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Everything posted by John
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Adaptive Limiter - how do I handle volume boost
John replied to craigr68's topic in Cakewalk by BandLab
Ok I think you are looking at the AL as a maximizer and not a limiter. -1dB for the cutoff is too high. To me a limiter is to prevent audio from exceeding a set dB. Unless its a true brickwall limiter and even then you can still have inter sample clipping if you push it hard. I would start reading about Loudness Unites and LUFS. -
Me either. I have never locked anything. What I do is save as an unedited version and save incrementally thereafter with a save as.
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Its a good idea to keep it as a default. Its sort of important to know if you have output or not.
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Sort of like dyslexia of the ears. ?
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Did you try the gain control on the Console?
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I have mine enabled.
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You may need to increase the buffer for WDM if its crackling.
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I'm sure some one will come to your aid soon. However I and many others view ASIO4All as not the best way to access your audio interface. If you have a proper low latency audio and it has real ASIO drivers please use those. If not use WDM drivers. As an aside that is what ASIO4ALL uses.
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Recording audio is not as CPU intensive as some think. What really are the CPU hogs are plugins and virtual instruments. Sample libraries need RAM. Audio is streamed from disk. USB audio devices are very competent especially if they are USB 2. Keep in mine we were doing all this back in the last century.
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I think you are approaching this with an attitude of too much humility. If you ran Samplitude and were able to create with it you have nothing to be concerned about. Cakewalk is a lot like Samplitude. It is object oriented without touting that. I have Samplitude X4 Pro. A desktop computer that is fast with at least 16 GB of memory will do well enough. Fast meaning 3 or 4 to 5 gHz speed will do the job. Graphics cards are not as important for DAWs any decent one with work. If you want dual monitors get one with adequate memory for two monitors. Speed is more important than cores or threads for the CPU. Though more cores are better than fewer. Cakewalk is a very easy DAW to get to know. Yet it is able to do just about anything any other quality DAW can do. Plus you have the forum for those times when you get stuck Also, welcome to the forum.
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I'm from Mars.
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How do I remove unwanted distortion from an audio track?
John replied to EDT's topic in Cakewalk by BandLab
RX 7 can remove clipping. It may also be done with Spectral Layers 6 from Steinberg. As mentioned Samplitude Pro X4 also can. True one can not unbake a cake. However, we are not talking about baking. Sound has a frequency(s) with all its overtones. These can be isolated and edited. Who thought we could adjust the note pitch with ease via Melodyne of recorded sound. Anyone heard of autotune? -
When you say you have tried going through the options in Kontakt what are you doing in detail? Also in Kontakt have you set the MIDI input port and channel correctly? The keyboard needs to be using the same channel. Channel = 1 to 16. In Kontakt there are channel banks i.e. a, b. c, d, and so on. Be sure its on a.
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Actually, members have asked for different or added functionality of many of the tools and features in Sonar and Cakewalk over the years. I'm sure once you get used to the way things work it will seem natural.
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Unable to watching youtube tutorials with WDM
John replied to Eric Swardt's topic in Cakewalk by BandLab
I have just the opposite setup. Cakewalk is using an ASIO driver for my digital mixer. Windows uses a separate system with a different driver. In the past I used the built in audio for Windows now I use an SB card. This gives me the use of any audio from MP4s (showing vidieos)or MP3s on the Windows system while all my DAWs and audio Editors use the digital mixer. The two never interact with one another. I have been doing it this way for many years without any problems. Plus I don't need to have my serious audio on when I'm not doing anything with it. -
I do know what you mean Larry.
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Sonar?
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One of the most common uses of EQ. Good point Jim.
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Before. EdIt I was thinking EQ not the compressor. So I think EQ first than compress. Sorry. Think this EQ is there to control tonal balance. That should be first. The compressor is there to control dynamics . That should be on the output and last. However, it really depends on what you want out of those two plugins. They can interact in interesting ways. Or you can use a dynamic EQ. Really we are at a point where you can do just about anything. It is all about the sound.
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I was hoping no one would ask that.? For many and they are right an RMS of around -12 dB is often used. However, my view is let it be what it is. I see no reason to adjust it up wards or not because it isn't at a proscribe level. To me if the over sound is good and nothing is close to clipping yet it is fine otherwise why doggedly force something where it really doesn't want to go simply to do what some others do? I think we should view any of the target dBs as guidelines not hard and fast rules.
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This is perhaps the most difficult subject to deal with. One can compress to the point any wisp of musical dynamic is gone. One way I think to think of this is putting each instrument in its own dynamic space without letting any other sound over power it. Try to avoid compression at this stage. I think -6 dB to 0 is much too narrow a band for peaks. What about RMS? Where should that be? I have talked about pulling all the faders to zero or infinity to start off mixing. Than bring each one up so it can be heard. Than adjust each so everything is heard well. ( A note on this is if you are doing many instruments of the same sort use a bus for making a section of them. Balance within the section the balance each section to one another.) Don't worry much about making it louder rather make it balanced. Here EQ can carve out frequency space so instruments aren't masking any other instrument. I think once you have things balanced then you can look to the right over all loudness (RMS) then give the peaks lots of room. I think it takes judgement.