سورس دیجیتال Transport & DAC

Bits is Bits

دوشنبه ۲ خرداد ۱۴۰۱
/ / /
Comments Closed
https://www.theabsolutesound.com/articles/bits-is-bits

it seems the computer audio problem is not bit errors and even correct bit-perfect digital data (from computer) does not give us good sound. 
the wadax new server (computer) does not use digital processing and it only change the digital wave shape.   

“The Wadax Reference Server I review in this issue raises some fascinating questions about the fundamental nature of digital audio. Unique for a server, the Wadax has three front-panel controls that allow the user to adjust the amplitude and shape of the digital waveform that represents the music. These controls don’t change the digital ones and zeros, but rather introduce an analog-like variability to the digital bitstream—a radical concept.

Digital audio was supposed to work perfectly or not at all; removing analog-like variability was its raison d’être. Yet early on in digital audio it became apparent that identical bitstreams could sound different if the digital samples were put back together with even the most miniscule timing errors—jitter. Although 30 years later this mechanism is fully understood, it came as a shock to a mindset that viewed digital-audio data as just another form of digital information that could be transmitted or copied endlessly without error. However, unlike other forms of digitally represented data, the end of a digital-audio system is an analog signal that is analyzed by our exquisitely sensitive hearing mechanism. 

Yet for all we’ve learned about digital audio, there’s much that remains a mystery. One such mystery is precisely how adjusting the waveshape’s steepness with the Wadax server’s “Speed” control changes the music’s sense of pace and rhythm. 

The analog-like variability of digital signals has long fascinated me. When I was working in a CD mastering lab in the late 1980s, one of my jobs was investigating technical problems with mastertapes that could lead to issues with replicated discs. One day I learned that a customer, a small, independent music label, was unhappy with the sound of the replicated discs we had made. I spoke with someone in the band, who described how the replicated disc sounded different from the mastertape. This was the first time a customer had complained about the sound quality of a replicated disc.

The sonic differences he described could not be the result of data errors on the disc. For starters, our QC department would have rejected any discs that had uncorrectable errors. CD error correction is extremely robust; it can completely and perfectly correct—not conceal through interpolation—up to 4000 consecutive missing or corrupted bits. Second, such errors would show up as audible glitches, not as, for example, a reduction in soundstage dimensionality.

The first thing I did was compare the data on the customer’s ¾” U-Matic CD mastertape with the data on the replicated disc, using a CD-ROM pre-mastering system. As expected, the data on the mastertape and the data on the replicated CD were identical.  

To the engineers I worked with, that was the end of the story. “Bits is bits,” they said, dismissing the musician’s claims. Because the replicated discs contained data identical to the mastertape, they reasoned, our company had done its job, and any sonic differences were figments of someone’s imagination. These guys were brilliant engineers. They had designed and built, from scratch, the two custom CD mastering machines in our factory—no mean feat. Yet, the audiophile in me was compelled to explore the question, so I cut a new glass master from the customer’s CD mastertape on our second, newly designed mastering machine and had discs replicated. This would enable me to listen to the two discs through the same CD player, something I couldn’t do with the CD mastertape and the replicated disc (the mastertape could be decoded only by a Sony PCM-1630 processor). After verifying that the second disc contained the same data as the mastertape and the first disc, I listened to both discs on my home system. The two discs did, indeed, sound different—the second disc sounded smoother and more dimensional. Without telling the customer what I heard (or about the different mastering machine), he reported that the second disc sounded like what he created in the studio. 

Now, I was really curious. I rented an analyzer that would measure the time periods of the pit and land structures on the CD. The analyzer graphically plotted the precise period of each of the nine discrete pit and land lengths that encode information. The first disc that sounded inferior had a much wider frequency distribution of the signals generated by the pits. The second, better-sounding disc, had a much narrower frequency distribution, indicating that the pit and land lengths were more precise. Moreover, looking at the raw signal from the CD player’s photodetector revealed that the pit-to-land and land-to-pit transitions were cleaner and sharper on the second disc. In essence, jitter was embedded in the disc itself in the physical pit and land structures. It wasn’t surprising that the second CD mastering machine produced less timing variation; its turntable was controlled by a vastly more sophisticated and precise rotational-servo system.

Although this exercise was illuminating, it still didn’t answer the question of how those timing variations on the disc made their way through an enormous amount of complex signal processing (the error-correction decoding alone is mind-boggling) to somehow affect the CD player’s analog output signal. 

it seems the computer audio problem is not bit errors and even correct bit-perfect digital data (from computer) does not give us good sound. 
the wadax new server (computer) does not use digital processing and it only change the digital wave shape.   

“The Wadax Reference Server I review in this issue raises some fascinating questions about the fundamental nature of digital audio. Unique for a server, the Wadax has three front-panel controls that allow the user to adjust the amplitude and shape of the digital waveform that represents the music. These controls don’t change the digital ones and zeros, but rather introduce an analog-like variability to the digital bitstream—a radical concept.

Digital audio was supposed to work perfectly or not at all; removing analog-like variability was its raison d’être. Yet early on in digital audio it became apparent that identical bitstreams could sound different if the digital samples were put back together with even the most miniscule timing errors—jitter. Although 30 years later this mechanism is fully understood, it came as a shock to a mindset that viewed digital-audio data as just another form of digital information that could be transmitted or copied endlessly without error. However, unlike other forms of digitally represented data, the end of a digital-audio system is an analog signal that is analyzed by our exquisitely sensitive hearing mechanism. 

Yet for all we’ve learned about digital audio, there’s much that remains a mystery. One such mystery is precisely how adjusting the waveshape’s steepness with the Wadax server’s “Speed” control changes the music’s sense of pace and rhythm. 

The analog-like variability of digital signals has long fascinated me. When I was working in a CD mastering lab in the late 1980s, one of my jobs was investigating technical problems with mastertapes that could lead to issues with replicated discs. One day I learned that a customer, a small, independent music label, was unhappy with the sound of the replicated discs we had made. I spoke with someone in the band, who described how the replicated disc sounded different from the mastertape. This was the first time a customer had complained about the sound quality of a replicated disc.

The sonic differences he described could not be the result of data errors on the disc. For starters, our QC department would have rejected any discs that had uncorrectable errors. CD error correction is extremely robust; it can completely and perfectly correct—not conceal through interpolation—up to 4000 consecutive missing or corrupted bits. Second, such errors would show up as audible glitches, not as, for example, a reduction in soundstage dimensionality.

The first thing I did was compare the data on the customer’s ¾” U-Matic CD mastertape with the data on the replicated disc, using a CD-ROM pre-mastering system. As expected, the data on the mastertape and the data on the replicated CD were identical.  

To the engineers I worked with, that was the end of the story. “Bits is bits,” they said, dismissing the musician’s claims. Because the replicated discs contained data identical to the mastertape, they reasoned, our company had done its job, and any sonic differences were figments of someone’s imagination. These guys were brilliant engineers. They had designed and built, from scratch, the two custom CD mastering machines in our factory—no mean feat. Yet, the audiophile in me was compelled to explore the question, so I cut a new glass master from the customer’s CD mastertape on our second, newly designed mastering machine and had discs replicated. This would enable me to listen to the two discs through the same CD player, something I couldn’t do with the CD mastertape and the replicated disc (the mastertape could be decoded only by a Sony PCM-1630 processor). After verifying that the second disc contained the same data as the mastertape and the first disc, I listened to both discs on my home system. The two discs did, indeed, sound different—the second disc sounded smoother and more dimensional. Without telling the customer what I heard (or about the different mastering machine), he reported that the second disc sounded like what he created in the studio. 

Now, I was really curious. I rented an analyzer that would measure the time periods of the pit and land structures on the CD. The analyzer graphically plotted the precise period of each of the nine discrete pit and land lengths that encode information. The first disc that sounded inferior had a much wider frequency distribution of the signals generated by the pits. The second, better-sounding disc, had a much narrower frequency distribution, indicating that the pit and land lengths were more precise. Moreover, looking at the raw signal from the CD player’s photodetector revealed that the pit-to-land and land-to-pit transitions were cleaner and sharper on the second disc. In essence, jitter was embedded in the disc itself in the physical pit and land structures. It wasn’t surprising that the second CD mastering machine produced less timing variation; its turntable was controlled by a vastly more sophisticated and precise rotational-servo system.

Although this exercise was illuminating, it still didn’t answer the question of how those timing variations on the disc made their way through an enormous amount of complex signal processing (the error-correction decoding alone is mind-boggling) to somehow affect the CD player’s analog output signal. 

it seems the computer audio problem is not bit errors and even correct bit-perfect digital data (from computer) does not give us good sound. 
the wadax new server (computer) does not use digital processing and it only change the digital wave shape.   

“The Wadax Reference Server I review in this issue raises some fascinating questions about the fundamental nature of digital audio. Unique for a server, the Wadax has three front-panel controls that allow the user to adjust the amplitude and shape of the digital waveform that represents the music. These controls don’t change the digital ones and zeros, but rather introduce an analog-like variability to the digital bitstream—a radical concept.

Digital audio was supposed to work perfectly or not at all; removing analog-like variability was its raison d’être. Yet early on in digital audio it became apparent that identical bitstreams could sound different if the digital samples were put back together with even the most miniscule timing errors—jitter. Although 30 years later this mechanism is fully understood, it came as a shock to a mindset that viewed digital-audio data as just another form of digital information that could be transmitted or copied endlessly without error. However, unlike other forms of digitally represented data, the end of a digital-audio system is an analog signal that is analyzed by our exquisitely sensitive hearing mechanism. 

Yet for all we’ve learned about digital audio, there’s much that remains a mystery. One such mystery is precisely how adjusting the waveshape’s steepness with the Wadax server’s “Speed” control changes the music’s sense of pace and rhythm. 

The analog-like variability of digital signals has long fascinated me. When I was working in a CD mastering lab in the late 1980s, one of my jobs was investigating technical problems with mastertapes that could lead to issues with replicated discs. One day I learned that a customer, a small, independent music label, was unhappy with the sound of the replicated discs we had made. I spoke with someone in the band, who described how the replicated disc sounded different from the mastertape. This was the first time a customer had complained about the sound quality of a replicated disc.

The sonic differences he described could not be the result of data errors on the disc. For starters, our QC department would have rejected any discs that had uncorrectable errors. CD error correction is extremely robust; it can completely and perfectly correct—not conceal through interpolation—up to 4000 consecutive missing or corrupted bits. Second, such errors would show up as audible glitches, not as, for example, a reduction in soundstage dimensionality.

The first thing I did was compare the data on the customer’s ¾” U-Matic CD mastertape with the data on the replicated disc, using a CD-ROM pre-mastering system. As expected, the data on the mastertape and the data on the replicated CD were identical.  

To the engineers I worked with, that was the end of the story. “Bits is bits,” they said, dismissing the musician’s claims. Because the replicated discs contained data identical to the mastertape, they reasoned, our company had done its job, and any sonic differences were figments of someone’s imagination. These guys were brilliant engineers. They had designed and built, from scratch, the two custom CD mastering machines in our factory—no mean feat. Yet, the audiophile in me was compelled to explore the question, so I cut a new glass master from the customer’s CD mastertape on our second, newly designed mastering machine and had discs replicated. This would enable me to listen to the two discs through the same CD player, something I couldn’t do with the CD mastertape and the replicated disc (the mastertape could be decoded only by a Sony PCM-1630 processor). After verifying that the second disc contained the same data as the mastertape and the first disc, I listened to both discs on my home system. The two discs did, indeed, sound different—the second disc sounded smoother and more dimensional. Without telling the customer what I heard (or about the different mastering machine), he reported that the second disc sounded like what he created in the studio. 

Now, I was really curious. I rented an analyzer that would measure the time periods of the pit and land structures on the CD. The analyzer graphically plotted the precise period of each of the nine discrete pit and land lengths that encode information. The first disc that sounded inferior had a much wider frequency distribution of the signals generated by the pits. The second, better-sounding disc, had a much narrower frequency distribution, indicating that the pit and land lengths were more precise. Moreover, looking at the raw signal from the CD player’s photodetector revealed that the pit-to-land and land-to-pit transitions were cleaner and sharper on the second disc. In essence, jitter was embedded in the disc itself in the physical pit and land structures. It wasn’t surprising that the second CD mastering machine produced less timing variation; its turntable was controlled by a vastly more sophisticated and precise rotational-servo system.

Although this exercise was illuminating, it still didn’t answer the question of how those timing variations on the disc made their way through an enormous amount of complex signal processing (the error-correction decoding alone is mind-boggling) to somehow affect the CD player’s analog output signal. 

That question remains unanswered to this day. Although our knowledge of digital audio has advanced enormously in the last 35 years, there’s still much to be discovered. The conundrum presented by the Wadax Reference Server is simply the latest example. It shows us the limits of our understanding by raising more questions than it answers. Robert Harley”

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جمع بندي من از كامپيوتر آوديو COMPUTER AUDIO

جمعه ۲۸ آذر ۱۳۹۹
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مجموعا با چالش هايي كه هست بهتره CEC TL0 3.0 رو داشته باشید . بقیه مدلهای CEC هم پیشنهاد نمیشه، فقط مدل TL0 3.0   .

ما سه چالش داريم در گرفتن صداي خوب از كامپيوتر :
اول اينكه فايل هاي دانلودي فعلا كيفيتاشون نامشخصه و تو بهترين حالت كه كسي بياد از روي CD ريپ كنه كيفيت فايل يه چيز متغيير هست بسته به خط و خش نداشتن دیسک ، سرعت خوندن ، كيفيت نرم افزار rip و CDRom اي كه اون سي دي رو ريپ ميكنه. حتی با بهترین حالت هم فایل ریپ کیفیت اش 20% کمتر از فایلی هست که خود استودیو داره. نمیدونیم چرا اما رومی میگه فایل که میره روی CD افت کیفیت داره و بعد ریپ هم طبق تجربه رومی بازم افت کیفیت داره و این موضوع به ما هم تو تست ها ثابت شده.
اين ادعا هست كه فايل ريپ شده همه اطلاعات روي سي دي رو در ممکنه در خودش نداشته باشه بخاطر عواملي چون ضعيف بودن فرمت ديتا روي سي دي و ضعف CDRom ها در تشخيص C2 Error و سالم نبودن دیسک.

تو سايت رومي (http://www.goodsoundclub.com/Forums/ShowPost.aspx?PageIndex=3&postID=25039) من نظرات دیگری ديدم . در هر حال اين حقيقت هست كه چون فرمت ساختاري اطلاعات روي Audio CD با ديتا فرق داره خوندن دقيق اون اطلاعات هم توسط Firmware هر CDRom اي با دقت بالا ممكن نيست ولی ناممکن هم نیست و شرایط خاصی میطلبه.

دومين مشكل اينه كه وقتي فايل خونده ميشه و از خروجي USB اين اطلاعات مياد بيرون كيفيت سيگنال بسته به انتخاب ما متفاوته ، مثلا نوت بوك هاي اپل خيلي بهتر از بقيه هستند و اونهمه كامپيوترهايي كه بر مبناي آئوديو هم جمع ميشوند صداهاشون متفاوته.
مجموعا من محصولات اپل خصوصا مك بوك پرو رو پيشنهاد ميكنم ولي هيچكدوم اينها براي آئوديو بهينه نشده. موزیک سرور هایی اومده مثل Weiss MAN301 که عالیه.
سوم اينكه اون دك يا مبدلي كه ديتاي USB رو به I2S يا SPDIF تبديل ميكنه هم كيفيتش مهمه و من كدهاي آسنكرون Gordon Rankin رو پيشنهاد ميدم مثل بركلي و ارت لگاتو يا دك هاي خود wavelength و یا Ayre.

ما بهترین نتیجه ای که گرفتیم این بوده که CD رو ریپ نکنیم و فایل رو از سرویس Tidal بگیریم با اشتراک ماهی 20 دلار و از نرم افزار Roon روی مک بوک استفاده کنیم بصورت Roon Ready با دک Weiss 502 Roon Ready Mode که تو اینحالت مک بوک و دک Weiss هردو به تایم کپسول اپل وصل میشه (هردو اتصال بصورت کابل شبکه و بخاطر نویز از وای فای استفاده نمیکنیم) و تایم کپسول اپل به مودم اینترنت وصل میشه.

اگر دک roon ready ندارید و مجبورید از ورودی usb استفاده کنید اينو پيشنهاد ميدم :

Macbook Pro quad core 15” Display model 2015  MJLQ2LL/A  ,  MJLT2LL/A

Wavelength Tube DAC/Weiss DAC

Skogrand USB Cable

استفاده از مبدل usb to spdif مثل Berkeley یا مبدل های شبکه به spdif پیشنهاد نمیشه.

اگر هم خيلي كم ميخواهيد هزينه كنيد يه نتورك پلير مثل Sonore microRendu بگيريد.
من خيلي خيلي خيلي وقت و انرژي گذاشتم به هيچ وجه پيشنهاد نميكنم براي كامپيوتر ائوديو خرج كنيد و راه هاي متفاوت رو امتحان كنيد. همينايي كه پيشنهاد دادم رو بگيريد.

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مونیخ 2017

چهارشنبه ۳ خرداد ۱۳۹۶
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شرکت Mojo Audio این اعتقاد رو داره که باید کواد کور کم وات سلرون باشه اونم خیلی روی این موضوع و طراحی Power Supply برای کامپیوتر کار کرده. و اینکه گفته باید نرم افزار و سیستم عامل سبک باشه.

این پسره Romaz هم تو Computeraudiophile forum هم اعتقاد داره باید فرکانس کلاک کم باشه و جریان cpu کم ولی Cashe سی پی یو زیاد باشه.

منم معتقدم اینا حرفشون درسته ولی شرط گرفتن صدای خوب با این سیستم کم وات اینه سیستم عامل و نرم افزار خیلی سبک و راحت باشه. اینجا لینوکس کمک میکنه با MPD یا Roon Bridge . با سیستم عامل سنگین و بهینه نشده نمیشه از برد و سی پی یو 5 واتی صدای خوب گرفت. سیسام پیشنهادی :

Fujitsu D3313-S5

4G RAM

JCAT Femto PCIe to USB Card

بدون هارد بدون مونیتور ، فقط و فقط یک اتصال شبکه و یک خروجی USB

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Oyaide EMI sheet

پنجشنبه ۷ اردیبهشت ۱۳۹۶
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سه شنبه رفتم از آقای امیدوار یه سری چیزای جالب گرفتم برای جذب امواج الکترومغناطیس که شرکت Oyaide ژاپن زده و خیلی هم قیمتش مناسبه. یه سری هم پد برای کنترل لرزش مکانیکال گرفتم از برند FO.Q ژاپن.

http://www.oyaide.com/ENGLISH/AUDIO/products_category/emi/pg530.html

http://foq.jp/

کلا این ژاپنی ها خوب چیزایی میزنن. حالا تست کنم ببینم چطور هستند. اولی که oyaide زده رو باید روی cpu یا مدارات های فرکانس بچسبونید تا با جذب اون تشعشع ها محیط اطراف تمیز بمونه. دومی برای کنترل لرزش برد و دیگر کامپوننت ها هست. یه سری پنبه هم گرفتم از Acoustic Revive برای زیر کانکتور برق و اینترکانکت که بامزه هست :-)))))

این پنبه ها برای جذب و دیفیوز امواج هستن.

https://www.acoustic-revive.com/english/psa100/psa100_01.html

در مورد روشهای کاهش نویز :

https://www.computeraudiophile.com/forums/topic/30376-a-novel-way-to-massively-improve-the-sq-of-the-sms-200-and-microrendu/?page=58

I have done extensive testing with battery supplies (not the LPS-1) in the past and compared against some of my lesser supplies like my iFI, HDPlex, Teradak, etc., even though my battery supplies have no leakage current, they never sounded that much better and in some cases, they sounded worse.  At least in these instances, leakage current wasn’t a big deal.

With some of my server builds in the past, I had gone to great lengths to minimize RF with such practices as using thick aluminum enclosures, using well insulated DC cabling and making sure no bare wire was exposed, using point-to-point wiring to keep wire lengths to a minimum, underclocking my CPU and RAM, independently powering my OS and storage drives, shielding my CPU, wrapping my cabling, RAM and SSDs in ERS paper, and even lining the ICs on my motherboard as well as the entire inside of my chassis with ERS paper and despite these painstaking measures, I found it astonishing that when I grounded the signal from my server with an Entreq grounding box, noise floor dramatically dropped further.

 

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Digital Signal or High frequency Analog Signal

یکشنبه ۳ اردیبهشت ۱۳۹۶
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Comments Closed

یه لینک هست لازمه بخونیدش اگر علاقه مند کامپیوتر آئودیو هستید:

http://www.audiostream.com/content/draft#q6v7bgt8IhVXd7Lh.97

http://www.audiostream.com/content/theres-no-such-thing-digital-conversation-charles-hansen-gordon-rankin-and-steve-silberman-p#wvWARBwiAseeDsXr.97

http://www.audiostream.com/content/qa-wavelength-audios-gordon-rankin#8qDow4jtOBexPvS7.97

http://m.electronicdesign.com/boards/480-mbitss-signal-integrity-becomes-issue-usb-20-designs

تو انتقال بيت پرفكت دیتای كامپيوتر مثلا کپی فایل از روی هارد به هارد یا هارد به فلش یا روی شبکه ما هرچقدر هم خطاي تايمينگ داشته باشيم اصلا مهم نيست چون تايمينگ ربطي به صحت اطلاعات انتقالي نداره اما خطاي تايمينگ تو انتقال ديتاي صوتي بخاطر پخش در لحظه Real Time كاملا روي كيفيت سيگنال صوتی اثر ميزاره. یعنی ممکنه سیگنال بیت پرفکت انتقال داده بشه اما بخاطر جیتر کیفیت صدا افت زیادی داشته باشه.

نکته مهم دیگه شکل موج سیگنال انتقالی هست که اونم کاملا روی صدا تاثیر میگذاره. به گفته طراح Ayre شما نباید فکر کنید سیگنال دیجیتال دارید منتقل میکنید باید فکر کنید سیگنال آنالوگ رو دارید با فرکانس بالا که خیلی هم حساس تره انتقال میدید.

برای همینه خیلی مقابله با نویز و جیتر هردو برای حفظ کیفیت سیگنال تو دیجیتال مهمه. مهم تر اینکه سیگنال دیجیتال هم باید خوب تولید بشه با پاورساپلای خوب.

شعار Bits are bits در مورد های فای صدق نمیکنه و در این های فای لعنتی حتی یه کابل شبکه بهتر هم میتونه کلی کیفیت رو تغییر بده.

دوستان صدای های اند یه جورایی Black Art هست چون سالها زمان میبره یکی بتونه مثل CEC یا Audio Note یه چیزی بیرون بکشه که بقیه نمیتونن. اصلا اصلا فکر نکنید طراحی سیستم صوتی به معنای های اند ساده است. برای همین پیچیدگی های طراحی دیجیتال هست که تنها ترنسپورتی که دیدیم کلا با همه ترنسپورت ها فرق داره CEC TL0-X هست.

من بعد از اینهمه سال های فای بازی معتقدم خیلی خیلی خیلی کم هستند طراحانی که بتونن صدای خوبی بگیرند. واقعا همونجور که رومی میگه 99% بازار پر شده از چیزهایی که صدای خوبی نمیدن.

 

Charlie Hansen:

All of this can be boiled down to a simple phrase. “All of the problems with digital are analog problems.”

This is the primary reason that digital audio has taken so many decades to come close to the sound of analog.

The thing to remember is that digital systems are not immune to degradation due to noise. They tend to be much more highly resistant to noise than analog systems, but noise in any system will cause performance degradations.

Gordon Rankin:

But since digital audio is a streaming system, the timing of the bits is critical. If the bit changes to the correct state but at the wrong time, this is equivalent to changing to the wrong level at the correct time.

Another area to tackle is what is referred to as signal integrity. The signal leaves a transistor or IC chip and it has to make its way across the PC board, component-by-component so that the signal is degraded as little as possible. When you are talking about what makes one transport a “good sounding” one, again we are talking about treating so-called “digital” products as very high speed analog circuits. The clock frequencies in these units are typically between 10 and 100 MHz. When considering a square wave, a convenient rule of thumb is that the bandwidth must extend in both direction (higher and lower frequencies) by a factor of at least 10x to preserve the waveform fidelity.
So designing a high performance digital circuit means that you are essentially designing high performance analog circuits that have a bandwidth extending up to at least 100 MHz, and in some cases all the way to 1 GHz. The traditional rules of PCB layout connectors, signal routing, ground planes, solder joints, PCB materials, and even PCB coatings break down at these high frequencies.

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Bit-Perfect Voyage MPD Image

پنجشنبه ۲۴ فروردین ۱۳۹۶
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تشکر بسیار از جادی عزیز که این سیستم بیت پرفکت لینوکسی رو راه انداختند. 3 جلسه 4 ساعته (مجموعا 12 ساعت) طول کشید و نوشتن مراحل کار با لینوکس برای راه اندازی برد Alix 2d2 کار راحتی نبود و بهتر دیدیم نتیجه رو که یه فایل ایمیج از لینوکس روی CF CARD هست رو براتون بزاریم.

جادی با دستور dd اومد از فلش کارت 2 گیگی سیستم عامل Voyage MPD که همه تنظیماتش کامل شده برای پخش بیت پرفکت (با فایل رونالد) یه ایمیج تهیه کرد من zip شده اش رو اینجا گذاشتم.

https://www.dropbox.com/s/c7d72vnz3m9p8sz/voyage.2017.04.11.image.zip?dl=0

بعد دانلود از زیپ دربیاریدش و اون فایل رو با دستور dd روی یه cf card بنویسید بیت به بیت. فلش کارت شما آماده بوت و کار هست.

ورژن Voyage 0.9.5 هست، ورژن MPD 0.17.6 هست ، ورژن کرنل Real Time لینوکس 3.14.12 هست. کل تنظیمات بیت پرفکت بودن کامل انجام شده و لینوکس و برنامه با هم حجمشون کمتر از 200 مگ هست.

یه تشکر دیگر هم داریم از رونالد :

https://lacocina.nl

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Alix 2d2 , Alix 1e , Sotm PCI to USB , RAVPower 12v LifePO4 Battery , Linux Voyage MPD , Macbook Pro , GMP Client , Ubuntu

جمعه ۶ اسفند ۱۳۹۵
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Special Thanks to Pascal Dornier (from PCEngines)

http://www.pcengines.ch/

اینا رسیده و هفته بعد دو تا سیستم راه اندازی میشه با کمک دوست خوبمون جادی که لینوکس رو خیلی خوب میدانند.

بعد از مک بوک من اومدم سراغ Alix و ممکنه بعدا سراغ BBB با RuneAudio لینوکس هم برم .

ببینید جمع بندی کامل من از همه مطالعاتم این شده کلا دو راه داریم.

یکی مسیر سرور کلاینت هست یعنی همین Alix هست با Linux Voyage MPD و یا BeagleBone Black با Linux RuneAudio که این حالت رو دارم تست میکنم همین هفته و نتیجه رو خبر میدم.

مسیر دیگه PC هست که حتما باید fanless باشه با هارد SSD متصل به mSATA . حالا یه سری میگن کم وات تر (زیر 10 وات) بهتره مثل Intel DN2800MTE Marshalltown یا Supermicro X11SBA-F با CPU Intel N3700 و یک سری هم میگن توان بالاتر بود مشکلی نیست مثل سری های 30 تا 80 وات Supermicro X11SSH-F Intel Xeon processor E3-1240 v5 یا Intel NUC DC53427HYE Core i5-3427U .

سیستم عامل برای PC هم عده ای معتقدند همین لینوکس Voyage MPD خیلی سبک خوبه و عده ای هم میگن ویندوز سرور 2012 و JPlay و استفاده از کد های Audiophile Optimizer .

بهتره CPU دو هسته یا 4 هسته باشه و برای CPU های کم وات و کواد کور سری های Core m3-7Y30 یا Atom یا Celeron یا Pentium N3700 یا N4200 که همه اینا زیر 7 وات هستند وجود دارند.

از کارت PCIe to USB JCAT Femto استفاده کنید. تو همه این حالات شما باید یه power Supply عالی داشته باشید یا حالا باتری عالی یا LPSU عالی و بدون اون نتیجه نداره.

http://www.thewelltemperedcomputer.com/Linux/Distro.htm

برای نصب نسخه 0.9.5 که بهترین نسخه هست :

https://www.hifi.ir/wp-content/uploads/2017/03/Voyage-MPD-0.9.x-Readme.pdf

http://zawiki.praxis-arbor.ch/doku.php/tschinz:linux_alix

http://cheap-silent-usb-linux-music-server.blogspot.com/

https://lacocina.nl/how-to-setup-a-bit-perfect-digital-audio-streaming-client-with-free-software-with-ltsp-and-mpd   (این وبلاگ برای رونالد هست)

http://acquisitionsyndrome.com/2014/07/bit-perfect-liva/

برای کانفیگ بیت پرفکت mpd باید اینجا رو دانلود کنید :

https://github.com/ronalde/mpd-configure

Hi there,

All those new (and maybe even older) to music player daemon (mpd), linux and alsa, I’ve published a free script which with a single command generates a configuration file for mpd, which turns it in to an bit perfect music streamer.

The only requirement is that you have a working linux installation with mpd installed. After opening up a terminal screen, paste the following commands to generate and display a mpd configuration. When multiple audio cards (including USB DACs) are found, the script will ask you which one you want to use:

Code:
## make the directory where you want to download the script
mkdir /tmp/mpd-configure
## change to that directory
cd /tmp/mpd-configure
## download and unpack the script and other files needed
wget http://lacocina.nl/mpd-configure -O - | tar --strip-components=1 -zxf -
## run the script (the resulting configuration file will be displayed on the screen
bash mpd-configure

The script can be used in a fully automated fashion, by setting command line parameters and/or environment variables. In the following example, the result is saved to the system wide mpd configuration file `/etc/mpd.conf`, uses the first available USB Audio Class interface, sets the `music_directory` to `/srv/media/music` and the mpd ‘home’ directory (including its ‘database’) to `/var/lib/mpd`. In the case `/etc/mpd.conf` already exists, the script will make a backup of it:

Code:
## become root if neccessary
[[ $EUID -eq 0 ]] || sudo su
## set the paths to the music and mpd data directories and run the script,
## saving the output to `/etc/mpd.conf` while creating a backup of that file
## in case it exists:
CONF_MPD_MUSICDIR="/srv/media/music" CONF_MPD_HOMEDIR="/var/lib/mpd" \
bash ./mpd-configure --limit usb --noprompts --output "/etc/mpd.conf"
## restart mpd to use the new file
systemctl restart mpd
## done (press ENTER)

Some background information:

Regards and enjoy the music,
Ronald

 

این لینک هم شاید بدرد خورد برای نصب :

http://www.computeraudiophile.com/content/533-geek-speak-how-build-beaglebone-black-mpd-music-server/

 

اینم بخشی از نظرات همین آقای Ronald :

Hi,

Based on my personal listening experience, a highly overpowered fanless system, with a high quality linear power supply, fitted with voyage mpd or another “linux audio” system sounds best, although the quality of the digital source files seems even more important and troublesome because we depend solely on the mastering, packaging and delevering effort the supplier of digital music provided us with. In computer and software design we do have choice and influence.

The difference with “linux for audio” systems and what is called “optimized for audio” Windows or Mac systems is that the latter are “best effort reverse engineering efforts” at most, while the former can, could and should be truly “designed for audio”. The suppliers of the audio optimizers claim they made hundreds or even a thousand of registry tweaks, but don’t mention the tens of thousands system properties they aren’t aware of or simply can’t access.

I’m an IT guy, and as such, don’t like perhaps and maybe when it comes to software design and implementation. On the other hand I’m a music lover, where emotion, personal interpretation and feeling are core values which I cherish and love. In these kind of discussions I get a bit annoyed by the fact that the domains of software, electronics and music get mixed up altogether.

Audio enthusiasts tend to use the same vocabulary for describing IT-systems and electronics. But they arent’t comparable. On the other hand, none of those guys and girls, who take their audio serious and are open minded at the same time, have ever looked for a “best effort reverse engineered” amplifier. When it comes to audio electronics, they go for the greatest design, coupled to a flawless, albeit affordable, implementation. The CAPS proposal is, as far as its hardware properties goes, exactly that.

The CAPS software properties on the other hand experience precisely the problem I described. “Audio Optimizers” for Windows or Mac do exactly what they preach, but we shouldn’t want that, nor do we need that. We should demand “designed for audio” software and are lucky enough to have such a system at hand, or at least something we could get to work that way. Because it is long standing, much debated, completely transparent and actively maintained, we can and should aim for perfect. Nothing less. We don’t have to reverse engineer it, modify 1.000 of its 40.000 registry settings. We can simply design it to work the way we want it to.

Good luck choosing your hardware! Run a designed for audio Linux together with mpd* —of course in bit perfect mode— and you’re ready to challenge the best analog setups, as long as your digital source files are made with love.

Regards,
Ronald

I started out with a diskless client (OS and software both run from the network) while the music resides on a NFS server. The first one was designed around an HP t5725 with an AMD Geode NX 15002 dual core) around 2005. I than stepped up to a Intel DN2800MT dual core Atom board, with which I was happy.

Until I heart my new Intel Core i5-3427U ULV NUC board, which I initially bought for playing back video (using xbmc). For playing back two channel digital audio (even at highest resolutions) it is overspecced by far, but it just sounds better.That is a result of the packing of the SPDIF frames inside USB URBs, which suffers from resources under load by design. To really fix that issue, we need a new discrete/i2s-like (open) standard.

As you can see in the changelogs of both the article and the script, that stuff is very recent. Of course Windows and Mac made progress the last years, as did Linux (and its alsa sound system, its UAC kernel modules/drivers and the music playing software mpd). But load on the system and its resources still have a negative impact on sound quality while maintaining stability and quality over longer periods of time takes up a lot of effort. So even today, those things need to be “managed” by reverse engineering on closed platforms.

Regards,
Ronald

لینک های رونالد :

https://github.com/ronalde/mpd-configure

https://lacocina.nl/bitperfect-audio

https://lacocina.nl/audiophile-mpd

https://lacocina.nl/how-to-setup-a-bit-perfect-digital-audio-streaming-client-with-free-software-with-ltsp-and-mpd

https://lacocina.nl/detect-alsa-output-capabilities

https://lacocina.nl/disable-pulseaudio

اینم یه بخش دیگه:

 

 A dedicated designed-for-audio computer (hardware part)
On the hardware side we’re looking for a fanless, diskless and headless industrial grade PC with two CPU cores and two network interfaces.Fanless: We aim to minimize noise and vibration.
Diskless: We don’t want spinning disks inside our box, because they cause noise, vibrations and power fluctations. The only disk inside the PC will be a small mSATA solid state disk, to store the OS and music playing software on. Those wille be loaded in memory at power up, after which everything is booted and executed from RAM. We will be storing user files, like audio files, settings and preferences, on a network storage device. This way, we’ll complete eliminate activity on the SATA-bus and controller.
Headless: We only want the PCIe/USB-busses and controllers be dealing with the handling of audio. So there will be now screens or input devices attached to our box and we have no need for resource hungry 2D/3D graphics systems and their complex and error prone drivers. Instead we’ll be controlling the OS and music playing software from native applications on other devices, like desktops, laptops, smartphones or tablets, or, if desired, with any webbrowser on the local network using a webserver on the music playing PC. Of course, that would require careful implementation and resource assignement, as we would want to minimize it’s influence on audio related processes.
Industrial grade We’re looking for a system that lasts at least ten years with extensive use, without active cooling and with minimum EMI/RF radiation and vibration. We want extended lifespan components from well established suppliers. We want a well build non-vented box with as few holes as possible. We only need holes for connectors, two for ethernet, two for usb and one for power.
Two CPU cores We want a dedicated core to which we will tie all processes related to audio with realtime priority. Here processess will run like audio file retrieval from the network including the network stack itself, the usb stack and the processes needed for the music playback software. All other processes, like logging, controlling and the optional webserver, will be tied to the second core.
Two network interfaces We want a single dedicated gigabit network adapter with hardware TCP/IP offloading for retrieval of audio files on the network. All other network traffic, like control sequences, will be redirected tot the second (built-in) network interface.While the C.A.P.S. proposals are great, things can be simpler, cheaper and even better.When your on a tight budget (aren’t we all?) buy yourself a fanless industrial Intel dual core Atom based based system, like the Logic Supply AG150, configured with 2GB RAM, an idustrial 32GB msSATA drive and a best-in-class Seasonic switching power supply for around $390/€260, and you have a great starting point for this purpose.More speed means less switching, so if you can afford it, you might want to spend around double that money and buy a fanless industrial Intel dual core i5 Haswell based system, like the Logic Supply ML320, configured with 4GB RAM, a 32GB internal mSATA drive and a best-in-class Seasonic switching power supply for ~$750/€560. This system features the-best-in class NUC-design, coupling the CPU directly, so without heatsinks, to the upper side of the box. The upper part of the box is a folded sandwich construction of thick aluminium and thinner iron, which is great because it not only keeps the cores cool but minimizes RF/EMI radiation and vibration as well.You might improve on the rather good basics by replacing the switched mode adapter with a proper linear audio supply. I still haven’t come around to listen to the effect of such an upgrade, and I’am currently working with a local engineer to get such a beast built.The use of a dedicated USB PCIe card designed for audio in the PC, like the ones offered by Sotm (~$300/€350) or Paul Pang / PPA Studio (~$130) (who also offers great audio PC’s and other tweaks) did do some good in the Atom based system, but did not have any audible effect in my Core i5 system. This probably is due to the fact that I didn’t use an external linear power supply to feed the cards.

Other tweaks, like dedicated audiophile SATA-controllers and cables, do not apply to our system, as we only use our solid state mSATA disk to boot the OS and music playing software. After that everything will be executed from RAM, thereby bypassing the SATA-bus and controllers completely.
A dedicated designed-for-audio computer (software part)
As Microsoft has a long and bad track record of proprietary, hidden and non-sustainable “standards” and technology while frustrating open standards, they are not the supplier I want to attach myself to. But there are those who do and some of them have created some nice offerings, which can be divided in two categories.The first type consists of stuff that’s meant to be used like a desktop, connected to a TV and input devices or touchscreen, like JRiver Media Center (~$50/€40) and the free (as in free beer) closed source and proprietary Foobar. Of course that price is without a valid Windows (desktop) license, which those users –knowingly of course– bought as part of an OEM-installation for about $100/€100. For reasons described in this article, I don’t like all-in-one solutions like these and I’m not interested in them.The headless ones (based on Windows Server) are –as designs– more to my taste, like Audiophile Optimizer (~$100/€80) and JPlay (~$130/€100). Apart from that, you will of course need to buy a proper Windows Server license, which is an art in its own, that will set you back more than $300/€300 (just an estimation).Apart from the price, the Windows based “products” all suffer from two intrinsic problems. The first one is that Windows seized supporting USB Audio after Class 1 was defined, back in 2006. A a result, there’s no native UAC2 support in Windows, which means you have to revert to third party (and closed source) drivers, which is something I’m surely not after. The other problem is that they can only go forward by going backwards, ie. by reverse engineering. Thereby they’re battling their supplier of choice, which seems silly in my opinion. Generally these “products” consists of registry tweaks and scripts that disable standard services or tweak the system in some way. Their developers bet they can get and keep the OS, drivers and software in control that way, and hopefully the 25.000 remaning settings, proprietary drivers and their updates don’t interfere with their plans. The same applies to Apple, although the underlying OS does offer more possibilities.

On the other hand, using free and open source software one can design and build a custom dedicated OS with playback software for a single purpose; getting the AES/EBU signal from the files on the network to your external UAC2 DAC in the best possible way.

Some of my fellow enthusiasts have created some great things based on free and open software. AudioPhile Linux is in active development and uses Arch, which is fitted with a custom realtime kernel and mpd. Voyage MPD, the first audio oriented system in a single compressed image, together with Vortexbox are geared towards small and cheap embedded DiY platforms, like Beaglebone and RaspberryPi.

Mine consists of a fully automated silent installation of a heavily customized (not reverse engineered) Debian with a custom compiled kernel based of the stock backported realtime kernel. It uses stock mpd and alsa modules and libraries and achieves great results.

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Lithium LifePO4 Battery 12.8v

دوشنبه ۲۵ بهمن ۱۳۹۵
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من کمی بازار رو گشتم تا باتری از نوع Lithium LifePO4 با همان lithium iron phosphate battery پیدا کنم و مارک RavPower پیدا شد. این نوع باتری 12.8 ولت داره و مستقیم وصل میشه به باتری ماشین و میتونه یک لحظه 600 آمپر جریان بده.

یعنی جریان دهی عالی داره در لحظه و اینو من میزنم به برد Alix2d2 ببینم چطوره. این شرکت Red wine هم از همین LifePO4 برای تغذیه سیستم هاش استفاده میکنه. برد ALix رسیده تهران اما هنوز دستم نرسیده. اون کارت PCI to USB شرکت SOTM اما رسیده دستم. اگر بیاد تو اسفند یه تست از ALIX MPD میگیریم.

https://en.wikipedia.org/wiki/Lithium_iron_phosphate_battery

صفحه نخست

 

 

 

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PC Engines ALIX 2d2 system x86 boards

دوشنبه ۱۸ بهمن ۱۳۹۵
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مدل alix2d2 رو سفارش دادم داره میاد. شد 120 دلار با همه هزینه هاش.

https://pcengines.ch/

https://www.hifi.ir/wp-content/uploads/2017/01/alix2.pdf

اصلا این alix2d2 چی هست من توضیح میدم. ببینید برخلاف یک PC یا نوت بوک که ورودی خروجی های زیادی دارند و cpu باید به همشون سر بزنه شما با alix2d2 هیچ ورودی خروجی I/O اضافی ندارید و یه فلش کارت دارید که سیستم عامل روش هست با حجم زیر 256 مگ و یه رم دارید 256 مگ و نه هارد دیسک دارید تو سیستم نه وای فای نه بلوتوث نه PCIe نه SATA کلا هیچی ندارید. فقط یک ورودی دارید که کابل شبکه هست و یه خروجی که  USB هست همین. دیتای موزیک روی شبکه در NAS هست و کنترل alix2d2 توسط هر کامپیوتر یا موبایلی که به شبکه وصله انجام میشه. یعنی تمام بار و پروسس سیستم روی کامپیوتر یا موبایل شما هست و هیچ کاری با alix2d2 نداریم. هم alix2d2 و هم NAS و هم موبایل ما به شبکه وصل هست و از طریق موبایل ما به alix2d2 دستور میدیم فلان فایل Wave یا هر فرمتی رو از روی NAS بخونه بیاره تو حافظه RAM و شروع به Play کنه بصورت Bit Perfect . همین خیلی ساده خیلی سرراست و راحت.

The ALIX board is a completely silent and fanless single board computer that only consumes 4 watts of power. The CPU is an x86 compatible AMD Geode running at 500Mhz; no need to compile special software. 256mb of RAM allows me to buffer FLAC files %100 to RAM before playing. The device has 2 USB ports, one of which is used to feed a USB DAC. There is no VGA, mouse, keyboard, or onboard video.

Voyage Linux is a stripped down version of Debian Linux desinged to run on embedded or low power devices, such as the ALIX. It can run off of a compact flash card as small as 128MB and runs entirely in RAM. Most importantly, it keps Debian’s APT package manager; installing software such as MPD and ALSA is only one apt-get command away. On the server it is configured with no audio software mixers, and MPD is given a direct hardware address of the USB DAC thus affording bit-perfect output.

A special version of Voyage Linux has been bee released by the Voyage team and aptly named Voyage MPD. This OS is aimed at the computer audiophile who wishes to use either the Alix, Soekris, or any x86 machine to serve up using MPD.

Unique features include:

* MPD 0.16~alpha2
* latest ALSA driver that supports USB Audio Class 2
(allowing 24bit and up to 192Khz sample rates)
* 2.6.33 real-time kernel

حالا بیاد ببینم نتیجه چطوره.

 

http://db.audioasylum.com/mhtml/m.html?forum=pcaudio&n=109926&highlight=alix&search_url=%2Fcgi%2Fsearch.mpl%3Fforum%3Dpcaudio%26searchtext%3Dalix

برای نصب لینوکس روی برد ALIX :

Installation was done using a USB card reader which is really easy using the installation script provided by voyage. After installing voyage on the cf card, you can just boot the system and it will start sending DHCP requests on eth0.

Put your CF card in your Alix board and start it. The first time it takes a couple of minutes because it will connect over dhcp and generate a ssh key pair.
Afterwards find out the IP Address and connect via SSH onto the machine. The default user is root and it’s default password is voyage.

فعلا بهترین نسخه Voyage MPD برای نصب نسخه 0.9.5 هست :

https://www.hifi.ir/wp-content/uploads/2017/03/Voyage-MPD-0.9.x-Readme.pdf

http://www.thewelltemperedcomputer.com/Linux/Players/MPD/MPD.htm

https://sites.google.com/site/computeraudioorg/linux-for-audio/setting-up-alsa

https://hifiduino.wordpress.com/2014/03/27/beaglebone-black-navigating-the-audio-maze/

http://mimizukobo.sakura.ne.jp/articles/voyagempd.html

http://lacocina.nl/

http://choerbaert.org/wiki/voyage_on_alix

http://zawiki.praxis-arbor.ch/doku.php/tschinz:linux_alix

http://cheap-silent-usb-linux-music-server.blogspot.com/

یه برد دیگه هم از همین سری سفارش دادم بنام alix1e که کارت PCI میخوره و براش کارت SOTM هم گرفتم که بشه ایزوله از برد کامپیوتر خروجی USB داد.

 

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