Hi,
I'm having an issue with midi tracks not being played back consistently. When I hit play and then stop and play again, there will be differences in the timing of notes.

I am aware that Round Robin can cause issues like that, but I don't think that this is active in the patches that I tried it with. I am also aware that instruments can be poorly sampled with differing start times across notes, but if that were the case, I'd be able to 'fix' it by manually moving midi notes around. But my problem is that any change I make will not prevent the track from sounding different next time, or in five playbacks or ten.

This example is Audio Imperia Jaeger with Kontakt 6.1.1. (Multi Patches -> 01 Strings -> Spiccato). But this also happens with Metropolis Ark 1 and 2, and also with Nucleus.

The midi looks like this, quantized 16th notes:


And it sounds like this:
https://voca.ro/1gLFw52vn9Fs

I start and stop playback a few times. It's subtle in this recording but you'll notice that it's not even, and that this happens in a non-predictable way. Sometimes it sounds as if there's 'swing' like in a drum sampler, and then there'll be random hiccups, and the hiccuping notes are different.

This is in Reaper 5.99, however it also happens in Studio One 4.14. Which leads me to believe that this is either a Kontakt problem, or a problem with these libraries, or that something is wrong with my audio setup. I use a Soundblaster ZxR on Windows 7, with the included Asio driver. Whatever it is, I suppose it must be on my end because this is a really noticeable issue in a piece with multiple instruments and I never heard anyone talking about that in reviews of these libraries.

 

Round-robin? Humanization? Multi-sampled real strings are not (mathematically correct) synths. Their amplitude envelope all differ slightly (each individual sample). That's also one thing that makes them "real" and more enjoyable to listen to.

Edit: Waitwut? Soundblaster? Ahh, that brings me back (to Soundblaster16. It sucked but it was fun times).

 

Last edited: Feb 15, 2022

Actually, you need another sound card that is full duplex capable, or a USB audio interface. The Soundblaster is very good for listening to music but not really suitable for Reaper and Studio One.

What Is An Audio Interface ?
https://audient.com/tutorial/what-is-an-audio-interface/

Audient iD14 MKII - USB 3.0 Audiointerface
https://audient.com/products/audio-interfaces/id14/overview/
www.thomann.de/de/audient_id14_mkii.htm

 

you are using very old hardware and this is a legacy system?

 

@Baxter
I ruled out Round Robin which is as far as I can see not turned on, same with humanization which the instrument has no settings for. My problem is that the playback of the samples is not consistent. The general rubato or whatever that you imply, it's something I can program - but it won't play it back consistently.

@BEAT16
Thank you for this. I didn't realize the Soundblaster ZxR would not be capable enough. With that said, what does Full Duplex capability have to do with this - doesn't that mean that a card can play back and record at the same time. This should play no role if I play back a midi clip, right?
Is there anything else that the card lacks for anything that would be connected to midi timing issues? Because it was quite an expensive card.
I was always wondering about the difference between sound cards and audio interfaces. I've gotten used to the really great sound of the ZxR and I'm worried that an audio interface won't sound as good. Also, won't an external interface be slower or somehow worse because it's not PCIE?

@clone
Yeah it's an old machine, but I've never heard of anyone having an issue like that which happened just because it's an old system. I know several producers who use machines that are similarly old, or even older, than my machine.

 

You can't rule out anything if you don't know the whole scripting of the Kontakt Instrument. Most probably the "problem" is intended to prevent sounding mechanical with repeating notes. Most modern orchestra libraries are doing something like this.

That is soo 90s, as with ISA or early PCI plug in "sound cards" ;-) Rendered audio isn't affected at all by the playback device, it's all computed by the CPU only.

 

I really doubt that the ZxR is not full duplex capable (24bit, 192kHz, PCIe, various digital ports). I've used smaller ones for making music with Cubase.

My Babyface is even older.

 

Cactus Music says:
I certainly don't believe a word of ocenskate's post,, notice the post count. If any of it where true there's be a lot of people using these cards. He's claiming performance you only get with the higher end cards like RME or Lynx. That would certainly be "to good to be true" at 1/10 th the price.

I'm not suprised by the playback latency. I'm not sure if Creative has spent any money on good drivers,,,ever,, but there's only one way to find out and that's perform a loop back test and see if the tracks line up. I highly doubt it. MY old Audicy card was off by a mile. It was the reason I joined this forum looking for answers..the answer has not changed,, buy a PROPER audio interface..not a sound card.. sound cards are for gaming and playback of movies and music.
Source: http://forum.cakewalk.com/Sonar-X3-playback-latency-issue-Sound-Blaster-Z-m3157458.aspx

Spoiler: Soundblasters are great for gaming, listening, exc but if your doing even simple DAW work, a soundblaster can not keep up unless your using ASIO4ALL.

Early on I had the unpleasant and unfortunate experience of using a soundblaster soundcard on Windows XP.
Soundblasters are great for gaming, listening, exc but if your doing even simple DAW work, a soundblaster can not keep up unless your using ASIO4ALL.
I highly discourage anyone from using ASIO4ALL.
and I don't believe it is just (as you say) a selected handful of "Parroted responses".
ASIO4ALL can become very unstable. trust me I know, ive used in in a pinch.
I also used to suggest it. I don't anymore. ive just seen to many conflicts, problems and issues that lead users in a circle thinking there is something wrong with their system.

you want to praise soundblaster and use it...be my guest.

Spoiler: "I don't recommend a Sound Blaster for DAW work."

"I don't recommend a Sound Blaster for DAW work." AND " I see that 1ms is also claimed for this card but I get nowhere near that." AND "A little research into recording with DAWs would go a long way, Mr. Grundberg"

These 3 comments suggesting against SB, and one suggesting the OP to do just "A little research", when these are the same parroted comments I've noticed over the last 15 years+ in an ever changing line of offerings. The sentiment made me want to comment just to straighten out what I see as a percentage of people who keep "rubber stamping" to not use SB in a DAW year after year, even though A 2010+ card is so different than say a 1995, 2000, or 2007 card etc.

When SB isn't working in a system, I can count on forum members to show up with several one sentence remarks saying not to use SB, when I, and many others are likely getting lower latency on some SB card versions than a lot of posters, likely getting 5-10 times the lowest rate possible they are getting, or yes, I'm saying 1/2 ms one direction, or 1ms round trip. I've also gotten .8ms before round trip. I can actually comfortably sing live (without direct monitoring) right into sonar using ANY effect I like, and at 1ms it feels live, with a few tracks. I'm not trying to start a war. I want to help, but I think some of the attitudes need to go, and a better understanding is line for newer Sound Blaster cards, on high end systems that are achieving better than some of the ultra pro solutions costing hundreds (not always of course). Also there's plenty of threads where other cards are having all the same troubles. It troubles me to read that and was tempted to make a video (may still do it), to show what is possible, except someone in our family is very ill, and I'm switching things around, and having issues with an SSD and little time. I may still do it as one install on an SSD seems to be getting this with Sonar Producer X3e in 32bit Windows 7 (patched). I'm sure there are people with configurations that are just not doing the trick, but this is also true of many other cards or USB setups etc. And I get to easily keep windows sound going, and with a special program can even record youtube videos direct into Sonar, or record Sonar out to windows. Not to say it can't be done with other cards. Just saying, SB's can keep up just fine.

That said, I have the Sound Blaster Titanium HD, and with Sonar X3e Producer and I'm getting, at best, 1/2 ms each way, by setting to 1ms, and sample rate of 88200 and 96000. No crackles or pops. I achieved this at 44100 at some point too, but trying to remember what I did. This is not with one track and one effect or soft synth either. The problem is understanding, and there are so many settings, I have obtained this several times, even with a good amount of plugins. But some things alter it, like what plugin I use, or the master clock, or the way it's configured in windows itself, which inputs/outputs and more. I got 1ms round trip and had several tracks like about 5-8 or so running CA-2A, BBE sonic maximizer, CW TTS 1, EQ's going etc, TH2 Producer, Slate virtual tape, and regular audio on an Asus i7 4770k at 4.5ghz and fast ram, with this SB card. So it's not a LIE. But make a few wrong settings, and it all changes. So people are likely getting different results here. Maybe I should post my exact settings sometime. I also can record in WDM/KS at 4-5ms, but prefer starting with ASIO at 1ms and winding up with 4-5 ms, then winding up with 10 or over.

As I started getting upwards of 4 or 5 tracks and loaded them up with multiple plugins in each rack, moog synthesizer (Yes, it was going), it was still good, but nearing the point of crackling. As my project grew, I had to then either freeze, or change to 2ms, then 4ms, and sometimes 5-7ms, but usually not. Had they been more like pure audio tracks, I could have likely gotten 16 tracks at 1ms-2ms. Funny thing is when I did a re-install to another SSD and did my best to set things right, I had forgotten some little things, and did NOT get the same results, but close. I'm still tweaking it and struggling, proving to me settings matter! Yet I still have a backup of the original. The most amazing thing (to me) I did was for fun to remix the original real 24 multi-track (21 tracks) of Madonna's "Lucky Star", and I needed some compressors, a bit of tape saturation emulation (could have used perhaps other plugins, but this worked), a touch of reverb on some things, EQ, BBE other various. And wanted to make it sound like the original for a game. At first it sounded really close, then I got lucky, and now it's so close, it sounds so much the same as the original, it's hard to discern differences in mix and dynamics/punch etc. However I couldn't pull off 1ms with 21 tracks and that much going on.

But 4ms or 5ms and 21 real audio tracks running plus a bunch of FX isn't shabby. If I'm wrong it was no more than 5 or 7, but I feel fairly sure I was at 4-5. That is astounding for the ever disliked Sound Blaster. But what bugs me is it's elusive on large numbers of tracks and that was my complaint, until I saw others getting worse results than me. I have no problems getting some mixes with 8 or so channels and effects at under 4ms and sometimes 1-2ms round trip. Sometimes it's true, it does not work out as well. But the new SB cards, when everything is set right, and you have a beast of a CPU, and use the right drivers, it really can achieve very low latency. I hope this helps some realize that SB's best cards have some pretty fast DSP hardware on them. I'm still experimenting and learning, and also curious what I might obtain with a different card. But not sure if I will beat the results I've obtained. And not everyone is going to get the exact same results unless they have the exact same hardware or better. I hope this helps open some people's eyes that SB shouldn't be simply tossed out, but a better look needs to be taken, IMO.

Source: http://forum.cakewalk.com/Sonar-X3-playback-latency-issue-Sound-Blaster-Z-m3157458.aspx

Spoiler: As far as I know, the Creative Soundblaster ZxR is more suitable for 5.1 surround (for games and movies

As far as I know, the Creative Soundblaster ZxR is more suitable for 5.1 surround (for games and movies), while
the audio interfaces wei Audient or Focusrite is mainly for music production and multitrack recording.

Latency
Consumer card's drivers (or hardware) are not made with audio and low latency in mind.
A dedicated card, or interface for audio will always offer better latency.

ASIO4ALL
You can try the driver ASIO4ALL (www.asio4all.com 1) - if you are lucky, it will recognise the Creative Soundblaster ZxR. Maybe the ASIO4all driver works better with those sound cards than Creative's own drivers do Gamers basically couldn't care less about ASIO and low latency in real-time application. So why would Creative, or any other brand, put much effort in making general purpose, or gamer cards work with particularly low latency.

Creative was first to offer ASIO drivers for consumer sound cards and their drivers are quite good.
For running VST-s it might be good enough. But a proper audio interface is definitely a step up.

Integrated soundcards still do not cut it, imho. There have been tremendous improvements since the first integrated soundcards became available and today most of them are perfectly good for gaming and listening music. But a Creative SoundBlaster or Asus Xonnar cards are still better quality. And of course proper audio interface is still a step up.
With Soundblaster you get a DSP chip that has Dolby codec, DTS codec, Surround processor,
virtual 3D, Bass boost, Noise and echo cancelation for mic .... a gamers dream.

With an audio interface you a DSP chip with low latency, direct monitoring, high sample rates, good converters ... a musicans dream.
A PCIe card for gamers is the wrong product for you. Firewire/USB audio interface is better. PCIe audio interface is best, but expensive.

Source: www.kvraudio.com/forum/viewtopic.php?f=16&t=487063

Spoiler: Sound Blaster sound cards, is an ideal all-round solution for your PC gaming and entertainment needs.

Sound Blaster sound cards, is an ideal all-round solution for your PC gaming and entertainment needs.

So not really good a "pro" sound card for recording I suspect. I have to admit though I haven't used this card, but most creative lab cards are not up to the job I'm afraid.

FYI zero latency bypasses the DAW, you use the sound cards monitoring software (if supplied) to get a direct output. Anything that goes through processing has some sort of latency. So you could play a track on Sonar with the lead guitar muted, and listen to the lead guitar played live via the sound cards software at the same time. So you would be hearing just the guitar at "zero latency". Sounds great but in practice most people don't bother as the latency is very little (assuming you have adequate hardware and it is correctly configured).

Also, there'll be some on-board soundchip (usually realtek these days) on your mobo, you may get better performance from that via asio4all than the sb.

You're not doing audio recording. You're doing midi recording and live playing of a software instrument. So the blurb doesn't actually apply to what you're doing. Soundblasters aren't made for this kind of thing, though sometimes you can get lucky. Use its own ASIO driver and see what happens.

Source: http://forum.cakewalk.com/Sonar-X3-playback-latency-issue-Sound-Blaster-Z-m3157458.aspx

 

It's most likely just different envelopes per sample. Strings are not drum hits with a transient right at the beginning. It has swell. All samples have different "swell-times"/attack envelope.
If you load a synth (with fastest attack) playing the same quantized MIDI it will be steady and exact, right? Robotic.
In strings you don't generally don't want that super-tight timing, hence why some people insert humanization and randomness (to timing, velocity, etc). It's why composers offsets the MIDI that is triggering slow-enveloped audio so that the fade-in reaches the highest amplitude on time. If this isn't fixed, the strings seems "dragging" and gives a feeling of being tired and slow (when they need to be snappy on the grid).

Rubato is fluctiating (emotionally enhancing) tempo overall, not on timing per se. It's not playing "sloppy".

 

Last edited: Feb 17, 2022

@waverider I'll tell you a secret, Kontakt is a sampler do you know how a sampler works?

 

@BEAT16
Thank you again. Most of these posts talk about latency which is not my issue. I don't care about latency and never understood what the big deal is. When playing an instrument live, then sure, I tried to bring it down, but it doesn't concern me when working with my DAW.
The question is, does the ZxR have anything that could explain the inconsistencies in the playback of the individual midi notes?
Would you be telling me that even with the cheapest audio interface I'd get better midi timing consistency than with my ZxR? If so, can you recommend a cheap one that I can still hook up with USB2 or USB3 (not UBS-C) and that won't sound much crappier than the ZxR?

@Baxter
It's not supposed to sound like this. Very slight and subtle humanization is one thing, but getting shifts in time that amount of swing settings (or dotted rhythms) randomly is just not right.

@Recoil
Yeah I do, but I don't see why you would ask me that question in that smug way? I am aware of things like Round Robin and poor sample start times, as written in my post.

 

@waverider try to load this problematic instrument, navigate to Instrument Options -> Mapping Editor, and check how many samples it uses, if only a few, then this is your answer that's quite normal.

 

Your Creative sound card has a "Brown Burr" chip inside. The Infrasonic - Quartet - PCI Soundkarte - has a Burr Brown Chip too.

Strong hardware points of the new product also include new promising converters (AKM AK4620B). We've already seen them in the ECHO Audiofire 4 and RME Fireface 400.

The AK4620A is positioned for professional sound cards, DAW, and other equipment. The codec supports up to 24-bit 192 kHz for recording and playback. Signal to noise ratio is 100 dB, dynamic range - 113/115 dBA.

The PCI sound card "Infrasonic - Quartet" is no longer manufactured. Only used. I also used it for years.

1.) Infrasonic - Quartet - PCI Soundkarte - Burr Brown Chip - 40 €

Original German Language:
www.ebay-kleinanzeigen.de/s-anzeige/infrasonic-quartet-studio-soundkarte-4-i-o/2007697541-74-1986
Transaleted with Google Translator
https://www-ebay--kleinanzeigen-de....l=auto&_x_tr_tl=en&_x_tr_hl=de&_x_tr_pto=wapp

2.) Developer Infrasonics - Info & Drivers (Go to " Products --> PCI Audio Interfaces)
http://www.infra-sonic.com

3.) Infrasonic Quartet Sound Card - Review
http://ixbtlabs.com/articles2/proaudio/infrasonic-quartet.html

4.) What is latency in music production?

Audio latency is one of the things that you will pay more and more attention to, once you're getting more skilled at music production.

Latency is the time delay it takes for your hardware and software equipment to read the sound signal that's being played, process it, and play it into your speakers.

To be more in-depth, when an audio signal is being played on any device, let's say microphone or guitar, for example, that sound is in analog form and being delivered into your system. Your software then converts it into a digital signal in order to be processed, and then changes it back into an analog, for it to be played into your speakers/monitors.

This whole process, having low processing power in your machine, not having the correct settings and tools optimized for lowest latency output, and a couple of other factors, are all potential reasons that might cause a high latency.

Having a better setup will of course help in reducing latency, but there are other things you can do to improve your latency.

One of the most important things you need to do if you want to start producing music is to get an audio interface. Buying an audio interface is going to be a lot more reliable than using the cheap sound card your pc or laptop has built-in. It also allows zero-latency recording, with its built-in "direct monitor" option. Just remember to update all of the drivers for it first.

Buffer size is the amount of time spent for processing the audio, measured in ms. Installing an audio driver like Asio will give you the option to increase or decrease your sample rate. Having a lower sample rate is going to improve your buffer size, but it's a lot more taxing on your system.

Use the lowest sample rate when you are recording, and set it to a higher rate when mixing, this will spend less of your computer's resources, and allow you to use more plugins without stutters and errors.

https://transverseaudio.com/tip/what-is-latency-in-music-production

 

Last edited: Feb 16, 2022

I asked because I would like a better idea of the overall picture. I was not guessing at what is wrong with it. Now I can do that.

I would like to suggest that you open the computer case, then reseat the ram. power off then a simple remove and reinstall in the same slots.

also, since your legacy sound card is plugged directly into the motherboard; I would remove/reinstall that the same way, too. You may want to switch it to another available slot, if you have one.
Your creative card may have a very small wiring harness and a clip on it. also make sure that is seated correctly if there is one.

 

^
Are you aware of negative track delay ? (measured in milliseconds) for example Metropolis Ark 1 has around -70ms for the legatos & -40ms for the staccatissimo samples,Metropolis Ark 2 has around -100ms for legatos & -30ms for the staccatissimo samples

With Audio Imperia you can go all the way to -250ms in playback.
Also have a look here :

Edit : @waverider ,then maybe it has something to do with your midi velocity settings ?
(not talking here about different round robins, but midi velocity "jumps" happening in playback.
Also did you change anything in your Kontakt settings ?
Also :

• What does DFD stand for?

DFD stands for Direct From Disk and is a technique for playing back large and very large instruments and samples without loading them entirely into RAM. In fact, only the first portion of each sample is loaded into RAM permanently; the rest is read from the computer's hard disk while playing the instrument. RAM is able to react virtually instantly, delivering the first portion of any sample the user requests, while the computer goes to fetch the next portion of that sample from the hard disk. With DFD switched on you can load samples with up to 2 Gigabytes each even with moderately equipped computers.
^ Hope the above helps you & have a great day

 

Last edited: Feb 17, 2022

Does the audio clip you have posted contain different takes (variants) of the same pattern? Did you bounce several variants and then consolidate them in one file?

I may be deaf but they sound IDENTICALLY.

It seems to me that you misled yourself.

Try to compare different takes visually.

 

Last edited by a moderator: Feb 16, 2022

Ok, I've moved the start of every run to the beginning and it looks like this:

Spoiler

As you can see, the waveforms look differently, so there's some sort of variation going on (maybe only due to the different velocities).

More important is how this sounds:

Spoiler: effin window

As you can hear, there is no timing deviation.

 

As I already wrote, it's most probably the Kontakt instrument itself "trying to sound natural" by modulating sound and timing.

The whole discussion about audio interfaces, converter chips and so on has absolutely nothing to do with this, as it doesn't affect the output of Kontakt playing a sequence of notes at all.

Also the word "midi" for a sequence of notes is a bit misleading, because "MIDI" is a standard to connect musical instruments with a physical cable. It's a rather slow serial protocol from the early 80s that indeed introduces some latency and jitter. But these problems don't occur in DAWs playing out sequences to VSTis, just in case someone comes up with "midi latency" or "midi jitter".

 

@BEAT16
Thank you for this, it is very interesting, with that said, I still don't see how that would pertain to my issue. All of what you mentioned might lead to lower latency, and if I understand correctly, latency involves everything I hear and not just an individual note etc (meaning, latency is the same for everything I edit on the screen, to put it in basic words, hope this is correct). So that means that with my older card I might get bad latency, but that would still not explain why upon midi playback, it will randomly play individual samples earlier or later.

@SpyFx ✪ ✓
Thank you, I am aware of these things, that's what I mean in my OP when I wrote

I can fix negative track delay and such by simply moving over midi notes, but that should then allow for the same playback every time I hit play, which it doesn't.

@lbnv
the recording is me hitting play and stop repeatedly, it's the same midi. There are subtle differences between the takes and when I play back in an actual project with going on melodies and stuff I hear it very clearly.

@Myfanwy
Thank you, I encountered Midi Jitter in my internet search but it seems to be only related to hardware connections.
The thing with a 'natural playing' explanation is that this is not natural at all. In no way would all players of an orchestra make the same decision at the same time "now we're gonna play this note too early, and then that note too late". Individual players might do that unwillingly because they're not machines (and then their teacher berates them for it, as did mine back in the day ), but not the entire orchestra which should be able to play a steady rhythm correctly without any hiccups.

@clone
Thank you, I must admit this would be a bit too much for me right now but I'll keep it in the back of my mind.

@No Avenger
Thank you for taking the time. What am I listening to - are you playing back all the takes at the same time?

 

Yep, all five runs at once, as shown in the pic (towards the end you can hear that one or two runs are a bit shorter).