If you're not sure if you should believe the statements in this blog which contradict much of the marketing hype, myth, and legend in the audiophile industry, feel free to check the references at the end of this blog.
In 1857, douard-Lon Scott de Martinville invented the phonautograph, which could graphically record sound waves. In early 1877, Charles Cros devised a way to reverse that process on a photoengraving to form a groove which could be traced by a stylus, causing vibrations that could be passed on to a diaphragm, recreating sound waves.
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In 1887, Emile Berliner invented the technically inferior disk phonograph. Disks warp and there was arch error and skating errors introduced. Certainly no comparison to the tangential tracking Edison cylinder player.
But since disks are much cheaper to produce than cylinders, and since disk fit nicely in display bins at stores and can include larger cover art and notes, they became the standard. And so began the long history of the recorded music industry being more about consumer convenience and optimal profits than about optimal fidelity.
In the early 1980s, when digital recording became readily available, studios converted from analog to digital to save money. For studios, this cost less for the equipment, required less space for both recording and archiving, and made it easier to mix and edit tracks in post-production. For consumers, there weren't many advantages. Most of the early digital recordings were produced with relatively low resolution and sounded so fatiguing they would make you want to tear your ears off.
Long before the DVD, SACD, or DSD formats were developed, the Bit Stream DAC chip was introduced to the consumer market as a lower-cost alternative to the significantly more expensive R-2R multi-bit DAC chip. Bit Stream DAC chips have built-in algorithms to convert PCM input to DSD, which is then converted to analog. Once again, the result was a huge cost saving at the expense of fidelity.
In contrast, not only do multi-bit R-2R DAC chips cost significantly more to manufacture than single-bit DAC chips, but they also require much larger and more sophisticated power supplies. If you were to make a 7.1 channel R-2R multi-disk player, it would cost several times the price of Bit Stream technology and it would be several times the size. Certainly not what the average consumer is looking for.
PCM recordings are commercially available in 16-bit or 24-bit and in several sampling rates from 44.1KHz up to 192KHz. The most common format is the Red Book CD with 16-bits sampled at 44.1KHz. DSD recordings are commercially available in 1-bit with a sample rate of 2.8224MHz. This format is used for SACD and is also known as DSD64 or single-rate DSD.
There are more modern, higher-resolution 1-bit DSD formats, such as DSD128, DSD256, and DSD512 as well as wide-DSD formats with 5-bit to 8-bit Delta-Sigma decoding which I will explain later. These formats were created for recording studios and comprise only a very small portion of the recordings which are commercially available.
In other words a DSD64 SACD has much higher resolution than a 16-bit 44.1KHz Red Book CD, roughly the same resolution as 24-bit 88.2KHz PCM recording, and not as much resolution as a 24-bit 176.4KHz PCM recording.
DSD encodes music using pulse-density modulation, a sequence of single-bit values at a sampling rate of 2.8224MHz. This translates to 64 times the Red Book CD sampling rate of 44.1KHz, but at only one 32,768th of its 16-bit resolution.
In the above graphical representation of PCM as a dual axis quantization, and DSD as a single axis quantization, you can see why the accuracy of DSD reproduction is so much more dependent on the accuracy of the clock than PCM. Of course, the accuracy of the voltage of each bit is just as important in DSD as PCM, so the regulation of the reference voltage is equally important in both types of converters.
Of course the accuracy of the clocking during the recording process which is done at several times the resolution of commercial DSD64 SACD and 16-bit 44.1KHz PCM recordings is significantly more important than the accuracy of the clocking of either DSD or PCM during playback.
There are other DSD formats which use higher sampling rates, such as DSD128 (aka Double-Rate DSD), with a sampling rate of 5.6448MHz; DSD256 (aka Quad-Rate DSD), with a sampling rate of 11.2896MHz; and DSD512 (aka Octuple-Rate DSD), with a sampling rate of 22.5792MHz. And most modern A to D and D to A Delta-Sigma converters do multibit wide-DSD with 5-bits to 8-bits decoding in parallel. All of these higher-resolution DSD formats were intended for studio use as opposed to consumer use, though there are some obscure companies selling recordings in these formats.
Of course when studios convert a 48KHz multiple format to a 44.1KHz multiple format or visa versa they introduce quantization errors. Sadly this is often the case with older recordings when they are released in a remastered 24-bit 192KHz HD version derived from DSD64 masters, such as the ones Sony and other companies used to archive their analog masters in the mid-90's. Note that the optimal HD PCM format which can be created from a DSD64 master would be 24-bit 88.2KHz. Any sampling rate over 88.2KHz or that is equally divisible by 48KHz would have to be interpolated (not good). But consumers demand 24-bit 192KHz versions of all their old favorites, so companies provide them, despite the known consequences.
Though modern sampling rates are high enough to fool the human ear, quantization errors still occur when translating from one format to another. For example, when Sony decided to archive their analog master libraries to DSD64 back in 1995, they were wrong to believe that these masters would be future-proof and able to reproduce any consumer format. The fact is, these masters could only properly reproduce a format that was divisible by 44.1KHz. So any modern 96KHz or 192KHz recording created from DSD64 master files have quantization errors.
This leads me to one of the many things that enrage me about the recorded entertainment industry. If 44.1KHz was the standard which was engineered to put aliasing errors in less critical audio frequencies, then why did they start using multiples of 48KHz?!?!?!? All they had to do was go with 88.2KHz and 176.4KHz as the modern HD consumer formats, and all of this mess could have been avoided. They made DXD, a 24-bit 352.8KHz studio format, equally divisible by 44.1KHz. What blithering idiot decided to put a wrench in the works with 96KHz and 192KHz HD audio?!?!?!?
The actual reason for the 48KHz multiple has to do with optimal synchronizing to video. So it makes sense to have sound tracks from movies recorded in a 48KHz multiple, such as the 24-bit 96KHz format embedded into 7.1 channel audio on DVDs and Blu-Rays. But since over 90% of all music recordings are sold in a 44.1KHz for Red Book CD or DSD64 SACD it is rather ridiculous to offer any HD music in 96KHz or 192KHz as opposed to the optimal 88.2KHz and 176.4KHz HD formats. But because naive consumers wrongly believe that the higher the sampling rate the higher the fidelity they demand 192Khz falsely believing it is better than 176.4KHz, so that is what record companies market.
Quantization noise is unavoidable. No matter what format you digitize in, ultrasonic artifacts are created. The more bits you have, the lower the noise floor. Noise floor is lowered by roughly 6db for each bit. So as you can imagine, 1-bit DSD has significantly more ultrasonic noise than even 16-bit PCM. This is part of why wide-DSD formats with 5-bit to 8-bit parallel Delta-Sigma decoding were created. With PCM, you have to deal with significant noise at the sampling frequency. This is why Sony and Philips engineered the Red Book CD to sample at 44.1KHz, which is over twice the human high-frequency hearing limit of 20KHz.
Since quantization noise is present around the sampling frequency of a PCM recording, a 44.1KHz recording has quantization noise one octave above the human hearing limit of 20KHz. This quantization noise needs to be filtered out, so all DACs have a low-pass filter at the output. Because the quantization noise is only one octave above audibility the filters used have a very steep slope so as to not filter out desirable high frequencies. These steeply sloped low-pass digital filters are commonly known as "brick wall" filters. This is why there can be an advantage in playing 44.1KHz PCM upsampled to 88.2KHz or 176.4KHz.
Though you hear a lot about "brick wall" filters causing an audible distortion in the top end of early Red Book CD players , the fact is that was only a small part of the reason early Red Book CDs and players had an unnatural sounding top end. Most of the hard, harsh, unnatural sounding high frequencies in early digital had more to do with flaws in the power supplies and flaws in the recording process, not "brick wall" filters.
Sorry to be the one to burst your bubble, but despite what many audiophiles may believe, less than one person in a thousand can hear anything above 20KHz as a child and there is almost no one over the age of 40 who can hear much above 15KHz.
Of course DSD64 is another story: above 25KHz the quantization noise rises sharply, requiring far more sophisticated filters and/or noise-shaping algorithms. See graphic below. When you filter the output of DSD64 with a simple low-pass filter, the result is distorted phase/time and some rather nasty artifacts in the audible range. The solution is noise-shaping algorithms which move the noise to less audible frequencies and/or higher sampling rates. This is why Double-Rate DSD and Quad-Rate DSD formats came into being. This is also why advanced player software, such as JRiver, offers Double-Rate DSD output. Using player software that upsamples DSD64 to DSD128 or DSD256 significantly improves performance by putting the digital artifacts octaves above audibility allowing more advanced noise-shaping algorithms and less severe digital filters. Note these extremely high sampling frequencies are why ultra accurate clocking is more important in the playback of DSD than PCM recordings. 152ee80cbc
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