Save that EQ curve and use it in the master section to correct the overall tonal quality of your speakers. It won't be perfect, far from it, but it should bring you closer to a flat curve.
Gary
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The main argument for 24 bit is more digital room for the audio, more dynamics, less need for compression while recording and more "headroom". Every article I have read about 24 bit vs. 16 bit recording uses the word "headroom" about every other sentence.
I guess you have to agree of a definition of "headroom". I think the general definition is "how much before clipping" -- 24 bit has more.
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Kevin, There is 144dB of Dynamic range in a 24 bit recording. There is 96dB of DR in a 16 bit recording.
So, what does this actually mean? With a 16 bit recording, if you could find an absolutely quiet space, you have 96dB of range before running out of range. A piano being played fortissimo is about 95dB, and is considered 'very noisy.' A full symphony orchestra is 110dB! Does that mean that you can't record a full symphony orchestra with a 16 bit recording? No, of course not. What it does mean though is that it's going to be about 15dB above dead silence. A rock band at a live concert is about 130dB, which means that the bottom of your dynamic range is going to be about 35-40dB, not dead silence.
A true 24 bit recording, and you're never going to have a TRUE 24 bit recording, because you run into electron flow noise down in the mid 130's, would allow you to go from dead quiet to well past the pain level. Imagine standing next to a 747 jet taking off, you'll understand 140dB!
This is the true difference between 16 and 24 bit. Headroom is measured from clipping down, dynamic range is measured from dead silence up.
Gary
I'm blessed watching God do what He does best. I've had a few rough years, and I'm still not back to where I want to be, but I'm on the way and things are looking far better now than what they were!
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Hi Gary. These are very complicated topics for a forum such as this, and I'm trying to keep my answers short and perhaps oversimplified for Musiclover, who appears to be someone just getting started in audio.
I agree you're not going to hear everything even in the quietest of cars, and good for you that you have one. I did not mean to mix for a car, but to compare the sound in all the places you can listen, and go back and make small adjustments in the mix. What the OP should be looking for is a glaring mismatch in levels. You can't fix everything, though; bass solos, for example, are going to disappear in a car (mine, anyway) no matter what.
The best most of us can do is to learn how our monitors and our listening environment for mixing are not perfect, and we adapt. Let's all help the OP to get close without burying him in details that he/she will pick up in time. To that end, I will be guided by the level of the question from the OP.
You made a great point that you might as well put it all out there, warts and all, since hiding in the mix doesn't work. There's a funny message circulating now on Facebook to the effect that Jimi Hendrix remarked that people were copying his playing, even his mistakes. Gotta love it.
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Matt, Understood. Kevin, something else that I do, and works for me and my style of recording. I record synth tracks only, no vocals and no acoustic instruments. Hence, I can go back and adjust as necessary, and record the track as many times as I want with the same MIDI sameness. First, I record each of my tracks separately, one at a time. I try to find the peak of the track, and have that peak somewhere between 0 and -3dBFS or -3dBr. That assumes that 0dBr is the same as full scale and clipping. I record each track so that each is in that same range. What this does for me is to give me the fullest dynamic range of the instrument, so when there is quiet, I'm not getting hiss from the lower levels and running into the noise floor. Synths are inherently noisy creatures, it seems.  After recording all the tracks, I adjust the relative volumes of the tracks to what I consider to be pleasing. I have an arranger keyboard, which means play a chord, it generates the backing sequence, you play the melody on top. Since each of the backing tracks are at a different volume, you need to play with your mixer to get that relative volume setting back. So, I go through and I set all the audio tracks to give me what I'm looking for. If I do it right, my main mix output should be just below 0dBr. There is no headroom. Now, what I'll do is to group all the audio together as a single group, and then lower the peak output to -3 or -6, or even -12dBr. At this point, I have not applied any effects whatsoever. Once I pull all the channels down by the amount I think I'm going to need, then I start applying the effects I want, EQ, reverb, compression or whatever I'm using. I keep an eye on the main outs to make sure that I'm not exceeding 0dBr and go into clipping. If I do that, I need to remove all the effects and lower all the sliders some more, to give me more headroom for mixing and effects. In the end, I should get a mix that has very good dynamic range (provided I don't mash it with compression), all the instruments should be clear and distinct, and I should have the peaks at between 0dBr and -3dBr. As to applying this to a live recording, I've never tried it, I'm not sure how it would work, but I'd be willing to bet Bob Harvey would have some good insight into this. Gary
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Gary: I only record in 16-bit -- but my next set of recordings will try to do so in 24-bit. Of course I only have 16-bit BIAB wma files -- but I am not "recording" those. Quote:
...have that peak somewhere between 0 and -3dBFS or -3dBr... ... If I do it right, my main mix output should be just below 0dBr. There is no headroom. Now, what I'll do is to group all the audio together as a single group, and then lower the peak output to -3 or -6, or even -12dBr. At this point, I have not applied any effects whatsoever. ...
I do the same thing, but I have been reading (can't remember where) that there is some sonic advantage to recording the instruments so you don't have to group them all and reduce the gain by 6-10 db. Something like reducing the gain increases the noise-signal ratio. I could be wrong, but that seemed to be another one of the arguments for recording in 24-bit.
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Kevin, Reducing the gain increase the SNR?
Well, I can see how you might think that, and I could also see how you would get there, but I don't think that's true for most cases.
If you reduced all the tracks by 50dB, and then increased the main outs by 50dB to bring the signal up to 0dBFS, then yeah, you're probably decreasing the SNR. But, you aren't. You're decreasing the signal strength enough to allow you the headroom for your effects, which still should bring the individual tracks up to an aggregate output of 0dBFS. If you effects require so much gain that you need to reduce the track level by 50dB, that's going to sound really weird, and probably not good. At least, not for the kind of music I record.
Mac, or some of the other more knowledgeable people may chime in and correct me.
Gary
I'm blessed watching God do what He does best. I've had a few rough years, and I'm still not back to where I want to be, but I'm on the way and things are looking far better now than what they were!
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I feel like I need to step in here and correct some misinformation that is being passed along.
24 bit does not give you any more dynamic range than 16 bit. Dynamic range is determined by the analog side of the signal processing, not by the converted to digital format.
What 24 bit recording does do is give you a better signal to quanitization noise ratio than lower bit depths.
For every bit depth there is an additional 6 dB of SQNR. You can go off and google and wikipedia this to get the complex math but I'll try to keep it simple here.
Here's the way to think about this. Let's pretend you have a pure sine-wave at a fixed peak-to-peak voltage. Let's pretend that we take this signal and we amplify it so that the the peak-to-peak voltage is the same as the peak-to-peak voltage that the A/D converter can handle. You can quantize this into 16 bits, or 2^16 values, or one of 65,536 possible values. The sine wave will have very tiny little stairstepped voltage values into which the wave is encoded.
Now, let's take the same signal, and quantize it into a 24 bit A/D converter, which has one of 16,777,216 possible values. As a result, the stair-steps are WAY smaller with 24 bit A/D conversion than with 16 bit conversion.
The 'stair-steppy-ness' of the signal is the quantization noise. Yet another way to think about this is to think of the smallest possible signal that can be encoded, a signal so quiet that the A/D converter switches between the lowest possible value at zero and the next highest value. It switches back and forth between 0000000000000000 and 0000000000000001. When those values eventually get sent back to D/A conversion, that twitching between the two values tweaks the output D/A and generates an unintended analog output noise: Quantization noise. (For those of you that know binary formats, and the difference between signed and unsigned stuff - bear with me here, just trying to make a point)
For a 24 bit recording, if we adjust the signal down so that the switching back and forth occurs between 000000000000000000000000 and 000000000000000000000001, the lowest two values, and as it goes back to D/A the tweak is much smaller and hence the quantization noise is much smaller. In fact the quantization noise will be roughly 48 dB lower than the 16 bit quantization noise.
Note - no magic with the dynamic range occurs.
The big benefit to recording with 24 bit recording over 16 bit is that one doesn't really have to worry nearly as much about using the full dynamic range of the A/D converter in order to get a nice signal to quantization noise ratio, and the little stair steps that occur in the digitized data are peanuts in comparison to 16 bit.
In other words, you can be quite a bit less careful about it, and just get on with recording.
Bottom line, real dynamic range has nothing to do with bit depth.
Other bottom line, disk space is nearly free these days and switching from 16 to 24 bit doesn't have a hard and fast noticeable cost these days. It's not even computationally expensive any longer with software that takes advantage of multiple computing cores.
If your input A/D on your audio I/O device has 24 bit capability, switch over to it if you haven't already done so.
Back to the OP's original question....
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Thanks very much indeed for all your very helpful adivce and tips. If I only take in a fraction of the advice that has been offered it has been well worth it. I will save this thread on my computer and then refer back to it, when I need to. Thanks again  Musiclover
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I was gonna post similar to Scott, but didn't have the time last night. To simplify it; 24 bit gives more steps between zero input (none) and the clipping point but it doesn't 'prevent' an already clipping signal... although it can be more forgiving when you push the barrier. It's really 'larger' chunks of data, which can buffer the impact a bit (compared to a smaller chunk that spikes at the right spot).
Like Scott said, it is preferable to use it when available.
Last edited by rharv; 06/07/12 05:00 PM.
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What really bamboozled me when I first made to trip from analogue recording to digital is the folks who insist on using analog audio terms to describe digital recording and processing. They just don't work. There are no real analogies to bit depth and sample rate in analog audio. They need to be understood without some ancient tape jockey babbling about headroom and frequency response. The way I look at it, packages of digital audio information are processed to make music. Greater bit depth means the packages are larger. A higher sample rate means the packages are being delivered more frequently. Clipping and digital distortion are not products of either bit depth or sample rate. Keep your track peaks below -0.25 and they are not going to make the nasty noises - it's that easy. 
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Quote:
They need to be understood without some ancient tape jockey babbling about headroom and frequency response.
Hi, all. I think I just recognized myself, there!
Actually, I completely agree with the 16/24 argument, but I think that, sometimes, we assume that better equipment makes better recordings, without acknowledging the difference made by the skill of the operator.
In my view, the differences between 16/24 are hard to detect, whilst the skill of the operator makes a huge difference. Some of us who remember working with tape, when you had a noise floor you could measure on the VUs and nothing was tracked without a compressor to preserve the s/n ratio without switching in the Dolbys, are quite happy to stick with 16 bit.
Just sayin....
ROG.
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Band-in-a-Box® 2026 is packed with major new features, enhancements, and an incredible lineup of new content! The program now sports a sleek, modern GUI redesign across the entire interface, including updated toolbars, refreshed windows, smoother workflows, a new dark mode option, and more. The brand-new side toolbar provides quicker access to key windows, while the new Multi-View feature lets you arrange multiple windows as layered panels without overlap, creating a flexible, clutter-free workspace. We have an amazing new “AI-Notes” feature. This transcribes polyphonic audio into MIDI so you can view it in notation or play it back as MIDI. You can transcribe an entire track (all pitched instruments and drums) or focus on individual parts like drums, bass, guitars/piano, or vocals. There's an amazing collection of new content too, including 202 RealTracks, new RealStyles, MIDI SuperTracks, Instrumental Studies, “Songs with Vocals” Artist Performance Sets, Playable RealTracks Set 5, two RealDrums Stems sets, and much more!
There are over 100 new features in Band-in-a-Box® 2026 for Mac®.
When you order purchase Band-in-a-Box® 2026 before 11:59 PM PDT on May 31st, you'll also receive a Free Bonus PAK packed with exciting new add-ons.
Check out the Band-in-a-Box® for Mac packages page to find the best package for you.
Holiday Weekend Hours
It's Victoria Day Long Weekend in Canada. Our Customer Service hours are:
Saturday, May 16: Closed
Sunday, May 17: Closed
Monday, May 18: 8:00am - 4:00pm
Regular hours resume Tuesday, May 19th!
Today's the Last Day of the Band-in-a-Box 2026® for Mac Special!
Order before 11:59pm PDT today (May 15, 2026) to save up to 50% off your Band-in-a-Box® 2026 for Mac® upgrade and receive a FREE Bonus PAK loaded with great new Add-ons to use with this new version!
Don't wait - order today!
Check out all the new features in the redesigned Band-in-a-Box® 2026 for Mac®!
Band-in-a-Box® 2026 for Mac - Special Offers End at 11:59pm PDT on Friday, May 15th, 2026!
Order before 11:59pm PDT on Friday, May 15th and SAVE up to 50% on most Band-in-a-Box® version 2026 for Mac Upgrade packages... and that's not all! With your version 2026 for Mac purchase, we'll include a Bonus PAK full of great new Add-ons FREE! Upgrade to the 2026 49-PAK to receive even more NEW Add-ons including 20 additional RealTracks... that's 222 NEW RealTracks available with version Band-in-a-Box® 2026 for Mac!
Upgrade to Band-in-a-Box® 2026 for Mac® today for as little as $49! Check out the Band-in-a-Box® packages page for all available purchase options.
Learn more about the Free Bonus PAK and 49-PAK here.
If you have any questions about which package is the best option for you, just let us know. We're here to help!
202 New RealTracks Released with Band-in-a-Box 2026!
With Band-in-a-Box® 2026, we've released 202 incredible new RealTracks (in sets 468-488) in a variety of genres—featuring your most requested styles!
Jazz, Funk & World (Sets 468-475):
Our new jazz, funk & blues RealTracks include a groovin’ collection of RealTracks and RealDrums! These include more requested “soul jazz” RealTracks featuring artists Neil Swainson (bass), Charles Treadway (organ), Brent Mason (guitar), and Wes Little (drums). There are new “smooth jazz” styles (4), which include a RealTracks first: muted trumpet, as well as slick new smooth jazz brushes options for drums. Blues lovers will be thrilled—there are more “classic acoustic blues” styles, including guitar (5), bass (4), and drums (10) with blues master Colin Linden, featuring understated and tasty background acoustic soloing, plus brushes drums and acoustic bass. There are also new electric blues RealTracks, including electric blues with PG favorite Johnny Hiland (3) and soulful electric slide guitar from Colin Linden (4). If you love funk & gospel, there are great new options this year, including gospel organ (3) from Charles Treadway, as well as new funk, tango, and rock ’n’ roll drums (3) and bass (1). And for big, bold arrangements, we have uptempo soul horns (4) featuring a three-part hip horn section with options for a full mix or stems of each individual horn — plus an accompanying rhythm section (4) of drums, bass, guitar, and electric piano!
Rock & Pop (Sets 476–482):
Our new rock & pop RealTracks bring a powerful mix of requested favorites, fresh genres, and modern chart-inspired styles! We have more of our popular “Producer Layered Acoustic Guitars (15)” featuring Band-in-a-Box favorite Brent Mason. We’ve continued our much-requested disco styles (10), and added new Celtic guitar (5) with a more basic, accessible approach than our previous Drop-D or DADGAD offerings. There are also highly requested yacht rock styles (17), inspired by the smooth, polished soft-rock sound of the late ’70s and early ’80s — laid-back grooves, silky electric pianos, warm textures, elegant harmonic movement, and pristine production aesthetics. Fans of heavier styles will love our new glam metal (13), capturing the flashy, high-energy sound of ’80s arena-ready guitar rock. We also have a set of rootsy modern-folk rock (18), with a warm, organic sound combining contemporary folk textures and driving acoustic strumming. And we’ve added lots of new modern pop styles (16) — the kinds of sounds you’re hearing on the radio today, featuring exciting new drums, synths, and cutting-edge RealTracks arrangements.
Country, & Americana (Sets 483–488):
Our new country & Americana RealTracks deliver a rich collection of acoustic, electric, and roots-inspired styles! We have new country pop (9) with legendary guitarist Brent Mason. There is also a potpourri (14) of bouzouki, guitars, banjo, and more, perfect for adding texture and character to contemporary acoustic arrangements. We’ve added funky country guitar (5) with PG favorite Brent Mason, along with classic pedal steel styles (5) featuring steel great Doug Jernigan. There are more country songwriter styles (8) that provide intimate, rootsy foundations for storytelling and modern Americana writing. Finally, we have “background soloing” acoustic guitar (12) with Brent Mason — simpler, but still very tasty acoustic lines designed to sit beautifully behind vocals or act as a subtle standalone solo part.
Check out all the 202 new RealTracks (in sets 468-488)!
And, if you are looking for more, the 2026 49-PAK (for $49) includes an impressive collection of 20 bonus RealTracks, featuring exciting and inspiring additions to add to your RealTracks library. You'll get new country-rhythm guitar styles from PG Music favorites Johnny Hiland and Brent Mason, along with modern-pop grooves that capture today’s radio-ready sound! There are also new indie-folk styles with guitar, bass, 6-string bass used as a high-chording instrument, acoustic guitar, and banjo. Plus, dedicated "cymbal fills" RealDrums provide an added layer that work very well with low-key folky styles with other percussion.
The 2026 49-PAK is loaded with other great new add-ons as well. Learn more about the 2026 49-PAK!
2026 Free Bonus PAK & 49-PAK for Band-in-a-Box® 2026 for Mac®!
With your version 2026 for Mac Pro, MegaPAK, UltraPAK, UltraPAK+, Audiophile Edition or PlusPAK purchase, we'll include a Bonus PAK full of great new Add-ons for FREE! Or upgrade to the 2026 49-PAK for only $49 to receive even more NEW Add-ons including 20 additional RealTracks!
These PAKs are loaded with additional add-ons to supercharge your Band-in-a-Box®!
This Free Bonus PAK includes:
- The 2026 RealCombos Booster PAK:
-For Pro customers, this includes 27 new RealTracks and 23 new RealStyles.
-For MegaPAK customers, this includes 25 new RealTracks and 23 new RealStyles.
-For UltraPAK customers, this includes 12 new RealStyles.
- MIDI Styles Set 92: Look Ma! More MIDI 15: Latin Jazz
- MIDI SuperTracks Set 46: Piano & Organ
- Instrumental Studies Set 24: Groovin' Blues Soloing
- Artist Performance Set 19: Songs with Vocals 9
- Playable RealTracks Set 5
- RealDrums Stems Set 9: Cool Brushes
- SynthMaster Sounds Set 1 (with audio demos)
- iOS Android Band-in-a-Box® App
Looking for more great add-ons, then upgrade to the 2026 49-PAK for just $49 and you'll get:
- 20 Bonus Unreleased RealTracks and RealDrums with 20 RealStyle.
- FLAC Files (lossless audio files) for the 20 Bonus Unreleased RealTracks and RealDrums
- MIDI Styles Set 93: Look Ma! More MIDI 16: SynthMaster
- MIDI SuperTracks Set 47: More SynthMaster
- Instrumental Studies 25 - Soul Jazz Guitar Soloing
- Artist Performance Set 20: Songs with Vocals 10
- RealDrums Stems Set 10: Groovin' Sticks
- SynthMaster Sounds & Styles Set 2 (sounds & styles with audio demos)
Learn more about the Bonus PAK and 49-PAK for Band-in-a-Box® 2026 for Mac®!
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