Several well-respected audiophiles in this forum have stated that the sound quality of hi-res streamed audio equals or betters the sound quality of traditional digital sources.
These are folks who have spent decades assembling highly desirable systems and whose listening skills are beyond reproach. I for one tend to respect their opinions.
Tidal is headquartered in NYC, NY from Norwegian origins. Qobuz is headquartered in Paris, France. Both services are hosted on Amazon Web Services (AWS), the cloud infrastructure services giant that commands roughly one third of the world's entire cloud services market.
AWS server farms are any audiophile's nightmare. Tens of thousands of multi-CPU servers and industrial-grade switches crammed in crowded racks, miles of ordinary cabling coursing among tens of thousands of buzzing switched-mode power supplies and noisy cooling fans. Industrial HVAC plants humming 24/7.
This, I think, demonstrates without a doubt that audio files digitally converted to packets of ones and zeroes successfully travel thousands of miles through AWS' digital sewer, only to arrive in our homes completely unscathed and ready to deliver sound quality that, by many prominent audiophiles' account, rivals or exceeds that of $5,000 CD transports.
This also demonstrates that digital transmission protocols just work flawlessly over noise-saturated industrial-grade lines and equipment chosen for raw performance and cost-effectiveness.
This also puts in perspective the importance of improvements deployed in the home, which is to say in the last ten feet of our streamed music's multi-thousand mile journey.
No worries, I am not about to argue that a $100 streamer has to sound the same as a $30,000 one because "it's all ones and zeroes".
But it would be nice to agree on a shared-understanding baseline, because without it intelligent discourse becomes difficult. The sooner everyone gets on the same page, which is to say that our systems' digital chains process nothing less and nothing more than packets of ones and zeroes, the sooner we can move on to genuinely thought-provoking stuff like, why don't all streamers sound the same? Why do cables make a difference? Wouldn't that be more interesting?
I guess parts is parts and pieces is pieces and it don’t matter in what order they go cause it’s all gonna sound the same.
Simply...No.
Most of the conversation so far has been on digital data transmission and receipt. While the fundmental design of a digital network insures that what was sent ends up being the same data at the end, factors such as the effect of the noise floor on creating data packet resend requests, can effect the timing of the data feed at the final end point. (In this instance, the network connection at the streamer.)
What hasn't been really talked about yet is how all the hardware and software "parts and pieces" come into play. One goal is to reduce (or isolate) the noise floor on the data transmission by using good hardware/software, like adequate power sources, quality connectors and wiring, capable networking equipment, etc. Reduce the noise floor and, hopefully, reduce the data timing difficulties at the end point. Thus it changes the sound (hopefully for the better.)
Then there is how all the parts and pieces come together after the final network connection at the streamer and/or DAC. Then the circuit design, the parts chosen, software operating system, firmware, DAC chips, shielding, power supplies, etc. all come together to take the digital data stream and convert it to an analog signal. Doing this well can be difficult. And it is where the greatest differences in the "sound" of a streamer/DAC come from.
And then you get to enjoy all the different types of chicken nuggets from places like McDonalds, Taco Bell, KFC, and Bojangles. Same chicken, different receipes. LOL.
I having to remember and Google. It was a few years ago.
The modified DLink was an Aqvox. I remember that because there was a Linus Tech Tip on YouTube that did the tare down. I also seem to remember there was a Paul Pang modified DLink that was somewhat dubious (I had to Google this one for the name.)
The Netgear may have been one of the early Silent Angel switches, but don't hold me to that.
And the Linksys, no clue anymore. I just remember the pictures because they painted the case a matte black and didn't even mask the open ports properly. On top of that, they just pasted their sticker over top of the Linksys sticket on the bottom.
honestly, I think there are folks who champion high end streamers whose arguments are the functional equivalent of those who call themselves “Creationists”. It’s essentially pseudo-science.
If you want to get into switches that are something more than repackaged off the shelf check out the Dejitter Switch X, they have some pretty interesting perspectives on how networks can affect sound quality.
Audiophile switch sellers purchase off-the-shelf switches, ditch the noisy switching mode power supply (SMPS) that came with it, repackage the switch with a small linear power supply (LPS), and sell it at a huge markup.
Your global statement seems untrue, what about Telegärtner, Network Acoustics, Ansuz…. While it’s true that very many (most?) have tried to modify off the shelf switches to find significant sonic improvements at a more economical cost, results have commonly been marginal if any sonic improvements. I followed “audiophile switches” reviews and forums looking for the lowest cost but sonically effective solutions, but seems the costlier alternatives simply perform consistently sonically better.
Why does a costlier switch perform better than a stock and/or a tricked out stock switch? In this copycat world, manufacturers rarely reveal their findings through their R&D efforts which is vexing for those who try to understand markup “value” - they will never get there.
I meant to add "Some" in front of "Audiophile switch sellers" and I never did, but you're right. Vendors like Silent Angel and those you mentioned appear to produce their own PCBs, thereby offering an original product, which is really the right / ethical way to go - especially at the $3-4,000 price point.
On a higher level, high-end switches still send network packets on their way just like Monoprice switches, and there are limits to how quiet their power supplies can be made.
Speaking of which, if you go to this page and scroll down you will find a pic of the Silent Angel Bonn NX's ($3,999) main PCB. Isn't that an onboard SMPS right there at #4?
How many times have you entered $50 to deposit in your bank account and it became $5000 because the banks server was 3000 miles away?
We are just talking music here, not a lot of data and not hard to do no matter the miles away the disk is. When you watch (stream) an F1 race 10,000 miles away in 4K, how many times does Hamilton’s red Ferrari gets displayed in blue? 4K streaming is much more difficult than streaming a 16/44 song or a hires song.
I setup very large enterprise databases and worked with Facebook/Apple/Yahoo/Comcast/banks and these companies have servers all over the world on AWS/Oracle/Azure/their own server farms and every transaction done is bit perfect, with thousands/millions of people concurrently accessing these servers and every transaction has to be bit perfect or we would have some big issues. And this occurs over dirty WiFi or using a $1 Ethernet patch cable. Now you take an audiophile that uses a $1000 Ethernet cable going to a galvanized streamer going to a $10000 dac, I’m not worried that a 1500 byte packet will make it to my system perfectly.
"The digital data rides on an analog signal that can pick up noise"
I think you're trying to explain Wi-Fi, which is a Layer 2 Data Link Protocol, as is Ethernet. Layer 3 Network Protocols, like TCP/IP, handle all the logical addressing, routing, forwarding, fragmentation and reassembly, error correction, and buffering, and diagnostics. Any 'noise' accumulated at Layer 2 simply isn't recognized at Layer 3, unless it corrupts a packet, in which case a checksum error is generated, the entire packet is discarded, and another packet retransmitted. Remember, thi is happening at 100 Mbit or Gigabit speeds, hundreds or thousands of time faster than even the highest audio signals. From a network challenge perspective, even 192KhzX24-bit audio is small beans.
I keep reading the same old stance from folks that either don’t stream or streaming for fun with their fancy analog or CD players. You know who you are :-)
It’s not about the data but about the context (emotional, physical, psychological) in which it’s delivered. Think about it before you eager to call out the cult-like belief in well designed streamers with premium parts without understanding why something works and works well over mass produced streamers.
I have compared streamers ranging from $500 to $25K. You don’t have to spend $25K to get a great sound, but pick a streamer that is well engineered to deliver bit-perfect digital output by implementing low-noise design, stable clocking, robust power supply and isolation (ethernet noise or power rail interference).
So why some of the streamers sound different? Because they prioritize aforementioned underlined elements that ultimately impacts how bitstream is distributed to your DAC.
The message should be; don’t fall for over-priced pseudo tech without due diligence be it a DAC, Streamer or Switches.
A complex topic when allied to digital music reproduction, made more difficult by S/PDIF.
Assuming sound engineering the bits at the output of the receiver in your device at the end of the Cat 5 cable are precisely the same as the bits stored at the source. If they were not the modern world would collapse as no transaction could be relied upon.
Setting aside the linearity of the DAC chip(s) there are then, however, two (digital) issues that arise; what data gets presented to the DAC processor (the component(s) inside the "DAC" box that generate the analog signal) and when is that data presented.
Unlike the TCP/IP ethernet or WiFi connection the S/PDIF has error detection but not correction so a truly terrible digital interconnect could, I suppose, result in bit errors, resulting in data loss.
My suspicion is that any unwanted artifacts that are caused by changing between decent digital cables are caused by jitter being introduced and not eliminated by processing before the data is presented to the actual DAC chip(s).
I did an experiment with different cables between my Aurender N100SC and the Esoteric K-01XD. My baseline is the Cardas Clear USB connection, and I do have an external Stanford Systems Research rubidium clock.
The three cables that I tested were an MIT from the 90's, a "Trubutaries" Video Cable - not high end, not even a specific digital interconnect, and a two meter audio junk interconnect from who knows when (or why).
I did not listen extensively but the MIT cable and the Video cable were indistinguishable. The junk interconnect however was clearly not up to the task, the sonic quality sounded like hard clipping was occurring!
The Esoteric K series DACs have buffering and clock synchronization before the actual DAC processor, that appears to have taken care of any jitter issues.
I suspect that the audio cable did not have the bandwidth to support the 1.2MHz 24 48 S/PDIF signal and so there were data errors. (S/PDIF has a parity bit, hence 25x48,000 = 1.2MHz).
BTW, I take exception to a total dismissal of switch mode power supplies. OK regarding wall-warts, but, for example, Benchmark have transitioned to use SMS to reduce noise. It all depends on the quality of the power supply.
Once again, my apologies if I am a bit off-topic, but an interesting question makes my mind diverge onto allied topics.
one of my favorite threads ever. starts with servers - one guy claims servers need a couple weeks burn-in before their sound is optimal - but then makes a hard right on page 3 to switches, where one guy explains that the reason you can’t hear a difference between a $20 switch and a $700 switch is that you have to spend at least $3500 on a switch before there’s a difference. Not making this up:
Ok I'll chime in...the equipment we buy whether its an 80 dollar wiim mini or the top of the line wiim, will in many cases will be wholly adequate because it isnt the weak link in the chain. Many of us or at least some of us have quite limited financial means to buy stupid expensive equipment and I put myself in that category. Choices are made based on bang per buck, feature set and yes overall performance too. Streaming hi bit rate audio is on most affordable equipment going to be just as good as physical media held in your hand. Are there issues with streaming? Well sure there are...sometimes its just not good, but most of the time it is superb. The physical media is better, more stable and provides consistent performance. Its also a lot more 'work' - getting up from something I am working on to change a CD is not a welcome interruption. Streaming avoids that event and just keeps going like the energizer bunny. Choose a stream and get to doing whatever it is you need to do while you enjoy some tunes in the background...maybe a fave comes on and you crank it up and take a 'song break' from your work. Or maybe its a kick back evening, a few drinks some good music from a favorite channel...at near concert levels. Oh yeah I have done a few of those. OR maybe its streaming in the kitchen while you make dinner. Streaming is compelling, both for its really high quality and for its convenience. I may never replace all those CDs I lost in the fire...but the jury is still out on that. As far as the equipment you use, choose what you like you only have to justify your choice to yourself. If a 30,000 dollar Streamer/DAC is what it takes to make you happy then go for it, my 339 dollar Wiim Ultra streamer and the emotiva amp that follows it along with a cheap 10" sub and a pair of Opal bookshelfs from Dayton audio represents a roughly 2000 dollar investment that makes my desktop system an amazing experience. Especially for the money. None of these components are the best in their category probably, but the whole is greater than the sum of the parts. And it just works. No fuss, Astonishing performance especially for piano music, which in my experience is extremely hard to reproduce accurately. Yet I find myself listening to mostly Piano music on this cheap little system. The drawbacks: it does not have the dynamic range to play at live levels for orchestral or rock concerts... but that isnt how I use this system. Living in an apartment with neighbors requires a certain amount of restraint in using the loud pedal. So...it is enough for the circumstances. And I think that is the real message here, we all have different living situations and have differing requirements and preferences. There is no 'one size fits all' in making these choices. My vote is for streaming for its convenience and not requiring storage space for the physical media. Since I lost most of that in the fire, replacing the media and the storage cabinets to hold it all is a sizable investment that is getting harder by the day to justify. I am trying to simplify my life not make it more complicated. I don't know if any of this resonates with those following this thread, but as Mark Knopfler would say "do your worst" :)
It certainly resonates with me. I believe the choices you’ve made show admirable self-awareness. More often than not it’s not so much what we do that matters as why we do it. Best wishes for a prompt return to normal after the tragedy you’ve experienced.
I cant tell the difference between Tidal MAX recordings and the same high res on the HD in my Aurender N20 > MSB Cascade > MSB M500 Monos > Estelon Forzas.
My general understanding is as long as the bits arrive faster than they need to be used up (filling the buffer), that a decent DAC will convert them nicely to analog.
I dont subscribe to the "hifi router" camp lol , nor switches or CAT6 etc.
USB was developed almost 30 years ago to replace and consolidate the old serial, parallel and PS/2 ports for PC peripherals, and later became ubiquitous in small electronics chargers.
How this pedestrian interface became quasi-standard in high-end digital audio is puzzling.
USB evolved tremendously over the decades and USB4 is a powerhouse, however many of today’s high-end DACs and streamers are still stuck in the USB 2.0 era.
I beseech you to understand what the word streaming means in digital networks, and how it differs from file transfers. At its simplest, streaming prioritises getting something out (timeliness), over getting it right. With streaming it is OK if some packets go missing, or get horribly corrupted, as long as there is a more or less steady stream of packets.
I beseech you to understand the difference between Internet Protocol (IP), Transmission Control Protocol (TCP) and User Datagram Protocol (UDP).
IP deals with basic network addressing. TCP works on top of IP and includes error detection and recovery, guaranteeing accurate delivery of messages and files but not timeliness. A TCP/IP transfer is not complete until the entire message has been transmitted, checked and corrected. If you want to 'stream' using TCP/IP without packet loss, you cannot start playback until the 'stream' has completed. In other words, the stream has become a file transfer!
UDP also works on top of IP but does not guarantee delivery. In general, UDP/IP is the protocol used for streaming, because it does not bother to stop and ask for retransmission. Think about multicasting where a single stream is fed to a large number of receivers. Not all receivers will get exactly the same packets because of transmission losses, which are not individually corrected!
I also beseech you to understand that Ethernet, on its own, does not guarantee packet delivery any more than pigeon post offers a delivery guarantee. Sure, most pigeons will get home most of the time, but some sometimes get lost, get picked off by hawks, get shot down, or die on the wing.
On top of all this, even USB does not guarantee accuracy when streaming. I recently posted in another thread:
"As of 2024, USB consists of four generations of specifications: USB 1.x, USB 2.0, USB 3.x, and USB4."
So there is no such single thing as USB. It is no longer even Serial! There are now nine families of USB connectors.For example, the USB-C connector has 24 pins and looks more like the purpose designed HDMI which is Parallel and eschews data packets.
USB was never designed for error-free streaming.
A stream pipe is a uni-directional pipe connected to a uni-directional endpoint that transfers data using an isochronous,[69]interrupt, or bulk transfer:
Isochronous transfers
At some guaranteed data rate (for fixed-bandwidth streaming data) but with possible data loss (e.g., realtime audio or video)
Interrupt transfers
Devices that need guaranteed quick responses (bounded latency) such as pointing devices, mice, and keyboards
Bulk transfers
Large sporadic transfers using all remaining available bandwidth, but with no guarantees on bandwidth or latency (e.g., file transfers)
Note the implications here. Audiophiles often believe that because files and messages can be transferred error-free, that implies streams are error-free. They aren't, but you do get your errors for free.
All of this is over my head but my lying eyes don't deceive me. Blu Ray players are head and shoulders above streaming. Period. From AI Overview:
Streaming services use compression techniques (like HEVC/H.265) to reduce file sizes and bandwidth usage, which can lead to some data loss. Blu-ray players, especially 4K, use less aggressive compression (H.264/AVC) to preserve more details.
Streaming services typically use lower bitrates (data transmission rate) than Blu-ray players. For example, a 4K Blu-ray might have a bitrate of 40-70 Mbps, while a streaming service might only offer 10-25 Mbps.
Due to lower bitrates and more aggressive compression, streaming can sometimes show more visible compression artifacts, like banding, blocking, or a loss of detail in dark or fast-moving scenes. Blu-ray, with its higher bitrates and less compression, generally produces a clearer, sharper, and more detailed image.
Blu-ray often includes lossless audio formats (like DTS-HD Master Audio or Dolby TrueHD), which provide better sound quality than the compressed audio formats used in streaming (like Dolby Digital Plus) according to Audio Science Review (ASR) Forum (of all sources).
Blu-ray discs can store significantly more data than a streaming service. A standard 1080p Blu-ray can hold up to 50GB, while a 4K Blu-ray can hold up to 100GB.Streaming services, on the other hand, compress the video into much smaller files, often in the range of 10-13GB for a 4K movie.
The argument that a Ferrari doesn't turn blue for a moment is specious. What streaming does is change the color tone from something like Rosso Barchetta to Rosso Berlinetta or Rosso Cino or even worse, Rosso Corsa.
When streaming audio, there's really no doubt that all the packets get there in time and in order but that's a numbers game. That's all that's being discussed and relies on everyone to just take their word for it as they can hear the difference.The signal in a CDP isn't compressed and travels a matter of inches while the streamed version is compressed and then uncompressed and travels around the world.
This numbers game presupposes that the resultant sound is not in the least affected when compressed and uncompressed and focuses only on the numbers. That's the very definition of a red herring:
A red herring is something that misleads or distracts from a relevant or important question. It may be either a logical fallacy or a literary device that leads readers or audiences toward a false conclusion.
How many here jumped on the MQA bandwagon only to see it shot down? That compression scheme corrupted the sound. How many are fully invested in streaming and find the need to validate it to others? There are so many previous threads on this that, in the end, went nowhere. Both sides stayed put.
A casual search on the internet shows more contradictions to arguments for streaming not being brought up but well known to those who can rattle off numbers and protocols like there's no tomorrow. It's in threads like this that one can find consensus to put forth their arguments without fear of being challenged. I certainly don't have the chops for that but there are so many recording engineers online that say otherwise as well but they rarely frequent sites like this. Some have in the past and when they do, the thread goes silent for awhile and then gathers momentum, picking up where it left off with the hopes of not hearing from them again.
It's not that those who prefer CDs and better quality endpoint devices think there's magic in our choices though we are told that. It's the ones who say that a compressed signal can travel through a dirty and noisy chain and remain its virgin self when uncompressed that abide by magic.
It seems to me, and in this bunch there is someone who will correct me if they think I am wrong... :) IT seems to me that there are very few things that determine the accuracy of the played signal out of our streamers. Number one is the source file. How good was it? SACD or at the other end an mp3 file? And then at the end of the digital transmission chain is the accuracy of the DAC that we use to decode the digital file and finally then our playback equipment...amplifier and speakers or headphones. Everything else in the middle is basically the same for everyone. Its all ones and zeros with error correction built in. So that part is the same for everyone. The difference is truly in our playback equipment and I am going out on a limb and say any audible differences within today's DACs are negligible and surely well below the threshold of hearing and definitively well below my ability to discern the differences. And even if I COULD hear the difference between my 329 dollar DAC and a 30,000 dollar DAC, my common sense and budget dictates I buy the 329 dollar unit. I know there are those here that could and would buy the 30k unit but I also suspect the reasons for doing so go above and beyond perceived better performance. Any of you guys going to admit to a leaning towards buying something for its looks? I have done so in the past...and probably will again. Any takers?
The other glaring variable in resolution is antique coaxial internet service versus fiber optic or microwave mesh. At my old house I was limited to coaxial and it was not pretty. The technician told me he had a job for life replacing coax connectors under the sidewalk. They lose continuity like clock work in summer heat.
There are fundamental differences between digital video and digital audio.
Digital video is always lossy because compression is used to make the bit rate manageable. 4K Blu-ray uses less aggressive compression than streaming, but there is still compression. The Motion Picture Expert Group has combined a range of lossy compression techniques which are realised as mpeg formats in various versions.
If no video compression were used, the bit rates are fairly easy to calculate. Take the bits for one video frame and multiply by the frame rate of your choice. The bits for one video frame are the bits per pixel times the number of horizontal pixels times the number of vertical pixels. At 24 bits per pixel and 4K resolution we need about 24 * 2,000 * 2,000 bits, or about 100-million bits per frame. With a 60-Hz refresh rate, that's about 6-trillion bits per second. Fortunately, there's not much change between most frames and the following one, and one compression technique is to just send the changes, with an occasional complete refresh.
Our eyes are much more forgiving than our ears.
Audio can be losslessly encoded at bit rates which are achievable with today's technology. However, while lossless encoding and decoding can always recover the original digital sequence if all the packets are delivered and error corrected, it does not stop packets getting lost or corrupted when streaming.
So on top of Qobuz using TCP/IP protocol, streamers and their operating systems has a bearing on delivering bits. Here is a read in regard to how Euphony OS (which I use) delivers data to dac, https://euphony-audio.com/hesk/knowledgebase.php?article=18
Thanks for the clarification. I hope I can remember it. Something tells me that this noisy streaming chain is purpose built for longevity and reliability as best they can while keeping the bottom line in mind. Purpose built but not that elegant.
After seeing your post of last night I worried local winemakers were boosting their product with methanol. I'm relieved to see you're OK this morning 🙂
... anyway, UDP is used only when latency is more important than accuracy: telephony, gaming, live sports. It is not used at all for audio streaming because latency isn’t that important, but accuracy is. Accordingly, both Tidal and Qobuz stream over TCP. Not sure what Spotify does, but no one cares.
Audio streaming presents a very, very light load in Ethernet terms, and error-free transmission is a given.
"Noise" and the so-called "jitter" are immaterial to digital-to-digital transmissions, and the vast majority of quality DACs readily handle noise and jitter before they can get to where they might negatively impact sound quality.
It’s probably safe to say that digital components (ie components whose inputs and outputs are both digital), and that have no onboard processing power, such as switches, routers, digital cables, etc. do not have a bearing on sound quality (again, all power supplies being equal).
Streamers, at least theoretically, should not either (digital in, digital out) but they do, or at least they can.
Streamers are computers in a pretty case, and even the weak-kneed Raspberry Pi-class CPUs employed by Aurender and the like have plenty enough power to handle relatively sophisticated DSP, thereby giving the designer wide latitude to transform the sound at will.
It isn’t that hard for DSP to make a streamer sound like a tube amp or, why not, a broken phone booth. Wanting to harness that power is understandable; what would you do if you made and sold high-priced streamers? Presumably, you would give your customers a taste of what you know they crave, just like the chef-owner who ladles fat and sugar in every dish up to just beneath the threshold of detectability. Pulling it off is a minor art form, but you don’t catch flies with vinegar.
DACs are half-analog, so of course they can make a difference in sound quality. So can upsampling, DSD conversion, and other signal processing taking place within the digital realm, but that’s a whole other conversation.
Personally, I believe system voicing is the job of the amps / speakers pairing. If any additional voicing is desired, it should be handled at the preamp level.
If every component in your system, including your digital chain - which should be absolutely neutral - intentionally dials in some level of euphonic distortion, pretty soon things become unmanageable. It’s like newspapers where every single writer is now a columnist whose important opinion on national affairs deserves to be heard, including food writers.
I know Qobuz claims to use TCP/IP but to many people, and it seems to many here, the internet is synonymous with TCP/IP. TCP is only part of the internet. I have tried to show that UDP/IP is equally ubiquitous, and that conclusions on accuracy drawn from TCP do not apply to UDP.
Qobuz also claims on their website that "An analog audio signal is composed of a sine wave" when they probably mean an infinite set of sine waves. They are careless with the truth.
Streaming is different from downloading, which can be bit-perfect using TCP/IP. The functional difference is that you can start playback of a stream before it is complete. Nothing in the world can guarantee the future will be error free.
TCP/IP only guarantees bit perfect transmission after the transmission is complete, and cannot guarantee how long that process will take.
Qobuz is very tight-lipped about the actual protocols used, which are proprietary. It is possible that they make a stream up from many small files which are transmitted using TCP/IP, but nothing guarantees all future files will be ready when playback gets to where they are needed.
The Euphony article you quote illustrates this well:
The best way to do this is to preload the complete song to RAM before playing
It does not address Qobuz but the major streamers have the same issues to face:
We don't know that much about Roon's internal workings
My advice to all on this thread…. Buy a Rega Planar 3, a Schiit Mani 2 phono stage and some records… Problem solved, no digital variants to worry about. Try it, you may love it!
Audio streaming presents a very, very light load in Ethernet terms, and error-free transmission is a given
It is not a given, not by Ethernet. Ethernet on its own does not guarantee packet delivery, nor timing, nor error-free status. You need higher level protocols, which may operate over Ethernet, if you want to guarantee delivery and error correction.
Ethernet timing is indeterminate, because unlike USB there is no central controller managing timing. However, as you point out, it is usually fast, depending on the version. Early Ethernet was under 3-million bits per second, though, which is not really enough for a CD let alone high res. Ethernet does have significant overheads.
You might care to rethink your statement that latency is not important for audio. It is the reason streaming was introduced.
Also most Ethernet network components including routers, switches and gateways do include processors and do a fair bit of work on each packet. They build up lists of Ethernet devices connected to each port and only forward packets to the necessary port(s)
My advice to all on this thread…. Buy a Rega Planar 3, a Schiit Mani 2 phono stage and some records
I would say get a universal disk transport with HDMI outputs, and some SACDs, Blu-ray audio disks, 4K opera recordings and CDs. You will always have your media, and it will always be as new because it won't wear out. Keep your fingers crossed that somebody will still be making transports if yours carks it!
Having said that, I have started buying records again ... but not if the music is also available on SACD!
So I asked the web about "ethernet packet delivery guarantee" and Google's AI overview came back with:
Ethernet, at its core, is a best-effort delivery protocol, meaning it does not guarantee that every packet will be delivered or that they will be delivered in the order they were sent. While Ethernet frames are encapsulated in IP packets, which in turn can be part of TCP or UDP protocols, the underlying Ethernet layer itself doesn't provide delivery guarantees.
Qobuz also claims on their website that "An analog audio signal is composed of a sine wave" when they probably mean an infinite set of sine waves. They are careless with the truth.
No, Qobuz is correct - analog audio is just a series of sine waves (and cosines). That's the Fourier Theorem. If you doubt that, just look at the squiggles on an LP - a series of sine waves. It's not infinite, though. To do that you'd need infinite bandwidth, which isn't needed and isn't possible.
Streaming is different from downloading, which can be bit-perfect using TCP/IP. The functional difference is that you can start playback of a stream before it is complete. Nothing in the world can guarantee the future will be error free.
But the future is "error free" because Qobuz is using TCP/IP and the file is at least partly cached before playback begins. And even hi-res audio requires a download speed of only around 10Mbps, so there's plenty of time for any damaged packet to be re-sent.
TCP/IP only guarantees bit perfect transmission after the transmission is complete ...
It isn't clear what you mean here. TCP/IP guarantees perfect transmission with the transmission of each packet. Each packet is "complete" unto itself.
Qobuz is very tight-lipped about the actual protocols used ...
My experience is just the opposite - I've found Qobuz to be remarkably accessible and transparent about its protocols. My information regarding how Qobuz works comes right from its US execs David Solomon and Dan Mackta.
My only connection with Qobuz is as a paid subscriber and to be clear, I have issues with the service. In particular, its files are often the "remastered" compressed versions, so my local files often sound better. But facts are facts, and Qobuz delivers bit-perfect audio direct to your streamer, provided your connection speed and network are capable to the task. And for audio, those requirements are minimal - almost trivial here in the third millenium.
I highly doubt bluray players/transports for blu-ray audio are going obsolete any time soon...
Panasonic is bound to keep it alive...Sony playstation consoles will be around forever...thanks to the ever growing number of gamer dudes. In fact, i believe sony has been killing other manufacturers off (yamaha, for instance) w.rt the bluray market. Essentially, they take a loss with ps5 player/console sales & make their money with game sales and subscriptions. But, to be fair, the ps5 is a very high quality bluray transport for audio. I wouldn’t know about video, am not a videophile.
The future of sacd could be bleak.
I would say get a universal disk transport with HDMI outputs, and some SACDs, Blu-ray audio disks, 4K opera recordings and CDs. You will always have your media, and it will always be as new because it won’t wear out. Keep your fingers crossed that somebody will still be making transports if yours carks it!
On the same note, buy a hires official studio master from qobuz, burn it on bluray disc and play it with your bluray transport....almost always/definitively sounds better than streaming the same album directly from qobuz or tidal...couldn’t be sure why.
@cleeds - I am curious whether the transfer protocol really matters. As you say,
“even hi-res audio requires a download speed of only around 10Mbps, so there’s plenty of time for any damaged packet to be re-sent.”
Therefore, regardless of whether a UDP/IP or TCP/IP protocol, doesn’t the entire data packet eventually arrive at the streamer where it is reclocked, sent on to the DAC, and then reclocked again (depending on whether a synchronous or asynchronous connection is used), and finally converted to analog and sent on?
I stream both Tidal and Qobuz, plus play files locally stored, through Roon and, in the context of a moderately high-end system, do not really perceive a sonic difference regardless of what is playing.
UDP can safely be ignored in the context of 2-ch digital streaming, together with all the AI-generated nonsense @richardbrandhas been cut-and-pasting in this thread. Both Tidal and Qobuz use TCP, and TCP does ensure bit-perfect transfer of your PCM-encoded music to your streamer (I say PCM because I don’t think either service offers DSD as of now).
SACD is different in that it is encoded in DSD64 format. Now there are multiple considerations associated with this, and folks who are interested may want to read this blog post by Benjamin Zwickel - the founder and designer of respected DAC manufacturer Mojo Audio - who is considerably more qualified than I to dissert on the subject.
... regardless of whether a UDP/IP or TCP/IP protocol, doesn’t the entire data packet eventually arrive at the streamer ...
It does if it's TCP/IP because any faulty packet is simply re-sent. That re-send isn't possible with UDP/IP, though, so an error is possible. The audibilty of that error is debatable, of course, and probably depends on the extent of the error in the first place.
Both of my main two DACs are NOS R2R types so only PCM for me, and my system sounds great through either of them. BTW, one of those DACs is Mojo Audio’s top DAC.
It does if it's TCP/IP because any faulty packet is simply re-sent. That re-send isn't possible with UDP/IP, though, so an error is possible. The audibilty of that error is debatable, of course, and probably depends on the extent of the error in the first place
Thank goodness at least one person here gets it!
Lots of people here claim to hear differences with digital streams and lost packets are one possible explanation.
On the same note, buy a hires official studio master from qobuz, burn it on bluray disc and play it with your bluray transport....almost always/definitively sounds better than streaming the same album directly from qobuz or tidal...couldn’t be sure why.
There is a very simple explanation. When you buy a download, you are essentially transferring a file. You are not listening to it as it transfers, so timing is not critical.
File transfers use the TCP/IP protocols. The internet is a packet switched network, and individual packets may take completely different routes through the network and eventually arrive out-of-order, or corrupted, or not at all.
At the end of the transfer, TCP/IP guarantees either that the file is a bit-perfect copy or that the transfer has failed.
How does the TCP/IP receiver even know when the transfer has finished? According to Google AI:
TCP/IP knows when a data transfer is complete through a four-way handshake involving FIN and ACK packets, ensuring both sides have signaled the end of the connection and acknowledged it. This process guarantees a reliable and ordered transfer.
Here's a more detailed explanation:
Data Transfer:
TCP establishes a connection, breaks data into segments, and uses sequence numbers and acknowledgments to ensure reliable delivery.
Initiating Closure:
When one side (client or server) has finished sending data and doesn't need to receive any more, it initiates the closure by sending a FIN packet.
Acknowledgement of Closure:
The other side acknowledges the FIN with an ACK packet.
Return FIN:
The receiving side, now also finished with data, sends its own FIN packet back.
Final Acknowledgement:
The original sender acknowledges the FIN from the receiving side with a final ACK packet.
Connection Terminated:
Once both sides exchange the FIN and ACK packets, the connection is officially terminated.
In essence, the four-way handshake (FIN-ACK-FIN-ACK) signals that the transfer is complete and both parties have finished sending and receiving data.
UDP/IP does none of this checking because it is attempting to keep a stream going
Qobuz is correct - analog audio is just a series of sine waves (and cosines).
Qobuz would be correct if that is what they said, but it ain't. My quote was precisely copied from their website "An analog audio signal is composed of a sine wave". This is ridiculously wrong.
Fourier theory says than an arbitrary repeating waveform can be constructed from an infinite series of sine waves, being the odd harmonics of the base frequency.
Fourier transforms are often used to convert between the time domain and the frequency domain, so much so that many audiophiles only think about frequencies.
The extreme case is a square wave which does require an infinite series. Unfortunately, transforming the transform to get the square wave back produces spike artifacts.
Can you share what you know about how Qobuz actually works "My information regarding how Qobuz works comes right from its US execs David Solomon and Dan Mackta".
So I asked Google AI "how does TCP/IP correct for packet loss and corruption"?
UDP/IP does none of these things. I have added italics:
TCP/IP uses mechanisms like checksums, sequence numbers, and retransmission to ensure reliable data delivery, handling both packet loss and corruption. TCP detects missing or corrupted packets and requests their retransmission, ensuring that data arrives in the correct order and without errors.
Here’s a more detailed breakdown:
1. Error Detection (Checksums):
TCP uses a checksum to verify the integrity of data during transmission. The sender calculates a checksum based on the data and sends it along with the data. The receiver recalculates the checksum and compares it to the received value. If they don’t match, it indicates data corruption, and the receiver requests retransmission from the sender.
2. Sequence Numbers and Acknowledgements:
TCP uses sequence numbers to track the order of packets. If a packet is lost or arrives out of order, the receiver can use these sequence numbers to detect the issue and request retransmission.
The receiver sends acknowledgements (ACKs) back to the sender, indicating which packets it has received successfully. If the sender doesn’t receive an ACK within a certain timeframe, it assumes a packet is lost and retransmits it.
3. Retransmission:
If a packet is lost or corrupted, the sender will retransmit it, ensuring that the receiver eventually gets all the data it needs.
The sender also implements timers to ensure that lost or corrupted packets are retransmitted within a reasonable time. If the timer expires without an ACK, the sender retransmits the packet.
4. Flow Control:
TCP employs flow control to prevent the sender from sending data faster than the receiver can handle. This helps avoid packet loss due to buffer overflows on the receiver’s end.
5. Congestion Control:
TCP also includes congestion control mechanisms to avoid network congestion, which can lead to packet loss. These mechanisms help regulate the rate at which data is transmitted, preventing the network from becoming overloaded.
In summary: TCP/IP uses a combination of error detection (checksums), sequence numbers, acknowledgements, retransmission, flow control, and congestion control to ensure reliable data delivery, handling both packet loss and corruption.
together with all the AI-generated nonsense @richardbrand has been cut-and-pasting in this thread
Google AI has access to all the nonsense on the web, including yours, but it is intelligent at processing the entire content. If the consensus it reaches agrees more with my understanding than yours, think perhaps that you just may not be entirely right.
What I have tried to do here is to back up three assertions with the best sources of information you just might believe, one being Wikipedia and the other Google AI.
You labelled this topic "We Need To Talk About Ones And Zeroes" and we clearly do, because there is so much misinformation surfacing here.
My backed-up assertions are
Streaming does not guarantee packet delivery nor bit-perfect accuracy
Ethernet on its own does not guarantee packet delivery nor packet accuracy nor packet timing
USB when used for streaming does not guarantee error-free delivery
I understand why audiophiles who have committed to streaming might react in horror to these assertions. I urge you to do your own research with an open mind.
There are much better formats than 2-channel PCM, after all!
You must have a verified phone number and physical address in order to post in the Audiogon Forums. Please return to Audiogon.com and complete this step. If you have any questions please contact Support.
