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Computer Audiophile


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About audiventory

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    Yuri Korzunov, software developer
  1. Technically higher tap number allow to make narrower band for higher sample rate (higher total band). Or higher tap number allow to make steeper filter. Ringing level and distribution, flatness of magnitude in pass band, suppressing level and deviation in stop band, phase linearity depend on filter design method. Tap count is result of design. There big number of such design methods. And filter building have element of art sometimes, in my opinion. Personally, I always try to maximally reduce tap count for given filter features. Because each of taps consume calculation resources. In general, necessary tap count is defined by solved task with certain initial conditions (band, sample rate, etc.). As example, 0 ... 30 kHz steep filter in DSD demodulator will have many taps with big probability. I see to filter features first.
  2. As rule, interpolator like spline is not used for 44.1x / 48x range conversion. Read details here https://samplerateconverter.com/educational/resampling But spline can be used in pro-audio processings like 44.1 to 44.2 or same ratio. From my experience, spline have more issues with non-linear distortions than multiple-capable resampling like [44.1 x N] to [48.0 x M] and back. In any kind of resampling difference between resamplers depend on resampling filter implementation and its settings. These things cause different distortion/noise level and spectral content. As example, in our resampling algorithm alphaC we use: integer multiplication/decimation only and individually designed filters for each combination of input/output sample rates. Our filters have settings: filter band modes (Optimized, Non-optimized, Non-optimized Wide band); linear/mimimal phase filter (in Non-optimized modes linear only). If new options is added to our algorithm, we prefer do it under hard factory control to minimization of probable degrading of existing fugures. Other manufacturers give more settings for adjusting. But even so limited settings number, as into our software, may consume many time to search the best sound of given audio system. In ideal case resampler must be "invisible". But it is impossible due ringing and real filter steepness. Difference between good resamplers is very subtle for ears. Unfortunatelly, I know nothing about such professional ear tests. In my opinion, audio optimization (proper band reducing to 20 kHz to reduce audible intermodulation distortions) is more audible at some audio systems, than difference between resamplers though.
  3. Learning of audio equipment and psychoacoustics is great idea. I suppose there are many new things, that may be discovered. And I understand how many efforts and time need to launch and support of such project. I like that to do @Archimago. He describe conditions of his measurements (that I read) in details. You can visit his blog "Archimago's Musings" http://archimago.blogspot.ru/ Our business is pure software. Fortunatelly our conversion software may be checked pure mathematically via programming and several ways. And we use internally developed software analyzer. It is not ready for market yet. To analyze DACs, amplifiers and other equipment need professional hardware+software tools: generators, analyzers, including measurement DACs. Because there is enough subtle differences between values. I don't research current market situation and available tool abilities. So I can't recommend certain manufacturers and models. To publishing of measurement results, the hardware should be periodically checked via reference tools and certified by official organizations. It allow to decrease measurement error probability. It is especially important for public information. As engineer, I don't see other approach in these things. Though, in my opinion, researches with non- or semi-professional measurement tools is allowable, but such researches should not be posed as serious evidence.
  4. After our today morning (by my local time) conversation, I got idea to write article about NOS DAC and done it: https://samplerateconverter.com/educational/nos-dac As rule DAC have several circuits and/or processings by resolutions. We know too few about certain DAC internal works. And we can learn it as "black box". And, probably, some resolutions cause minimal distortions (no any warranties). We can check it by measurement tools. If we can't do it, we can to do sound check (comparison) of different resolutions. This way I described here https://samplerateconverter.com/content/how-improve-sound-quality
  5. Absolutelly. In design we look for optimal figure combination according general aim, not the best for all figures.
  6. I'm software developer (audio converter AuI ConverteR). I have no practical experience of DAC development. It was reason why I long time prepared and wrote the article However I have practical experience of project management, where software and hardware was developed and integrated. There is necessary skill "know how to alter hardware or software inside to achieve a target figures of full system". Digital sigma-delta modulators and resamplers are things that I works directly. These things are same in software and hardware implementations. And hardware implementation of these processings is software actually. However, I suppose, that prcactical DAC developers can give more details. It is need to separate: high resolution, oversampling, NOS, R2R. Each of the points solve own task. High resolution (high sample rates) allow to reduce downsampling aliases in Analog Digital Converter (ADC) when record performed, accounting real implementation of analog filter. Also it allow to reduce digital filter steepness to reduce ringing. Oversampling allow to improve filtering aliases by analog filter in complex with digital filter. However the digital filter ringing artefatcs are there. R2R convert digital code to analog voltage. But it have non-linearity issue. With NOS we try rid ringing artefacts of digital oversampling filter. However it is need to feed the DAC by high sample rate. And there are 2 ways: record on the high sample rate or pre-upsampling. Pre-upsampling add artefacts. And there is question: what oversampling is better - into DAC or into external inline/offline software? There is no univocal answer. However, oversampling/filtering implementation on PC have more abilities for developers due bigger computing resources.
  7. Understanding Sample Rate

    Impulse response so terrible for delta impulse only. Music have a few places, where the ringing is visible. Ringing may be reduced by wider transient band on higher sample rate. But higher sample rate have wider band and can cause more audible products of intermodulations. Reducing of pre-ringing cause increasing post-ringing energy on the pre-ringing's one. Design of resampling filters is compromise.
  8. Understanding Sample Rate

    I'd recommend be very careful for with blind tests.The tests demands serious approach in methodic design, condition control and number of checks. Read details here https://samplerateconverter.com/educational/hifi-blind-test At the end of the page [References] you can see example of prober blind test protocol "One of careful test examples" Ringing is response at output to delta-impulse at input. Read more https://samplerateconverter.com/content/what-ringing-audio "Time smear" is not technical term, but it like to ringing, me seems. Phase is time moment of sine (harmonic). Phase is measured in degrees or radians. It is relative value. Depending on frequency it may be scaled on time in seconds. Phase shift between input and output like to time delay, but phase is relative shift in degrees or radians. Linear phase filter have linear dependency of the phase shift by frequency. For linear phase filter, phase shift between input and output is different for different frequencies, but time delay for all frequencies is the same. Minimal phase filter have non-linear dependency of the phase shift by frequency. And the time delay for different frequencies is various.
  9. Understanding Sample Rate

    I suppose you mean analog filter of DAC. Absolutelly brickwall filter in infinite time restore 100%. But we have not such filter and existing time of spectrum harmonic is limited.
  10. Check my review about R2R Ladder, SDM PCM and DSD DAC concept comparison https://samplerateconverter.com/educational/r2r-ladder-dac-vs-sigma-delta
  11. Understanding Sample Rate

    Higher sample rate give abilities to work in higher frequency range of analog filter, where it have higher alias suppression. See picture "Sigma-delta modulation and analog filter" here https://samplerateconverter.com/educational/r2r-ladder-dac-vs-sigma-delta
  12. Understanding Sample Rate

    This is limited band, because "between..." term there. May be you meant unlimited precision? The precision do not depend on sample rate and bit depth. Analog filter have abour 20 .. 48 dB per octrave (range = 2 times between low and high borders). When we use digital filters we works with -200 dB at stop band of filter. So 200 dB / 48 dB ~ 4 octaves = 20000 Hz * 2^4 = 320 000 Hz. I.e. there is need 300 kHz transient band to proper suppressing of aliases. Thus sample rate should be 320 kHz * 2 = 640 kHz. We can apply digital filter with transient band 20 ... 22 kHz from 0 to -200 dB. And it will almost ideal system. Of course, we can "play" with the transient band to reducing of digital filter ringing. However, may be issues with implementation 0 ... -200 dB and, may be, 0 ... -125 dB (for 24 bit) filter "on chip". Because there may be resource limitation of the chip. -125 dB is taken because electrical noise of analog circuits about -120 dB is expected.
  13. Understanding Sample Rate

    Sample rate guarantee nothing. But if limit frequency band of "accuracy" (0 ... 20 kHz, as example), higher sample rate give easier abilities to proper analog filter building. Real digital-analog conversion always have errors.
  14. Understanding Sample Rate

    Distortions is very useful thing in development of audio tools. It allow to aim and exactly control results of work. Though binding of figures and subjective perception is not simple work.
  15. Understanding Sample Rate

    Digital to analog conversion is like to binding of sample values (its tops) by a "steel-edge ruler". The "ruler as curve". It is interpolation. It made analog filter. Bit depth cause curve form error. We can observe it as noise at spectrum of digital signal. And analog one too. By Naquist theorem sample rate must be 2 times more, than restored curve frequency (if the curve is sine). But accuracy of restoring depend on sample number that bend the "ruler". In ideal need infinite sample number. And analog filter must be "infinite brickwall". Thus sample rate increasing is easier way to more accurate restoring of original sine at given frequency. Bit depth too. But it is no need to worry more than accuracy of spectrum. Because it show any distortions more clear than oscillogram (in proper settings for considered case). Sometimes need to consider spectrum development in time (time-frequency diagram).