Ideally the DAC runs the show with its clock ticking the samples through it precisely and data is pulled asynchronously to avoid any jitter and buffered to avoid buffer under-run. Spdif, TOSLink or I2S are synchronous interfaces introducing multiple clock masters. Streaming music is designed to work asynchronously making it possible to play audiophile music from some data center e.g. AWS (Amazon Web Services) where music data is just one of many categories of data and what matters is that data stays error-free and is delivered as requested with lowest latency. Music data is not even particularly hard to do over the internet nowadays in lossless CD quality or better.
Sorry @ThinkingBetter, You missed the whole point and are going to confuse a lot of people. The digital audio data is packetized and sent asynchronously but those packets are reassembled into a data stream that is identical to a SPDIF, I2S, AES stream as if from a cable interface. The real digital audio clock, which was already present and embedded before the stream was packetized, is then extracted from the digital audio just as it is from SPDIF, I2S, AES. So no new digital audio clocking is taking place. DACs are not expected to provide digital audio clocking. That is done at the ADC. DACs extract the clock and we don't want a clock to be unrelated to the clock imbedded in the digital audio stream.
@@mikeh2520 No, the point is exactly that your DAC is where the precision of timing matters and it should serve as the clock master. Thus of course it's expected to provide digital audio clocking. Any one-way synchronous connection to a DAC such as SPDIF from a separate box will introduce another clock master and you will have potential jitter issues and need PLL in the DAC. USB and Ethernet allow asynchronous data transfer giving full timing control to the DAC. Any master clock or "re-clocking" further upstream of the music data really should be avoided when the architecture is optimum. Unfortunately, a CD transport is based on solutions from an era where the CD drive ran its own DAC and was clock master thus you end up with two clock masters. Also still more often than not, your streaming player runs a separate clock and you again have two clock masters. To make matter worse, often operating systems such as Linux, Mac OS, Windows, Android and iOS run their PCM plumbing at 48kHz and you end up with sample rate conversion before you push the music via a synchronous line potentially both yielding SRC distortion and jitter. Best is to run in exclusive mode on the OS streamer/player side and with asynchronous USB connected DAC.
@@x-techgaming He didn't. He said AWS. Both Tidal and Qobuz utilize Amazon Web Services (AWS) for their operations. Qobuz hosts and serves its content to consumers worldwide through AWS infrastructure. Similarly, Tidal leverages AWS for various aspects of its service delivery.
Because video tape (3/4" Umatic, later Beta/VHS) was the first common storage system for the first commercially available Analog to Digital converters, 44.1Khz sample rate was chosen for audio CDs because it was the lowest usable sample rate compatible with both NTSC and PAL video standards, allowing for encoding with around 3 samples per video line per audio channel. I sold those systems. Living in Nashville at the time I was also exposed to others like the SoundStream which used four-channel, 16 bit, 50 kHz sample rate into Honeywell 5600e Tape Drives.
Can you explain why 44.1 in compatible with NTSC and PAL? I know that 48 is to synchronize with 24 images per second film (16mm, 35mm, etc), but I don't get the math for vcr tapes.
Never be sorry for rambling on a bit Paul. I'm a mechanical engineer not a sound engineer, but I enjoy listening all the same, there's always something to be learned.
Paul's just being polite! There probably isn't enough of this, in the World, today! I'm retired, from the Aerospace Industry. Investments, in Real Estate, have enabled me to buy a complete, PS Audio system; I'd by the largest speakers, don't have an adequate ~ or really, any ~ place to put them! All of the recommended, quality AudioQuest cables, connectors, sounds great! I almost wish that I was still working, all day, & so couldn't use my cell phone. When it's turned on, there are continuous Cold Calls, Robo Calls, Sales Calls, etc., many of which are useless, scams, just bad, a nuisance, unscrupulous. Most of these employ: "Spoofing," which means that they use someone else's phone number, to call, me. When I call the number back, contacted are: Police Stations, Hospitals, Nurses, & so on. Schools. When I briefly explain what happened, they say, "I'm sorry." I nicely thank them, and say, "You don't have to be sorry, it's not your fault." I've been complemented, at times {or, sometimes, criticized}, for being polite, even, `too polite.' I say that it's better to be polite, nice, than to be rude, brusque, uncaring, inconsiderate, & so on!
It is really simple in home stereo. The only clock that matters is the one that times the DAC. Typically, there are two clocks. One for 44.1 and another for 48kHz and their multiples. For any given track the appropriate clock is selected. I only listen to redbook, and use a single clock at 5MHz optimized for 44.1 playback. Improvements in phase noise of the oscillator will improve the sound. Lower phase noise results in natural sound, low level detail, better sound stage and a stronger bass foundation. You listen to the last clock that drives the DAC. A master clock is not needed. If the FIFO buffer before the DAC is working to time the DAC and isolate noise, the other clocks have no impact in playback. They are all in a different time domain than playback and only count to the extent that they introduce noise into your ground plane. The clock used to create the sound track determines the quality of the recording but you cannot do anything about that other than to choose good recordings.
This is what I would have thought, I.e. that even a small amount of buffering before the DAC should make the other (upstream) clocks irrelevant. This could of course be different in a live setting where multiple digital sources may need to be synchronized in real time.
An alternative, the bluish preparation on the left, makes the walls melt, is relaxing and more digestible than LSD (better than most blotter LSD I tried). The acidified preparation on the right (lemon juice), sobers my mind and is a better quality of stimulant than amphetamine, no heart palpitations nor lack of sleep to recover from ..I feel strong while walking and wake feeling great, as with cannabis. The blue liquid lingers in the brain's biochemistry, being metabolised for two or three weeks, after a 6 to 12 hour trip. Not what I want, most of the time ..I like the milder natural stimulant version best, still much stronger/ effective than coffee. It is also more difficult, for the bad guys to put Fentanyl in fungi. I keep a hot sauce bottle in my pocket (like a mickey), a sip every two or three hours, to stay alert and energetic. That was handy, while foraging active Canadian species of mushrooms in nature and preparing outdoor beds/ patches. The mushrooms no longer smell like dirty feet, dehydrators are considered necessary equipment. I do not have one but am the only person, checking for Spruce Bud Worms, by breaking fungi into smaller pieces (myco community slow to accept important information). Their metabolisms bio-synthesize nerve-toxins, from Tryptamine-producing fungi. I had the Wood Lover's Paralysis experience in Ottawa, I was the one given the Amanita cap ..I figured the crunchy bits would only be more protein, a whole night of convulsions and hypothermia ensued. In my trip, it was soooo cold, I felt like I was flying over the North Atlantic ..so cold. AAaaah cr*p, I did it to myself again, another learning experience. I drank some of the mycelium, along with the liquid at the bottom, that make the experience somewhat difficult, compared to the clear liquid above the very bottom. I was wondering, why are the walls starting to move again? ..DUH Dean! ..the blue Psilocybin polymer, has a higher molecular weight, than ONE of the two Psilocin monomers it is composed of. Confirmed, I prefer the liquid above the very bottom. The mycelium are not quite clones, they are inbred mycelium (bred with itself ..an IBL, inbred line). I do not feel so much like a caveman painting on walls, when having the best liquid/ mixture at the top. I can be functional and in public, using Psilocin ..Psilocybin requires a time and place to trip comfortably.
I never bother over 'clocks' in my studio except the one that keeps track of how much money I'm making. My DAW does its business as it should, and every other piece of gear follows along without question, and so in the end I like to render to my final track digital master as a 44.1 kHz lossless WAV file in Stereo although sometimes Quadrophonic or Dolby 5-Point Surround mixes depending on the client or mood if it's my own stuff. No clocks involved although at some point I will have to do so because I'm incorporating video shooting and editing in 2025. Here, you give something, so I return the favor. This was the first song off the first album I did 12 years ago when I first 'opened'. I was barely anything at that time but this was the result. The TH-cam compression and other little things they do haven't affected anything in the years it's been on here. Which is surprising. Enjoy! - th-cam.com/video/J-4KhwBvrps/w-d-xo.htmlfeature=shared
This isn't the standard terminology: what Paul is talking about are the internal clocks of units and their clock generators (or preferably, a clock each for the 44.1kHz and 48kHz groups or "families"). The clocks of all these units (such as in a recording studio) may then be synchronized by a (usually external) master clock. Alternatively, the inferior but more common way of doing it in home audio, the internal clocks of downstream units may be slaved to the source's.
The DAC does not rely on a clock because the clock is imbedded in the AES, SPDIF or I2S stream which feeds the DAC. Open the cover of a good DAC and you will not find a single oscillator module. There are some DACs and other devices that attempt to re-clock the stream but that is known to introduce bigger problems than it attempts to solve. Think of the clock and the stream data as a bike chain on a sprocket. The imbedded clock matches the the data like the teeth match the chain. If you change that relationship samples can be falling at one side or the other of the original transition. Analog to Digital Converters do rely on a clock and that clock is very important because that is the timing reference for encoding which will be a permanent part of the encoded audio. Mastering engineers rely on very high quality master clocks such as the Antelope Isochrone Trinity. Antelope discovered that they can make a better clock by introducing randomised jitter which actually gives better results than jitter that has some correlation to the signal.
@@stupdasso USB audio is packetized data with no timing reference and it needs a clock for re-assembly and buffering of the packets. The digital audio clock is still extracted from the digital audio just like SPDIF. Look at the PCM2704 data sheet for a good explanation.
@@stupdasso USB transmits audio in packet form with no timing reference included. To put those packets back together as a digital audio stream, a clock is needed to manage the buffering of the packets and the reassembly of them. The digital audio clock is then extracted from the digital audio stream exactly like the AES, SPDIF and I2S stream clocks are. So with a USB interface on a DAC, you have a clock but that clock is not a digital audio clock. It is simply managing the timing of the packet to digital stream reassembly. The clock signal is still always embedded in the original digital audio but needs some management to get it into shape to be extracted.
Async USB is the way to go. When i hear people who wants to use I2S... it is a standard for inter-chip communication over very short distance...AND it's another of those bad idea where the DAC is slave clocked from the input signal...(just as SP/DIF & AES). "Open the cover of a good DAC and you will not find a single oscillator module"
@@guyboisvert66 You are wrong about this. For a digital USB interface, a digital audio stream is taken and cut up into packets and sent asynchronously by the USB protocol meaning that no timing is sent with a packetized audio network connection. That is the only way that the USB or the audio over ethernet works. The packets are then buffered and reassembled and converted back into the original digital audio stream which at that point looks the same as a SPDIF or AES signal. The clock is then extracted from that digital stream exactly as it is done with SPDIF etc. There is no reclocking being done at all. The clock is still synchronous with the original digital audio after reassembly. Don't confuse this. Have a look at the high end DAC here at this link. It is an Audio Note Level 5 DAC. Notice the digital board only has a couple of input transformers, relays, an input receiver chip some glue logic and a DAC chip (and some voltage regulators). briansmith1.smugmug.com/Quad-Cire-Level-5-DAC-Mentor- And if you still have ideations about USB audio having some special superior re-clocking, then take a look at the circuit in this thread which is what the typical USB input is on high end DACs. Notice the SPDIF output of the circuit which gets switched in as a source with relays on the DACs. www.diyaudio.com/community/threads/usb-to-spdif-converter.62537/ The 12 MHz crystal here is not a clock doing any sort of digital audio stream re-clocking. Look at the data sheet if you still have doubts.
I couldn’t help but notice the notes on your left in reference to tariffs. Considering the potential changes coming in how we, and other countries impose them, does this or will this affect how you do international business? Will this also affect your prices too? Just wondering how all this could affect PSA, if at all.
Perhaps the missed part of the question is whether a master clock can positively impact performance as a separate, dedicated component. The answer in my experience is yes but it comes at a real cost.
Using 10MHz master clock nowadays would lead to worsen sound quality. As present digital equipment working with higher frequency crystals, it needs circuit to multiple the clock signal, hence increase noise. It's better to using internal clock with specific designed frequency as common frequencies using in Audio DAC and Streamer nowadays are 12, 22.579, 24, 24.576, 25, 38.4, 45.158, 49.152, 50, 54, 90.316, 98.304, and 100MHz.
Stomach is settled, am comfortable with walls moving a bit, two hours in ..time for a witching-hour broadcast, the midnight mix. th-cam.com/users/livevLcJwQTbboc?feature=share Dilla's unedited/ unfinished works, are now among some of his most celebrated. I also like to share the unedited recording of a mix, as proof I am not using AI. Humans make some mistakes but they also accentuate the more well-put-together segments of the mix.
Very most impacting element on synchronous digital transfer or communication. No stable clock in the network, no business. No network, no biggie. Master clock is not about belief, go to church for that. You place the master clock typically at the most solid system that all other systems are connected to. All systems must be able to work as slave and self-clocked as a backup. Sophisticated systems can map multiple Master clocks for contingency. It‘s like a orchestra where there‘s only one director, otherwise you play your own rhythm or you seek an alternative musician to pick on his rhythm to align to. If you only have a single digital device, this is of no relevance, if you have two, they need to be mapped but can‘t always, i.e. pc+dac, or cd-transport+dac, typically a PC will not follow others, a clocking card may establish a Master, but it‘s left to the dac to buffer-sync on the the USB/FW port before amping and feeding the analog outputs. That works at home hifi. If you go all device separate, stream, file, server, rate-conversion, dedicated clocking system, digital patchpanels, have digital routing, compressors, mixers, etc. I don‘t explain what you already know. Your clocking strategy is key here and the gear shall support it. All in all, I agree with Paul, while very relevant in complex chains, not in home context of a 1 max 2 digital items serial connection.
Yes, but in stereo playback the DAC runs asynchronously, and does not need to be in sync with anything else. Perhaps in a video system you want the picture and the sound in sync the notion of a master has some function, but now we're talking about a far inferior sound system.
@2:11 "If you have a transport..." If you play digital content, then you have a transport. There are stand-alone boxes that handle transport functions. And there are all kinds of other boxes that have transports included. If, for example, you purchase a $50 computer, and use it to feed your stereo, then you are using the transport that is built in to that $50 computer. If you have a device, any device, that can play digital music, then that device has a built-in transport. It might not be very good. But it is a transport. Every CD player, every computer (from the last 30 years), every smart phone, etc, has a transport.
What is annoying is that high quality lab grade clocks, including Rubidium, if you want to get really excessive, tend to be 10MHz, converting that to for 44.1 or 48 KHz sample rate audio is a right pain.
Paul, you're definitely wrong on this - master clocks can make a HUGE difference, as they are designed to much closer tolerances than the ones built into equipment for cost reasons.
Yes but in this case the 'master clock' is there simply to improve the clock that counts, the one that drives the DAC. The system designer should simply have built the DAC with a SOTA clock to do its job. Instead they leave the door open for an 'optional' master clock simply to charge more money. The notion of a master clock is for situations where multiple cooperating systems need to be synchronized. A DAC runs asynchronously from everything else in the chain and has no need for a master.
Never be sorry Paul as an x sound engineer, let me tell you I enjoyed it.🙂👍🎧
Ideally the DAC runs the show with its clock ticking the samples through it precisely and data is pulled asynchronously to avoid any jitter and buffered to avoid buffer under-run. Spdif, TOSLink or I2S are synchronous interfaces introducing multiple clock masters. Streaming music is designed to work asynchronously making it possible to play audiophile music from some data center e.g. AWS (Amazon Web Services) where music data is just one of many categories of data and what matters is that data stays error-free and is delivered as requested with lowest latency. Music data is not even particularly hard to do over the internet nowadays in lossless CD quality or better.
100%. I was going to respond to the video, but I won't bother since you said everything that needed to be said.
Sorry @ThinkingBetter, You missed the whole point and are going to confuse a lot of people. The digital audio data is packetized and sent asynchronously but those packets are reassembled into a data stream that is identical to a SPDIF, I2S, AES stream as if from a cable interface. The real digital audio clock, which was already present and embedded before the stream was packetized, is then extracted from the digital audio just as it is from SPDIF, I2S, AES. So no new digital audio clocking is taking place. DACs are not expected to provide digital audio clocking. That is done at the ADC. DACs extract the clock and we don't want a clock to be unrelated to the clock imbedded in the digital audio stream.
@@mikeh2520 No, the point is exactly that your DAC is where the precision of timing matters and it should serve as the clock master. Thus of course it's expected to provide digital audio clocking. Any one-way synchronous connection to a DAC such as SPDIF from a separate box will introduce another clock master and you will have potential jitter issues and need PLL in the DAC. USB and Ethernet allow asynchronous data transfer giving full timing control to the DAC. Any master clock or "re-clocking" further upstream of the music data really should be avoided when the architecture is optimum. Unfortunately, a CD transport is based on solutions from an era where the CD drive ran its own DAC and was clock master thus you end up with two clock masters. Also still more often than not, your streaming player runs a separate clock and you again have two clock masters. To make matter worse, often operating systems such as Linux, Mac OS, Windows, Android and iOS run their PCM plumbing at 48kHz and you end up with sample rate conversion before you push the music via a synchronous line potentially both yielding SRC distortion and jitter. Best is to run in exclusive mode on the OS streamer/player side and with asynchronous USB connected DAC.
Please don't say audiophile and Amazon Music in the same sentence 😭
@@x-techgaming He didn't. He said AWS. Both Tidal and Qobuz utilize Amazon Web Services (AWS) for their operations. Qobuz hosts and serves its content to consumers worldwide through AWS infrastructure. Similarly, Tidal leverages AWS for various aspects of its service delivery.
Because video tape (3/4" Umatic, later Beta/VHS) was the first common storage system for the first commercially available Analog to Digital converters, 44.1Khz sample rate was chosen for audio CDs because it was the lowest usable sample rate compatible with both NTSC and PAL video standards, allowing for encoding with around 3 samples per video line per audio channel. I sold those systems. Living in Nashville at the time I was also exposed to others like the SoundStream which used four-channel, 16 bit, 50 kHz sample rate into Honeywell 5600e Tape Drives.
Can you explain why 44.1 in compatible with NTSC and PAL? I know that 48 is to synchronize with 24 images per second film (16mm, 35mm, etc), but I don't get the math for vcr tapes.
@@FrancisEVidal If the formatting holds up:
...... Active lines/field Fields/second Samples/line Resulting sample rate
PAL 294 50 3 294 × 50 × 3 = 44,100 Hz
NTSC 245 60 3 245 × 60 × 3 = 44,100 Hz
Excellent point I was racking my brain as he was talking trying to remember the details on this
Thank you so much for answering my question! What an honor. Now I know that I should focus on improving cables or devices instead of clocks 🙂
Never be sorry for rambling on a bit Paul. I'm a mechanical engineer not a sound engineer, but I enjoy listening all the same, there's always something to be learned.
Paul's just being polite! There probably isn't enough of this, in the World, today! I'm retired, from the Aerospace Industry. Investments, in Real Estate, have enabled me to buy a complete, PS Audio system; I'd by the largest speakers, don't have an adequate ~ or really, any ~ place to put them! All of the recommended, quality AudioQuest cables, connectors, sounds great! I almost wish that I was still working, all day, & so couldn't use my cell phone. When it's turned on, there are continuous Cold Calls, Robo Calls, Sales Calls, etc., many of which are useless, scams, just bad, a nuisance, unscrupulous. Most of these employ: "Spoofing," which means that they use someone else's phone number, to call, me. When I call the number back, contacted are: Police Stations, Hospitals, Nurses, & so on. Schools. When I briefly explain what happened, they say, "I'm sorry." I nicely thank them, and say, "You don't have to be sorry, it's not your fault." I've been complemented, at times {or, sometimes, criticized}, for being polite, even, `too polite.' I say that it's better to be polite, nice, than to be rude, brusque, uncaring, inconsiderate, & so on!
It is really simple in home stereo.
The only clock that matters is the one that times the DAC. Typically, there are two clocks. One for 44.1 and another for 48kHz and their multiples. For any given track the appropriate clock is selected. I only listen to redbook, and use a single clock at 5MHz optimized for 44.1 playback.
Improvements in phase noise of the oscillator will improve the sound. Lower phase noise results in natural sound, low level detail, better sound stage and a stronger bass foundation.
You listen to the last clock that drives the DAC. A master clock is not needed. If the FIFO buffer before the DAC is working to time the DAC and isolate noise, the other clocks have no impact in playback. They are all in a different time domain than playback and only count to the extent that they introduce noise into your ground plane.
The clock used to create the sound track determines the quality of the recording but you cannot do anything about that other than to choose good recordings.
This is what I would have thought, I.e. that even a small amount of buffering before the DAC should make the other (upstream) clocks irrelevant. This could of course be different in a live setting where multiple digital sources may need to be synchronized in real time.
The Master Clock I pay most attention to is her "It's 11PM, time for quiet! I'm going to bed!
I set my clock back on November 3.
from 48 to 44.1 khz
Good morning all and welcome to another episode of Paul McGowan, ASMR.
my love for you is ticking clock..... Bezerrrkerrrr!
An alternative, the bluish preparation on the left, makes the walls melt, is relaxing and more digestible than LSD (better than most blotter LSD I tried). The acidified preparation on the right (lemon juice), sobers my mind and is a better quality of stimulant than amphetamine, no heart palpitations nor lack of sleep to recover from ..I feel strong while walking and wake feeling great, as with cannabis.
The blue liquid lingers in the brain's biochemistry, being metabolised for two or three weeks, after a 6 to 12 hour trip. Not what I want, most of the time ..I like the milder natural stimulant version best, still much stronger/ effective than coffee.
It is also more difficult, for the bad guys to put Fentanyl in fungi.
I keep a hot sauce bottle in my pocket (like a mickey), a sip every two or three hours, to stay alert and energetic. That was handy, while foraging active Canadian species of mushrooms in nature and preparing outdoor beds/ patches.
The mushrooms no longer smell like dirty feet, dehydrators are considered necessary equipment. I do not have one but am the only person, checking for Spruce Bud Worms, by breaking fungi into smaller pieces (myco community slow to accept important information). Their metabolisms bio-synthesize nerve-toxins, from Tryptamine-producing fungi.
I had the Wood Lover's Paralysis experience in Ottawa, I was the one given the Amanita cap ..I figured the crunchy bits would only be more protein, a whole night of convulsions and hypothermia ensued. In my trip, it was soooo cold, I felt like I was flying over the North Atlantic ..so cold.
AAaaah cr*p, I did it to myself again, another learning experience. I drank some of the mycelium, along with the liquid at the bottom, that make the experience somewhat difficult, compared to the clear liquid above the very bottom. I was wondering, why are the walls starting to move again?
..DUH Dean! ..the blue Psilocybin polymer, has a higher molecular weight, than ONE of the two Psilocin monomers it is composed of. Confirmed, I prefer the liquid above the very bottom.
The mycelium are not quite clones, they are inbred mycelium (bred with itself ..an IBL, inbred line).
I do not feel so much like a caveman painting on walls, when having the best liquid/ mixture at the top. I can be functional and in public, using Psilocin ..Psilocybin requires a time and place to trip comfortably.
I never bother over 'clocks' in my studio except the one that keeps track of how much money I'm making. My DAW does its business as it should, and every other piece of gear follows along without question, and so in the end I like to render to my final track digital master as a 44.1 kHz lossless WAV file in Stereo although sometimes Quadrophonic or Dolby 5-Point Surround mixes depending on the client or mood if it's my own stuff. No clocks involved although at some point I will have to do so because I'm incorporating video shooting and editing in 2025.
Here, you give something, so I return the favor. This was the first song off the first album I did 12 years ago when I first 'opened'. I was barely anything at that time but this was the result. The TH-cam compression and other little things they do haven't affected anything in the years it's been on here. Which is surprising. Enjoy! - th-cam.com/video/J-4KhwBvrps/w-d-xo.htmlfeature=shared
You could add a DIY guide to each, as a training and incentivising guide to create gear or modd it.
Just when you think you’re getting a decent understanding of audio. 😅 fml
No kidding !
This isn't the standard terminology: what Paul is talking about are the internal clocks of units and their clock generators (or preferably, a clock each for the 44.1kHz and 48kHz groups or "families"). The clocks of all these units (such as in a recording studio) may then be synchronized by a (usually external) master clock. Alternatively, the inferior but more common way of doing it in home audio, the internal clocks of downstream units may be slaved to the source's.
The DAC does not rely on a clock because the clock is imbedded in the AES, SPDIF or I2S stream which feeds the DAC. Open the cover of a good DAC and you will not find a single oscillator module. There are some DACs and other devices that attempt to re-clock the stream but that is known to introduce bigger problems than it attempts to solve. Think of the clock and the stream data as a bike chain on a sprocket. The imbedded clock matches the the data like the teeth match the chain. If you change that relationship samples can be falling at one side or the other of the original transition. Analog to Digital Converters do rely on a clock and that clock is very important because that is the timing reference for encoding which will be a permanent part of the encoded audio. Mastering engineers rely on very high quality master clocks such as the Antelope Isochrone Trinity. Antelope discovered that they can make a better clock by introducing randomised jitter which actually gives better results than jitter that has some correlation to the signal.
USB
@@stupdasso USB audio is packetized data with no timing reference and it needs a clock for re-assembly and buffering of the packets. The digital audio clock is still extracted from the digital audio just like SPDIF. Look at the PCM2704 data sheet for a good explanation.
@@stupdasso USB transmits audio in packet form with no timing reference included. To put those packets back together as a digital audio stream, a clock is needed to manage the buffering of the packets and the reassembly of them. The digital audio clock is then extracted from the digital audio stream exactly like the AES, SPDIF and I2S stream clocks are. So with a USB interface on a DAC, you have a clock but that clock is not a digital audio clock. It is simply managing the timing of the packet to digital stream reassembly. The clock signal is still always embedded in the original digital audio but needs some management to get it into shape to be extracted.
Async USB is the way to go. When i hear people who wants to use I2S... it is a standard for inter-chip communication over very short distance...AND it's another of those bad idea where the DAC is slave clocked from the input signal...(just as SP/DIF & AES). "Open the cover of a good DAC and you will not find a single oscillator module"
@@guyboisvert66 You are wrong about this. For a digital USB interface, a digital audio stream is taken and cut up into packets and sent asynchronously by the USB protocol meaning that no timing is sent with a packetized audio network connection. That is the only way that the USB or the audio over ethernet works. The packets are then buffered and reassembled and converted back into the original digital audio stream which at that point looks the same as a SPDIF or AES signal. The clock is then extracted from that digital stream exactly as it is done with SPDIF etc. There is no reclocking being done at all. The clock is still synchronous with the original digital audio after reassembly. Don't confuse this. Have a look at the high end DAC here at this link. It is an Audio Note Level 5 DAC. Notice the digital board only has a couple of input transformers, relays, an input receiver chip some glue logic and a DAC chip (and some voltage regulators). briansmith1.smugmug.com/Quad-Cire-Level-5-DAC-Mentor- And if you still have ideations about USB audio having some special superior re-clocking, then take a look at the circuit in this thread which is what the typical USB input is on high end DACs. Notice the SPDIF output of the circuit which gets switched in as a source with relays on the DACs. www.diyaudio.com/community/threads/usb-to-spdif-converter.62537/ The 12 MHz crystal here is not a clock doing any sort of digital audio stream re-clocking. Look at the data sheet if you still have doubts.
I couldn’t help but notice the notes on your left in reference to tariffs. Considering the potential changes coming in how we, and other countries impose them, does this or will this affect how you do international business? Will this also affect your prices too? Just wondering how all this could affect PSA, if at all.
When I record tracks that I never finish it's in 96/24. 😂 I highly rec the rme babyface pro fs even if you don't finish recordings.
Perhaps the missed part of the question is whether a master clock can positively impact performance as a separate, dedicated component. The answer in my experience is yes but it comes at a real cost.
Spring forward fall back, right?
Using 10MHz master clock nowadays would lead to worsen sound quality. As present digital equipment working with higher frequency crystals, it needs circuit to multiple the clock signal, hence increase noise. It's better to using internal clock with specific designed frequency as common frequencies using in Audio DAC and Streamer nowadays are 12, 22.579, 24, 24.576, 25, 38.4, 45.158, 49.152, 50, 54, 90.316, 98.304, and 100MHz.
Paul then in broadcast digital sampling rate is/was 32 KHz.
Stomach is settled, am comfortable with walls moving a bit, two hours in ..time for a witching-hour broadcast, the midnight mix. th-cam.com/users/livevLcJwQTbboc?feature=share
Dilla's unedited/ unfinished works, are now among some of his most celebrated. I also like to share the unedited recording of a mix, as proof I am not using AI. Humans make some mistakes but they also accentuate the more well-put-together segments of the mix.
Very most impacting element on synchronous digital transfer or communication. No stable clock in the network, no business. No network, no biggie.
Master clock is not about belief, go to church for that.
You place the master clock typically at the most solid system that all other systems are connected to. All systems must be able to work as slave and self-clocked as a backup. Sophisticated systems can map multiple Master clocks for contingency.
It‘s like a orchestra where there‘s only one director, otherwise you play your own rhythm or you seek an alternative musician to pick on his rhythm to align to.
If you only have a single digital device, this is of no relevance, if you have two, they need to be mapped but can‘t always, i.e. pc+dac, or cd-transport+dac, typically a PC will not follow others, a clocking card may establish a Master, but it‘s left to the dac to buffer-sync on the the USB/FW port before amping and feeding the analog outputs. That works at home hifi.
If you go all device separate, stream, file, server, rate-conversion, dedicated clocking system, digital patchpanels, have digital routing, compressors, mixers, etc. I don‘t explain what you already know. Your clocking strategy is key here and the gear shall support it.
All in all, I agree with Paul, while very relevant in complex chains, not in home context of a 1 max 2 digital items serial connection.
Isn’t the master clock used as a reference for all other digital clocks in the system so they are in sync?
Yes, but in stereo playback the DAC runs asynchronously, and does not need to be in sync with anything else. Perhaps in a video system you want the picture and the sound in sync the notion of a master has some function, but now we're talking about a far inferior sound system.
@2:11 "If you have a transport..."
If you play digital content, then you have a transport.
There are stand-alone boxes that handle transport functions. And there are all kinds of other boxes that have transports included.
If, for example, you purchase a $50 computer, and use it to feed your stereo, then you are using the transport that is built in to that $50 computer.
If you have a device, any device, that can play digital music, then that device has a built-in transport. It might not be very good. But it is a transport.
Every CD player, every computer (from the last 30 years), every smart phone, etc, has a transport.
What is annoying is that high quality lab grade clocks, including Rubidium, if you want to get really excessive, tend to be 10MHz, converting that to for 44.1 or 48 KHz sample rate audio is a right pain.
All that for "Master clocks? Don't use them!" lol!
"Condominiums?
Never use 'em".
Paul, you're definitely wrong on this - master clocks can make a HUGE difference, as they are designed to much closer tolerances than the ones built into equipment for cost reasons.
Yes but in this case the 'master clock' is there simply to improve the clock that counts, the one that drives the DAC. The system designer should simply have built the DAC with a SOTA clock to do its job. Instead they leave the door open for an 'optional' master clock simply to charge more money.
The notion of a master clock is for situations where multiple cooperating systems need to be synchronized. A DAC runs asynchronously from everything else in the chain and has no need for a master.