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· Warner Music Group Licenses MQA Digital Remastering Technology    posted May 20, 2016 (show)
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  • From EnjoyTheMusic.com

    05 / 09 / 16

    Warner Music Group Licenses MQA Digital Remastering Technology

    Warner Music Group Licenses MQA Digital Remastering TechnologyWarner Music Group (WMG) has announced a long-term licensing deal with music technology MQA. The agreement – the first between MQA and any major music company – will increase music fans' access to hi-resolution music globally. The agreement paves the way for recordings from WMG's diverse roster of acclaimed artists and its world-renowned catalogue to be made available in studio master quality through MQA distributors. MQA is a technology that remasters a digital audio file to reduce the data/bandwidth needed to and then adds a flag / authentication to ensure a bit perfect delivery. By delivering the sound of the remastered studio masters from WMG's iconic labels, MQA will enable the listener to step into the magic of the artists' original performances. MQA music is currently available via High Res Audio, Onkyo Music, e-Onkyo, 7digital, 2L and Technics Tracks. Over the coming months, MQA will expand to more streaming platforms and into download stores worldwide. "This collaboration is a giant step forward for MQA and music fans everywhere" says Bob Stuart, creator of MQA. "We have been working tirelessly to ensure labels, studios, artists, services and playback partners understand the potential of our technology and the responses have been overwhelming. MQA is about bringing the most authentic sound to music lovers all over the world and WMG is our first major partner to help drive this mission forward." Craig Kallman, Chairman and CEO of WMG's Atlantic Records division, said "The digital music era has been all about convenience. It is fantastic that we can listen to virtually any song, anywhere, any time. In that process, however, convenience has trumped sound quality, and we have gotten further away from the sound that artists work so hard to create. MQA makes hi-resolution music easy to stream or download to any device. Music fans will love it when they hear it, and WMG is thrilled to be partnering with MQA to take the next step in bringing hi-resolution music to consumers across the globe."
· Open Meeting for local audio clubs at Ralph Glasgal's    posted May 16, 2016 (show)
  • Free to all
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  • All day Saturday, May 21st, 2016

    The Audiophile Society will conduct this meeting at the home of Ralph Glasgal,
    4 Piermont Road, Rockleigh, NJ. Ralph's phone number is (201) 784-0614.

    Ambiophonics has been around for some time, but its inventor, club member Ralph Glasgal, continues to make exciting breakthroughs in the field of psychoacoustics. After a recent visit to the famous Ambiophonics cathedral at Chez Glasgal, we knew it was time for everyone to drop by Ralph's to hear what he's been up to. And when we say everyone, we mean it: All the audio societies in the area are invited, including Connecticut, NJ, Philadelphia and the Audio Syndrome from Long Island. This will be a big meeting.

    The main Ambiophics system has gotten more spectacular — so good and so sophisticated at this point, with so many options, that we're insisting on single rather than group demos, with Ralph giving everyone a one-on-one tutorial. Chef Ralph has cooked up a demo menu for you to choose from (mostly Chesky recordings, with classical available on request). He'll have LPs (mono, stereo and SQ quad) spinning on the ELP laser turntable, along with CDs and SACDs on the Oppo. You'll hear all forms of Ambiophonics, including the latest Envelophonics, and will be able to control the parameters on a remote control. The massive SoundLabs electrostats are still there, now augmented by a pair of JBL line-source speakers designed by Don Keele, Jr. For budget-minded audiophiles, Ralph will give a mind-blowing demo of a new Ambiophonic mobile app playing through Bluetooth speakers. It's scary good!

    This will take some time, so we're doing something totally different: You can show up any time you want from 11:00 a.m. through 9:00 p.m. Take a number, get your demo, then hang out, eat, drink, schmooze. Think of it as an open house. Several demos, including MBLs in the home-theater room, will be running constantly. Ralph has gone all out on the catering for this, booking Ragazzi's Restaurant to provide food throughout the day. David is in charge of the coffee, soda, cookies, crumb cakes and dessert, so whenever you arrive, there will be plenty of food and drink. As a further enticement, a large collection of XLR interconnects and AC power cords will be available for free. This will be a tremendous gathering of audiophiles form all over the area, so please come out and have a great time! RSVP as usual to dnemzer@earthlink.net or reply to this.

    Directions to Ralph's house: Take the Palisades Parkway south from the Tappan Zee Bridge or north from the GW Bridge or Route 80. Take exit 2 off the Parkway and turn left going South on Route 9W. Turn right onto Closterdock Road, go down the hill to Piermont Road and turn right. Drive north on Piermont Road about 3 miles to the first driveway on the right after the "Entering Rockleigh" sign. The driveway is a half-mile long, straight up through the woods. Drive up to the house, turn around, and park on the right side of the driveway facing downhill.
· New free file tagger for WAV files    posted May 9, 2016 (show)
  • Great for Computer Audiophiles
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  • MP3tag
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  • The main feature of v2.76 is full support for the WAV audio format. This includes support for RIFF INFO tags (which are surprisingly limited) and writing ID3v2 tags (including user-defined fields and cover art) as dedicated RIFF chunks. The new version reads, writes and removes those tags and – as usual – also reports meaningful technical information about the file.

    As far as I know, convenient ways to tag WAV files are rare.
· B&W Sold to Silicon Valley Startup    posted May 9, 2016 (show)
  • Bloomberg Article
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  • Comments:
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  • § Any details?
     ⤷ posted May 10, 2016 by JimOfAllTrades
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  • § https://www.audioholics.com/news/b-w-purchased-by-silicon-valley-startup-cheers-or-tears
     ⤷ posted May 10, 2016 by JimOfAllTrades
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· Mahler in Space    posted May 2, 2016 (show)
  • Super-Conductor
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  • I was just surfing the net and found this. Who knew that Gustav's 1st and 7th Symphonies found their way into our favorite space opera?

    John P.
· Brian Kheel - Sad News    posted Apr 21, 2016 (show)
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  • With profound sadness I have to inform NJAS of Brian Kheel`s death this evening [Wednesday 4-20].

    He truly loved the comraderie of all the audiophiles.

    Funeral is Thurs. 4/21 at 1 pm at the Highland Park Conservative Temple, 201 South Third Avenue, Highland Park.
    We will be receiving condolences at our house, 345 Felton Avenue, Highland Park from 4pm on and on Friday morning. If you have any questions, you can call our friend Jerry Trub at 732-501-1643.
· Reconciling DAC Performance    posted Apr 14, 2016 by JimOfAllTrades (show)
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  • In the previous post we talked about how to read WAV files and what 16-bit, 24-bit and 32-bit samples actually look like. Please review that article if you haven't already done so.

    Let's again talk about 24-bit samples, but let's reverse the byte order so that the biggest value comes first and let's revamp the scale so that negative values are treated like positive values so that 7F FF FF is the maximum value. For example, 10 00 00 would now be 16x256x256 = 1048576. The maximum value would be 8,388,607. This is all better treated in dB's. 24-bits has a dynamic range of about 144.5 dB, which is huge. Very few electrical devices actually have a signal to noise ratio that good or better, which means that the least significant bits are encoding or decoding mostly noise. How bad is the situation?

    The first two bytes encode 16-bits of information and the last byte encodes 8 (of course). 16-bits is about 96 dB of dynamic range. The best pre-amps/amplifiers are about 120 dB of dynamic range, which means that the final nibble of a 24-bit signal saturated by noise. It would be something like 72 45 6?; where "?" represents the noise.

    Microphones get upwards of about 75dB of dynamic range and also have noise floors generally around 15dB, which means that you only get the full dynamic range of a mic when recording sound above 90dB or so, and 75dB is only about 13-bits! This is less than 16-bit redbook CD and is one reason 16-bits was though more than adequate for music. Our sample above would be more like 72 4? ??. This is why MP3's can sound passably good. You only need to encode the top 1.5 bytes and add some noise shaping to get to 13-bits. When engineers compress MP3's further to take into account the limitations of human hearing then you get about 10x compression ratios over redbook CD, down to about 128K per second, as opposed to 1.4M per second for CD.

    So, how can you tell if the smaller bytes are recording noise or not? The more stringent way of asking is "how do you count the number of noisy bytes per sample you are getting?". This is a little like asking "what would perfect third bytes contribute to the signal?". Let's say you were recording a 1kHz full-scale sine wave. How do you know if the final bytes are signal or noise? The obvious way to do this is to model the incoming signal and subtract a mathematically perfect version from the real-world version. The difference would be closer to 0 the better the resolution got. Fortunately there is a tool for this called the dynamic signal analyzer or DSA, which will perform continuous FFT calculations against a real world signal. FFT's, or fast Fourier transforms, essentially attempt to match real-world signals against perfect sine waves. Real world FFT's have the limitation of being "discrete", and sometimes are called DFT's. This means they introduce their own measurement errors. However, when you get a spectrum from a DFT you can see the main signal as a peak. The rest is the remaining signal with the main signal subtracted out. On a DFT spectrum the "noise floor" really looks like a floor, and the peak to floor ratio is the signal to noise ratio. Spectra are traditionally shown in dB scale, so it would be readily apparent if you only had 16-bits or 24-bits of signal, the noise floor would be nearly 50 dB different. Since the best equipment at room temperature gets about 120 dB of dynamic range, you might see about 20 dB difference in the noise floor. Resistors, for example, produce hissy shot noise that depends on their temperature. The same is true of any electronic part that has resistance. That's just about anything in a circuit. You can reduce noise in an electronic circuit by cooling it, but this will change the electrical properties of resistors, and that can result in circuit failure unless the circuit is designed to be cooled. Take the cooling into account though and throw the circuit into a very cold liquid and the noise will drop considerably. This isn't practical for audio though.

    You might be thinking, "Okay, I'll use some feedback to remove noise from the circuit". Actually, no. It's really the opposite. Feedback will help maintain circuit stability so that it doesn't blow itself up, and help maintain linearity in the face of distortion, but feedback will also amplify noise. So feedback is always a trade-off between noise and distortion.

    What do you gain, then from using 24-bits? It might not be the signal so much as the parts quality, but aside from that you still get a more accurate signal representation. The digital "quantization" noise is reduced. This will probably help the signal quality in the 4kHz to 15kHz region most, as this region is often way down in the spectrum and outside of the background noise of the recording. For example, mic pre-amp noise generally peaks at 60Hz and extends to about 2kHz. Above 2kHz a mic can have over 100dB of signal to noise ratio and often reduced self-noise too. So, essentially, 24-bits will give you more accurate overtones and transients. If you suffer hearing loss at high frequencies then you probably won't get much out of it.

    More importantly, 24-bits makes life a lot easier for recording engineers. They can use 24-bits to have much better "head room" on their recordings. They don't have to worry as much about pegging the meters when loud music played, or getting all the mic pre-amps at exactly the right levels, or dialing in the peak limiters to a T. Almost everything will be captured without clipping and with enough dynamic range that it becomes purely a post-processing event. Everything can be done during the mix down. Remember, it's 40 dB more head room than 16-bit recording, which is like have a camera lens you can zoom in and out much further. Framing the shot becomes easier.

    The next article will be on condenser microphones.
· How to Read a Wav File    posted Apr 9, 2016 by JimOfAllTrades (show)
  • The standard uncompressed music format
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  • "WAV" or "RIFF" is a file format for storing data from Microsoft. Developed in the days of 8-bit computing and modified a few times since, it has some quirks and limitations. The biggest limitation is that it has no truly standard "header" for track/album/artist information. You can add such information to the file in the "header" spot provided for by the standard, but since it's completely unstructured, hardly any device or software that reads WAV files will bother to pull the header information. One of the few devices to do this is the Oppo Bluray player. That's if there's anything to pull--most files won't have it. The biggest quirk is the convoluted "header" in the file for getting format, channel and track information. Devised for internal use only at MS, the file header is very messy and confusing by modern standards. MP3 and FLAC are much better, though MP3's are notorious for needing their headers fixed. This is sometimes called meta-data.

    To keep going, read the introductory article "How Many Bits in a Byte".

    To read a WAV/RIFF file, you need to open it with a "Hex" reader. This is short for hexidecimal reader. Most readers will show both the bytes as hex values and the alphanumeric characters that the bytes represent.

    You will probably see in the top row the characters "RIFF" first and sometimes later "WAVEfmt". This lets you know you are on the right track. A computer reading the file can also use this to determine it's a WAV file.

    Somewhere after that you will see "data". In the hex reader this will be "64 61 74 61", for "d a t a". Spaces are meaningless and can be ignored. They just help one read out the bytes correctly.

    However, to the left of "data" will be "nn mm", either "10 00" or "18 00" hopefully, which means 16-bit or 24-bit respectively. Between "fmt" and the bit length id'er, you might see "00 EE 02 00" or "00 EE 01 00", which are 192k and 96k respectively. 44.1k is "44 AC 00 00". To the left of that you will probably see "02 00", which means two channels, or "01 00" for mono.

    Where's the actual song encoded? To the right of "data" cross off four bytes. Everything after that is data. To decode the rest we need to know the format. Here are the common formats:

    S16_LE
    S24_3LE
    S24_LE
    S32_LE

    Redbook CD is S16_LE. This means that each channel is signed, two bytes and little byte first.

    In the data area, the first two bytes will be the left channel "nn mm" and the next two will be the right channel "pp oo". For example "45 67 89 AB" represents 6745 for the left channel and AB89 for the right channel. These are the decimal values 26437 and 43913, except any number above 7FFF or 32767 needs the correction X - 65535, which means 43913 => 43913 - 65535 = -21622. If the CD player outputs a maximum voltage of 1 Volt then the left would be 0.806817835 Volts, and the right would be -0.659871212 Volts. The minimum required voltage resolution is 0.000030519 Volts or 30.52 uV (microvolts).

    S24_3LE is a non-standard but somewhat common 24-bit format, where each 24-bit sample is just 3 bytes, as in "oo nn mm". Note that the only change is with the little byte, the two bigger bytes have the same meaning as with 16-bit formats. That means that when you go to 24-bits you increase "resolution", not "loudness". A 16-bit sample "nn mm" is the same as the 24-bit sample "00 nn mm". For example "45 67" is the same as "00 45 67" in terms of CD output voltage. To play 24-bits you need a player capable of 0.12 uV (microvolts) resolution.

    S24_LE and S32_LE are essentially the same format because they both use 4 bytes per sample. The only difference is that S24_LE enforces 00 padding. Most DAC's are S24_LE natively before converting to delta-sigma internally. This just means they can handle all types of input natively with minimal fuss. 00 padding means that an S24_3LE sample "23 45 67" will become "00 23 45 67" when converted to either S24_LE or S32_LE. The zeros have no impact on the output voltage.

    The next article will be on verifying DAC performance.
· How Many Bits in a Byte?    posted Apr 9, 2016 by JimOfAllTrades (show)
  • Digital Media FAQ No. 1
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  • We've heard of 1-bit, 16-bit, 24-bit and 32-bit DAC's, but this can be a bit vague. Besides, when you store media digitally, you're talking bytes, not bits, as in Megabytes and Gigabytes, etc.

    Almost all modern computers are 64-bit with some 32-bit exceptions. DAC's really aren't any different. They almost all operate at 32-bits, regardless of what their resolution is. This is because the first half of a DAC is essentially a computer and the second half is a line-level analog source component. The bridge between the two is some sort of digital to analog converter or translator, hence "DAC". There are many choices as to what it could be. The simplest is simply a series of resistors, each double the resistance of the previous one followed by a low-pass filter of some sort, usually fourth order or more.

    The use of a ladder DAC (doubling resistor series) shows exactly how bits and bytes work. Bits are essentially meaningless without bytes. Bits are just placeholders for powers of 2, hence the doubling. A 16-bit DAC, therefore, would have 16 levels of doubling, which is a lot. The maximum value bit is about 65K greater than the least value bit. In a 64-bit computer there are 64 levels, which is a huge ratio. Bytes, though, represent the minimal amount of useful information. One bit it some information, but it's not useful enough to make significant distinctions. As you bundle bits together you trade space for resolution. When computers were first being developed, 8 bits seemed like the natural boundary point. Had the choice been made today, 32 bits probably would have been chosen. The trouble comes from needing to string bytes together to make meaningful minimal statements because there aren't enough bits in a byte to use just one. This breaks the definition of a byte and leads to all kinds of problems. It's a major problem in the computer world.

    What does it mean that there are 8 bits in a byte? It means there are 256 distinct byte states, but since one bit is often used for redundancy, that cuts it down to 128 states. This is because 2 to the eighth power is 256 and 2 to the seventh power is 128. Since a Byte represents the most minimal distinctions, all alphanumerical and punctuation characters, not to mention computer control characters, must fit into 128 states (or the maximum number of such characters is 128). This is good enough for modern English, but the advent of international computing, and symbols needed for science, math and logic, has resulted in Unicode, or multi-byte characters. Unicode now exceeds 16-bits, which would be two bytes per character. Because the original English character set is one-byte, the "character boundary" needs to be defined before any text is represented, and strings must be distinguished from control codes. When this isn't done properly, computer code can easily be hacked.

    Now we get to the first punchline. A byte is a numerical value from 0 to 255, and 256 = 16x16, so each byte is represented by two hexidecimal (nibble) characters 0-9, A-F, where 10=A, 11=B ... 15=F. 0 = x00 and 255 = xFF, or just 00 and FF. 15 is 0F and 16 is 10. Because a byte is a minimal character, it's always of this pattern. However, if you need more than one byte, you need to define the order in which they appear. For redbook CDs, the 16-bit channel information is designated S16_LE, which is somewhat clear, but sidesteps bytes. It's really saying "2 bytes, with the smaller one first", that's because "L" means "little", so the little one comes first. A single 16-bit music sample can be from 0000 to FFFF numerically, but "S" means signed, so while 0 = 0000, -0 = FFFF and all negative values are between 0080 and FFFF, and all positive values are between 0000 and FF7F. Keep in mind that 0080 is about 32K in "LE" notation, and should be really read as 8000 = 8 x 2^12. Each nibble being 4-bits, the three to right of the 8 make 12. So, in the next article, we'll go over how to read a "wav" file.
· Audio Industry Going Haywire??    posted Apr 8, 2016 by JimOfAllTrades (show)
  • Modern Snake Oil Getting Out of Hand?
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  • This isn't a good thing for the industry.
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  • The high-end audio industry already has a poor reputation among engineers and often the general public. While good speakers can be appreciated, and tube amps and turntables get a passing nod, a lot of what's produced receives endless ridicule.

    The last thing we need is a product that make claims about tachyons and sucking electrons out of the air to improve your electrical efficiency. Tachyons haven't been shown to exist and not all scientists think they are needed to explain any known physical phenomena. Actually, few think they are helpful to physics at all, but if the public is gullible enough, maybe they'll buy the marketing going on here.

    There are a lot of constraints on any explanation for consistent observations in audio. Products like this will just make people more dismissive of the entire subject.
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     ⤷ posted Apr 14, 2016
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  • § Interesting. Obviously someone has been watching too much Star Trek. Sucking electrons out of the air ought to have interesting effects on atmospheric quality. The scary thing is what goof actually funded this product!
     ⤷ posted Apr 24, 2016
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· For the folks in the club who belong to the Mahler Cabal...working now    posted Apr 2, 2016 (show)
· USB Cables on the March!    posted Mar 27, 2016 by JimOfAllTrades (show)
  • Many opinions on why they sound different.
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  • Recent PF Article
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  • Postive Feedback magazine recently ran an opinion piece on USB cable "differences" from a broad survey of cable makers and industry insiders. A few theories can be downplayed immediately:

    Not an issue:

    1) Bit correction. When there's a checksum error the data is re-polled so the datastream is bit-perfect on arrival at the DAC, otherwise you'd get horrific noise coming out that would sound like very loud static or electrical shocks, and would be at the highest frequency of the output sample rate (around 22K for CD, etc). This would be easily noticeable if a problem.

    2) Reclocking. There's no realtime USB streaming standard, so all USB data must be buffered by the DAC. Data can even arrive out of order and still be okay. USB is not a standard anyone should have used for audio, but it's cheap and ubiquitous (always a bad combo). DAC's can buffer entire songs these days and reclock internally. A better cable will neither help nor hurt the situation.

    3) Squarewaves are analog 1's and 0's. This is the same as 1) above. The actual signals are far more robust than just using single squarewaves as 1's and 0's. These are well known digital signal transmission problems and bit-perfect transmission can occur at much high rates than used by audio (by several orders of magnitude).

    What remains:

    1) Noise transmission. USB is not designed to work with analog gear. It's meant for ink jet printers and keyboards, etc. The various chipsets can be noisy. You can hear this if you use unshielded computer boards near an AM radio.

    2) Jitter?? Not likely since the DAC must buffer the signal, but noise can interfere with a reclocking circuit. If a device says it's "fixing" a USB signal, keep in mind that a USB interface is a lot more complicated than a normal wire. Devices are meant to "talk" to each other, and this sideband traffic occurs at the same time the audio streaming is done. So your computer might be talking to the keyboard or printer and that traffic will be sent to the DAC as well. At best, an intermediate device might block the traffic from getting any further. It might also handle some of the buffering.

    Jitter has always been the go-to reason for every digital ill, but fixing it is essentially impossible because you can't fix the jitter built into the recording from the start. With zero timing issues (a standard deviation of 0 for timing deltas) on the playback side, the recording was still most likely digitized with jitter from the start. What you want isn't zero jitter, it's a "jitter follower" that duplicates the jitter from the ADC (a practical impossibility to be sure). On the whole, you can reduce total jitter by about 50% at most from the playback perspective (assuming the typical DAC and typical ADC have about the same jitter). You also need voltage linearity to be on par with clock accuracy, otherwise the gross error will just be the larger of the two. You can think of your gross error as a rectangle where the width is jitter and the voltage linearity error is the height. Your delivered signal can be anywhere in the box following a normal distribution to some standard deviation. The mean error is the vector length from the center to the corner, so reducing jitter will not be too useful if the linearity error doesn't also go down in proportion. However, as a practical matter, the ADC box combined with the DAC box will result in the two largest sources of error dominating.

    It would be nice if the audio press stopped suggesting that everything can be fixed on the DAC side.
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  • Comments:
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  • § Actually the main reason they are different is because audiophiles have different ears and manufacturers need to compensate for that fact.
     ⤷ posted Apr 24, 2016
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· To 384K and Beyond!    posted Mar 27, 2016 by JimOfAllTrades (show)
  • ADC's Available
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  • What's Out There??
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  • Benchmark ADC1 USB

    P.S. Audio NuWave Phono Convertor

    M2Tech Joplin Analog-To-Digital Converter

    http://www.enjoythemusic.com/magazine/equipment/0213/m2tech_joplin.htm

    Anything else??

    There are a lot of 192K ADC's on the market, but audiophiles have moved beyond that to 384K and such. There's a problem though in that there's really no first run music at very high bit rates.

    There's actually a whole slew of problems. The current industry standard for the most part is 96K, which is just double the old 48K standard. 192K is double that, and 384K is double that. Many studios have switched to 192K for mastering, but almost none have even looked at 384K. We need to start to list the reasons why:

    1) The file sizes become needlessly large.
    2) The recording chain would need a lot of 384k compatible devices (many are not even 192k ready). This includes compressors, equalizers, mixers, the mixing and mastering software and bit clocks. Or studios have to go back to analog chains after they spent twenty years converting to digital.
    3) You can run into a lot of RF issues at high frequencies. You are starting to record noise at high frequencies (above 100k or so) that used to get filtered out naturally, and it gets harder to filter out as you go higher in frequency.
    4) A fair number of 192K (not to mention 384K and DSD) ADC's aren't true 192K devices, but simply double the bit rate of a 96K stream. The only way to test is to feed HF test signals into the device and see the output. What's the point of 192K if it's really only 96K?
    5) Studios need to be compatible with each other since recordings are often done in multiple locations.

    What's actually happening is the remastering craze of taking analog master tapes and converting them to high-rez digital at progressively high bit rates. This isn't likely to improve on fidelity much since the tapes don't have much bandwidth and many of the recordings are old and not done with fidelity in mind. Finding good "master" tapes and re-engineering them for high-rez digital has as many issues or more as studios adopting a new high-rez standard.

    Why do audiophiles like the new re-issues? It's hard to say how the masters were treated. Many will get some degree of digital "enhancement" before being re-issued. We can start with the basics: re-eq'ing, auto-tuning, wow-flutter removal, phase-removal, compression, noise removal, etc. There's a whole bunch of modern digital tools that go beyond these basics too. It's likely then that the re-issue will sound more modern than any record or CD that was part of an "official" studio release.

    Probably the best fidelity anyone's going to get these days of new recorded music are "bootleg" concerts recorded with inexpensive but reliable portable recording gear and microphones. Many of these are at least 192K "compatible" even if they might cheat and only do 96K or even just 48K (how would anyone know otherwise?). (And a fair number of "192K" DAC's just downsample to 96K anyway). We haven't even addressed the issue of 24-bit word lengths and how the industry standard is generally 5-bit delta-sigma conversion. Caveat Emptor.
· LP Surprise    posted Mar 22, 2016 (show)
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  • So, I was digitizing an LP last night (The Motels' "Little Robbers" - yes, I like The Motels and I am not ashamed to admit it), and thinking, gee, this LP sounds pretty good for a pop record from 1983, with solid bass, natural highs, and fairly wide dynamic range, when I noticed on the liner notes that it was mastered by the late, great, Doug Sax. Cool!
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  • Comments:
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  • § I always liked "Only the Lonely".
     ⤷ posted Mar 22, 2016 by JimOfAllTrades
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  • § I always liked that record. There were A LOT of snappy tunes on it, as I recall. Martha Davis was an underrated lead singer.
     ⤷ posted Apr 2, 2016
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· New Patent for "High Definition Vinyl"    posted Mar 22, 2016 (show)
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