سورس دیجیتال Transport & DAC

Asynchronous USB looks to be the perfect solution

پنجشنبه 21 جولای 2016
/ / /

این متن عالیه :

http://www.thewelltemperedcomputer.com/KB/USB.html

http://www.empiricalaudio.com/computer-audio/technical-papers/

USB

Universal Serial Bus (USB) is a serial bus standard to interface devices. USB was designed to allow many peripherals to be connected using a single standardized interface socket and to improve the plug-and-play capabilities by allowing devices to be connected and disconnected without rebooting the computer (hot swapping). Other convenient features include providing power to low-consumption devices without the need for an external power supply and allowing many devices to be used without requiring manufacturer specific, individual device drivers to be installed.

http://en.wikipedia.org/wiki/USB

A nice story about the development of the USB interface: The D/A diaries: A personal memoir of engineering heartache and triumph by Hitoshi Kondoh.

USB is a computer bus like any other but for some reason it inspires people to make all kind of funny products.

Wedding

USB wedding ring

Introduction

USB audio is very popular.
One of the reasons is that USB audio is part of the USB standard and as a consequence native mode drivers are available in all the popular OS (Win, OSX and Linux).
Connecting a USB audio device is a matter of plug&play.

USB audio is a flexible solution as any PC offers USB.

If you use a laptop this is probably the way to go if you want to improve on the on-board sound card.

The audio is routed to the USB.
This is a matter of choosing the USB audio device in your media player.
The on-board sound card is bypassed; in fact you don’t need a sound card at all.

The USB audio device is your (outboard) sound card.

 

Today the resolution of USB audio ranges from 16 bit/ 32 kHz to 32 bit/ 384 kHz.
A lot of DACs are still limited to 16 bit/ 48 kHz max.

 

The data transfer from the PC to the DAC can be done in adaptive or in asynchronous mode.
In adaptive mode the DAC adjust its timing to the rate the data is pouring in.
In asynchronous mode the DAC keeps its timing constant and controls the amount of data send by the PC. By design asynchronous mode eliminates input jitter.

Resolution

A lot of people think USB audio is limited to 16 bits/48 kHz max.
A lot of (cheap and sometimes not so cheap) USB DACs are indeed limited to this resolution.

This is because the manufacturer decided to use a simple and cheap of the shelf hardware solution.

Another common misunderstanding is the specification of the bus (USB 1,2 or 3) and the USB audio standard (1 or 2).

USB Audio Class 1 standard (1998)

This standard allows for 24 bits/96 kHz max.
The standard itself doesn’t impose any limitation on sample rate.
Class 1 is tied to USB 1 Full Speed = 12 MHz
Every millisecond a package is send.
Maximum package size is 1024 bytes.

2 channel * 24 bit * 96000 Hz sample rate= 4608000 bits/s or 576 Byte/ms
This fits in the 1024 byte limit.
Any higher popular sample rate e.g. 176 kHz needs 1056 bytes so in excess of the maximum package size.

 

All operating systems (Win, OSX, and Linux) support USB Audio Class 1 natively.
This means you don’t need to install drivers, it is plug&play.
All support 2 channel audio with 24 bit words and 96 kHz sample rate

USB Audio Class 2 standard (2009)

It is downwards compatible with class 1.
USB Audio Class 2 additionally supports 32 bit and all common sample rates > 96 kHz
Class 2 uses High Speed (480 MHz). This requires USB 2 or 3.
As the data rate of High Speed is 40 X Full speed, recording a 60 channel using 24 bits at 96 kHz  (132 Mbit/s) is not a problem.

From mid-2010 on USB audio class 2 drivers are available in OSX 10.6.4 and Linux.
Both support sample rates up to 384 kHz.
It is unclear if Microsoft is going to support USB Audio 2.
You need a third party USB class 2 driver on Windows.

Companies like Thesycon or Centrance have developed  a USB Class 2 Audio driver for Windows.

Using High Speed USB for playback  there are no limits in resolution.

USB Speed

  • Superspeed – 10 Gbps USB data rate (USB 3.1)
  • Superspeed – 5 Gbps USB data rate (USB 3.0)
  • High Speed – 480 Mb/s with a data signalling tolerance of ± 500ppm (USB 2).
    This means every 125 µs a SOF packet arrives with a allowed deviation of ± 0.0625 µs..
  • Full Speed – 12 Mb/s with a data signalling tolerance of ±0.25% or 2,500ppm. (USB 1&2)
    This means every 1ms a SOF packet arrives with a allowed deviation of ± 500ns.
  • Low Speed – 1.5Mbits/s with a data signalling tolerance of ±1.5% or 15,000ppm (USB 1&2)

USB receivers

The data send over the USB must be transformed to a format a DAC (the chip doing the DA conversion) does understand. This can be SPDIF or I2S.
This is the task of the receiver chip.

Adaptive mode 16 bit units often use the Cmedia or TI (PCM270x) based chip sets. These are not programmable and usually only support 16 bit  and 32, 44.1, 48 kHz sample rate.

An example of how this chip-set performs compared with asynchronous USB can be found below.

24 bit adaptive mode DACs needs a programmable design (TAS1020 or other USB Audio Controller).

This chip enables 24 bit/ 96 kHz over USB.

Chips like the TAS1020 are limited to full speed.

You can’t do high speed as needed for USB audio class 2.

96 kHz is the upper limit when using native mode USB Audio Class 1 drivers.

A USB audio class 2 or a custom driver is needed to run 176/ 192 kHz and higher.

An example is the Tenor TE8802L by Galaxy Far East Corp.

 

  • USB2.0 Audio Class v2.0 and v1.0
  • 2-Inputs support by one I2S pairs with 128/256 Fs.
  • 2-Output support by one I2S pairs with 128/256 Fs
  • Adaptive/Asynchronous Mode supported
    • High-Speed mode support Adaptive/Asynchronous
    • Full-Speed mode support Adaptive only
  • Resolutions support 16/ 24Bit with sampling rates support 44.1/48/88.2/96/176.4/192KHz
  • Built in one IEC60958 professional 24 bit/96KHz S/PDIF RX, with I2S pins as well.
  • Built in one IEC60958 professional 24 bit/192KHz S/PDIF TX, with I2S pins as well.

 

Today you can buy complete USB-receiver modules like XMOS.

Basically a USB to I2S or SPDIF converter.

A couple of these interfaces can be found here.

Transfer modes

Data is exchanged over USB using one of the four possible modes:

  • Control Transfers: command and status operations,
  • Interrupt Transfers: device requires the attention of the host
  • Bulk Transfers: large volumes of data like print jobs
  • Isochronous Transfers: time sensitive information, such as an audio or video stream
    • Guaranteed access to USB bandwidth.
    • Bounded latency.
    • Stream Pipe – Unidirectional
    • Error detection via CRC, but no retry or guarantee of delivery.
    • Full & high speed modes only

Transfer modes explained in detail.

Isochronous transfer

When the computer sends the audio stream to an USB port, if first reads the data from the hard disk and caches blocks of the data in memory.

It is then spooled from memory to the output port in a continuous stream (Isochronous mode).

Frames are sent out every millisecond.
This happens whether there is any data in the frame or not.
The rate at which the frames go out is determined by a oscillator driving the USB bus.
This rate is independent of everything else going on in the PC.
In principle this guarantees a constant flow of the frames.
In practice the frames might not be filled properly with data because some program simply hogs the CPU or the PCI.
Anti virus polling the internet at high priority are a well known example.

 

Isochronous transfer can be done with three possible types of synchronization modes in the USB audio device.

Synchronous, adaptive and asynchronous synchronization

There must be some kind of synchronization between the PC and the DAC to avoid buffer under/overrun.

Synchronous

The clock driving the DAC is directly derived from the 1 kHz frame rate.
This mode was used by the early USB audio devices.
They were limited to 48 kHz and pretty jittery.

Adaptive

In this mode the timing is generated by a separate clock.
A control circuit (sample rate guesser) measures the average rate of the data coming over the bus and adjusts the clock to match that.
Since the clock is not directly derived from a bus signal it is far less sensitive to bus jitter than synchronous mode, but what is going on the bus still can affect it.
It’s still generated by a PLL that takes its control from the circuits that see the jitter on the bus.

adaptive1

Asynchronous

In this mode an external clock is used to clock the data out of the buffer and a feedback stream is setup to tell the host how much data to send.

A control circuit monitors the status of the buffer and tells the host to increase the amount of data if the buffer is getting too empty or to decrease if it’s getting too full.
Since the readout clock is not dependent on anything going on with the bus, it can be fed directly from a low jitter oscillator, no PLL need apply.
This mode can be made to be very insensitive to bus jitter.

async1

 

The warm reception in the audiophile world of asynchronous USB as developed and promoted by Wavelength inspired other brands to offer asynchronous USB DACs .

 

Asynchronous mode is not better by design but by implementation because you can implement a top quality (low jitter) clock in the DAC.

There is actually a good example of this case of its the implementation of the clock thats important, not the asyncness itself that is important. The recent inexpensive Musiland devices use an asynchronous protocol but then use a frequency synthesizer to generate the local clock rather than use a fixed frequency oscillator. The result is jitter that is actually worse than some of the better adaptive implementations!

John Swenson

The best way to get the most out of a dac chip is to put 2 audio oscillators right next to the dac chip. Buffer the oscillators and send them back to the USB controller to use to create the I2S (or other audio data stream L/R justified, DSP whatever) and this will give you the best response and the lowest jitter.

 

What many companies are doing is using the Frequency Synthesizer to create the audio oscillators. Basically these are frequency multipliers that can create any frequency and in the case of the TAS1020 down to 4Hz resolution. The problem with a Frequency Synthesizer is that the jitter can be as much as 100x that of a fixed oscillator. When enabling the oscillator in the TAS1020 also adds noise to the audio data stream because of the noise it fixes to the power supplies.

 

So choose wisely what you buy and ask the correct questions.

It’s not about the code… though all of ours is different, it may have an effect on the sound. But more so it has to do with the hardware and how that functions.

Thanks
J. Gordon Rankin

 

Not everybody agree that asynchronous is better.

Centrance, manufacturer of adaptive mode solutions, is one of them.

Some manufacturers may lead you to believe that Asynchronous USB transfers are superior to Adaptive USB transfers. This no more true than saying that you “must” hold the fork in your left hand. If you know what you are doing, you will feed yourself with either hand.

Michael Goodman, Chief Product Architect

Async USB provides a simpler way to implement a low jitter DAC relative to adaptive mode USB. For the cost of a small number of lines of firmware code, you reduce the amount and complexity hardware needed and potentially reduce the cost of the hardware needed for a high quality result. Most thinking engineers appreciate simplicity and the potential for low cost designs that deliver the goods.

You can find plenty of bovine excrement in the marketing of all kinds of high-end gear. Marketing products using buzz words without supporting detail or test results works when the audience is technically ignorant.

Old Listener

The perfect solution

Asynchronous USB looks to be the perfect solution.
You configure your PC for bit-perfect output and the DAC takes care of the timing totally independent of the timing of the PC.
But there are posts on the Internet claiming that even in case of an async USB DAC what is happening upstream is still affecting sound quality.

 

Almost all recent offerings of quality DACs have asynchronous USB input.

 

One issue with USB is that it sends regular bursts of info like the start of frame packet – “The SOF packet consisting of an 11-bit frame number is sent by the host every 1ms ± 500ns on a full speed bus or every 125 µs ± 0.0625 µs on a high speed bus”. If the timing of this shifts or is variable, this could elicit a different & variable reaction from the USB receiver & translate into a different & varying jitter or noise spectrum. Making the PC end as solid & stable as possible without undue processing could be one factor in ameliorating this variation. It might not be the low level of jitter that we notice but the variation in jitter – that’s one reason why I say that the measurements we currently run seem not to be capable of picking up these issues or we are not directing them to the correct target for measuring.

Jkeny

Vendor specific

Some companies don’t use USB audio in isochronous transfer mode.
They implement their own solution using bulk mode transfer.
Bulk mode is asynchronous by design.

As it is bulk mode,

  • No guarantee of bandwidth or minimum latency
  • Error detection via CRC, with guarantee of delivery.

In case of isochronous mode it is exactly the reverse.

As long as the DAC is the only one connected to an internal hub, bandwidth is in general not the problem using USB high speed mode.

 

Inherent to a vendor specific solution is that he either supports your OS or not.
The advantage of USB audio is that it is natively supported by Win, OSX and Linux.
However in case of USB audio class 2 on Win you need a third party driver too.

Anyway this solution does audio over the USB without using the USB audio of the operating system.

 

Measurement

Jim Lesurf did a nice experiment.
He measured the analog out of a DAC Magic when feed by its own adaptive mode USB and by a asynchronous USB to SPDIF converter (Halide).

The differences between adaptive (USB direct) and asynchronous (Halide) are clear.

According to the author not only measurable but also audible. [5]

Archimago [11] measured the jitter performance of a adaptive mode USB and a async mode.

adaptive mode USB

 

Asynchronous mode USB (CM6631A

Indeed, the jitter performance improves with asynchronous USB

USB cables

Cable length between full speed devices is limited to 5 meters. For a low speed device the limit is 3 meters.

As the signal degrades proportional to the length of the cable, a short cable is often recommended.

Other says this can put a source of RFI (the PC) to close to the USB-DAC.

Audiophile USB cables

As file based audio is gaining momentum and many believe asynchronous USB the way to go there is a growing market for audiophile grade USB cables.

The question of course is why a cable can have any impact on sound quality.
Some say that improved jitter performance of a cable can make a difference.
Others say that the reason we use asynchronous USB is exactly to have zero input jitter at the DAC so all what is happening upstream is irrelevant cable included.[1]
Audiophile USB cables are becoming as controversial as high-end power cords.

The Limitations of digital audio processors and cables create timing errors known as jitter, which remove portions of the audio signal and replace them with noise and distortion. Cables tend to round off the square waveforms of the signal, making them less clear to the processor, thus increasing jitter. This rounding effect varies greatly among cables and a truly superior digital audio cable can make great improvements in sound quality.
http://www.wireworldcable.com/categories/usb_cables.html

 

Another manufacturer talking some marketing bull shit?
They do have a point.
Digital is indeed sending fully analogue electrons over a wire.
And indeed, the block pulse degrades with the length.
A good digital cable is one who minimizes this degradation.

The USB 2.0 specification lists a maximum cable length of 5 meters (~15 feet). This is marginal with the best of cables, and many “audiophile grade” cables will run into problems even with far shorter lengths. There have been many credible reports of improved sound quality with some cables, but these have almost all been in systems using Class 1 Audio, with a maximum data rate of 12 MHz. When the data rate is boosted by a factor of 40x to 480 MHz, there are very few “audiophile” cable companies that have the tools and experience to ensure good results.
Computer Audio Playback Overview – Ayre

This is an easy test.
Connect your high speed USB device, e.g. a hard disk using your audiophile grade USB cable. If hi-speed mode (480 Mbps instead of 12 Mbps, the old USB 1 standard) fails, it is a bad DIY digital cable not even compliant with USB 2 standards.

Configuring

A clear and well written step by step guide to setup and USB DAC using XP, Vista or OSX can be found at the Ayre website.

Setup for Win7.

Setup for Vista.

Check

You can check if your asynchronous USB DAC is really asynchronous.
Audio devices supporting asynchronous transfer mode should have an extra ‘endpoint descriptor’ with
bmAttributes = 0x5 (USB_ENDPOINT_TYPE_ASYNCHRONOUS).

Svyr has more

A simple way to find out is to use the Thesycon USB Descriptor Dumper

Drop out.

Some users complain about dropouts when playing USB audio.

This might be due to different devices sharing the same USB-Hub.

If your audio and your graphics card are on the same hub, the bandwidth required by the graphics might cause the audio to stutter.

Anti-virus programs polling the internet with high priority might interrupt the audio too.

Trouble shooting

Trouble shooting USB audio is covered here.

References
  1. Universal Serial Bus – usb.org
  2. USB audio spec and jitter – John Swenson
  3. How USB Works – Tech-Pro.net
  4. USB in a NutShell – Byond Logic
  5. Time for a change? – Jim Lesurf
  6. Universal Serial Bus Device Class Definition for Audio Devices 1 – Universal Serial Bus (1998)
  7. Universal Serial Bus Device Class Definition for Audio Devices 2 – Universal Serial Bus (2006)
  8. USB audio standards – Computer Audio Asylum
  9. How can USB performance impact audio quality? – Computer Audio Asylum
  10. Confirming whether your DAC is asynchronous as claimed or not – svyr
  11. MEASUREMENTS: Adaptive AUNE X1, Asynchronous “Breeze Audio” CM6631A USB, and Jitter – Archimago’s Musings
  12. USB made simple – MQP Electronics Ltd
  13. Fundamentals of USB Audio – Henk Muller, Principal Technologist XMOS Ltd. -June 27, 2012
  14. CYCLIC REDUNDANCY CHECKS IN USB – USB Implementers Forum
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Computer Audio

جمعه 15 جولای 2016
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نتایج من تا این لحظه (بعد از 30 ساعت مطالعه) این بوده در عمل بهترین گزینه USB هست (دقت کنید فقط USB 2 و USB 3 خوب هستند و گزینه USB1 مناسب نیست) بصورت Asynchronous و استفاده از یه کابل خوب مثل پیوریست و اینکه استفاده از ری کلاکر و واسط های حذف نویز اصلا خوب نیست. بهتره از مک استفاده بشه با نرم افزار بیت پرفکت Audirvana با تنظیمات مربوط. کل دیتای فایل موزیک باید تو حافظه باشه.

بعضی ها معتقدند بهتره از Toslink اپتیک برای ایزوله کردن استفاده کنیم که من حس میکنم راه حل USB (با پیشرفت بیشتر USB در آینده) راه حل بهتری خواهد بود. خود تورستن هم میگه این راه حل خیلی Jitter داره و اصلا خوب نیست.

راه دیگه هم Firewire هست که عملا نه DAC سازان رفتند سمتش و نه تو مک بوک خیلی بهش توجهی شده. این راه حل خوبه اما خیلی برای ماها عملی نیست و بیشتر بدرد استودیوها میخوره و برای پهنای باند بالا خوبه. راه حل PC با کارت صدای AES یا Firewire و سیستم لینوکس هم خیلی در موردش به نتایجی کسی نرسید که بخواد توضیحی به ما بده. عملا بهترین حالت استفاده از پورت یو اس بی MAC هست با Audirvana. خوبی دیگر Usb نسبت به Firewire در سادگی و نداشتن درایور هم هست.

هارد چه SSD چه معمولی خیلی فرقی روی ماجرا ندارند اما SSD کمی بهتره. برق یا باطری روی کامپیوتر مهمه و خودم ترجیح میدم از یک Power Supply خوب استفاده کنم.

موضوعی که مهمه اینه که تو انتخاب DAC باید به نکات زیر دقت کنیم :

  • اون DAC حتما ورودی USB 2.0 یا USB 3.0 داشته باشه
  • ورودی USB مبدل DAC حتما باید حالت Asynchronous گوردون داشته باشه مثل Wavelength یا Ayre یا Berkeley USB Alpha
  • باید ورودی USB اون DAC سیگنال رو مستقیما به چیپ DAC برسونه و نیاد اول به SPDIF تبدیلش کنه چون کل ارزش USB به همون Jitter کم Asynchronous mode هست و اگر بخواد تبدیل به SPDIF بشه دیگه کار خراب میشه.

دقت کنید نقش کابل USB برای کاهش نویز از PC به DAC بسیار مهمه.

از سایت رومی این لینک رو دیدم که مصاحبه های سال 2013 هست.

 

http://www.positive-feedback.com/Issue41/ca_intro.htm

Larry Moore and Eric Hider of Ultra Fi Audio Designs

Andreas Koch of Playback Designs

Tony Lauck

Steve Nugent of Empirical Audio

Gordon Rankin of Wavelength Audio

Jon Reichbach of Sonic Studio/Amarra

Vinnie Rossi of Red Wine Audio

John Stronczer of Bel Canto Designs

Daniel Weiss of Weiss Digital Audio

Vincent Sanders and John Hughes of VRS Audio Solutions

Kent Poon of Design w Sound

Charles Hansen of Ayre Acoustics

Pete Davey of Positive Feedback Online

رومی میگه جای نظرات Dan Lavry,  Ed Meitner, Michael Ritter and Pflash Pflaumer هم خالیه. شرکت های Lavry Engineering و  EMM Labs و berkeley audio design .

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Asynchronous USB DAC Bit-Perfect NOS Tube Non-Oversampling No Filter

جمعه 15 جولای 2016
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نتایج ما از یک سورس دیجیتال خوب تو میکرو :

  1. در صورت استفاده از کامپیوتر ورودی یو اس بی USB (نسخه های 2 و بالاتر) بهتر از SPDIF و بقیه ورودی ها هست اگر USB خوب پیاده سازی بشه بصورت آسینکرون Asynchronous USB و حذف نویز کامپیوتر . اینو هم تورستن میگه هم گوردون طراح Wavelength
  2. بیت پرفکت بودن یعنی آپ سمپل نشدن NOS (Non-Oversampling) و عدم استفاده از فیلتر دیجیتال و بطور خلاصه هیچ تغییری روی دیتای دیجیتال ندادن. این موضوع رو هم Peter Qvortrup طراح Audio Note و 47Lab میگه و هم رومی و تورستن (طراح AMR) و طراح Lampizator (آقای Lukasz Fikus ) خیلی روش تاکید دارند و طراح Wavelength (آقای Gordon Rankin) هم همین اعتقاد رو داره. همه اینها با دستکاری دیتای دیجیتال و هر نوع پردازش حتی ساده مخالفند. تمام DSP ها کارشون دستکاری دیتای دیجیتال هست.
  3. عدم استفاده از op-Amp و فیلتر خروجی و در مقابل استفاده از خروجی لامپ و ترانس که Tim طراح EAR و طراح Audio Note و طراح Lampizator بهش اعتقاد دارند.
  4. برق Power Supply خوب
  5. حذف کامل جیتر Jitter

در مورد این DAC USB شرکت Wavelength Audio بیشتر بخونیم :

Cosecant USB DAC by Gordon Rankin

http://www.6moons.com/audioreviews/wavelength3/cosecant.html

  – The Cosecant v3 connects the USB controller to a DAC module. The USB firmware to run that DAC module resides there so each module has its own developed code. The output of the module connector is sent directly to the 6GM8/ECC86 dual triode output tube, which drives the transformer-coupled output.
– The Cosecant has an external power supply with IEC connector to isolate the power from the DAC and its audio transformers.
– This is my own design and I wrote the software (1,800 lines of code for the USB controller to do what it does) like no one else does on the planet.
– The DAC featured in this review uses Asynchronous USB mode as do the Crimson. The USB interface is bidirectional, and has built-in error correction and buffering at both ends; it is an asynchronous interface. Clocking synch problems associated with SPDIF are not present with USB. The result is that the data on the disk is identical to what is leaving the DAC all the time. At start-up, the DAC tells the computer it can handle 16 bit audio at 32K, 44.1K and 48K. Since the USB receiver only has to handle these 3 frequencies, the clocking to the DAC has almost no jitter. SPDIF actually has to be synched to the exact frequency of the transport (i.e. if the transport is working at say 44.0896K instead of 44.1K the DAC has to sync to that frequency). Therefore the jitter problems of SPDIF are all but eliminated. The result is a zero error protocol to link between computer and DAC, with ultra low jitter.
– None of these DACs use any type of operational amplifiers (opamps) in their design. The DAC module is the only solid-state portion of the overall DAC.
– The output stage of each of these products is the key design element which is responsible for their overall sound.
– These DACs incorporate the TDA1543N2 (select top 5%) DAC chips with passive I/V using Shinko Tantulum resistors in a configuration that does not use analog or digital filters. Some people call these NOS DACs or what I call zero DACs. The data input is the data output without any up/oversampling or other manipulation, which seems to make for a very analog presentation.

http://www.goodsoundclub.com/Forums/ShowPost.aspx?PageIndex=1&postID=8265#8265

Well, I admit that I was clueless on the subject but I was intrigued and I consulted my engineering recourses.  It turned out that what I proposed above was not exactly accurate: for instance regular computers do not use I2S bas interface…

To make the long story short: For a computer the USB port is a devise of the same hierarchy as sound card and there is not different for a computer to where output stream: to a driver on the sound card or to the driver the sit behind the USP port designation. With identical drivers used the ASE/EDU interface has fundamentally lower jitter interface, where the min jitter for USB is not even included in the USB standard. So, with identical design (more of the time used) the USB has no chance to be superior.

There is a catch however. With proper design, when USB is made not for utilitarian purposes but for high quality transfer, the USB has internal chance to be way more advanced. The ASE/EDU is forward-only interface where stream flows only in one direction with data mixed with clock marks. The USB is full duplex or bi-directional interface where stream flows in both ends. This enables designers to put a clock that might everything on the receiver side (DAC) and let this clock to manage the USB’s and even reader timing. USB sends data in burps by requests of by scheduled timing, this marks all might be managed by receiver side clock. In ASE/EDU the receiver shell recognize the timing marks and PLL or re-clock data. In USB there are no needs for PLL or re-clocking as the data arrives at the marks of the original clock. The USB in this case acts like an elephant that sticks a long trunk to another devise and sucks juice… according to my consultants this USB implementation is the most proper way to do the things. Or course no one knows HOW the USB is managed even if a DAC has USB port…

The Cat

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Bit Perfect Digital Audio

چهار شنبه 27 آوریل 2016
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خب من رفتم تو ebay که Apple MAC mini 2011 MC816LL/A بخرم چون شنیدن از مک بوک بهتره اما دیدم نو نداره و دست دوم هم بدلم نمیشینه. دلیل mac mini 2011 این بود که usb 2.0 خروجی میده و DAC 4 EAR هم ورودی usb 3.0 نداره. بعدش فهمیدم USB 2.0 با USB 3.0 فرقی نداره و دیدم بهترین کار اینه همون mac mini 2014 رو اکبند بگیرم که usb 3.0 داره.

زنگ زدم مجتمع پاینخت که مک مینی بگیرم اما فروشنده گفت فردا بهم خبر دقیق قیمت 3 مدلش رو میده و 3 روز دیگه دستم میرسه.

شب که اومدم خونه یهو یادم افتاد خواهرم یه  Macbook pro 2014 داره که استفاده ای هم ازش نمیکنه و تقریبا تازه هم خریده بود ، رفتم اوردمش خونه دیدم به به هم رتینا هم mac os x 10.9 هم هارد ssd و usb 3.0 که گفتم همین عالیه. قسمت چیز خوبیه بخدا …

http://www.bhphotovideo.com/images/images2500x2500/apple_mf840ll_a_13_3_macbook_pro_notebook_1128848.jpg

خلاصه سرتون رو درد اوردم قسمت شد همین مک بوک پرو رو برای سورس دیجیتال استفاده کنیم. من به پیشنهاد تورستن طراح AMR میام از نرم افزار Audirvana Plus 2 برای پخش روی مک استفاده میکنم و تنظیمات لازم برای بیت پرفکت بودن شامل استفاده از مد Integer در mac os x 10.9 هست.

عبارت Bit Perfect یعنی هیچ بلایی سر سیگنال تو دیجیتال دامین نمیاد و همون جور که رومی دوست داره سیگنال منتقل میشه. برای تنظیمات PDF زیر رو دانلود کنید:

http://www.hifi.ir/wp-content/uploads/2016/07/MAC-OSX-audio-players-Integer-Mode.pdf

دیگه فقط اوردن کابل ها می ماند از برند Purist Audio که اونم آرمن از مونیخ انشاالله برام میاره. سیستم آماده میشه برای شنیدن موسیقی. یه Power 3000 purePower هم انشاالله تا قبل مردادماه برسه همه چی عالی میشه.

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iFi Audio AMR Audio by thorsten loesch

دوشنبه 25 آوریل 2016
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Retro-Stereo50-01

 

LS35_Main

این آقای thorsten loesch یه طراح باهوش حساب میشه تو این های اند و تو سایت رومی هم ای دی داره و بحث میکنه.

این آدم تو حوزه دیجیتال خیلی کار کرده و مقالاتی داره که خیلی ها از جمله طراح Lampizator هم اونارو خونده و از تجربیات تورستن استفاده کرده.

این طراح تو کلاس قیمتی پایین سیستم های زیر رو ارايه داده که من  iDSD و 3.0 iUSB رو خریدم. یه ست جالب خونگی هم داره که با بلندگوی BBC فروخته میشه. هر کدوم از این کامپوننت ها یه کاری انجام میده.

توضیحات بیشتر تو سایت http://ifi-audio.com/ هست. نماینده این برند هم آرمن هست 09121198262 . قیمت ها بین 300 تا 500 دلار هست.

 

http://ifi-audio.com/wp-content/uploads/2016/02/banner-iCAN-SE.jpg miCAN SE_EN_11 idac204 http://ifi-audio.com/wp-content/uploads/2015/07/miusb3.001.jpg iPhono2-1 Micro-iDSD-02 MicroiCANPic01 MicroiLinkPic03 MicroiTubePic01 Read More

ALAC, FLAC, AIFF , WAV Music Streaming by Bitperfect Network Media Player

جمعه 8 آوریل 2016
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https://gigaom.com/wp-content/uploads/sites/1/2010/06/mac-mini-06-2010.png http://static1.squarespace.com/static/53f2ba7ce4b0e9d91b048708/t/53f7b8d2e4b0e954f994ba60/1408743638527/alphausb.png?format=1000w

 

من باید دیتای دیجیتال فایل روی کامپیوتر رو به DAC 4 برسونم .

امروز کمی تحقیقات کردم و فعلا به این نتیجه رسیدم برای سیگنال خوب دیجیتال اگر بخوام سخت افزار بخرم باید بین 7000 تا 20000 دلار پول بدم که خیلی احمقانه است و تازه نمیدونم تو اون سخت افزار داره چه اتفاقی میفته.

ظاهرا بهترین گزینه مینی مک هست با Power Supply خطی افترمارکت و تنظیمات لازم برای Bitperfect . این Bitperfect یعنی هیچ تغییری روی فایل در حال پخش تو دیجیتال دامین ندهیم.
حالا اومدیم مینی مک رو تنطیم و با نرم افزار Audirvana موسیقی رو پخش کردیم میرسیم به ریکلاک و بافر Berekley Audio Alpha که این کارش حذف jitter سیگنال هست بدون دستکاری سیگنال دیجیتال. یه همچین چیزی باید توی خود DAC باشه (مثل AMR یا EMMlabs) اما من حدس میزنم DAC4 EAR خیلی روی این موضوع مانور نداده. البته حدس میزنم و فعلا این الفا رو درموردش تحقیق میکنم و قطعی نیست برام.

http://www.co-bw.com/Audio_OSX_Optimal_Audio.htm

http://yourfinalsystem.com/custom-modifications/yfs-music-servers

http://www.positive-feedback.com/Issue72/mojo_audio.htm

http://www.mojo-audio.com/mac-mini-upgrades/

http://www.coreaudiotechnology.com/products/music-servers/mac-mini-music-server/

http://www.positive-feedback.com/Issue54/mach2.htm

این تورستن که طراح AMR هست و کلی تو این حوزه دیجیتال سروصدا کرده مطالب جالبی نوشته در مورد اینکه چه نویز ها و مشکلاتی سر راه این انتقال سیگنال تا DAC هست.

من در مورد تورستن مفصل مینویسم و این آدم جزو طراح های خیلی باهوشه. آرمن این آدم رو کشف کرد و تو سایت رومی هم در مورد این آدم و کاراش زیاد صحبت شده و مهم اینکه قیمت های این AMR هم خیلی مناسبه.

http://www.psaudio.com/ps_how/how-to-build-a-music-server/

http://www.6moons.com/audioreviews/players/4.html

برگردیم به مطلب تورستن در مورد انتقال دیتا از فایل به DAC :

http://www.amr-audio.co.uk/html/dp777_tech-papers.html

http://www.amr-audio.co.uk/html/dp777_tech-papers_OSX-Integermode.html

Beyond bit-perfect: The importance of the Player Software And MAC OS X Playback Integer Mode

Damien PLISSON, Audirvana developer

Abstract

In computer audio, the player software replaces the CD drive as the transport feeding the DAC. Ensuring bit-perfect output of the original audio signal is only a pre-requisite, while minimizing jitter and RF interferences are still strongly needed.
This paper explains the main factors impacting sound quality on the computer side, and the means that have been implemented in Audirvana player and the AMR DP-777 DAC to boost the audio experience to the next level above the normal iTunes.
These main means are bit-perfect, sample rate switching, asynchronous transfer and Integer Mode.

Introduction: bit-perfect as the only goal or the myth of the flat-square world

In the world of digital audio, the caveats of the CD player are well known, namely the read errors and the jitter induced by its mechanical transport.
It is widely thought that computer sources are immune to these issues, given that they are faithful to the original signal, that is are bit-perfect.
But unfortunately the digital world inside a computer is not a flat-square world composed of perfectly timed zeros and ones. The audio signal chain goes through different elements whose each can alter the sound quality.
In this paper we’ll look in details at these, and see what a “source direct” solution can be to minimize the adverse effects, and achieve very high sound quality, better than nearly all the CD transports.

1. Sources of non-quality

Assuming the output is bit-perfect, the computer as a source creates two main sources on non-quality:

Software-induced jitter

Digital signal is in fact an analogue waveform composed of two states separated by a voltage threshold (1 if above, 0 if under).
As presented in [MeitnerGendron91], the receiver detects the value change the moment the analogue value crosses the threshold. In addition, the shift from one state to another is not instantaneous but more slope like.

So a slight change in the reference voltage of the source will lead to a slight temporal shift in the value change detection.

voltage-jitter
Figure 1: Reference voltage induced jitter

So fluctuations in the source reference voltage create jitter, as explained in details in [HawksfordDunn96]. This is the same on the receiver side with measurement threshold fluctuations from its power supply and/or ground instability. Moreover the computer can still cause this as the grounds are linked most of the time through the same signal cables.

Computer load means rapidly changing power demands from the CPU and its peripherals, with peak demands that are directly related to the software behaviour.

Radio-Frequency & other interferences

In addition, computation, disk access, … activities mean complex current waveforms are carried on electrical lines and thus generate electromagnetic interferences. Apple computers are now made of “unibody” aluminium cases that are good protectors from inside RF interferences. But this is not sufficient as the cables connected to the computer act as antennas. And these current waveforms are also going back through the computer PSU, polluting the mains power supply.

2. The hidden audio filters of OS X

As a modern operating system OS X needs to offer shared access to the devices including the audio output to all running applications. But this is done at the expense of pure sound quality:

Audio mixer

Fortunately when only one application is playing audio, it doesn’t affect the signal and thus is at least bit-perfect in this case.

Sample rate conversion

In this shared model the device sample rate is not switched to match the original signal’s, but it is this last one that is sample rate converted.
In addition a suboptimal algorithm is used to minimize the CPU load of this real-time operation.

Digital volume control

OS X offers through its mixer volume control (e.g. the one offered in iTunes). But as it operates on the digital signal, any volume value different from 100% means loss of bit-perfect and precision loss (e.g. a volume value of 25% means 2 bits precision loss).

3. The data transfer to the DAC

First way to connect to the DAC is to use the build-in TOSLINK output of the Mac. But this one should be dismissed for being too jittery for serious use.
Strong improvement comes by using “computer connection” to the DAC, being either USB or FireWire.
FireWire has long been the interface of choice for the pro-market as it is made by design to guarantee continuous streaming of AV data on large number of channels. Anyway its complexity of use (installation of driver required, hot plugging even strongly advised against by some manufacturers because of its potentially harmful issues, …) and its unclear future have made USB the widely used choice.
The first type of USB devices are called adaptive (or synchronous), meaning the DAC clock is slaved to the computer’s continuous stream of data.
More recent and advanced USB devices use asynchronous transfer mode where the DAC controls the flow of audio data, buffers it, and uses its own stable-low-jitter clock. Thus it is immune to short interruptions of USB stream (e.g. bus reset, other device burst transfer, …), and much less prone to computer jittery clock.

This combines the advantages of both worlds: ease of use of USB (no drivers), and stability of FireWire. This is a great step towards sound quality, but it is not decoupling completely the DAC from the computer, and the interferences, software-induced jitter still apply, starting by following the ground loops.

4. The player software impact

First of all the player should ensure bit-perfect reproduction of the signal by:

  • Adapting the DAC sample rate to each track native to avoid any unwanted sample rate conversion
  • Taking exclusive access (“hog mode”) of the device to prevent other opened applications from interfering

Furthermore, as we have seen in section 1, the computer load (and its variations) has an impact on sound quality. Minimizing such current demands and sources of interferences is key:

  • Loading tracks before playback (“memory play”) to reduce disk access and its audible, power and RFI impacts
  • Minimizing synchronous CPU load taken for the audio data streaming operations. In addition to reduce jitter, this also helps to reduce audible RF interferences patterns, especially in low frequencies

5. Further optimization at driver level: Integer Mode

Audio playback in OSX is usually performed through a high-level framework, the Audio Units processing graph [AppleCoreAudio]. The first optimization of an audiophile player is to bypass these overhead facilities and address directly the CoreAudio lowest layer: the Hardware Abstraction Layer. (See figure 2)

player-vs-audiophile
Figure 2: Usual OS X file player vs Audiophile concept

In normal mode, all data exchanges performed across the user/kernel boundary are in PCM 32-bit float format, easing the different audio streams mixing process and associated soft clipping. [AppleHAL_1]
Note that it is anyway still bit-perfect up to 24bit definitioniv.

Integer mode

Addressing directly the HAL [AppleHAL_2] gives the possibility to bypass the two main overhead processes of the above standard mode:

Field Programmable Gate Arrays

  • Mixing buffer
  • Float to DAC native format conversion

gate-array
Figure 3: Float vs Integer Mode

In Integer Mode (see figure 3) the player software supplies a stream already formatted in the native DAC format, thus optimizing synchronous CPU load at the driver level.
These operations performed inside the driver, in the kernel space, in real-time are on the critical path for sound quality as they are the most synchronous, happening at the very immediate moment of the data transfer to the DAC. So optimizing it is of great benefit, and this is only applicable to compatible DACs that offer this non-standard mode.

Conclusion

The computer is a great music server but also a source of jitter and other RF interferences that are detrimental the sound quality, even when bit-perfect reproduction is ensured.
The player software needs to optimize and streamline the audio path to minimize these adverse effects essentially linked to the processing load synchronous to the audio streaming. Achieving “source direct” in addition to “bit-perfect” is key.

This is what I’ve tried to get in the Audirvana player by streamlining to the maximum the real-time operations that are limited to simple data streaming in Integer Mode, while all the other processes (loading from disk, decoding, converting to DAC native format) are done

Float mode

offline in a preparation phase, before playback. This is called full memory play. Best results are achieved when feeding an Integer Mode, asynchronous USB DAC like the AMR DP-777 that can take advantage of all these optimization features.

References

[HawksfordDunn96] Bits is Bits ? in Stereophile 03/1996
[MeitnerGendron91] Time Distortions Within Digital Audio Equipment Due to Integrated Circuit Logic Induced Modulation Products, Ed Meitner and Robert Gendron, presented at the 91st AES Convention, New York, October 1991, Preprint 3105
[AppleCoreAudio] CoreAudio Overview: What is CoreAudio ? in Mac OS X Developer Library
[AppleHAL_1] Audio Device Driver Programming Guide: A Walk Through the I/O Model in Mac OS X Developer Library
[AppleHAL_2] AudioHardware.h documentation in Mac OS X Developer Library

Replacing the HDD by a SSD removes the directly audible mechanical noise but not the other issues as it still requests important current waveforms to transit on lengthy wires. And the OS overhead is still present.
OSX Audio low level subsystem typically requests data in 512 frames chunks, that is at a frequency of ~86Hz for a 44.1kHz sample rate.
Note that bit-perfect playback can still happen if all effect filters (including software volume control) are deactivated. Thus stock iTunes can be bit-perfect.
32bit float is composed of 1 sign bit, 8 exponent bits and 23 bits for the mantissa. Thus giving 24 bits of significant precision.

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انتخاب DAC/Amp برای هدفون سنهایزر

یکشنبه 3 آوریل 2016
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آرمن توجه مارو به کارهای آقای Thorsten Loesch جلب کرد تو حوزه طراحی دیجیتال و فونو استیج . این ادم خیلی هم مورد توجه رومی بوده و من هنوز تو شرایط مناسب سورس دیجیتال اش رو نشنیدم اما همه شواهد حاکی از کاردرست بودن این آدم داره.

این طراح شرکت AMR رو اداره میکنه که آرمن نماینده این شرکت در ایران هست و من DAC پورتابل سری Ifi این شرکت رو بنام iDSD برای هدفونم گرفتم و الان هم DAC و هم هدفونم داره برای بالای 300 ساعت آب بندی میشه. از نظر امکانات که بی نظیره و صداش هم بعد از اب بندی باید شنیدنی باشه. اینم لینکی که آرمن فرستاد :

http://www.enjoythemusic.com/magazine/equipment/0914/ifi_audio_micro_idsd_dac_headphone_amplifier.htm

مصاحبه تورستن

http://www.audiostream.com/content/qa-thorsten-loesch-amrifi#Pl5kjC5pyBfDhvgL.97

With AMR products we are largely free to design in technologies or exotic parts and even purely old stock parts that are no longer manufactured with scant regard for cost or time to implementation. New Old Stock DAC Chip’s not made for over a decade or two, no sweat. New Old Stock tubes over 50 years old – sure, put them in. Take a year to perfect a small aspect of the circuitry? No, make that two years, well if that is what it takes, then that is what we do.

With iFi we must live with time and cost constraints; use readily available technology and we do not have years to perfect a single-design, but need to be able to release new products much quicker.
Read more at http://www.audiostream.com/content/qa-thorsten-loesch-amrifi#MYmVAXK5UYVa0R9l.99

خود هدفونم تو هدفون ها سنسیتیوته و امپدانس معقولی داره و جزو هدفون های بد درایو محسوب نمیشه. منتظرم ببینم بعد 300 ساعت چطور میشه.

اما AMR در سطح محصولات های اند خودش مانند CD-77 هنوز DAC/Amp هدفون نزده و فعلا DAC/Amp خودش رو تو سری ifi ارایه میده.

اما Lampizator هم برندی شده الان و یه هدفون آمپ زده که توجه منو جلب کرده. من قبلا در مورد Lampizator نوشته بودم :

Lampizator DAC

این آمپ هدفون اسمش Head DAC هست : http://www.lampizator.eu/Fikus/HEAD_DAC.html

in this private website I summarized my 20 years of experience in designing, building and modyfying the stereo hi-fi equipment. Oryginally inspired by Audio-Note’ Peter Quvortrup philosophy of constant upgrading by better parts, I started a new era in my audio hobby – DIY. Then encouraged by Thorsten Loesch page I started learning about all things related to digital.

فعلا در مورد آمپ دک لامپی اطلاعاتم تکمیل نیست. وایسیم ببینیم به کجا میرسیم. حدس من اینه اگر توقع بازه فرکانسی زیاد خصوصا برای bass نداشته باشیم و یه انتخاب مناسب روی آمپ هدفون داشته باشیم نتیجه خروجی خیلی از نظر قیمتی بهتر از ست خانگی های اند خریدن است.

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مقاله ای در مورد دیجیتال

سه شنبه 27 اکتبر 2015
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آقای آرمن الکساندریان یه لینکی فرستادند که خوندنش جالبه :

http://www.audiostream.com/content/qa-thorsten-loesch-amrifi#rfBlVq51WfamE4jS.97

منم خوبم دیگه چرا اینجوری دارید نگاه میکنید منو … خب کارام زیاده نمیرسم به اینجا ، اقتصاد هم تعطیله مجبورم بیشتر کار کنم و آئودیو هم که میدونید خیلی دوست ندارم .

چقدر این پاییز 94 با صفاست ، این جمعه ای که گذشت با یکی از دوستان بعد از دو سه ماهی نرفتن شمال برنامه جور شد رفتیم جاده چالوس هوا بسیار مطبوع 20 درجه جاده خلوت برگ های پاییزی جای همه خالی خیلی خوش گذشت.

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پروژه DDDAC 1794 NOS

جمعه 5 جولای 2013
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تو سایت Twogoodears مطلبی نوشته شده درباره ساخت یک DAC :

http://twogoodears.blogspot.com/2013/06/digital-skyrocketing.html

امکان ساخت پروژه تو لینک زیر هست خیلی شفاف توضیح داده و بد نیست همون DAC 1541 استفاده بشه .

http://www.dddac.com/dddac1794.html

The Digital Azimuth or DDDAC frenzy in Koln

Hi, folks… here is an essay by Reinhard, a chronicle of last weekend near Koln, with new and old friends  enjoying music in Klauscope’s Goto system glorious sound with latest Doede Douma’s and Bernd’s DACs and PSUs… Jean and Andre, James and his girlfriend and others guests from France – BUT me – were partying… I deeply missed the gathering, indeed… thanking my german pal, here is the gap filled, for posterity;-)”Dear Stefano,
first of all congratulations on your personal ameliorations in sound within your own Gotorama with “digital stream”… This is wonderful to read. And this is really awesome, isn’t it? And I know exactly how lucky you feel now… so, especially regarding this matter: welcome to the club (again)…
Let me add that we all were very sad that you missed out and that it was a pity that you were not able being part of our latest session here at “The Goto-Triangle” at Cologne just the other week, especially because of this specific theme…

This is Klaus’s system, ready for testing…
Well, right in the beginning let me thank you for allowing me to put the relevant research and listening results of this special event onto your page, so that we all are at least able to fill in this gap a little via some infos and pictures… and indeed, the results are really breathtaking, rather dramatically superior to that what you experienced on your visit last december, another really big (musically fulfilling) step forward:
1. simply because of possessing and being capable of playing the new ISO-file-format, ripped from SACD with much higher resolution than 24-192 (the amount of data is about as double as much, and you may have an idea of how small the “stairs” in digital reproduction become, building a nearly “perfect” sine-curve and how the sound gets nearly completely integrated because of that) and
2. heavy progresses within the DDDAC-areas “Quadriga” and “Octopus” in combination with Doede’s and Bernd’s corresponding power supplies for the DACs and USB… you may imagine: it’s hard to believe, but within HiFi-heaven, where we clearly are already for quite some time, there is another stairway going up (you remember: it was you, last time using the expression: “stairway to heaven” (not to quote Led Zeppelin but when entering Klaus’s “Nirvana”… and what a stairway that is…
and

Reinhard’s Quadriga, the DDDAC1794 24 Bit NOS

and picture 6, in combination with Bernd’s 18 kg power supply

What about the panel?

They were, from left to right: Triodedick, Doede Douma, Jean Hiraga, Klaus Speth and André Klein.

1. of course the whole time we held a place free for you, Stefano, but unfortunately you missed it out
2. Jean Hiraga, the “Mount-Everest” in HiFi, who meanwhile has become our long time follower and whose satisfaction, contentness, nodding with his head and grinning on his face has become more and more fulfilling… (I know that we all once started at zero, being uninformed when we were young, so if there should be somebody out there not knowing about Jean, please google… the ocean of his longtime hardcore researches and reports is way too vast to present it all here (to name just two things: class A and tube amps development…), and to make it short, Jean is the one – together with Shinichi Tanaka from Goto Society in Japan –
who have constructed, listened to, measured and reported about the most and highest end systems all around the globe for quite some forty years now…
3. Doede Douma – the “new magician” in the row… there is no longer any need for me to introduce him further, because since our last session he has established one of the most inspiring and creative sites in the net for DIYers, not only especially regarding DACs,
but as well amps, and speakers, and…
I enjoy his sites that much because I – as a non physician and non electrotechnician – understand most of what is written there, and I can clearly hear everything that what is all about… to me – it is just exactly perfect !!!
4. just the same seems to have experienced our first time guest, “Triodedick”.
Klaus supervising, w/Doede Douma and Dirk aka “Triodedick”, doing what they like most: “conjuring tricks”;-)
Triodedick is very kind and quite famous, not only among DIYers. He is a real specialist, also coming from the Netherlands (like Doede), a long time friend and colleague of Doede in studying physics and electrotechnics at Utrecht University and sharing our HiFi-hobby ever since. If you like, enjoy his sites:
and here you might find Triodedicks report – in four parts, not only for DIYers – about Doede’s “magic” DAC, the DDDAC 1794 24 Bit NOS DIY DAC (part 1 to 4)…
5. and André Klein, a very nice guy from Metz in northern France with very specific knowledge and skills, joined us again… some of you might know him from the regular European Triode Festivals in France: he is an extreme tube lover and connoisseur and the proud owner of a system based on Western Electric WE15 in combination with self-built WE amplifier model 46-B for which he won several prizes… he is an EMT-turntables-and-cartridges afficionado, as well…
5. and James, our astro-physician from Australia, with girl friend, still in the process of exploring and planning his system (having spent some 70.000€ in the past and still not content), but knowing already the very best systems in Japan and ours…
James with his girl friend, sitting and listening in the – for Hi-Fi-Fans – “center of all centers”
6. a befriended couple from Jean from France, which simply couldn’t stop getting astonished and overwhelmed… you know, how people react, when they listen to Klaus’s system for the first time. They are struck in awe. And exactly the same happened here: you couldn’t talk to them…
Jean, André, and french guest
7. and lastly, Klaus and his wife Moni,  (I do not want to produce redundant information material here, so when you google, you will find enough of what Klaus is doing for the last 40 years…)

… and this is me, Reinhard…

What was it all about last saturday ?What was the theme… the main theme?The very theme?
This meeting was some sort of fulfillment (that way, that  w e   h e r e   nod with our heads, and not anybody else) of about three till four years of research within the digital area, and here especially within the DAC-area, i.e. digital to analogue converting… and this proved to be extremely exciting!
DDDAC1794 24Bit NOS in all it’s glory, with nice display showing bits, resolution, voltage… programable…
… latest since Douglas Sax, yes, the guy from Sheffield Lab with the direct vinyl cuts… some years ago at the CES in Vegas officially claimed that vinyl is simply produced because people demand it – and  n o t  because it is better than digital – nearly everybody awoke and got curious, so did we, and especially Doede…
Suddenly everybody wanted a DAC, and everybody got a DAC, and everybody was unhappy with the results, first with the small ones, then with the big and better (?) (or at least more expensive ones) up to 7.000 €… no, I do not name them, but they are leaders on the market and very famous…
…the result was always, I repeat  a l w a y s, that way, that we were all still unhappy and more or less (mostly more) plagued by a somehow unpleasant, unprecise and muffled sound with digititis (the deficiency of getting a signal that way reproduced that the “stairs” in digital are not or hardly annoying and/or recognizable during listening – mostly caused by jitter), the results were always clearly way beneath vinyl reproduction.
We nevertheless immediately recognized the “advantages” of a DAC even if we were not happy with the implementation and realisation at the time… as is to this very day with external, i.e. not self built DACs, independent from low or high resolution (by the way the results with higher resolution are not as good in comparison to 16 Bit)…
… and exactly this was the hour of Doede, “living” in the digital world from the first moment on at university since 1976, he managed to grew up (Hi-Fi-wise) with the fulfillment of his splendid ideas, clearly and precisely, and it pregnantly shows… well, to make a long story short:
(if you should like it long, you may visit his homepage here):
Doede developed two DACs, his DDDAC1543 for 16 Bit, and later on with the upcoming of high resolution and corresponding downloads on the market his DDDAC1794 24 Bit NOS. But he not simply developed a DAC, or DACs. What makes his DACs that special is the possibility of multiplying the DAC-Chips which ameliorate the sound  a n d  reduce the jitter. As the results were really spectacular within his DDDAC 1543 he went from 12 to 24, to 60, to 120, and finally to 240 chips (maximum amount: 16 x 16 = 256)…
This is Reinhard’s DDDAC1543 16bit DAC with 120 chips, also in combination with Bernd’s power
supply – getting attested on saturday the most “punch” of all.
The sound still and still and still and still got better, dramatically better, and within the DDDAC1794 24 Bit NOS till some months ago he went from one deck to four decks (nickname “Quadriga”) and then to eight decks, nickname “Octopus”, here the sound got much better as well, but because of a 24 bit chip as a basis and an already pretty much sublime sound to start with the need of multiplying chips is not that high as with 16 bit chips… although we will test further.
During the last three years it became obvious that the results of Doede’s research were that good that the ultimate tests could only be hold at Klaus’s…why ? because Klaus’s is not only the best but the only place capable of bringing the 500 horse powers of the newly constructed DACs onto the road… I mean onto the road without squeaking… and that’s exactly the point, if it is squeaking – like on nearly every other system – you either loose music or the music is not reproduced the way you like it…
This was exactly our theme on saturday: testing these different DACs with different amounts of chips and decks without squeaking, and “with” and “against” each other, and of course with all sorts of music and resolutions, ranging from “simple” 16 bit up to 24 Bit 88, 96, 176, 192 kHz, and as the icing on the cake with for the first time ISO-files, the direct DSD.format (DSD64 and DSD128) with about the double amount of data as with 24-192…… and ooohhhh, not to forget, all this in comparison with a big lorry battery, and Bernd’s and Doede’s newly finished controlled power supplies, and a new controlled power supply from Mundorf…
Doede’s controlled power supply, to the right

Mundorf controlled power supply (for Bernd’s controlled power supply see above)
Testing results
Of course the highest amount of chips and decks were the clear winner, in combination with the superb controlled power supplies, and the Kimber USB silver cable that Klaus and I have in use..
I’ll do not enter into a description of the sound, there is not much to say, it is realistic and natural, and it is simply music, that’s what we ever wanted but never got – till now (and here ISO brings us to exactly that level in reproduction what we ever have been looking for – to me this sounds like master tape)
We can listen now to music that way that the human brain is no longer bothered by whatever things that do not belong to the music and which are simply wrong and/or artificial. Let me say: the result was: listening to music the most pleasent way out of any system I ever heard (exept mine), no matter what music and/or what loudness… (but you have to keep in mind, that I have not heard Doede’s eight deck DAC – the “Octopus” – on my system…)
USB-cables
Wow, wow, wow, that was a program… as it turned out it was nearly too much to handle… it was not of course, but after some twelve hours there was no time left to properly check out different USB-cables within this “circus”… and I have to admit, the results were so satisfying right from the beginning, that there was no real need to urge.
USB-cables… this is indeed a theme on its own… all sound different… and this within the pure digital area where only 0s and 1s reign!!!Ultimately we want to get rid of PC- or MAC-norms, i.e. USB, Firewire or whatever, in order to get completely rid of these “uncontrollable” influences. This will be our next step, to check out the PC offal and take off the digital signal directly from the motherboard, but this is still a way to go… meanwhile we have to bear with USB. And actually we use Kimber USB silver cable. Klaus and me, we both have it running (it took more than 150 hours to burn in to the point)… those of you who might want to read more about the experiences that as well others have with it, have a look here:
This cable is really good, and it has a very acceptable price, at least it may serve as a good start to compare and try out more…
Controlled power supplies
The controlled power supply (everywhere: in pre-amp, channel deviders, power amps, DAC, and later on for hard drives and within the PC) is also a chapter on its own. I remember reading Jean and others – long time ago – that there is no better power supply than batteries… and of course at that time we didn’t dare to say anything against it, neither did we try it out for ourselves… but now we really wanted to know, precisely…
Klaus got a large lorry battery, weighting some 40 kg or so, and we listened with it to music in combination with the DACs … well, it was fine, it was ok, it was quiet, and we finally presumably would have been constantly convinced if Bernd would not have constructed a new power amp, originally planned as power amp prototype for the Goto-drivers, with two trafos, one for +, and one for -, with some nine kg each, so both about 18 kg…. it was not at all constructed and intended to supply the DACs, it was just a funny idea and happened by accident to be tested out in this combination.
This testing came out as one of the greatest surprises in HiFi-history: driving some 500 g DACs (each) with an 18 kg controlled power supply!!! As it turned out it is exactly that what is needed to get the most and best music out of the DACs (the same results are valid in all other amplification areas – as indicated above)… Within the chain of the amps exactly this result is “nothing new” and known for years, but I have to admit, if I would not have heard it with the DACs, I would not have believed… and listening to these results Doede immediately began to construct a power supply for his own DAC, with really dramatic results, see picture above.
ISO
Before I start with ISO, just let me tell you that just the other day I read in “a very friendly forum”, where they are not shy to emphasize to unknown “newbies” like me, like in a canon or choir, that “friendlyness is… culture”, that ISO-files are not a “new sensation”. Well, that’s right. ISO-files per se are not a new sensation. If one is able to generate a perfect PS3-rip.Stefano, I ask you, who on earth is capable of doing that?
The “new sensation” is fortunately no longer in getting the ISO-files ripped, but in getting them properly played!Who dares to say that he is capable of reproducing ISO properly ? Knowing, hearing that what they are listening to is still not that what they are looking for? Anyway, I have biggest doubts that anybody else on the globe is able to even come close to Klaus’s results,  e s p e c i a l l y  with ISO… the Goto drivers with ISO really live up to their true potential, now they really can show what they are capable of!!!
For those who are familiar with generating ISO-files from SACD and “perfect” PS3-rips (it is indeed not that easy and not that long possible at all), and know not only how to rip but play them  p r o p e r l y, ISO-files might indeed be no “new sensation”.But I wonder if all those who are that much talking about are personally capable of at all, and I wonder of the amount of people on the globe who have these SACDs, the corresponding programs to rip them, and know how to do it properly, i.e. 100% perfect… (and not just only produce a rip like the hundreds of millions faulty ones flowing in the net containing artifical mathematical error corrections which is not music and which at least I do  n o t  want to listen to…)
Even if we take a “perfect rip”,  then we all know from tape, vinyl, CDs and SACDs how difficult – or should I say better: impossible (at least so far) it is to make ultimate and especially proper use of the corresponding infos of these specific media. We are long time researchers in trying to reproduce “that what is on CD/SACD” and not simply playing a CD/SACD. And we can assure that nobody on the globe is totally capable of – neither are we… we are struggeling to get there, and we have come closer and closer, but we are still not “there”, till this very day, but we have come closest possible now – with ISO…
Let me finish with a phrase, that I read on the “very friendly forum”, that I talked about (thanks to the one who quoted it), which really made me thoughtfully. Somebody quoted Jiddu Krishnamurti, who wrote in “Freedom from the Known”, paraphrased: “if someone says ‘this’ is ‘it’, run… “
Here I am, with that, what I have heard, and I have to say: yes, “this” is “it”. So Stefano, please, help me: Cologne Cathedral is built, and I am right in the middle… listening to the humility pipe with 16,4 cycles… I know of no better place… and I am happy… so, should I really run, go somewhere else? … and if yes, where?
Jean Hiraga, Doede Douma
Stefano, you know what it takes to bring these two guys, e s p e c i a l l y  t h e s e  t w o  g u y s, to such a deep understanding and satisfying smile –  only heavenly HiFi-delights are capable of…
Within this specific context somebody on “the very friendly tracker” asked me:
“OK “rhlauranna”, but what shall we learn from your links to Stefanos site?”
First, that I am the last one to tell “them” what “they” should learn, it’s up to them…
Let me answer the following: if someone would have told me some four or five years ago, that “my” music after 45 years in HiFi ultimately would be defined by machines and mathematical algorhithms, I would have declared him completely mad… But as as it turns out: it is exactly like that…
… and it looks like that:
PC, and MAC, and NAS, and I-Pod, and W-Lan, and cables, and… ONLY the pix is blurred, folks;-)))
Music rules…
Thanking Reinhard for his review… the links and pixes are a great support for anyone wishing to deepen the topics… and YES, sure the digital-made-right is far from being fully understood… and these guys are the daredevils who will reach the highest peak!
P.S. – some pixes are still out-of-context… will fix it soon… just wishing to share it ALL – the enthusiasm and feeling – asap.
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دو مقاله در مورد تبدیل دیجیتال به آنالوگ از Lynn Olson

پنجشنبه 20 ژوئن 2013
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http://www.positive-feedback.com/Issue65/dac.htm

http://www.positive-feedback.com/Issue66/dsd.htm

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تاثیر فیلتر دیجیتال بر پاسخ فاز خروجی آنالوگ یک سورس دیجیتال

چهار شنبه 29 دسامبر 2010
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مقاله زیر رو طراح Ayre نوشته که بنظرم خیلی خوب مطلب رو توضیح داده. قبلا هم گفتم برخی طراحان سورس دیجیتال مانند طراحان برندهای Meridian ، EMM Labs ، Spectral ، Playback Design که فیلترهای دیجیتال رو کدنویسی میکنند سعی میکنند کاری کنند پاسخ فاز بهتر بشه و سیگنال تو Time Domain وضعیت بهتری داشته باشه.

بخونید :

http://www.hifi.ir/wp-content/uploads/2010/12/Ayre_MP_White_Paper.pdf

Ayre is pleased to announce the results of an intensive research program into the audible performance of various digital filters (including “non-oversampling” filterless designs). Extensive listening tests have shown that our new custom DSP-based “Minimum Phase” filters provide a significant leap forward in both musical naturalness and ease of listening. All of our current digital products use these filters, and upgrades are also available for all Ayre disc players still in production.

The Evolution of the Digital Filter

To illustrate the performance differences between various digital filters, included are graphs of their two key parameters—their frequency response and their transient (impulse) response. These graphs allow us to trace the evolution of the digital filter,
clearly showing the changes that have led to improved sound quality.
In the beginning, there was the common linear-phase, “brickwall” digital filter. This type of digital filter is used in 99+% of all modern digital equipment, for both recording and playback, including the “Measure” position of the original Ayre disc players.
On paper, it looks nearly perfect. There is no phase shift, so it is called a “Linear Phase” filter. However, a filter can only achieve a “Linear Phase” response by introducing pre-ringing. This means that before every single musical transient, there is a “pre-echo”. In nature, there is no such thing as a “pre-echo”. All events must be “causal” in the real world—the cause must precede the effect.

A sharp, “brickwall” filter like this typically introduces about 20 cycles of pre-ringing and 20 cycles of post-ringing. It is very unnatural sounding, as the effect (pre-echo) precedes the cause (musical transient). This time-smear is interpreted by the ear-brain as both a lack of image precision in the soundstage, and also a subtle smearing of the musical sounds together.

Improving the Transient Response
The first approach to solving the problems of a conventional digital filter was to use a filter with less ringing, often known as a “slow roll-off” filter. This type of filter was used in the “Listen” position of the original Ayre disc players. By reducing the “sharpness” of the “knee” in the filter’s frequency response, the filter’s transient response is vastly improved.
Now there is only about one cycle of pre- and post-ringing. The penalty (remember, there is no such thing as a free lunch—only intelligent tradeoffs) is that there is more “leakage” (aliasing) of high frequencies above 22,050 Hz back in to the audio band. Still, this only affects very high
frequencies and the levels are low enough not to cause audible problems.
This “slow roll-off” filter reduces the time smear by a factor of ~20x compared to conventional digital filters. The net result is a much more musically natural sound, as the ear-brain is very sensitive to time-related distortions. This filter provides an outstanding compromise between frequency response and transient response, and for ten years was the mainstay of Ayre’s digital audio filters.
A natural extension to this idea is to eliminate the digital filter altogether. In theory this provides the best transient response possible from a digital playback system. There are certain audible advantages to this approach, but they are highly system-dependent and come at the expense of two separate problems.
The first is a loss of high frequencies starting at -0.75 dB at 20 kHz and reaching -3.2 dB at 20 kHz. The second is that the aliased “image” frequencies are injected into the audio signal, creating non-harmonically related distortion that increases in level as the frequency increases, reaching over 50% at 20 kHz unless additional analog filtering is employed. Careful listening tests revealed that these drawbacks outweighed the gains in improved transient response, especially given the CD format limitation with its 44.2 kHz sample rate. We therefore continued examining different approaches to digital filtering.

Eliminating the Pre-Echo
In Peter Craven’s 2004 AES paper, he proposed that the playback DAC should include a digital filter that had a corner frequency below the half-sample rate of 22,050 Hz. This would filter out any ringing (pre- and post-) that was introduced during the recording process and thereafter embedded on the disc itself. He named this an “apodizing” filter. It is a mathematical law of any filter, digital or analog, that the steeper the frequency cutoff, the more ringing it will have. In an attempt to avoid the problems of this ringing, Craven proposed using a “Minimum Phase” filter instead of the conventional “Linear Phase” filter. While this means that the phase response now varies, especially at high frequencies, there is no longer any pre-ringing. Furthermore all pre-ringing from the recording process has been filtered out, and the new playback filter only has post-ringing.
Now a giant step forward has been taken in the musical naturalness of digital audio reproduction. The unnatural pre-echoes have been completely eliminated. All of the filter’s ringing occurs after the musical transient. This is just the way that sounds occur in nature. Every sound made in the real world will have post-echoes after the original sound, so the ear-brain system more easily accepts these post-echoes from the digital filter as natural. Note that the post-ringing of a “Minimum Phase” filter is greater than that of a “Linear Phase” filter with the same
frequency response. The energy that had been contained in the pre-ringing of the “Linear Phase” filter has simply been delayed until after the transient. (Remember—there are no free lunches.) But redistributing this same total energy leads to significant gains in musical realism.
This type of digital filter is not available in off-the-shelf chips. Instead, it must be implemented in custom DSP filters. In the case of the new Ayre MP (Minimum Phase) disc players, we use sophisticated FPGAs (Field Programmable Gate Arrays) to create the desired custom filter, and the chip is easily reprogrammable should future improvements be made. This filter type is used in the “Measure” position of the new Ayre MP disc players.

The Best of Both Worlds
While the “Apodizing” filter proposed by Craven solves many of the problems with digital filters, careful listening tests conducted at Ayre showed that the multiple cycles of post-ringing still created an artificial brightness and an overall confusion to the sound. We therefore sought to combine the best aspects of Craven’s minimum-phase digital filter proposal with a slower roll-off that reduced the overall amount of ringing.
The resulting filter has no pre-echoes, and only about one cycle of post-ringing. This filter is implemented in the “Listen” position of the new Ayre MP disc players. The result is simply the most musically natural digital playback available today.
In addition to the radical improvements provided by the digital algorithms themselves, several man-months were spent conducting thorough listening tests to optimize all other aspects of the filters. The powerful customprogrammed FPGAs used in the MP series allow performance far beyond that available from off-the-shelf solutions. The mathematical calculations are conducted with 32-bit coefficients, using 64-bit accumulators to ensure the greatest degree of signal precision.
A 26x oversampling rate was determined to provide the highest level of musical realism. All of the filtering calculations are accomplished in a single pass through an FIR (Finite Impulse Response) filter. This is in contrast to conventional designs that employ a cascade of 2x FIR filters, thereby losing critical precision of the audio data as it is passed to each successive filter section. Finally, the dither algorithms for both the “Listen” and “Measure” filters were chosen on the basis of careful listening tests to provide the most realistic music reproduction possible.
The Ayre MP filter provides a significant step ahead in digital audio reproduction. The CX-7e provides this advanced technology for CDs, while the C-5xe universal disc player takes this to the limit of today’s recording technology at 292 kHz and 24 bits. In addition, this exclusive advance in digital audio reproduction is included in the Ayre QB-9 USB DAC, achieving the most natural and realistic sound available from computer-based audio systems.

http://www.stevehoffman.tv/forums/archive/index.php/t-186820.html

http://www.stereophile.com/content/meridian-8082808i2-signature-reference-cd-playerpreamplifier-measurements

http://www.stereophile.com/features/106ringing/index.html

http://hifiduino.blogspot.com/2009/05/wm8741-digital-filters.html

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