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GuyBorlander

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  1. I don’t get exactly what you mean by this. The version that works for me in regards to it being seemingly perfectly in key with my baseline is the version with the gap before and after , I just print this gap of silence before the beginning and after the Audio file ends. I understand the concept of an anchor now but I cant visualise what you mean exactly with setting the anchor into the wave form? thanks for your patience,
  2. Thanks for the reply - I’ve not heard of term anchor before what do you mean? Cheers
  3. just letting you know that when I drag a perfectly cropped version of the audio sample where it starts immediately and cuts off right when it finishes gives me the GSHARP3 +29 cents result but when I import a version with a gap of silence about a second before the sample starts and after it finishes gives me the G3 -35 cents result. Maybe the algorithm is more "accurate" when there is a a gap of silence before and after giving it time to "think" when scanning. The version with the gap before and after which gives the G3 -35 cents result sounds perfect against the bass line but the GSHARP3 +29 cents result is out of tune completely... maybe something to get the developers to look into. Kind regards, Joe
  4. SOLUTION The 1810c has routing options within the mixer software, which allows the user to set up custom mixes for each pair of physical stereo outputs on the interface. These custom mixes can be created by highlighting the Main, Mix 3/4 or Mix 5/6 buttons and adjusting the faders and mixer settings. I could see that in my setup, i was currently sending both the Stereo 1-2 and Stereo 3-4 outputs of Logic to the 1810c Main out. In order to get the setup working properly, i would need to set different mixes for Main and Mix 3/4 on the software mixer. I downloaded the 1810c manual from the Presonus website and skimmed through the relevant info on routing. I then set up the software mixer as shown in the pic attached. For the Mix 3/4 setup, I reversed the mutes on the first two DAW (Logic) channels. DAW1 was muted and DAW 3 was unmuted. (The setup of Mix 5/6 wasn’t relevant at this point.) This meant that the Presonus physical outputs were now matching the outputs selected in Logic. I could see that the Presonus mixer settings were also responsible for the feedback I was experiencing. All the input channels were turned on with raised faders on each of the mixes. This meant that the output of the SPL would be feeding back into itself. Muting all the mixer inputs stopped this problem and Logic could be used to control input monitoring.
  5. "The tuner might not have a long enough note to lock onto the pitch of a short section in your example. You could try flex pitch, (or Melodyne), for better tuning by individual note features." great idea I will extend the note with polyphonic time stretch so it doesn't change the pitch and will let you know if it works. is there an explanation for the optimise setting on quick sampler giving me inconsistent results? one time it detected a G3 and adjusted cents by -35 and then when I dragged the same sample in another time it detected GSHARP3 and adjusted cents by +29...
  6. the tuner has been giving me a real time reading as the sample is playing through but it isn't telling me how much the entire vocal has been pitch shifted away from a perfect AFlat which is the key of the track. it is clear to me that this vocal has been pitch shifted up a few cents because my bass line written in the ket of AFlat sounds slightly off. There is one note that is sung in the vocal which sounds particually off during a certain part of my written bass line - I cut this note out of the audio sample and played it on a new audio channel with the tuner plugin attached; the results are inconsistent every single time... its surprising because its just half a second worth of audio information. (the real time reading of the entire vocal sample being played having inconsistencies is understandable, but surely not with half a second of one note being played.) I use my ears initially and finally of course that is what I'm supposed to do - but it would be great to have consistent results from these meters.
  7. the tuner plugin is giving me a real time reading which is great but it doest tell me the key of the sample so I can't use it for what I'm trying to do - also I've realised that the quick sampler optmize setting is actually inconsistent every time I drag and drop in the audio file; I'm just going to have to use my ears to pitch this sample against my sub bass - nothing else seems to be working better than just using my ears. not sure if there is a solution to this. Kind regards, Joe
  8. Will this find the accurate key of the entire vocal sample or will it just lock into the last note of the sample as it plays out? I’ve used it whilst tuning individual notes before but not to detect the key of an entire sample.
  9. the audio is from an acapella b side ripped from vinyl highest quality possible - it’s a clean recording good performance too. what tuner plugin can I do this? Maybe Next time I can do this first and then just manually deducted cents in quick sampler by ear. Thanks for the reply
  10. Hey! I have been writing a sub bass line for an acapella I ripped from off of an old dance record - the baseline is in key but there is one note on the bass line that feels slightly off I came to the conclusion that the vocal has been shifted slightly higher in pitch I dropped the vocal into quick sampler with the optimise option and it has deducted the vocal by -35cents… the bass line also now sounds perfect and none of the notes feel slightly off. 1. Am I right by saying that the optimise function examines the file and then gets it perfectly in key to whatever key closely matches the file… how accurate is it? Now I know this vocal is definitely slightly pitched up I’m thinking of manually deducting cents to this vocal by ear and see if I can get an even better result- but this optimise function seems bang on and I will probably leave what it has done. I’m pretty blown away by how good it is. 2. How was this even designed in the first place? thanks for your response as usual guys. kind regards, joe
  11. thank you so much des you have really helped me - will I be hearing the processed version of the audio file and definetly not the original doubling over the top? I'm going to set this up later and I'm sure it will work. also I don't have a headphone jack to xlr adapter so I will get one - I've never plugged In my headphones to the interface.. will this change the signal from studio monitors to headphones simply by plugging it in? (and yes my outputs one and two on interface are plugged into my monitors) and will I have to play with the gain on line input 3&4?
  12. headphones are plugged into back of my Mac mini, I'm having a hard time understanding what you've written. I'm new to all of this - I have only used plugins since I started I don't know how to do a physical set up like this
  13. Hi! I'm trying to set up my SPL transient designer hardware unit to use as an effect in Logic... it has two line in and line out plugs on the back my audio interface is presonus 1810c and has plenty of line in and line out plugs. I also have plenty of XLR cables. the desired end result: I want to be able to have a stereo audio file on loop and in real time as its playing back turn the knobs on the transient designer and listen to the processed signal only with the original audio file muted... basically turning this external unit into a plugin. I can then hopefully print a new processed audio file with the desired effect. I also want to be listening back on my headphones - when I change output device to headphones rather than my audio interface - this stops me from setting up the I/O plugin correctly... I've tried so many times to get this right and I just can't (I need to be able to pull this off for my next project, I need quick easy access to this unit and use it like a plugin) please help! if someone can solve this for me you'll be an absolute legend! all I've achieved so far is nasty feedback noises and frustration. kind regards, Joe
  14. Wow thank you for this incredibly in depth response! I’ve taken this into account… I ran some of my own tests I put one of my 32bit samples into sampler with “original” setting and drew out a midi note playing the sound back at original pitch - I then did a duplicate track and exported new version at 24bits . also made sure both sounds are identical volume dBTP (reading of 0.00dBTP) I played them both back together at the same time but with the duplicate (24 bit exported audio file version) phase inverted ; the result is that they cancel each other out completely so that tells me they’re “practically identical” like you’ve said.
  15. I agree would also like to know a solution
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