I think this is what is happening here. The original rip has been similar enough with streamed content with node type streamers streaming from the usual suspects. Now with the Auralic set up there is a difference that is noticeable from different streaming sources. My choice is to use Amazon for hires content and I have now decided to add to the Amazon library their ‘version’ of all my locally ripped content. I will then probably add the same for the LP’s I have. The aim is to have all music ‘stored’ in the cloud with my original content archived.
Edit: I don’t think I can face the hassle of re ripping all of my stored cd content and don’t want to invest in a high end ripping system.
This may be a question covered previously (or slightly off topic) but if you rip a CD to an uncompressed / lossless file, would it not be an exact replica of what’s on the disc? I don’t understand this point of increasing quality vs the master - like taking a picture and increasing the resolution
I’m clearly no expert but it seems to me that if you ‘up-sample’ an MP3 you cannot add what isn’t there and the resulting larger file - whether FLAC, WAV or whatever - won’t be an improvement on the MP3.
Doesn’t the same apply to up-sampling 16 bit red book??
I’ve been buying a lot of Qobuz hi res music this year and when I play these files from my local server they sound better than the same file from the cloud. I think this is largely b/c I’m on a legacy streamer. I’m under the impression this is less of an issue with the new streamers.
I’m not sure how this relates to the difference between sources that I refer to, but to answer your question, no, or yes, or maybe. The differences noticed or often discussed are to my mind the differences between one manufacturers implementation of ’ re assembling’ the digital transmission against another’s.
I have my own thoughts and doubts about resampling or upsampling and I still haven’t decided what I think. At what stage is a manufacturer of the streamer/dac combination giving us the source file presented in its cleanest state, and at what stage does the streamer/dac start to present something different but arguably better sound quality.
Moving to a hi-res format like 24-bit, this dynamic range increases to 144dB – assuming the equipment is actually capable of working in true 24-bit.
It is a common misconception that 24-bit audio simply records louder and quieter sounds than is possible with 16-bit audio, but this is not the case. Instead, the same range of loudest to quietest is measured, but with 24-bit sampling it is done with considerably more steps than with 16-bit. This means the absolute value of the waveform at any given point can be much better represented.
Upsampling doesn’t add anything in the way of original audio that wasn’t recorded back into what you are listening to … it can’t do that as that information simply isn’t there, but one of the things it does do is effectively move the sampling frequency of the audio file being played further up and away from the audio band so that enables the use of much gentler (and less compromising) audio filters in the reproduction chain after the DAC.
Dunc, can you explain to this digital ignoramus how going from 16 bit to 24 bit is thought to be an improvement but going to just the 1 bit (dsd) is also thought to be an improvement?
I will do my best, i guess you could go as simple as counting to 100, 1st off count in 10 and then one’s, you still get to 100, but if you write it down as you count obviously the single numbers will be much more,this then gives you 100 points compared to 10 points that can be fine tuned.
I think thats sort of right
PCM uses the bit depth to record the amplitude at a particular time slice - at 48kHz there are 48,000 slices per second. The greater the bit depth the finer the graduations for the amplitude steps - like moving from 10mm increments to 1mm. It also allows the studio engineer to stay away from the lowest bits reducing something called ‘quantisation distortion’ where the low number of bits makes the graduations coarse. e.g. if we look a the LSB (least significant bit) there are only two amplitude levels allowed; 1 or 0, if we add another low bit we get four levels; 00,01,10,11. Once get to 8 bits we have 256 levels of amplitude, as you can see the scale is not linear and the more bits the higher the resolution.
Can you check if your CD rips are FLAC, ALAC or WAV, AIFF?
Not looking to start a tussle. Just asking.
How you described the sound of your CD rips was frequently how folks in my AXPONA sessions described the FLAC playback.
At AXPONA I usually get a full room when I run my FLAC v WAV listening sessions.
I even blew a few minds comparing original FLAC MQA to the same file uncompressed to WAV MQA. (Yes, the DAC still authenticated the MQA in the WAV file and decodes.)
The more “1” 's are present the higher the amplitude of the sine wave is,the more “0” 's the lower.
That’s why it is called 1-bit coding. The sampling rate is much higher than PCM,some 2.8 MHz.
The density of the pulses over time can be used to reconstruct the analog audio signal.
No doubt there is much more to it but this basically the technique used.
HTH.
Missed this thread when it started. Re the original observed difference, the question was raised as to whether the same master. This is a fundamental question, which some 20 years ago highlighted to me the difficulty in comparing different sources unless you can be sure the master is the same:
I was having a listening session (pre-streaming) with someone else who had brought round some of his CDs, one of which duplicated one of mine. On playing mine, my visitor noticed something odd, not sounding as remembered, at first thinking it was my system different from his. The long and short of what followed was that the two CDs were distinctly different. On scrutinising the packaging and label, the only discernible difference was that one was made in Germany, the other in UK.
That is why DSD is commonly claimed to be more ‘analogue’ in it presentation.
The other claim is it can be D to A’ed with a capacitor to integrate the pulse-train (yes, integrate like in calculus).
On the last bit I personally have not tried using a capacitor and the bitstream/ pulse-train but I also do not recall hearing of any commercial DACs that support DSD doing so either. I imagine the challenge is the channels are muxed into a single bitstream so you might just get mono.
Also almost always DSD releases pass through a pass where they are DXD which is multibit and very close to PCM.
