@littlewierdo said:
Not to mention the memory footprint. A full orchestral piece requires 60 plus gigabytes of ram for me, and that is really nothing in comparison with what it could be. There is a reason why 44.1 at a 16 bit depth rate was chosen for CD audio, this is considered just above the maximum quality that anyone can hear a difference. Anything above this is really unecessary, unless you are time stretching samples (think slow motion video, the more frames you have, the smoother it will be). When it comes to orchestral, I dont think time stretching samples is something many people are apt to do.
1. No disrespect, but have you read the thread? This is about LIVE instrument playability quality, not RECORDED/MIDI music file playback sound quality or time stretching.
To summarize, there is a negative effect of latency on pianist's sense of hearing and motor cortex processing as impacted by low sampling frequency. That is, if you press a key and you expect to hear the sound in 3 Milliseconds, but the sound instead comes to you in 6 Milliseconds, that will be as if the sound board has moved from 3 feet away to 6 feet away. A bit wierd to say the least. And that assumes no other latency elsewhere in the sound processing chain. As there always is this latency in every component, the cumulative effect of latency might mean that the piano sounds as if the sound board is 10 feet or more away. And that's before we start to consider the effect of jitter, which can be described as the sound board constantly shifting back and forth several feet, with this perceived movement happening from note to note or even during the resonance of a single note. Aye...very bad indeed for your brain to be 'hearing' the grand piano vibrating back and forth.
Also note that as this jitter is likely random, good luck trying to achieve consistent playback between pieces. And that is true even for playback of MIDI notes in which instruments are closely placed.
There is a way to elilminate the 'sound shifting': By roughly doubling the frequency from 44 to 96 Khz, the latency of the VSL library is cut in half, and that allows for a more consistent and authentic playing experience. And by doubling again from 96 Khz to 192 Khz the latency is halved again and we have more 'latency headroom' to stay under the audibility threshold.
Don't take my word for it, but do take VSL's deafening silience on this topic as being problematic. I suggest if my explanations are not sufficient, please research and understand other trusted sources.
2. As to RAM requirements, this piano at 24 bit 44.1 Khz currently has a RAM requirement of 4-6 GB. Doubling the sampling frequency will half the latency and only require a total of 6 GB * (96/44.1) = 13 GB. Most laptops with any sort of multimedia claims will have 16 GB RAM or more.
To editorialize a bit, VSL is doing an enormous dis-service to themselves and their customers by attempting to ignore the issue. Unfortunately, an uneducated customer at some point becomes a frustrated customer...and that's a net negative for musical creativity and classical music as a whole (LONG LIVE POP MUSIC?).
I have, and its chalk full of misinformation and jumps back and forth between audio quality and latency, I was speaking to the audio quality side of the discussion (my response back to you, did YOU read the thread?).
Latency is a huge, convoluted mess of a problem that cannot be nailed down to just one thing, as some have tried to do in this thread. Audio interfaces do not "perform" better with certain resolutions over others, and yes, I confirmed this with a friend of mine who designs audio interface circuitry for ASUS.
Also conveniently missing from this conversation is the significant increase in processing power (and thus delay) required to play back audio with effects at higher sample rates. RAM access times and storage to RAM access times are another big factor, and then, probably the biggest elephant in the room not even discussed is, where the majority of the latency comes from, the MIDI / USB / Firewire interface.
I work with sounds from all sorts of sources, ranging from Orchestral Tools crazy 24 bit samples to, by comparison, EastWests lighter sounds and latency is of ZERO concern to me. It is within expected tolerances of 15-20 MS, all of which is easily compensated for by Kontakt, Play, and Vienna's engines, which have settings that work very well for this. Very rarely do I have to go back and correct timing, unless I am working with something that needs to be REALLY tight (big percussion sections or many staccato notes that need to be really tight).
Finally, most DAWs, on the playback side of things, have a track delay setting that virtually eliminates this problem. My guess is, you havent set a predelay setting in your DAW, look into it, this conversation becomes a moot point once you discover this setting. Then you run into a different problem, calculating the predelay, or determining what the predelay setting should be, because it is different for each manufacturer of soundfonts, and can even be different from library to library (the general default numbers to set in your DAW are 30, 60, or 120 ms). From there, you can add additional time depending on your standard latency, so I usually add about 15 ms and everything I record and playback is spot on the beat.
