Cubase SX vst overload
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- KVRian
- 1256 posts since 6 Sep, 2003 from Seattle/Oklahoma
...good pointHanafiH wrote:3ms is the wavelength of a note played at E4 in air.
Wish I could hear well enough to distinguish individual wavecrests.
Amazing that my ears detect the difference between an F4 and E4. Maybe I get a job with the circus.
nF
- KVRian
- 1325 posts since 6 Mar, 2001 from London, UK
Probably better off reading some technical background over at RME:
http://www.rme-audio.com/english/techinfo/lola_lomo.htm
By increasing the latency, reliability increases and the risk of drop-outs decreases. On the other hand every decrease of the latency time leads to higher system load, because the number of indications for an upcoming data transfer increases. For example the Hammerfall triggers an interrupt every 1.5 milliseconds at a latency setting of 1.5 ms (translating to a buffer size of 64 samples), that means 666 times per second!
But not all that is technically feasible is also sensible. A practical example for this: a bass player 3 meters away from the drummer hears the drums with a delay of 9 ms (speed of sound 340 m/s.) Even close to the crash cymbals (1 m, tinnitus guaranteed) there are still 3 ms. With respect to those numbers, a latency of 6 ms seems to be way sufficient. In practice, even with a fixed latency of 10 ms, you can work wonderfully. Anyone can prove this on its own: just delay the audio output of a keyboard on purpose. The difference between the delayed and the non-delayed signal can be felt clearly, but one gets used to it quickly and can play groovy without any problem.
This claim will for sure be doubted by a lot of studio professionals, who declare a production with a variation of few milliseconds in the MIDI timing unusable. They are right, but they don't mean a fixed delay, they mean a varying delay, as caused by Latency Jitter.
Latency Jitter means variation in the latency
http://www.rme-audio.com/english/techinfo/lola_lomo.htm
By increasing the latency, reliability increases and the risk of drop-outs decreases. On the other hand every decrease of the latency time leads to higher system load, because the number of indications for an upcoming data transfer increases. For example the Hammerfall triggers an interrupt every 1.5 milliseconds at a latency setting of 1.5 ms (translating to a buffer size of 64 samples), that means 666 times per second!
But not all that is technically feasible is also sensible. A practical example for this: a bass player 3 meters away from the drummer hears the drums with a delay of 9 ms (speed of sound 340 m/s.) Even close to the crash cymbals (1 m, tinnitus guaranteed) there are still 3 ms. With respect to those numbers, a latency of 6 ms seems to be way sufficient. In practice, even with a fixed latency of 10 ms, you can work wonderfully. Anyone can prove this on its own: just delay the audio output of a keyboard on purpose. The difference between the delayed and the non-delayed signal can be felt clearly, but one gets used to it quickly and can play groovy without any problem.
This claim will for sure be doubted by a lot of studio professionals, who declare a production with a variation of few milliseconds in the MIDI timing unusable. They are right, but they don't mean a fixed delay, they mean a varying delay, as caused by Latency Jitter.
Latency Jitter means variation in the latency
- KVRAF
- 4749 posts since 15 Jul, 2001 from Holmfirth, West Yorkshire, U.K
I didn't say HE was the problem
I said his Lo-End soundcard may not help.
sorry I didn't write war and peace about it.
over to you.
I said his Lo-End soundcard may not help.
sorry I didn't write war and peace about it.
over to you.
Crackbaby wrote:Topaz: Perhaps something inside you should say NO before you submit and tell the topicstarter that he is the problem. It was his third post..
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original flipper original flipper https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=8999
- KVRAF
- 2544 posts since 14 Sep, 2003 from Essex
Hi
Were having some fun here - no?
I remember my first mega-post; the first response was a picture of WAR and PEACE, followed by Devonb (I seem to remember - but I might be wrong - if I am I apologise and will buy you a piece of software of your choice-but it must be under $5) statin Squids pisses on me (or something like that) -but I found it amusing.
I always remind myself when things are getting heated up that if I was sitting in a bar or on a porch with the person who I was having difficulties with (in a thread) it simply would not happen - for a 1,001 different reasons.
It is very easy (well, I would like to think so) in open conversation to give and take - but this type of communication is staggered and not suited to keeping up a flow of communication.
Flipper.
Were having some fun here - no?
I remember my first mega-post; the first response was a picture of WAR and PEACE, followed by Devonb (I seem to remember - but I might be wrong - if I am I apologise and will buy you a piece of software of your choice-but it must be under $5) statin Squids pisses on me (or something like that) -but I found it amusing.
I always remind myself when things are getting heated up that if I was sitting in a bar or on a porch with the person who I was having difficulties with (in a thread) it simply would not happen - for a 1,001 different reasons.
It is very easy (well, I would like to think so) in open conversation to give and take - but this type of communication is staggered and not suited to keeping up a flow of communication.
Flipper.
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- KVRAF
- 2608 posts since 26 Aug, 2002 from here
now lets add a few little extras to the mix
latencies reported by soundcards are unfortunately not true - they list the time the computer takes to send the data - add another couple of milliseconds from the convertors and usually about 1 ms from the card itself (it features a number of buffers to ensure no underruns) - see this SOS article "the truth about latency"
but then what are you listening to all this delay on - cos if your another 3 meters from your speakers you have another 10 ms latency
so all this talk that you cant notice 10 ms latency only applies if your on headphones - cos if your 3 m from your speakers your looking at a real world latency of 23 ms - and trust me you will hear that - still its variable latency that makes it difficult to play and even a soundblaster can cope with that !
latencies reported by soundcards are unfortunately not true - they list the time the computer takes to send the data - add another couple of milliseconds from the convertors and usually about 1 ms from the card itself (it features a number of buffers to ensure no underruns) - see this SOS article "the truth about latency"
but then what are you listening to all this delay on - cos if your another 3 meters from your speakers you have another 10 ms latency
so all this talk that you cant notice 10 ms latency only applies if your on headphones - cos if your 3 m from your speakers your looking at a real world latency of 23 ms - and trust me you will hear that - still its variable latency that makes it difficult to play and even a soundblaster can cope with that !
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- KVRAF
- 4644 posts since 28 Nov, 2002 from Chicago
add to that the latency inhernet in the response time of a MIDI keyboard and the delay incurred by the MIDI connection, typically worth another few ms at least.
Someone shot the food. Remember: don't shoot food!
- KVRian
- 1325 posts since 6 Mar, 2001 from London, UK
Well I’m glad to see so many people bothered to read RME’s piece on Latency Jitter.
Latency Jitter only affects software synths that are played in real time, not sequencers, not audio record and playback. Because the (typically) ASIO host cannot know when the player is going to hit the keyboard, but ASIO requires the sound to be output in gobbets - full buffer loads - the actual latency experienced between hitting the key and hearing the sound can be anywhere between the nominal card latency and double the nominal card latency. Consequently the actual latency experienced by the soft synth’s player jitters – or wanders about.
Many musicians can tolerate playing a keyboard with a constant latency up to 30ms, what no player can tolerate is a constantly shifting latency. That’s why people like Devon feels driven out of their wits when playing at 12ms. The nominal latency is below the Haas threshold and should not make any difference at all, but because the actual latency varies unpredictably between 12ms and 24ms, ho cannot possibly adapt a consistent playing behaviour.
There’s only two solutions to this problem: you either buy very expensive hardware like the RME Hammerfall that can operate very short nominal latencies without loading the PC’s cpu and force the real-time latency jitter below the threshold of the player’s perception (the I just can’t take anything more than 3ms school) OR you can get an audio card that has the profoundly unfashionable double-buffering featuring the SW1000XG has (that’s why its latencies are twice what the buffer sizes would predict) trading in your latency for a 100% jitter-free performance.
Otherwise, you just can’t play a soft synth in real time with sample accuracy - you have to find a sweet spot between what your PC can cope with supporting and what your ears can ignore.
Question: Are soft synths as good as hardware synths?
Answer: it depends on the hardware you use.
Latency Jitter only affects software synths that are played in real time, not sequencers, not audio record and playback. Because the (typically) ASIO host cannot know when the player is going to hit the keyboard, but ASIO requires the sound to be output in gobbets - full buffer loads - the actual latency experienced between hitting the key and hearing the sound can be anywhere between the nominal card latency and double the nominal card latency. Consequently the actual latency experienced by the soft synth’s player jitters – or wanders about.
Many musicians can tolerate playing a keyboard with a constant latency up to 30ms, what no player can tolerate is a constantly shifting latency. That’s why people like Devon feels driven out of their wits when playing at 12ms. The nominal latency is below the Haas threshold and should not make any difference at all, but because the actual latency varies unpredictably between 12ms and 24ms, ho cannot possibly adapt a consistent playing behaviour.
There’s only two solutions to this problem: you either buy very expensive hardware like the RME Hammerfall that can operate very short nominal latencies without loading the PC’s cpu and force the real-time latency jitter below the threshold of the player’s perception (the I just can’t take anything more than 3ms school) OR you can get an audio card that has the profoundly unfashionable double-buffering featuring the SW1000XG has (that’s why its latencies are twice what the buffer sizes would predict) trading in your latency for a 100% jitter-free performance.
Otherwise, you just can’t play a soft synth in real time with sample accuracy - you have to find a sweet spot between what your PC can cope with supporting and what your ears can ignore.
Question: Are soft synths as good as hardware synths?
Answer: it depends on the hardware you use.
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- KVRAF
- 2973 posts since 10 Sep, 2003 from Karlskoga, Stockholm, Sweden
topaz wrote:Dekkie wrote:Dear gurus,
Audigy Platinum
KX drivers
Here's the problem:Dek
Here's the problem: Dek (= Dekkie or Audigy?)topaz wrote:I didn't say HE was the problem
I said his Lo-End soundcard may not help.
Perhaps Dek is a short for something, please explain if you can.
HH and everybody else (but devon and topaz ...
By the way, i belive i have mentioned this before: If you want to try some new rythms for drums, increase the latency a lot and then play something. Since you dont know what you will hear (because of the latency) the rythm can become really cool and a little unusual.