I am still waiting for this items: A50 and P50 ~
me too, i hope we will have some good news soon
it expected to be announced on chinese new year
I am still waiting for this items: A50 and P50 ~
me too, i hope we will have some good news soon
it expected to be announced on chinese new year
Hello. Amazing thread and really interesting contributions.
I am between the D50 and a Pro-Ject Pre Box S2 Digital, which has its own thread here, it is around 150€ more expensive and has a headphone amp. But I haven't heard anyone comparing both. The D50 is now 200€ on Shenzenaudio and is a good value, but seeing some people here say it is U-shaped and harsh or fatiguing makes me wary. So far everybody says the Pro-ject is not fatiguing. Yes, it's more expensive but I thought they be comparable. I plan to connect optical or coaxial from a blu-ray to powered speakers in an auditorium.
Has anyone heard both?
I wonder what the problem is with this ultrasonic noise, is it possible to cause hearing damage? I know at least it will mess with the performance of an amp.
this is a really excellent explanation, especially DSD parts. very easy to understandI think it doesn't have impact to our hearing, but could have impact on some amps, like overheating or maybe possible oscillations in some designs. But it can have also impact to audible frequency range by means of intermodulation distortion. For example two peaks in inaudible frequency range, which differ in 7kHz (for example 50 and 57 kHz) generate intermodulation distortion at frequency 7kHz and that's audible if level of that distortion is sufficient.
Look at these pictures: https://audiophilestyle.com/blogs/entry/428-ifi-idsd-micro-measurements/
Author of HQPlayer measured here iFi micro iDSD DAC frequency response up to 5MHz, when sweep 0 - 22.05 kHz was used as input signal. The 1st picture show filterless DAC output and the next 2 pictures show influence of "standard" and "minimum phase" digital filter types - they are provided as option by the iFi DAC. These filters are used yet in digital domain as part of PCM input oversampling - yet before the delta sigma modulator circuit itself. They lower the unwanted aliasing effect of oversampling. Then the next pictures show how the high frequency noise is affected when that 0 - 22.05 kHz sweep is at first upsampled by computer software to 768kHz, DSD256 or DSD512 and then sent to DAC digital input. In such a case the DAC chip has "less work" to oversample the input signal up to MHz range sample rates, which are required on delta sigma modulator input.
The first upsampling (or oversampling) stages have the most significant impact on sound quality and powerful computer can do that job in much higher quality than resource constrained DAC chip. That's the reason why upsampling players like HQPlayer exist. In a special case, when DSD signal is provided on input of for example Burr Brown TI DAC chips, even the delta sigma modulator of DAC chip is skipped and only the DSD signal conversion to differential analog signal and low pass filtering is used. That's the technical reason why DSD can sound better than PCM with delta sigma DACs - also when DSD input is result of PCM source file software conversion.
Now look at page 32 of the TI Burr Brown DSD1793 DAC chip manual: http://www.ti.com.cn/cn/lit/ds/sles075b/sles075b.pdf
You can see that the DAC chip does not provide suitable analog output signal for audio applications. It requires analog LPF stage, some example implementation appears on the picture. Each filter implementation is imperfect, because each filter has imperfect amplitude and phase response so it causes some level of distortion. Therefore less filtered output may sound to us better - if it doesn't cause audible distortion and unwanted effects downstream.
Delta sigma modulator circuits convert PCM input to 1 bit or multi bit delta sigma signal output. The delta sigma signal is of PDM type (pulse density modulation). So information is not coded by amplitude but by phase differences. DSD is 1bit two level delta sigma signal, sample rate is in MHz range. Delta sigma signal (including DSD signal) doesn't require anything more than analog low pass filtering to convert it to analog signal.
The nature of DSD signal is very simple and can be explained for example on a very quickly blinking LED lamp. Imagine a LED lamp which is able only of full light or no light and nothing between. The lamp provides 2 level high frequency pulse signal - a 'pulse' or 'no pulse' can be repeated in any amount and order. Imagine it is blinking (or not blinking) very very very quickly so our eyes are unable to distinguish individual pulses. Our eyes provide low pass filtering in such a case. The LED lamp outputs some number of pulses within some very short time interval. For example 61% of that time interval the signal level was "on" and 39% of time it was "off". It results to perceived light intensity of 61% between the minimum "off" level and maximum "on" level. This is the principle how any analog signal can be coded into 1 bit PDM signal, the only requirement is enough high pulse sample rate in relation to frequency spectrum of the analog signal.