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Ajax

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  1. Hi PAR, FYI the Society of Sound recordings, a joint venture between Peter Gabriele and B & W speakers, were originally distributed at 24/48 and sounded amazing. Their files were available as downloads and you received 12 albums of "new" artists for 50 GBP. It was Gabriel's way of giving back by helping new artists get a start. Good bloke. The several albums I received were very good in terms of the music, however, the sound quality was simply stunning, certainly the best recorded sound I have heard to this day. I'll never forget my then 12 year old son coming into my home office one night and exclaiming "that's spooky Dad, it's like she's in the room with us". At that time (about 2011) their web site quoted an English professor in psychoacoustics who stated that the bit rate was more important than the sample rate, however, I note more recently they have increased the sample rate to 96. I assume as you suggest that their ADC equipment was capable of 96, and with the increase in bandwidth and better internet speeds why not take out some "insurance" and ensure they left nothing on the table, however, from what I was hearing through my then Benchmark DAC1 and ADAM A7 active speakers, I doubt it was really necessary.
  2. Hi Chris, Didn't know you had Benchmark gear. Many many years ago I thought you did a review of the original Benchmark DAC 1 and added it in your CASH list. It's not there anymore so did I imagine it? Are you planning a review of the DAC3? Apologies for the off topic but I just bought 2 x DAC2s for less than US$1,000 each. Best HiFi bargain going around in my very humble opinion.
  3. Thanks Ralf, I wasn't a big fan as a young man and actually appreciate his early Cars' music more today. Great beats and licks that are fun with plenty of drive, that contrast with the clever but self deprecating lyrics.
  4. Thanks Chris, I thought your review was well balanced and thoughtful , you give credit where credit's due, as well as highlighting Neil's lack of technical understanding of how digital audio actually works. I've been listening to Neil's music for many many years, since I was a young boy in the late 60's, and what consistently shines through is his desire for authenticity, to be real, and I disagree with earlier contributors that suggest he was in Pono only to make money. Neil is an artist, an idealist and above all a humanitarian. His very nature makes him exactly the wrong person to take the lead in such a demanding project. I think you hit the nail on the head when you cited problems started when John Hamm's influence was replaced by Bob Stuart's. The record companies' profits have been decimated, first by Napster, then by Apple and finally by the streaming companies and their desperation to wrestle back control is a large part of the reason why MQA has enjoyed it's (limited) success to date. I think the whole pono fiasco highlights Neil's imperfections (an idealist surrounding himself with mates and not professionals) as well as his attributes, trying in his own way to create a better music industry. His contribution to music cannot be underestimated and his efforts to provide consumers with better sounding music (I can't stomach the expression "hi-res", it's a nonsense to me) should be applauded. At the very least he had a go at fixing what he saw was a wrong where others merely whinged, unfortunately, when Hamm went it was always going to end badly.
  5. Hi Ralf, It has iTunes built in so yes to ALAC
  6. Nice article Bluesman. Your personal experience and knowledge shines through and mimics my own thoughts. However, I've just done the bloody opposite. I'm half way through renovating a 3300ft house and driving myself and my wife crazy. They say never say never but believe me never again! For the past 10 years my office system has consisted of my lap top as the source (Audirvana & Tidal) feeding a Benchmark HDR DAC 1 with balanced out to ADAM A7 Active speakers. This system fits your desire for a small footprint and excellent sound quality but fails your "low cost" criteria. However, I bought the system 10 years ago and when you amortised it over such along period (with no sign of deterioration due to both components having excellent build quality) it is IMO actually very cost effective. Despite reading thousands of audio reviews and CA threads I've never thought the need to upgrade until last month when I bought a used Benchmark DAC 2 DX for $US 900 (I don't need the analogue inputs) and gave the DAC1 to my son for his home recording studio. As an aside it is one of the joys of our hobby to buy a product manufactured in the US, from a gentleman residing in Austria, and then arranging for it to be transported to Australia.... while paying 1/3 of the cost to purchase it new here. The second hand audio market (s) provide an excellent way of obtaining great gear at reasonable costs. Here knowledge is everything. I have also purchased inexpensive DACs from SMSL (Chinese) that you mention in your list and concur that the sound quality / cost ratio is outstanding and for someone starting out pairing one with JBL 305 active speakers is a no brainer. A great little system can be had for < US500. To your list of equipment "bargains" I would also add the Auralic Mini (Streamer/DAC). It ticks all the boxes - small footprint, good sound, flexible, user friendly, Local HD, Tidal, Spotify Connect, USB, Coaxial, Optical inputs. Really versatile piece of kit, which is also great to take travelling. Auralic have their own proprietary app, Lightning DS, however, they have discontinued the Mini (?) but if you hunt around you can pick one up for around US$350 (I have 3). I'm sitting on my balcony controlling my music via my iPhone (as a remote) and listening through my outdoor speakers. Very cool. Well done again on an excellent article. All the best, Ajax
  7. Hey Bluesman, Greetings from Sydney (Australia). Congratulations on selecting such a worthwhile topic. I have been purchasing used high quality audio gear for many years now. Today my son turned 20 and he received a Rega Planar 3 TT, which only cost me US$500 + Project preamp. I also bought myself a Benchmark Media DAC2 DX (made in the US but bought from a gentleman in Vienna) for my office system. Using Tidal as the source it will act as a pre-amp to a pair of Adam A7X active speakers. $2k complete system (second hand) - we are NOT talking budget here but great HIFI at a very affordable price. Like previous members I would also like to see you include some emphasis on usability as we all live in the real world, with kids and wives or partners. On this front nothing in my experience nothing competes with the Auralic Mini - for $US350 you can stream Spotify Connect and Tidal and for another $US200 install an internal SSD drive for your local tunes. It simply shows up under "devices as "living room" so nil tech knowledge required. I'm looking forward to the second edition of your blog after such an excellent start. All the best, Ajax
  8. Ajax

    Some commonsense

    Hi Paul, You are correct in that the Benchmark upsamples everything. The Benchmark DAC 1 (now over 10 years old and replaced by the DAC 2 and DAC 3 series) resampled all frequencies to 110khz. Here is a link to an article by John Siau giving the reasons behind such an obscure number.. https://benchmarkmedia.com/blogs/application_notes/13127453-asynchronous-upsampling-to-110-khz The upsampling frequency was subsequently increased to 210khz in the DAC 2 and DAC 3 as the chip was changed from an AK type to ESS. Some more information here. https://benchmarkmedia.com/blogs/application_notes/inside-the-dac2-part-2-digital-processing
  9. Ajax

    Some commonsense

    Hi Jabbr, I agree that ideally we should replay at the same resolution as the recording (pre digital music was mastered specifically to be replayed on turntables by limiting the base so the needle did not jump out of the grooves. Today we limit the band width of the audio frequency so we don't end up with artefacts). My reasoning of recording in hi-res and playing back at 16/44.1 was simply to achieve more head room during the recording process, which with due care is obviously not essential, especially today with so much compression being added. The point of the article is that mathematically 16/44.1 is adequate for music playback, IF as George points out the recording and mastering has been done with sufficient care. My hearing is limited to 12khz (I'm 63) so for me personally there is no need to record at higher frequency rates (above 44.1), however, maybe there is benefit at recording at 24 bits, however slight. I went to a hi-fi show in Melbourne, Australia about 4 years ago and heard the Devialet ensemble being demonstrated using only CDs and it was truly stunned by the sound, despite the poor acoustic environment of hotel show rooms. There were lots of competing gear with massive power amps and exotic cables but nothing to my ears came close. I bought the demo system, which included Atohm GT1 speakers and it now sits in my living room. However, the best sound I have ever heard was my office system, which consisted of a Benchmark DAC1 (I just purchased a DAC2 second hand for $US900) driving a pair of Adam X7 active monitors listening to Gwyneth Herbert recorded and produced by Peter Gabriel's 'Society of Sound". This was recorded and distributed at 24/48. One night about 7 years ago my then 12 years old son (now an accomplished musician) came into say goodnight and said "that's spooky Dad, it's like she is in the room with us". The point is John states (and my experience confirms) that extremely higher frequency rates aren't required, especially for old buggers like me, and if you don't play your music at pain levels you also don't need more than 16 bits. As John says, it is all in the maths. It is important to remember the reason why we have digital audio in the first place is because a couple of very smart mathematicians, Shannon & Nyquist, developed a theorem that simply put states that if you record at twice the highest maximum audio frequency of the music then that is sufficient for perfect fidelity and no actual information is lost. The other reason for reproducing the article is that the majority of music is available at 16/44.1, whether via CD, downloads or streaming, and manufactures should therefore be concentrating on improving the playback of that resolution, preferable using a combination of software and hardware, not offering us more and more exotic gizmos in an effort to differentiate themselves from other manufactures..
  10. Ajax

    Some commonsense

    Hi Peter, Thanks for your thoughtful response. I have always known that you have considered 16/44.1 adequate when upsampled in your XXHighEnd software prior to converting in your NOS1 DAC. MIska promotes a similar philosophy with his HQ Player as does Damien with Audirvana+ software. John Siau states there are only 3 things that can benefit from hi-res, none of which he sees as being practical: 1. An increased high frequency limit - majority of equipment can't produce frequencies over 22khz, and we can't hear over 22khz, so why bother. 2: An increased immunity to the clipping of inter sample peaks - this can be overcome by upsampling in the player software prior to converting in the DAC.  3: An increased SNR - you would have to play the music at an uncomfortably loud level to gain only a very minimal benefit. The bottom line is that here is really little need for hi-res material (and in deed there is a very little "true" hi-res' available as most offered today is upsampled music recorded on tape from an era when hi-res was simply not available (pre 85). My interpretation of Mark and John's blog, and everything I have personally heard is that when material is well recoded in hi-res (24/96) and then expertly mastered with dithering and noise shaping down to 16/44.1 for distribution it meet the requirements of even the most fastidious audiophile.
  11. I have often wondered how humans can "hear" the difference between a 16/44.1 audio file, that has been recorded at say 24/96, and expertly mastered using dithering and noise shaping down to 16/44.1, and the original 24/96 file. Many of you would have heard of Dr. Mark Waldrep (Dr. Aix), an expert in the recording and production of high res audio and John Siau, principal designer at Benchmark Media. This following article reinforces my own personal experience being that I cannot "hear" the difference between CD (redbook 16/44.1) and high res (24/96 and 192) when played back through my Benchmark Media DAC 1 HDR or Devialet 200 systems. I wonder if those that hear a difference do so, not because their equipment is highly resolving as many claim but quite the opposite, their equipment is in fact deficient in it's ability to resolve 16/44.1 correctly so that they "hear" a difference when playing back a 24/96 file. My conclusion is that assuming the original file is well recorded and produced at say 24/96, which allows for some extra headroom and the capture of all possible frequencies up to 48hz (nyquist frequency) then you will not hear the difference if that file is well mastered using dithering and noise shaping down to 16/44.1, assuming your DAC and its filtering system is of sufficient quality. My hearing is limited to around 12khz so in my own case an original master using 24/44.1 (22 kHz nyquist frequency) would most likely be sufficient. Here is the article: "I'm really fortunate to have friends that are smarter than me. In many aspects of life, it's great to be able to discuss and learn from people with different areas of expertise and life experiences. So when I authored a recent blog on "highly resolving" systems and whether they are necessary (you can read that article by clicking here ), it was a welcome surprise to hear from one of the smartest people I know, John Siau, the principal designer of both analog and digital systems at Benchmark Media. I consider their DACs to be among the finest on the planet — equally at home in my mastering room all these years AND in my home system. Benchmark supplied their DACs and power amplifiers for all of the 5.1 HD-Audio demos that AIX Records assembled at AXPONA shows. Benchmarks' hardware and my recordings delivered a music experience unlike any demo at any trade show in the history of trade shows. Why do I feel confident in making that boast? Because the entire signal path from source recording to speaker output was truly capable of maintaining high-resolution specifications — a truly "highly resolving" system. Read on and see how John's email clarifies the discussion with specifications and insights from a world class designer. Below is John's contribution to this important topic (reprinted here with his permission): Mark, It is not a matter of fancy cables, and esoteric tweaks. It is all about the math (MW bolded). If anyone hears the difference, it probably will not be on the basis of the frequency response. There are exactly three potential advantages provided by high-resolution audio: 1: An increased high frequency limit 2: An increased immunity to the clipping of intersample peaks 3: An increased SNR Items 2 and 3 are audible under the right circumstances. Item 1 may never or almost never be audible. Here is my reasoning: 1: High Frequency Limit Very few listeners will have transducers that extend beyond the 22 kHz limit of the CD format. The exception will be listeners with good headphones or very good speakers. But, most listeners can't hear beyond 20 kHz. You will need the rare combination of someone will excellent hearing beyond 22 kHz who is listening through good transducers. You will also need significant musical content above 22 kHz (at sufficient levels). This is going to be too small a group to be statistically significant. The size of this group is likely to be 0. You need to put some young but trained ears into some good headphones and play the right source material. 2: Clipping of intersample peaks The worst-case clipping is caused by a tone at 1/4 of the sample rate that is shifted 45 degrees relative to the sampling clock. This tone can reach 3.01 dB over 0 dBFS before the maximum digital codes are reached. This 0 dBFS + condition can happen 1000s of times on a single CD track. When upsampled in a sigma-delta D/A converter, these intersample peaks can cause clipping in the D/A converter. This often causes a DSP overload that creates a burst of IMD. This artifact is audible, but completely avoidable. The Benchmark DAC2 and DAC3 converters do not have this artifact at any sample rate. Most other D/A converters have this artifact. This will tend to make high sample rates sound better unless you are using Benchmark DAC2 or DAC3 converters. Here is a case where a better system makes all sample rates sound good. Most systems will make 44.1 sound worse than it should. 3: SNR: A few quick calculations will show that the listeners will need playback systems with at least an 87 dB SNR and they will need to play the audio at peak levels exceeding 93 dB SPL. Otherwise, it is mathematically impossible to hear the noise advantage of anything beyond 16 bits. If the listener's system has a playback SNR of 87 dB and it is playing at a peak SPL exceeding 93 dB, they would be listening for a very small 1 dB difference in the noise floor. If the listener's system has a much better SNR, then the task will be much easier (assuming they turn the volume up high enough the hear the 16-bit TPDF noise). If their system is absolute state of the art, it will have a 130 dB SNR. If they set 0 dBFS to 130 dB SPL, the 16-bit dither noise will be reproduced 37 dB above the threshold of hearing and would be very audible. On such a system, you could easily hear the differences in word lengths until you reach 23 bits. But, you would go deaf listening to the music unless the music had an unusually high crest factor or was recorded at a very low level. You would need to turn the system down to listen to the music. The more you turn it down, the less difference you will hear. Here is the math (nothing here that you do not already know): SNR in dB for a digital channel is (6.02*N)+1.76 measured over the entire Nyquist bandwidth of the channel. If the signal is TPDF dithered, subtract 4.77 dB Therefore, for TPDF channels we have: (6.02*N)+1.76-4.77=(6.02*N)-3.01 dB At 16 bits this gives 93.31 dB. We will round to 93 for the sake of discussion. In order for a listener to detect a difference on the basis of SNR, all three of the following conditions must be met: 1. The recorded noise must be lower than the channel noise of the 16-bit system (-93 dBFS). 2. The channel noise of the 16-bit system must be played at a level that exceeds 0 dB SPL. Therefore, the SPL at 0 dBFS must exceed 93 dB. 3. The noise produced by the playback system must not be more than 6 dB higher than that of the 16-bit system. This noise summation will produce just detectable noise difference of 1 dB. Therefore the SNR of the playback system must exceed 93-6 = 87 dB. Item 1: You are controlling item 1 with your choice of source material. Item 2: The listener may or may not choose to exceed a peak SPL of 93 dB during the test. The peak SPL chosen by listeners will be higher on uncompressed material due to the higher crest factor. You are choosing the source material and can experiment with different crest factors. Item 3: This is a big problem! Many "high-end" systems are not capable of delivering an 87 dB SNR. There are almost no speaker systems that can do better than about a 105 dB SNR because of amplifier limitations (the Benchmark AHB2 is a notable exception). A few state-of-the-art systems can deliver a 130 dB SNR to headphones if all of the components are properly gain staged (Benchmark DAC3 driving a Benchmark HPA4 is one example). A Benchmark AHB2 power amplifier driven by a Benchmark DAC3 can deliver a noise-free 130 dB SPL if it is driving speakers that have a sensitivity of at least 104 dB at 2.83 Vrms. We have a few customers with such systems, but there are less than 10 examples worldwide. Here is a chart that I created. It sums the digital channel with the playback system noise to calculate the SNR that will be delivered to the listener. I included bit depths of 8 through 24 bits. Find your playback SNR or peak playback SPL (whichever is lowest) on the X axis and then find the y value for each bit-depth curve. If the lines are separated at your SNR or SPL, then you will be able to hear a change in the noise floor. Example 1: If you have a noise-free playback level of 100 dB SPL (find 100 dB on the X axis), you should be able to hear changes for bit depths up to 19 bits. Beyond 19 bits, there will be no audible (or measurable) improvement. Example 2: If you have a system that can achieve a noise-free playback level 130 dB SPL, and you are playing it that loud, find 130 dB on the X axis. You should be able to hear the noise differences for bit depths up to 23 bits. Beyond 23 bits, there will be no audible (or measurable) improvement. Example 3: You have a system that has a SNR of 87 dB and you are playing it at a peak SPL of 100 dB. Take the lower of the two (87 dB) and find this on the X axis. Any bit depth above 16 bits will produce a 1 dB reduction in the noise relative to the 16-bit system. Longer word lengths may be just noticeable, but 17 bits will give the same performance as 24 bits. Example 4: 100 dB SNR playing at a peak level of only 80 dB. Find 80 dB on the X axis. Changes should be noticeable up to a bit depth of 14 bits. I should also note that a given D/A converter may deliver different performance at different sample rates. They often use entirely different filters and may have entirely different THD performance. The filters may have distinctly different phase response and this can be perceived as a change in frequency response. Early oversampled converters performed poorly at higher sample rates due to poor stop band attenuation. To mitigate this problem, I would suggest separating the word length tests from the sample rate tests. Change one parameter at time. Test 44.1/16 against 44.1/24. Then test 44.1/24 against 192/24. This will tell you if the audible differences are related to word length, sample rate or both. Once this has been established, you could look at 18, 20, and 22 bit depths to determine the threshold of audibility. You could also run a separate test to determine the audibility of sample rate is linked to intersample clipping or to bandwidth. John Siau
  12. My Grade 5 teacher, Dick Hurn, use to hit me with one for talking. He would make you go up the front of the classroom, turn around and bend over, and watch 30 other kids laughing at you while he belted the crap out of your arse. The only way to "save face", that is not wince and cry out, was to put a writing pad down your pants (if you had the time). Bloody hurt if you didn't.
  13. One of my hobbies (personality?) is to strive for the best possible deal I can get no matter what the situation. e.g. Living in Australia it is very time consuming, exhausting and expensive to travel anywhere. A few years ago my son did a ski racing course just outside Turin (Italy). It is 24 hours from Sydney to Milan (9 + 3 + 12). I'm a big fellow (6'2" and 220lbs) so I would spend a lot of time checking out flights, plane types and availability to get the bet possible deal AND comfort. The best I found was to catch a fight out of Sydney so that we could make the 1am flight out of Asia (any asian carrier via Bangkok, Singapore, HK, KL etc) as at that time of the morning the Asia - Europe flights were never full. We would get on this second flight last so that we could see where the empty seats were. Often I would get 4 seats across in the middle at the back and my son (then a small 13) would get 3 on the side. I would take 2 sleeping pills (he got a half) and after 8 hour sleep on a 12 hour flight arrive in good shape. Business class was about US$5k return and economy US$1k, however, as long as we got a sleep that was all that mattered to us. With Audio I am always on the look out for such deals, which usually come in the shape of used gear, but not always. The opposite to bling has to be the Benchmark Media range. Their engineering is superb, however, their appearance is at best bland. In stereophile over the years JA has raved about the measurements of their range of DACs and in particular the performance of their AHB2 power amp. In a "normal" domestic situation a DAC 3 / AHB2 combination would drive just about any speaker to say 90db with sonic excellence. That front end would only cost you $5k new ($8k if you needed more power and bridged 2 x AHB2s). If you shop around you can get a very similar "sound quality" by purchasing the older DAC2 with the AHB2 (used) for about US$4k. Just add source and speakers. Using the DAC3 with its remote (as a pre amp) you could hide these two pieces of gear out of site if you were hung up on appearances. I know US$10k is a lot of money to some people for a hifi system, however, I still have my beloved 10 year old Benchmark DAC 1 as part of my home office system. I just turn it on each day and it does it's thing, unbelievably reliable and still sounds great. That's value. It acts as a preamp for a Nord Hypex Ncore power amp driving a pair of ATC SCM 19 speakers. There are plenty of other examples of high quality manufacturers available at "reasonable" prices to those who have the knowledge. Knowledge is the difference between paying silly money, $50K +, and reasonable money <$10k. My 3 cents. All the best Ajax
  14. Hi Mike, Lots of ways to skin this cat as you have seen from the responses to date. We have the Yamaha NX N500s in our bedroom attached to the TV and they work well and will do everything you ask plus more. You can pick them up for around $350 second hand. If you are looking for better sound quality, while still retaining simplicity, I would look at a Pre/ DAC + Active Speakers. You mentioned stretching to $2,200, which will be sufficient for great sound, especially if you are happy to buy used. Have a look at the SMSL and Topping DAC range from China on Amazon for the front end $200 - $500 and combine with Active (not powered) speakers. Just google but I can recommend the Adam A7X for $1,000 second hand plus a sub. Alternatively a SMSL 8a DAC $200 with a $500 second hand power amp and the excellent KEF L50s. $1,500 new and 1,000 second hand. There are so many combinations it can get confusing.
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