[rb2013]

While the discussions on my thread about the new class of XMOS USB processors rages on along with all kinds of USB gizmos to fix it's follibles.  A new paradigm of computer audio is not here.  This is different then the UpNP/DLNA - it that this new Audio over IP Ethernet std will allow you to use any audio player and will have compatibilty amoung many audio devices - all operating on a LAN!
 
So here is what I feel is going to be the new computer audio std:
AES67 Audio over IP Ethernet!
 
http://www.pro-tools-expert.com/home-page/2015/11/4/aes67-what-is-it-and-why-should-you-care
 

 

What Is AES67?

1. It is an interoperability standard
2. It is for audio transport only
3. It isn't a complete system. AES 67 is a feature or option in a wider audio system which can fulfil other tasks such as routing, monitoring, discovery or system control.

 

Why All The Fuss About AES67?

It is deployable - It is very limited in its scope. It does audio transport and that's all, by audio transport I mean moving audio around a network

There are many things which make AES67 exciting. One of the most significant is that it is very limited in scope. The two technologies which have received the most attention on the blog are Dante, as implemented by Focusrite in their RedNet systems, and AVB which Avid are using in their S3L live sound system. The biggest differences between these two systems are that Dante is proprietary and operates on layer 3 of the OSI 7 layer model. AVB is an open set of standards and it operates on Layer 2. While AVB is a fantastic technology, it is broad in scope and its future definitely lies in more areas than just pro audio. The biggest limitation on its uptake in pro audio is probably the relative indifference end users have to which technology they use combined with the fact that Ethernet switches used on an AVB network have to have specific AVB features, ruling out old or non-AVB switches. AES67 is an interoperable stream with moderate latency, each data packet contains roughly a millisecond of audio and the total latency is 6ms, though lower and higher latencies are available as an option. It supports both multicasting and unicasting - multicasting makes all streams available everywhere but requires good quality switches. Unicasting is useful for point to point scenarios over distance where the quality of switches can't be guaranteed. AES67 speakers are coming, the Genelec 4020A prototype has AES67/Ravenna inputs eliminating the need for any extra network node/AD-DA hardware. Just plug the RJ45 into the back of the speaker - install designers will love that! Just as important as what AES67 does is what it doesn't do. It offers no routing and control protocols, no online metering or remote control and no easy web GUI management. 

 
Ok what does AES67 and it's current implementations like DANTE and RAVEENA offer:
 
1- Near zero latency - even over a LAN and long distances
2 - ASIO support and Apple support, as well as Linux.  So you can use your current favorite audio player or software.
3 - Full layer 3 TCP/IP support - no special swtiches needed.
4 - Current PC Ethernet port works just fine, or add a PCIe card.
5 - 1 GB massive data throughput
6 - As with all 1GB or higher LAN - native galvanic isolation
7 - No USB gremlins like clocking and Async packet noise or loss
 
Well you may say great - but this is ll vaporware - I can't get this right now.  Can I?  Oh yes you can, but in a rather expensive fashion, but as this AES67 protocol is only 2 years old the adoption is occring right now.  Mainly amoung the Pro-Audio companies, but I firmly believe high end consumer audio products will see this as the replacement for USB and maybe i2s on the backs of DAC's and DDC's - just a plain old RJ45 jack is all that's needed (and the AES67 compatible Ethernet internal board).
 
So here are two products available now for audio using AES67 as their main connection (with SPDIF as the legacy connection)
 
Merging Technologies NADAC: http://nadac.merging.com/

 
 

 
From the positive-feedback review
http://positive-feedback.com/audio-discourse/impressions-the-merging-technology-nadac-mc-8-dsd-dac/
INPUTS
AES INPUT

1. Connector: gold-plated female XLR
2. Input impedance: 110 Ohms
3. Sample rate: 44.1 kHz—192 kHz

S/PDIF OPTICAL INPUT

1. Connector: Toslink
2. Sample rate: 44.1 kHz—96 kHz

S/PDIF COAXIAL INPUT

1. Connector: gold-plated RCA jack
2. Input impedance: 75 Ohms
3. Sample rate: 44.1 kHz—96 kHz

NETWORK INPUT

1. Connector: Neutrik EtherCon RJ45
2. Bitrate: 1 Gb/s (Gigabit Ethernet only)
3. Sample rate: 44.1 kHz—384 kHz, DSD64, DSD128 and DSD256

WORDCLOCK INPUT

1. Connector: BNC
2. Input impedance: 75 Ohms
3. Termination: 75 Ohms, software selectable
4. Sample rate: 44.1 kHz—192 kHz

MISCELLANEOUS

1. Enclosure material: Premium machined and anodized aluminum
2. Dimensions: 435mm (17.125") W x 435mm (17.125") D x 95mm (3.75") H
3. Weight: 11 kg (24.2 lbs.)
4. AC voltage: 100V-240V/47-63Hz (IEC socket)
5. DC voltage: 10V-14V (Hirose HR10A-7R-4S)
6. Power consumption: < 30W
7. Front panel display: OLED, 160x128 pixels, 16-bit colors

General Description and Considerations
The NADAC MC-8 is a very handsomely sculpted, quite solid audio design. It is designed around Merging Technologies' implementation of digital audio processing derived from their long experience in professional settings, which goes back into the 1990s. Merging has been particularly involved in the development of solutions for DSD processing on the A/D and D/A side of the equation. This include DXD, a 384kHz/32-bit PCM standard that allows for high resolution without the issues that DSD introduces to productions that require a great deal of slice-and-dice in the digital domain. (Analog manipulations are not a problem, since they can occur prior to the final feed to DSD.)
 
From http://www.digitalaudioreview.net/2015/05/mergings-ethernet-nadac-impresses-at-munich-high-end-2015/

Their NADAC’s main point of difference is a biggie: it doesn’t receive data via USB but uses Ethernet, which apparently offers far more accurate clocking capabilities when used in tandem with the Munich-developed RAVENNA network protocol. This ain’t no DLNA/UPnP cop out. Ethernet transmission also makes it useful for those needing to put some serious distance between DAC and host PC.​

Once connected to a network, the NADAC itself dictates data transmission rates instead of the computer. In other words data handling is asynchronous.​

All current digital audio format trends are met head on: PCM up to 384kHz, DXD, DSD64, DSD128, and DSD256. The business end of D/A conversion shows Merging Technologies’ roots: they’ve opted for an ESS’ 9008S chip​

 
Now RAVENNA:
is a open standard implemenation capatible with AES67:
http://www.ravenna-network.com
 
What is RAVENNA?

 

OPEN AUDIO OVER IP

RAVENNA is an open solution for transmitting audio over IP. Designed to meet the exacting standards of the Broadcast industry, RAVENNA delivers high-quality, multi-channel audio over a standard IT network.

 

WHY AUDIO OVER IP?

There are primarily two reasons why users consider using Audio over IP today:

• Flexibility in signal routing
• Low cost - the small audio industry ‘borrows’ technology from the much larger IT industry that has already been debugged and commoditised over many years, this greatly reduces the cost of devices such as network switches.

 
Traditional audio distribution is often implemented using large cross-point switchers or patch bays. These are specialised and expensive devices which take a long time to install and maintain. They do have the advantage of minimal delay and do not change the signal content or format.
 
A network system can be rapidly connected together using standard off-the-shelf parts which are much more compact and can be implemented in either a centralised or decentralised fashion.
 
As network systems originally had no need to consider real-time transmission of data (files were re-assembled from a collection of packets after they had all arrived), there were some disadvantages to overcome. These are largely done now and the advantage of a cheap scalable infrastructure that may already exist in many facilities is becoming too persuasive too ignore.
And is gaining wide adoption:

 

USER BENEFITS​

Flexible Profiles (Multi-Format Support)

RAVENNA Profiles enable users to customise the audio stream for their application. Based on standard networking technology, RAVENNA can support a variety of audio formats within its payload.

RAVENNA supports a variety of different data formats used in professional environment. For audio applications, 16 and 24-bit integer as well as 32-bit full bit-transparent AES/EBU data formats in combination with all relevant sampling rates (32 … 192 kHz) are supported. Since RTP is used as transport protocol, virtually any desired data format (i.e. 32-bit floating point, DSD and DXD high-res formats and others) can be transported across a RAVENNA network. This is not limited to audio data, but includes video data as well as control data. Although only one data format is permitted per individual stream, different streams with different data formats can coexist on the same network concurrently.


 

Quote:

 IP Technology (OSI Layer-3)

As an IP-based solution, RAVENNA is based on protocol levels on or above layer 3 of the OSI reference model. IP can be transported on virtually any LAN and is used as the base layer for communication across WAN connections (including the internet). Although Ethernet will be deployed in most cases as underlying data link layer, IP is in general infrastructure-agnostic and can be used on virtually any network technology and topology.

Quote:

 Phase-Accurate Synchronisation

Professional audio applications demand tight synchronization between all devices and audio streams. While playback synchronization in most applications requires sample accuracy, it has been the goal for RAVENNA to optionally provide superior performance by providing phase-accurate synchronization of media clocks according to AES-11; this would render the separate distribution of a reference word clock throughout the facility or venue obsolete.

In RAVENNA, synchronization across all nodes is achieved through IEEE1588-2008 (also referred to as Precision Time Protocol or PTPv2), another standard protocol which can be operated on IP. PTPv2 provides means for synchronizing local clocks to a precision in the lower nanoseconds range with reference to a related master clock - provided that all participating switches natively support PTPv2. But even without native PTP support, the achievable precision - while varying depending on size and bandwidth utilization of the network - will be more than sufficient to reach sample accurate synchronization across all nodes. Sample-accurate synchronization can even be reached across WAN connections, when local master clocks are synchronized to GPS as a common time domain reference.

RAVENNA supports WINDOWS ASIO, MAC and Linux:

To do this, Merging developed Ravenna, a TCP/IP protocol and application set for audio over Ethernet, later fully supported in the Audio Engineering Society's AES67 standard (HERE), which Merging Technologies had helped to produce. In sum, this allowed Merging's family of devices (Horus, HAPI, and NADAC, for example) to use ASIO (PC/Windows) or Core Audio (Mac) drivers to communicate and network with one another reliably, over much longer distances and in more complex topologies. This would have to be something other than a consumer-oriented "one-to-one-with-a-short-cable" basis, obviously. Since Ravenna is fully compatible with the AES67 standard, all Merging Technologies Ravenna devices will interoperate with those of other manufacturers using AES67. This is a key point for professional settings, since large and complex implementations are not cheap, and are not easily replaced or upgraded. 

 
 
 
NOW TO THE OTHER AES67 SOLUTION- DANTE:

What About Dante?

Dante is Layer 3 and as such doesn't have the same switch compatibility constraints as AVB. AES67 support is announced and a firmware update will allow Audinate Transport Protocol and AES67 Transport protocols to coexist on the same network. The reason why you might want to do this is because AES67 is a lowest common denominator between networks and using Dante's native transport protocol might provide performance improvements when moving data around a purely Dante network. When sending audio between mixed AES67 compatible networks an AES67 stream can be used, sacrificing a little latency for improved flexibility. Dante publishes the availability of AES67 streams on the network so they can be used by 3rd party network technologies with AES67 providing the transport and Dante looking after the system control.

 
 

Audinate Announces Support for AES67 Standard

https://www.audinate.com/article/audinate-announces-support-aes67-standard
 

Amsterdam, Netherlands 4 February, 2014 - Audinate announced today that it plans to incorporate AES67 transport in its Dante™ media networking solution. Dante has rapidly become the market leader and the dominant media networking solution for audio networking. 

https://www.audinate.com/solutions/dante-overview
 
Quote:

 

Economical and Versatile

One cable does it all. Dante does away with heavy, expensive analog or multicore cabling, replacing it with low-cost, easily-available CAT5e, CAT6, or fiber optic cable for a simple, lightweight, and economical solution. Dante integrates media and control for your entire system over a single, standard IP network.
Dante systems can easily scale from a simple pairing of a console to a computer, to large capacity networks running thousands of audio channels. Because Dante uses logical routes instead of physical point-to-point connections, the network can be expanded and reconfigured at any time with just a few mouse clicks.

 
Quote:

Unicast or Multicast

Dante audio channels can be configured as unicast or multicast as appropriate, to make best use of available bandwidth. Unicast provides a direct point-to-point stream for unique channels; multicast sends an audio stream to multiple devices simultaneously.

https://us.focusrite.com/ethernet-audio-interfaces/rednet

RedNet

Studio quality sound meets digital audio networking…

RedNet is Focusrite’s flagship range of modular Ethernet-networked audio interfaces that harnesses the power of Audinate’s tried and tested Dante digital audio networking system to bring studio quality sound to any modern audio application.

Designed with multiple audio applications in mind – from Live Sound rigs to Multi-room Recording Studios, Houses of Worship, Audio Distribution Installations, Post Production environments and anything in-between – fundamentally RedNet is an extremely scalable, near zero latency audio distribution system that can be used to expand I/O channel count, interface digital components, and/or bridge between Pro Tools|HD or MADI and the Dante audio network.
Incorporating Focusrite’s most advanced AD/DA conversion to date, rock-stable JetPLL clocking and premium multi-layered board circuitry, RedNet is no exception to the company’s philosophy that ‘Sound is Everything’. With every design detail meticulously engineered, RedNet is a step above its I/O competition, providing some of the most transparent and pristine audio quality available – with the added benefit of the systems’ revolutionary networking capabilities.
 
DANTE can be run from your PC or MAC using a $29 Digital Virtual Soundcard and your PC/MAC's RJ45 Ethernet port.  That is obviously low cost.
It can be downloaded here:
https://www.audinate.com/products/software/dante-virtual-soundcard?option=com_virtuemart&product_id=49&category_id=13&page=shop.product_details&flypage=flypage_cart.tpl







Advantages over UnNP:
1) Use your current audio player - Foobar, JRiver, iTunes, whatever and they are compatible.  Dante and Ravenna provide ASIO Windows drivers.
 
2) Multi-Cast or Uni-Cast both available
 
3)Interface with Thunderbolt 2&3 and USB 3.1 - http://www.audiomediainternational.com/recording/feature-investigating-interface-protocols/04655

One connection protocol that is ubiquitous on both Apple- and Windows-based machines is Ethernet. Currently mainly in use for distributed audio and large-scale networked systems, the protocol has been championed by Merging Technologies. “There are a number of audio-over-IP protocols now established in the pro-audio market that use Ethernet as the connection – the main players being Ravenna, Dante and Livewire,” details Paul Mortimer, managing director of Merging’s UK distributor eMerging. “Compatible devices can be connected using a simple point-to-point connection or via an existing standard IT network infrastructure. “The main advantages of Ethernet-based formats are the ability to run much longer distances between devices; being able to take audio signals from one source and route to many destinations; and sample accurate clocking from one master device on the network. Thunderbolt 2 and USB 3 offer the ability to connect to Ethernet, so would also be compatible with networked audio devices. With the introduction of AES67, all of these audio-over-IP formats will talk to each other, so enabling one harmonious compatible format.”

  4) Easier to set-up and more reliable? UnNP has it's well documented issues...

 

[rb2013]

http://www.computeraudiophile.com/f6-dac-digital-analog-conversion/new-digital-analogue-converter-merging-technologies-home-audio-market-22967/index3.html
 

dallasjustice

Junior Member

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Originally Posted by Miska
...to be more precise, I expect jitter performance around similar figures as HDMI without the asynchronous feedback option...






I am talking about the advantages that AES67 and PTPv2 offers over the current way ethernet is being used in this niche industry.

In the context of AES67, you are talking about multiple DACs all sync'd together over IP in large distributed systems. Even if timing is off one nanosecond in the sync, it won't be audible and it's not jitter in the way we typically define it in 2CH playback. The sync timing is different from the jitter in each DAC's conversion. There's nothing about how ethernet is used in this context which causes jitter in the playback of each device. The data packets are checked on both ends, so the data each DAC receives is exactly the same as the data the server sends. The timing of that data is irrelevant to the conversion inside each device.

Another major advantage AES67 offers is that it can support MCH setups much better than DLNA. I'm not aware of any high end audio quality DLNA renderer with an ASIO driver or anything more than 2CH.

​

 

 

[Luckbad]

I have an RJ45 to XLR adapter on the way to see if I can get this working on my PC before diving deeper.

Thanks for the thread!

I'm actually more interested in how this develops than most other things in computer audio.

 

[mhamel]

From the pro side of the audio world, and very well-regarded:  http://burlaudio.com/products/b2-bomber-dac 
 
Supports direct input via Ethernet/Dante.

 

[ginetto61]

Hi !
thanks a lot for the extremely interesting information.
I am afraid that will be a high end and very expensive technology ?
just the Focusrite pcie adapter is 999 USD

 
http://www.bhphotovideo.com/c/product/1020657-REG/focusrite_rednet_pcie_card.html
 
Lucky the rich people.  I envy them deeply 

Regards,  gino

 

[rb2013]

I have an RJ45 to XLR adapter on the way to see if I can get this working on my PC before diving deeper.

Thanks for the thread!

I'm actually more interested in how this develops than most other things in computer audio.

Me too

 

  Hi !
thanks a lot for the extremely interesting information.
I am afraid that will be a high end and very expensive technology ?
just the Focusrite pcie adapter is 999 USD

 
http://www.bhphotovideo.com/c/product/1020657-REG/focusrite_rednet_pcie_card.html
 
Lucky the rich people.  I envy them deeply 

Regards,  gino

This is just the start - these units like the Merging HAPI and Focusrite REDLAN stuff - is designed for many functions we don't need for home audio.  Once a Chinese developer designs a simple DANTEor RAVENNEAor maybe XMOS/AES67 Ethernet to SPDIF and/or AES DDC - it should be no more expensive then the $180 F-1, or Pro3a.
 
The nice thing is the std is open and a true std - so I expect someone like XMOS to develop a FPGA chip like solution - just like for USB.
 
Remember the first Empirical Off Ramp?  It was like $800 for 96k
 
I had one of these after going through about a half dozen Fireface devices (Your Apogee Rosetta is one) and settled on the RME Fireface 800.

 
It was $1600 ten yrs ago.  They still sell it!  DidN'T I need all these channel and mic inputs - no - but it was the best way to play hi res (192k) audio at the time.  This was before Async USB audio.  USB 2.0 iso was limited to 96k and even that was buggy back then.
 
So the bottomline is we will see many devices for our needs and much cheaper. Can't wait!
 
Cheers!

 

[rb2013]

  From the pro side of the audio world, and very well-regarded:  http://burlaudio.com/products/b2-bomber-dac 
 
Supports direct input via Ethernet/Dante.

Sweet DAC!  $2300 not bad.
 
When you get a chance take a look at this discussion on CA I posted (#2) between Miska (developer of the UpNP HQPlayer) and a fellow named 'Dallas Justice'.
 
It's good discussion of the clocking issues with AES67 ethernet vs Async USB.
 
Cheers!
 
PS Great review on the B2 Bomber ADC/DAC - Wow! 
http://tapeop.com/reviews/gear/79/b2-bomber-adc-dac/

The sibling B2 DAC sounds great too — very full and thick. Next to my Dangerous DAC and my Lavry DA10, I would characterize its sound as “warm” or even “analog” — round and present in the mids, with a really nice, wide stereo image. The all Class-A, discrete op-amp design was based around a new set of passive filters, chosen for their extremely flat phase response — and no capacitors (for great bass). Supposedly, this allows the Burl to overcome some of the inherent limitations of older designs that could sometimes yield a flat or thin sound by rolling off the bass and/or making the top end edgy. The intent was to aim for the reproduction ability of great analog tape, and I think Burl nailed it. But neutral it ain’t, and that’s where I came into conflict with it. I mix through my conversion chain (as I would hope everyone does), and I don’t want my DAC adding additional coloration. If I were mixing through the B2, I feel like I would compensate in such a way that everything would come out with less bass, a bit carved in the mids, and too bright. Maybe the B2 sounds too good, or it’s possible that I’m just way too familiar with my current chain 

As I'm a tube DAC guy, very, very tempting...

 

[ginetto61]

  Me too

 

This is just the start - these units like the Merging HAPI and Focusrite REDLAN stuff - is designed for many functions we don't need for home audio.

Once a Chinese developer designs a simple DANTEor RAVENNEAor maybe XMOS/AES67 Ethernet to SPDIF and/or AES DDC - it should be no more expensive then the $180 F-1, or Pro3a. 

 
Then you give me hope ! 

  so i have to wait a little.
because your comments on the sound of these interfaces make me want to jump on one ...
 

The nice thing is the std is open and a true std - so I expect someone like XMOS to develop a FPGA chip like solution - just like for USB.

Remember the first Empirical Off Ramp? It was like $800 for 96k

I had one of these after going through about a half dozen Fireface devices (Your Apogee Rosetta is one) and settled on the RME Fireface 800.

 

 

It was $1600 ten yrs ago. They still sell it! 

Did I need all these channel and mic inputs - no - but it was the best way to play hi res (192k) audio at the time.

This was before Async USB audio.   USB 2.0 iso was limited to 96k and even that was buggy back then.

So the bottomline is we will many devices for our needs and much cheaper.

Can't wait!

Cheers!  

 
I have to wait.   Actually looking at the pcie card it looks very well built but nothing out of this world.
I have to wait.  I hope that maybe next year ... i want a rj45 to AES really badly.
Thanks a lot again
gino

 

[rb2013]

Interesting discussion on CA:
http://www.computeraudiophile.com/f22-networking-networked-audio-and-streaming/lan-input-dacs-21722/

 

[rb2013]

DANTE AES67 Ethernet compatible BURL B2 Bomber DAC (courtesy of @mhamel):
$2300
 
http://www.sweetwater.com/store/detail/B2BomberDAC?adpos=1o1&creative=54989979481&device=c&matchtype=&network=g&gclid=CO_PgbDVu8wCFYqPfgodWr4Jtw
 
 
http://burlaudio.com/products/b2-bomber-dac
 

As a compliment to the B2 ADC, the B2 DAC punches you in the chest with low end while the 3D spaciality and stereo spread give you amazing detail throughout the spectrum. Add to that a sweet tone that is easy on your ears, and you have a unit that you will instantly fall in love with! Both the B2 ADC and B2 DAC feature identical, incredibly low jitter clocking, precision metering and stepped attenuators. Couple the B2 ADC with the B2 DAC and you have the B2 Bomber Master Signal Chain, a force to be reckoned with!​

Features:
• 44.1k Hz to 192k Hz, 24 bit, 2 channel DAC
• Proprietary custom design BOPA1, all discrete op-amps
• Passive filters
• All class-A
• Zero capacitor signal path
• Audiophile quality 6 position attenuator with standard headroom settings
• High definition metering with simultaneous RMS and peak indication
• Dante, 2 AES, SPDIF and Toslink input
• BNC word clock input with two outputs of extremely low jitter clock
• Frequency response at 48kHz sample rate is 10Hz to 22kHz, +/- 0.1dB
• Frequency response at 96kHz sample rate is 10Hz to 30kHz, +/- 0.1dB, -0.3 dB @ 40kHz
• Frequency response at 192kHz sample rate is 10Hz to 30kHz, +/- 0.1dB, -1.5 dB @ 90kHz
• 115dB Dynamic Range
• -96dB THD+N, full scale output = +22dBU
• Rugged, Made in USA design​

 
 
 
http://tapeop.com/reviews/gear/79/b2-bomber-adc-dac/

 
 
http://tapeop.com/reviews/gear/79/b2-bomber-adc-dac/
 

 With the B2 DAC, you get nearly transparent, yet very musical, conversion that is ideal for monitoring; and with the B2 ADC, you get the lush color and mojo of Class-A, transformer-based analog circuitry with a sound that is very reminiscent of analog tape decks. While these two converters meet very different goals, they function as a killer combination that makes recording to digital a musically satisfying analog trip and monitoring back to analog a listening experience you can trust and enjoy. These are the most exciting converters to hit the market in years, taking us into a new era in which digital recording may just have finally caught up with its analog ancestors.

 Next the B2 DAC travelled uptown to the mastering room of Howie Weinberg at Masterdisc to spend time with Matthew Agoglia. Matt ran the Burl through its paces against their DCS DAC, which they clock off of an Antelope Audio 10M (Tape Op #68). Keep in mind that the DCS cost about $10,000 fifteen years ago and has been a standard in mastering studios for well over a decade. On top of that, the Antelope system runs close to $8000. “Overall, the Burl (whether clocked to the 10M or internally) has a more neutral, smooth and transparent character compared to our DCS. The DCS has a color in its midrange, a tightness in the bass, and a subtle crispness in the highs. We could say that the DCS is more curvy, sounding different in different areas of the frequency spectrum, while the Burl is very smooth and linear, sounding very similar throughout the frequency spectrum. In particular, the Burl’s low end was actually a bit more extended, with sub frequencies a bit clearer, while the DCS had a very pleasant low end focused around 80–120 Hz. The Burl also sounds a bit wider than the DCS. When clocking the DCS off the Antelope 10M, we get that larger-than-life sound that some describe as “hype” — not necessarily a bad thing in mastering because you don’t end up adding too much EQ or other processing to achieve your results. I wondered if I might be inclined to EQ/process more with the Burl handling my DAC duties because I’d want to hear more excitement. Note that when I clocked the Burl to the 10M, it definitely took on more of the excitement I heard with the DCS, bringing the two converters closer in sound. Please keep in mind that we are talking subtle differences here. The Burl at $2500 is a bargain!”

In my own critical listening at The Farm (my mixing room in Brooklyn), I compared the B2 DAC up against my HEDD, and I found them so similar that I can’t honestly say that the differences I heard are terribly significant. Both converters are crystal clear, and there is no difference in the amount of information I was hearing. In terms of listening pleasure, the Crane Song excels at delivering a strong, focused center image, so for mixes where the interest lies in the center, I liked the HEDD a little bit more. Conversely, the Burl presents a wider and somewhat smoother image, so for mixes where there are a lot of interesting things happening on the sides, the Burl was a little more enjoyable.

 

[rb2013]

Well it's here...$29 Audinate DANTE DVS (Digital Virtual Soundcard) and a DAC like the BURL, NADAC, HAPI or REDNET.
https://www.audinate.com/products/software/dante-virtual-soundcard
 
Say hello to the Unicorn!

 
I added the Bolding:
http://www.computeraudiophile.com/f22-networking-networked-audio-and-streaming/lan-input-dacs-21722/

 

09-13-2014, 10:50 AM

#3​

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Superdad

Music lover; Owner, UpTone Audio

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Hi Forrest: I think Tranz is asking about DACs with direct Ethernet connection. And to qualify that even further, ones that don't have a whole computer running an OS (no matter how small) acting as a host. Thus, devices like sMS-100, Aeries, RasberryPi, and CuBox-i are all just scaled down or purpose-built variations of a Mac mini or CAPS--they just run a Linux-variant and sometimes act more as end-point for a stream than being the player itself. Software such as LMS, NAA, Vortex, or any DLNA set up are mostly what enable those "computers" to play via Ethernet. Yet they are not really perfect in that similar problems and issues still exist--sometimes just scaled down. USB, clocking, power supply, processing and ground plane noise, lack of transparency in s/w-- none of those thinks go away completely.

So Ethernet input DACs without a computer inside are still a rare beast. Even more so if you want one that does not require DLNA. The Sonore Rendu (and Logitech Squeezebox and the like) as an Ethernet>S/PDIF or >I2S device comes pretty close, as the processor that acts as its "renderer" has embedded code and is not running an OS. But one still has to think about server and controller s/w to feed it (if using DLNA).

Of course what a lot of us want is just to be able to use our favorite player s/w--be it A+, JRiver, Amarra, iTunes, HQ Player, XBMC, Foobar, whatever--and to direct its output to a port that looks to the OS exactly like a USB "sound card" so regular Windows or Mac drivers work (ASIO, Wasapi, CoreAudio, integer mode, etc.), but is actually sending it out over Ethernet (through the LAN switch, hi-res, DSD, etc.) to a DAC that just plays it.

Sounds good, eh? A pipe dream? No actually. Bur it can not be done entirely in sofware.
John Swenson and I have been working on just such a solution for a very long time. We are trying to move it closer (its been idling on the back burner this year but is moving again), and it is a solution that may take three forms business wise: as a licensed module, as a DIY board set, and as a retail product (but it won't outputUSB--that requrires it be a host and thus a computer processor involved; and our own crazy DAC ambitions are going to take a LOT of money to capitalize). I can't reveal more of the details at this time, and honestly you may not see anything for at least 6 months more.
Plus I am told that others may be woking along the same lines. So keep your eyes peeled. We are all watching for that unicorn-- the elusive Ethernet DAC!

Regards,
Alex C.

 

 

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09-13-2014, 08:48 PM

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Superdad

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Originally Posted by Elberoth
Linn, naim, lumin etc.






Yep, nice also. But again, a bit closed on software that can be used--iTunes, Audirvana, HQ Player all ruled out. I'm just not a UPnP/ DLNA fan.
We think people would enjoy a DAC (or device for a DAC) that lets you use ANY favorite player s/w, outputting to a USB port seen as a super-standard UAC2 sound card, but have the data then go out onto the LAN and find a waiting DAC or device. And that device will not have to have a big microprocessor or any reliance on the server/controller/renderer model and its layers.

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09-13-2014, 08:36 PM

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Superdad

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Originally Posted by boatheelmusic
PS Audio PWD or DS with Bridge.






Expensive. Works only with their DACs. Clocks in the wrong place (should us DAC's clocks as master). Requires messing about with UPnP/DLNA, so certain audio player software gets ruled out. All the usual limitations of a DLNA "renderer.". Quite fine for some, but not what I am chasing.

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09-13-2014, 09:09 PM

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Superdad

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Originally Posted by jabbr


Seriously, all modern digital design these days uses some type of internal processor (somethings got to interpret the firmware) ... so exactly how is this better than designing a "better" Cubox or mac mini type box that you'd actually be able to run whatever your preferred software is ... or becomes?






Because I think the split model--computer sending stream over galvanicly isolated Ethernet to a DAC where it is received without further processing or conversion back to a problematic USB output/input--sounds better. (I'm not going into details of actual circuit implementation here, but we are talking something done right with extremly low jitter and timing based on the DAC's master clocks as it should be.)
And it vastly reduces the sonic impact of much that goes on in the sending computer.

We think a lot of people don't want to have a computer with their music library right in the room/on the shelf with their audio gear. Keep and manage/rip/download your music on the big family computer in the den, use your favorite player s/w and remote control for it (even if that is just Apple Remote app on iPad/iPhone, or a screen sharing VNC app), and sit in front of your speakers with complete wireless playback control.

I respect Linn, Naim, Lumin, Auralic, SOtM, Sonos, Squeezbox, etc. for what they have done. But we think a more open, simple, compatible, and sonicly transparent solution is possible. Equal to or better sounding than the best direct computer>USBDAC implementations. A tall order we know. We'll just have to wait and see how it turns out.

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Originally Posted by Superdad
Software such as LMS, NAA, Vortex, or any DLNA set up are mostly what enable those "computers" to play via Ethernet.






You shouldn't put LMS or UPnP/DLNA in the same sentence with NAA, because those are vastly different... NAA is closer to AirPlay than LMS or UPnP/AV.
  So Ethernet input DACs without a computer inside are still a rare beast. Even more so if you want one that does not require DLNA. The Sonore Rendu (and Logitech Squeezebox and the like) as an Ethernet>S/PDIF or >I2S device comes pretty close, as the processor that acts as its "renderer" has embedded code and is not running an OS. But one still has to think about server and controller s/w to feed it (if using DLNA).




UPnP is horribly complex and bad design and DLNA spec on top of it makes it even worse. So that's ruled out. Anything that is based on IP protocol on top of ethernet requires a computer, one form or another. Doing anything above IP protocol without proper OS is extremely bad idea, because you'll spend a decade trying just to perfect IP stack implementation while you end up building OS of your own while doing so which would end up being much worse implementation than anything already existing (unless you have thousands on man years to spend)...
  We are all watching for that unicorn-- the elusive Ethernet DAC!




In addition to NAA and building a NAA inside a DAC you have bunch of pro-audio gear (with ADC too) already on this area. So nothing new, but depends on what you want.

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Originally Posted by tranz
Always enjoy reading your input. Do you have examples of the pro-audio gear?






For example:
Merging Horus
Focusrite RedNet
Both work with ASIO capable applications.
  Do you have a little more background on the DLNA issues, and why that would be bad for the 'perfect' audio path?




UPnP is based on complex three-party handshake based on mDNS, HTTP and XML. In UPnP, Renderer is the instance that actually performs audio reproduction is also the actual player. So it doesn't provide means for isolating audio recording/reproduction from the player functionality. Also the media streaming is based on HTTP which is really inefficient for the purpose. UPnP Media Server is pretty much just a web server with optionally media transcoding capabilities and the Renderer is player that makes requests to that web server. Control Point tells the Renderer what it should fetch from the Media Server.

Control Point doesn't really have any control over "how" the actual playback is performed, only "what".

When you want to play FLAC from MinimServer, Renderer is the one that performs all decoding and playback, MinimServer just provides the FLAC file as-is over HTTP when asked by the Renderer. If Renderer doesn't know how to play FLAC, playback fails. With more advanced Media Server, it could figure out it needs to convert 192/24 FLAC to something supported by Renderer for playback, but it's all up to black magic between the two what that intermediate format would happen to be. It could be for example MP3...

Because UPnP/AV spec doesn't define any media formats, Media Server and Renderer could have nothing in common. For that reason, there's DLNA that defines restricted small subset of formats that are "mandatory" or "optional". DLNA part doesn't cover such things as DSD at all, so a DSD and DLNA are completely unrelated. So a strictly DLNA compliant system could transfer your DSD files as MP3 or 44.1/16 WAV (mandatory) between the systems, and you wouldn't even know... (other than wonder why it sounds bad)

Renderer doesn't really support multiple alternative outputs per renderer either.


For example NAA makes remote audio devices appear just as if they were locally connected, player behaves the same as if the playback would happen locally. So it's more like the Merging/Focusrite devices above.

Signalyst - http://www.signalyst.com
Developer of HQPlayer

 

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tranz

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LAN Input DACs

Originally Posted by bplexico
Cool! Just like the MSB Analog DAC.






I wonder which protocol is less noisy and more accurate. The Ravenna of the NADAC or the UPnP of the MSB Renderer input.

Slightly related, I recently read a review of the Antipodes server, where the designer was quite certain of an UPnP setup creating more noise due to to the stack complication and response and request overhead especially with big files.

The MSB Renderer likely has a 256MB RAM if based on the ABC PCB Renderer board and so has to request for data packets many times per song if all packets arrive without issue (hence why wired LAN is likely better than wireless connections to the music library)

http://www.audiostream.com/content/a...-server-part-2

The ABC PCB board: http://www.abc-pcb.com/abc_docs/MR-MOD-DS-111E.pdf

The Ravenna protocol used in NADAC :http://www.merging.com/uploads/asset...0V8%20rev7.pdf

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