guymrob

I believe the authentication flag resides inside the first 16 bit data, probably the MSB, if someone can change that, sample rates display on a MQA DAC can be altered🤣

All MQA certified DACs display the samples are actually authentication flags, i.e. this is similar to Roon displaying 'Authentication' in the signal path. It does not report the actual samples that goes to the DAC chip. This can be very misleading by looking at the DAC's display and assumed it actually 'unfold' to '192k' or even '384k'. To a average consumers, they will take it as it is. In my opinion, this is 'cheating', it does not reflect the inner working of the DAC😂

Sounds like Sony is going to kill off stereophonic if it really takes off. It will generate enormous income for Sony and the music industries. A new way of listening to music?

Hi @Miska I’ve L2, I saw stereophile review that the frequency response when playing back DSD is capped at 19kHz with a sharp cut off while PCM 24/192k frequency response all the way to ultrasonic range. Is this error in measurement or a flaw in the Spring design?

I think the SQ of NOS vs OS DAC boiled down to the transient response. There’s article on a Holo Spring DAC reviewed by Stereophile https://www.stereophile.com/content/holoaudio-spring-kitsun%C3%A9-tuned-edition-level-3-da-processor-measurements in a NOS mode, the DAC is essentially filterless in the digital domain, the impulse response shows virtually no pre and post ringing while a conventional linear 8x OS digital filter creates a lot of pre and post ringings. In my listening test which I own a Spring DAC, I prefer listening to NOS rather than OS mode when playing back Redbook contents. The effects of aliasing happens above the audio range and these will get filtered down once it reaches to the amp and speakers.

I’ve a quote from KR March 2018 Stereophile that can probably sum up what most of us think: “Today, I’m less enthusiastic about MQA than I was. High-resolution streaming and downloads are now readily available …. MQA requires the purchase of compatible equipment, and holds the potential to eventually control all signal processing, such as room EQ. I don’t see a need for it, therefore, and I hope it doesn’t force the elimination of high-resolution, non-MQA downloads.” — Kalman Rubinson. “Multichannel MQA”, Stereophile, March 2018

Here is a review from Stereophile 'MQA Benefit and Cost' by Jon Iverson: https://www.stereophile.com/content/mqa-benefits-and-costs

The sampling rate of SDM(256x @11.22MHz) is so high that aliasing is unlikely will occur if there's nothing above 100kHz. The aliasing into the audio range is a PCM problem not related to SDM, that's the job of a DAC.

Since SDM is basically a 1 bit converter(high bit-rate DSD), it requires noise shaping prior to decimation to Hi-Res PCM. For instance, if the SDM ADC operates at 256x, noise shaping pushes virtually all the noise(depending on type of order; whether is 5th or 7th order modulator). See below: 64x of 44.1kHz, noise starts to rise around 22.05kHz and above frequency band 128x of 44.1kHz, noise starts to rise around 44.1kHz and above frequency band 256x of 44.1kHz, noise starts to rise around 88.2kHz and above frequency band If decimation is used to convert high bit-rate DSD at 256x to Hi-Res PCM, to capture everything between the noise shaped area, a PCM sampling of 176.4kHz is all needed with virtually no noise in output.

Here is a quote from the latest Stereophile April 2018 issue: ...I don’t believe that, over long term, MQA is in the best interest of audiophiles. I just hope it’s not too late — Jon Iverson, “As We See It” Sudden change of heart?

I’m more concern that the DACs we buy today especially MQA ones may default to MQA filters(for whatever reasons) when playing back non MQA contents. It is through technical testing one can see the effect of aliasing and distortion patterns here. There’s no tool at the moment to access MQA encoders other than the music companies themselves. Because of this, it is difficult to fully test an MQA DAC without generating a test signal from the encoder. The only test at the moment rely on MQA music contents. DSD and PCM have long been able to qualify technically with test signals for each hardware release from the manufacturers. This guarantees high standard of specifications and the rest is just listening.

Has anyone able to playback using Roon (ROCK) or using Auralic Ares without having issue with Ammaero USB based chipset? The last time using XMOS, I got issue on hissing noise.

Hi @Miska When it come to DSD decoding, I came across two different architectures at the DAC side. A true 1 bit SDM like those off selves DAC ICs example AK4490, PCM1792A and CS4398 etc, vs multibit, example Holo Spring DAC uses 6 bit, 32 levels structure and ESS Sabre DACs probably uses 6 to 8 bit 64 levels structure. Are there differences in term of SNR, distortion and sonic reproduction?

This is just part of it. Having a time-domain compensation means it needs to use ‘leaky filter’ which causes aliasing problem. This is trade off, nothing is unique here. As I quote 'However, optimizing the digital chain’s behavior in the time domain involves using a very “short” antialiasing filter at the A/D conversion, and a similarly “short” reconstruction filter when the digital data are decoded. The more you constrain the data in the time domain, the less you can do so in the frequency domain. These filters are therefore “leaky,” as you can see in the measurements accompanying the Aurender review in this issue, and will thus allow ultrasonic images to fold down into the baseband. Such filters are not new ’ by John Atkinson Read this at: https://www.stereophile.com/content/mqa-some-claims-examined

Hi @Miska This is puzzling question I always wanting to ask… DSD is always converted to analogue at its native sampling frequency. In PCM, 44.1kHz is always over-sampled to 352.8kHz inside the DAC then convert to analogue, this provided excellent signal to noise ratio across the audio range. DSD64 noise-shaping is only effective up to 22.05kHz, after that ultra-sonic noise will start to appear. If DSD64 is over-sampled to DSD256 or even DSD512 inside a DAC, the ultra-sonic noise will be shifted even further away by a few octaves! This means, DSD64 recording is all it needs to get a good signal to noise ratio above 22.05kHz if it is oversampled in a DAC. Native DSD128 and DSD256 recordings will shift the ultra-sonic noise further away by a factor of 2 to 4 times but the files sizes becomes too large. Is there any reason why DSD is never over-sampled or up-sampled in a modern DAC?