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发表于 2018-5-3 13:19
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来自 山东青岛
headfi上Rob的原文:
So here I am talking about the WTA filter, and I have assumed that the reader is aware of the WTA filter and why the recovery of timing of transients is important - so I assume you know already that more taps gives better accuracy, and much better sound, and that moving from 8 FS (output at 352/384 kHz) to 16 FS (output at 705/768 kHz) is seen as a benefit in SQ due to the improved time resolution of the output.
The FPGA is jam packed - I use all of the available DSP cores, all of the memory, and 99.8% of the logic too. The downside to all this is now the FPGA uses a lot of power, around 800 mW.
The output from the WTA 1 is now passed to another WTA filter (WTA 2) that takes us from 16 FS to 256 FS (output at 11.289.6/12.288 MHz) so now the WTA is working to a resolution of 88 nS.
The filter option is actually a big departure for me - I go for what is technically correct (assuming it sounds best), and don't give options. Now the HF filter is a technically correct option - as using the HF filter with HD files can reduce HF noise from the recording, which is not music but noise shaper ADC distortion and noise - if this gets into the analogue parts it can cause more noise floor modulation, so removing it will make it sound smoother. So having the HF filter is a technically valid option. But the 256 FS filter is always the more accurate option, and will recover the timing much more accurately than not using it. So why did I add this as an option?
Several reasons. I thought it would be cool for people to actually hear the effect of the WTA going from 16 FS to 256 FS. Normally, I do lots of listening tests, and so build up knowledge to allow better designs and future improvements, so I thought it would be good for one to hear the effects of 16 FS to 256 FS. What you hear is an immediate change in the ability to perceive the starting and stopping of notes. This quality is very different to the usual WTA benefits (better timbre, pitch, instrument separation and focus etc) in that being able to perceive the starting edge of a signal (the initial pluck of a string or how the piano sounds when instantly hitting the key) all depends upon timing accuracy going from uS down to tens of nS - so I thought it would be interesting to actually hear what I am talking about directly. Now this filter will be called the incisive option, as calling it Hugo is a bit confusing. Its incisive because you can now perceive the starting and stopping of notes much more clearly - and when the brain can't perceive the leading edges, then it becomes a blur and things sound soft.
The second reason for the filter options is that the incisive revealing nature of the filter does make it sound brighter. Now it is absolutely technically more accurate; it only sounds brighter because the brain can now more accurately perceive the starting and stopping of notes, and the starting and stopping transients have a lot of high frequency energy. When the brain can't perceive something, it simply ignores it, so it then sounds unnaturally soft, in that this is not truly transparent. But sometimes when you have say a bad bright recording, or say hard headphones, having a filter that allows you to hear high frequency energy may be a bit too much. But for sure you are using an aberration to hide another problem. So my advice is this; if you use the 16FS option (orange or red) all the time, then consider getting a warmer set of headphones, or trying out EQ. Normally you should be using white or green - I run with green all the time as its useful with 192 recordings |
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