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Everything posted by MLXXX

  1. Yes that would provide an interesting set of comparisons. _______ (Although motherboard sound can provide 5.1 or more analogue output channels, I think in my testing I'd limit myself to Front Left and Front Right.)
  2. That may be your experience, rmpfyf. It isn't mine. If I find the time in coming weeks I might record the output of one of my pc's with its onboard sound and upload it (level-matched) and the source digital file, let's say with both at a 48kHz sample rate (a very common rate these days). I'd then invite forum members to express their opinions as to the extent of audible differences, e.g. "extreme", "stark", "mild", "barely noticeable". That could be an interesting exercise.
  3. It's possible you will hear a slight difference in certain parts of certain recordings; especially if you are able to set up an immediate A B comparison. Many years ago pc motherboard chips could give noticeably poor sound, but that is much less likely with today's motherboards.
  4. Hugo is an entertaining movie in itself, and the 3D is very well done. Wikipedia tells me, "Hugo received 11 Academy Award nominations (including Best Picture), more than any other film that year, winning five: Best Cinematography, Best Art Direction, Best Sound Mixing, Best Sound Editing, and Best Visual Effects.". Not having to wear 3D glasses would certainly be convenient. I usually wear glasses when watching any movie, and I find I forget about the extra pair of glasses resting on my nose when I watch a 3D movie. However I've noticed that a lot of people who ordinarily wear pr
  5. I am mystified by your line of argument. You appear to be suggesting that audiophile electronic devices should be tested based on some sort of averaged out amplitude over time. The closest approach to that I can think of a is a music power rating for a power amplifier. More generally, such an approach would be quite arbitrary, as there is no agreed figure as to what "averaged out" RMS level recordings of music contain. Pop music tends to use a narrower dynamic range than classical music. However some movements in a piece of classical music might be loud almost throughout; or con
  6. There may have been some confusion. Let me try to clarify. 1. The discrepancy in level of the lead-in tone between the raw files kukynas supplied for the RME and Topping was just under 0.2dB and under very good conditions such a difference in level could, of itself, be just audible for some listeners. (A lot of people though wouldn't be able to hear such a small difference in amplitude even with a continuous tone and even with immediate A B switching. And older people might not be able to hear the 11.025kHz tone at all.) 2. However, if kukynas's files are adjusted i
  7. Recording the cartridge preamp output of a track on an LP for multiple playings would deliver variations that might swamp other variables such as arm pivot height. A minor amount of record warp, a little bit of of bounce of the stylus in the groove, and no two playings would deliver exactly the same waveform captures by an ADC. (People sometimes refer to groove wear but I understand that should be minimal with a non-worn stylus and correct stylus weight and anti-skating.) And then there is the question of audibility of a very small arm pivot height adjustment. More likely to be po
  8. Kukynas's files as supplied by him have not been normalised to the same reference level at 1kHz. Below are measurements I've done using the RMS analysis tool available in Audacity. (To load this tool, use the add plug-in option under the Analysis tab.) The RMS measurements can vary in the last two decimal places depending on exactly how much of the waveform you select. The Topping file is 0.153dB quieter at 1kHz, and 0.196dB quieter at 11.025kHz than the RME file. 0.196dB can be rounded to 0.2dB or a fifth of a decibel. (For my personal listening tests I
  9. The FCA1616 audio interface provides 8 balanced DAC outputs. I chose 2 of these on my unit at random for my tests. (I never use this interface for listening to music per se, only for making and quickly reviewing recordings of live music.) I've never looked into the DAC filter design, e.g. whether it uses a slow roll-off filter. (I've seen largely favourable comments on the net about the DAC performance.) I might try to get an answer by practical means, e.g. using an ADC operating with a 96kHz sample rate to see how the reconstruction filters of the FCA1616 DACs seem to apply for a
  10. Yes of course. It is almost too obvious to say, but if the treble has been rolled off by a DAC, that roll-off will remain after the DAC signal is amplified, assuming of course a standard amplifier with a frequency response that is flat.
  11. Yes whatever the extent of high frequency attenuation at the output of the DAC was, it would duly find its way to speakers, after being amplified. Actually I'd assumed the interconnect cable was only 3m in length; 5m is getting a bit long! I may pick up shorter lengths of cable and retest. (Another aspect is that I used a preamp input of the interface, not a line input, as the line input gain was very low.) I'm happy though that the audible quality stood up well with some many trips through the interface for file 4, including going through 50 cumulative metres of inter
  12. The picture below shows how the files looked on the DAW (digital audio workstation) display on the laptop I used for the re-recordings. The Behringer FCA1616 interface was connected via a USB cable to the laptop. The DAW software was Tracktion Waveform Pro 11. The analogue cable used for each of the two channels was 5 metres of stereo cable with 6.5mm plugs at each end. The capacitance of that cable would have contributed slightly to the attenuation at 11.025kHz. The slanting text was added later and identifies the files with the same labels I have used in previous posts.
  13. The number of round trips involved in creating file 3 was 5 (a total of 10 conversions between digital and analogue domains). File 4 involved 10 round trips (a total of 20 conversions between digital and analogue domains). This table shows the progressive attenuation of the burst of 11.025kHz sine wave at the start of file 1, relative to the burst of 1kHz sine wave at the end of file 1: 11.025kHz
  14. Hi, well the proof of the pudding is in the eating! I've just finished analysing the re-recordings. At 11.025kHz there was a loss of 0.55dB ± 0.03dB in each channel for each round trip through the FCA 1616 interface, compared with the 1kHz level. It was merely the same slight attenuation each time! I'll provide detailed figures when I have more time to post. (Will also divulge how many round trips were involved for files 3 and 4.)
  15. As shown above, I have had to correct my measurements for @kukynas's files. Although the amplitude of the 1kHz signal could be measured in various ways, the 11.025kHz signal required not only the highest sample value to be ascertained but the peak value of the reconstructed waveform. There is a normally hidden analysis tool for Audacity (an RMS analysis) that gives the proper result.
  16. This tangential discussion introduced by andyr was on the topic of finding an explanation for a "wow" change to the apparent position of a female vocalist. Such a dramatic change would not arise because of a minor inaccuracy in an amplifier. (In fact it ought to be possible to substitute 10 different standard amplifiers from different manufacturers and hear either no, or only slight, changes in the apparent sound stage.) Assuming the people involved in andyr's anecdote didn't accidentally bump one of the speakers when changing amplifiers, we are left with the conclusion that at
  17. In my reading of previous posts, no one has suggested a single measurement to explain depth. As I say, all the amplifier needs to do is to reproduce the signal accurately. If it does that, any depth cues in the recording mix will be properly revealed. [Subject of course to speakers or listening room acoustics not "messing" with the sound.]
  18. As the question of what to measure for an amplifier that seems to alter the sound stage radically is still being discussed, I'll add in my 2 cents worth. First off, width and depth are perceived by the ears in conjunction with the brain. You could program software to mimic that perception function though it wouldn't be straightforward and might only be feasible for tracking one instrument or vocalist at a time. Information needed to perceive depth includes: * compared with what could be expected if the source were close, the higher frequencies seem dull * compared with w
  19. Yes it is true that the effects are not separated, but surely it is a more arduous test for audio to be put through the gauntlet of two domain conversions, than just to be subjected to one or the other conversion. There is no suggestion of "compensating errors" as between the DAC effects and the ADC effects. Thanks.
  20. I don't think the ADC ever "misses" the continuous 11.025kHz sine wave at the start of the file The DAC or the ADC (or both of them, each to some extent) merely attenuates that sine wave to some extent. It is still there at each re-recording, just at a progressively more and more reduced amplitude (relative to the amplitude of the 1kHz sine wave at the end of the file). Anyway we shall see, if someone does the calculations. (If no one else does, I will.)
  21. If a DAC (digital to analogue conversion device) were of very poor audio quality then even a single use could be expected to result in quite noticable degradation. Similarly, If an ADC (audio to digital conversion device) were of very poor audio quality then a single use of it could be expected to result in quite noticeable degradation. File 2 involved two conversions: DAC followed by ADC File 3 involved 2 times "X" conversions: X times (DAC followed by ADC) File 4 involve 4 times "X" conversions: 2X times (DAC followed by ADC) File 4 has been reported in
  22. The scenario was: 1st pass: Play file 1 on interface DAC connected through an analogue balanced cable to interface ADC to record file 2. 2nd pass: Play file 2 on interface DAC connected through an analogue balanced cable to interface ADC to record the next generation file. ... Xth pass: Play immediate prior generation file on interface DAC connected through an analogue balanced cable to interface ADC to record file 3 ... Yth pass: Play immediate prior generation file on interface DAC connected throu
  23. Thanks, guesswork is just what I was looking for! [I note It would be possible to get an answer by calculation, by measuring the extent of the progressive attenuation of the 11025Hz tone with each rerecording.]
  24. As previously disclosed in this thread, for headphone listening to the test files I brought out my rarely used Asus Xonar Essence ST "audiophile" sound card (still sitting in an old pc). It had plenty of grunt to drive my Sennheiser HD800 phones. I didn't feel the headphone sound was noticeably different compared with plugging into the headphone socket of my integrated amp (which was fed via HDMI audio from a pc). The slight nuances of difference in the recordings that I perceived were noticeable with either way of driving the phones. The integrated amplifier is a Pioneer VSX-11
  25. I agree with the above. Headphones although they can reveal certain detail tend not to do well at presenting the soundstage in a way that seems realistic. I will use both listening methods if attempting a careful comparison.
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