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  »  New  The commercial music servers...  Touch screen remote...  Didital Things  Forum     37  350951  01-10-2008
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  »  New  DAW drives configuration and backup strategies...  Not expensive to recover DATA, avoid Corporate Recovery...  Didital Things  Forum     3  40802  10-05-2009
  »  New  Weiss Engineering DAC202..  Attenuation...  Didital Things  Forum     5  54555  06-21-2010
  »  New  Pacific Microsonics Model Two: What Platform, Software ..  XLR to RCA adaptor. Watch out...  Didital Things  Forum     1  28650  03-17-2011
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  »  New  Memory Player Box?..  Maybe I will not order the Pure Teflon capacitors after...  Didital Things  Forum     2  48876  11-03-2011
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03-08-2017 Post does not mapped to Knowledge Tree
xandcg


Rio de Janeiro, BR.
Posts 218
Joined on 09-07-2014

Post #: 141
Post ID: 23062
Reply to: 23061
There are other reasons.
I do not use USB audio and so I cannot comment on its quality properly.

There are also others objectives going on this thread about the Amir Alix project aside of audio quality, like the usability provided by Rune Audio and/or Volumio, and the maintenance predicability of NetBSD.

Also, Amir said he is not savvy on Linux/BSD OSs and then he should find something will fit better for his comfort as OS user.



Think for yourself, do not be sheep.
03-10-2017 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 361
Joined on 02-11-2009

Post #: 142
Post ID: 23063
Reply to: 23062
Flac is not bit-perfect
designer of spectral audio and inventor of HDCD claims:
"Keith Johnson, inventor of HDCD, claims that only WAV files guarantee perfect sound reproduction which can’t be achieved with any compression, even lossless compression. He says that any file processing even in the digital domain alters it and that executing FLAC decoding affects the sound."

It seems just .Aiff , .Wav and .raw are linear PCM and no other format is bit-perfect .
https://en.m.wikipedia.org/wiki/Audio_Interchange_File_Format

Wav and aiff are the same but with minor difference in saving structure. Pcm data is the same in both aiff and wave.
http://www.hifi-advice.com/blog/audiophile-insights/digital-info/lossless-audio-flac/


www.amiraudio.com, www.hifi.ir
03-11-2017 Post does not mapped to Knowledge Tree
xandcg


Rio de Janeiro, BR.
Posts 218
Joined on 09-07-2014

Post #: 143
Post ID: 23065
Reply to: 23063
Flac is just a contender.
I am not surprised this kind of bizarre affirmation coming from a guy whom aim to live of royalties from copyrighted material.

WAV or WAVE (aka Audio for Windows) is owned by Microsoft and IBM, and AIFF is owned by Apple Inc.

I do not know the current legal state of WAV, but if you want to commercialize AIFF audio files you shall pay royalties for Apple. And you probably will only really know how it works (if it do modify or not the data) if you do some agreement with Apple.

Flac is open source (GNU GPL) a kind of license called copyleft. What basically means you pay nothing to distribute Flac material, and depending on what you (he) do code-wise with it this code also became GNU GPL automatically...

Flac is not indeed a "file format", it is just a contender. If you use compression (yes, you can do it without using compression), a md5 hash will be taken from the original data -> it will be inserted into the Flac contender. When you play it, it will be uncompressed -> a md5 hash will be taken again, but now from the uncompressed data -> the md5 hashes (of original file and the uncompressed one) will be compared -> if everything match the file will play, otherwise it is marked as corrupt.

There is no difference between the original PCM data and the uncompressed one (not even a single bit) unless it was made by some weird custom software/algorithm instead of the Flac one. It is indeed more easily to have the file damaged during a download from the internet (or while you copy it from one storage device to another) than on the Flac compress/decompress activity, because most internet downloads (and copy tools) do not hash the data and compare them later.

If you really want to make a test, get any raw PCM data (or any other data) -> create a md5/sha256/sha512 hash of it (store/write the result) -> compress to Flac (or even 7z, what is the most aggressive compression algorithm commonly available) -> decompress -> do the hash again, but now against the uncompressed data -> compare the result of both hashes.

Cheers!



Think for yourself, do not be sheep.
04-13-2017 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 361
Joined on 02-11-2009

Post #: 144
Post ID: 23138
Reply to: 23065
Flac is bit-perfect
 xandcg wrote:
I am not surprised this kind of bizarre affirmation coming from a guy whom aim to live of royalties from copyrighted material.

WAV or WAVE (aka Audio for Windows) is owned by Microsoft and IBM, and AIFF is owned by Apple Inc.

I do not know the current legal state of WAV, but if you want to commercialize AIFF audio files you shall pay royalties for Apple. And you probably will only really know how it works (if it do modify or not the data) if you do some agreement with Apple.

Flac is open source (GNU GPL) a kind of license called copyleft. What basically means you pay nothing to distribute Flac material, and depending on what you (he) do code-wise with it this code also became GNU GPL automatically...

Flac is not indeed a "file format", it is just a contender. If you use compression (yes, you can do it without using compression), a md5 hash will be taken from the original data -> it will be inserted into the Flac contender. When you play it, it will be uncompressed -> a md5 hash will be taken again, but now from the uncompressed data -> the md5 hashes (of original file and the uncompressed one) will be compared -> if everything match the file will play, otherwise it is marked as corrupt.

There is no difference between the original PCM data and the uncompressed one (not even a single bit) unless it was made by some weird custom software/algorithm instead of the Flac one. It is indeed more easily to have the file damaged during a download from the internet (or while you copy it from one storage device to another) than on the Flac compress/decompress activity, because most internet downloads (and copy tools) do not hash the data and compare them later.

If you really want to make a test, get any raw PCM data (or any other data) -> create a md5/sha256/sha512 hash of it (store/write the result) -> compress to Flac (or even 7z, what is the most aggressive compression algorithm commonly available) -> decompress -> do the hash again, but now against the uncompressed data -> compare the result of both hashes.

Cheers!


Ronald van Engelen is a linux guy who wrote a bit-perfect script file for mpd music player.

https://lacocina.nl/bitperfect-audio

Ronald van Engelen wrote:

"While flac is a bit perfect encoder/decoder for digital audio, the extra decompression that takes place when playing back the audio within a flac file, adds to the total load of the playback computer, which is something we’re trying to avoid. Therefore flac should not be considered suitable for usage in a bit perfect audio playback chain, although it is a great tool for efficient and accurate archiving and transport."





www.amiraudio.com, www.hifi.ir
04-13-2017 Post does not mapped to Knowledge Tree
xandcg


Rio de Janeiro, BR.
Posts 218
Joined on 09-07-2014

Post #: 145
Post ID: 23139
Reply to: 23138
Anything add something to the total load of the processor.
Amir,

Anything running in a computer add something to the total load of the computer. The flac decompression is negligible, unless you are using something über low powered - better read quite old. It should not add much more than 1% load to just one of the cores of my computer when playing some 192/24 files I have - what is imperceptible computer-wise.

The maximum real thing the decompression will add is very little latency. However, on the contrary of recording practice where latency is a real thing, it is irrelevant to playback (unless you have serious latency problem, what is indeed a consequence of another serious problem). Does not matter if your play back will start 3ms or 8ms after you hit the play button, what is what it means in this case.

If the script you told about is this one, it does exactly what you should never do (personal opinion) with digital audio files, what is re-sample it. If you, for some reason, want to play all given files in a specific sample rate and bit depth, and you are using Alsa, you are doing it wrong from the very beginning.

There is Jack what is a professional Unix/Linux audio backend, what play only in user selected sample rate and bit depth - and do not modify (read destroy) the source file. And I am not ever talking about the fact Jack is a proper software written in C/C++ (fast), and not a fancy bash/sh script <- Linux folks have fantasies with bash scripts.

If you take a deep look on any Linux distro you will find tons of very complex bash scripts what does things you can do with just a few lines directly calling the commands.

Basically, those scripts try to cover all imaginable scenarios before running something to make it simple to the user. However, 99% of the covered cases are just useful for advanced users what in 99% of cases will change/customize everything anyway. But the real hidden objective is to create bash scripts everywhere, just for the sake of show off their bash skills.

You do not need anything extra to play bit-perfect when using mpd, because it already do it very properly. Mpd is that kind of software: set it up and forget. It rarely breaks, and rarely change anything that needs user intervention.

Cheers!

EDIT: Just for the sake of the transparency, those are two screenshots of my desktop while running top (a small software to monitoring resource usage¹). There are also several other unrelated services running around since it is not something purpose built.

On the top left you can see the *total* CPU usage. The first one is "idle", and the second one is playing a 44/16 flac file. Those number are not static, they change at every minute, but very little if you do not do anything more.

Idle.

Playing.

¹ On the idle file is also possible to see the same but per core at he right side, on Conky behind the terminal emulator.

Before you ask about the righ RAM usage, it is FreeBSD what basically does not flush the memory until the point it need the space for anything - the real usage should be around 700MB. If you have memory you should use it, doesn't you? :-D



Think for yourself, do not be sheep.
04-13-2017 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 361
Joined on 02-11-2009

Post #: 146
Post ID: 23140
Reply to: 23139
Mpd config
I do not use flac , i just use AIFF.

My friend configured voyage mpd 0.9.5 with ronalde (http://lacocina.nl/) bit-perfect script.Alix 2d2 with voyage mpd sounds not bad before break-in and after break-in i will report about the sound.I guess even good pc with good linux could not beat CEC TL0


www.amiraudio.com, www.hifi.ir
04-13-2017 Post does not mapped to Knowledge Tree
xandcg


Rio de Janeiro, BR.
Posts 218
Joined on 09-07-2014

Post #: 147
Post ID: 23141
Reply to: 23140
The current mpd version is 0.20.6!
Amir, I did an update in the previous post.

Why using so old mpd? The current mpd version is 0.20.6!



Think for yourself, do not be sheep.
04-13-2017 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 361
Joined on 02-11-2009

Post #: 148
Post ID: 23142
Reply to: 23141
New MPD is written by C++
 xandcg wrote:
Amir, I did an update in the previous post.

Why using so old mpd? The current mpd version is 0.20.6!

New MPD is different from older versions.Some reported 0.9.5 is better sounding


www.amiraudio.com, www.hifi.ir
04-13-2017 Post does not mapped to Knowledge Tree
xandcg


Rio de Janeiro, BR.
Posts 218
Joined on 09-07-2014

Post #: 149
Post ID: 23143
Reply to: 23142
Load.
Are you running that script on the fly? If you are concerned with added load that script add in a lot more load than the flac decompression.



Think for yourself, do not be sheep.
04-13-2017 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 361
Joined on 02-11-2009

Post #: 150
Post ID: 23144
Reply to: 23143
Mpd config script
 xandcg wrote:
Are you running that script on the fly? If you are concerned with added load that script add in a lot more load than the flac decompression.

I never used it's flac upsampling script. I just used ronalde mpd config file.


www.amiraudio.com, www.hifi.ir
04-23-2017 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 361
Joined on 02-11-2009

Post #: 151
Post ID: 23180
Reply to: 23144
Is there any good digital designer?
it seems digital design is harder than analog design.

good Q&A Charles Hansen, Gordon Rankin:
http://www.audiostream.com/content/draft#Jwd0w77ZLMYORVJa.97
http://www.audiostream.com/content/theres-no-such-thing-digital-conversation-charles-hansen-gordon-rankin-and-steve-silberman-p#HKAQvHqIkeqmVDhM.97

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."







www.amiraudio.com, www.hifi.ir
04-23-2017 Post does not mapped to Knowledge Tree
rowuk


Germany
Posts 454
Joined on 07-05-2012

Post #: 152
Post ID: 23181
Reply to: 23180
More moronic blah blah?
It seems that brainlessness knows no limits. Both Mr. Hansen and Mr. Rankin speak nonsense. Wasting time on text like this is like reading the brochures from extreme audio equipment. All the things that "audiophiles" expect someone to say, get said with absolutely no indication that they understand what they are talking about.

 Amir wrote:
it seems digital design is harder than analog design.

good Q&A Charles Hansen, Gordon Rankin:
http://www.audiostream.com/content/draft#Jwd0w77ZLMYORVJa.97
http://www.audiostream.com/content/theres-no-such-thing-digital-conversation-charles-hansen-gordon-rankin-and-steve-silberman-p#HKAQvHqIkeqmVDhM.97

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."







Whenever I feel blue, I start breathing again.
04-24-2017 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 361
Joined on 02-11-2009

Post #: 153
Post ID: 23184
Reply to: 23181
Haha :-)))
 rowuk wrote:
It seems that brainlessness knows no limits. Both Mr. Hansen and Mr. Rankin speak nonsense. Wasting time on text like this is like reading the brochures from extreme audio equipment. All the things that "audiophiles" expect someone to say, get said with absolutely no indication that they understand what they are talking about.



I 100% agree there is no useful information in the text but this Q&A is telling me  "digital design is not easy".

both Gordon and Charles are good designers in comparison by others. gordon rankin was the first man who converted PC to a good transport.

I know this forum is not a place to writing and expanding moronic ideas but i should say non of audiophiles in this forum do not know "whats the secret of transport design?".
which of you knows why CEC TL0-X sounds completely different to other transports?




www.amiraudio.com, www.hifi.ir
04-27-2017 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 361
Joined on 02-11-2009

Post #: 154
Post ID: 23197
Reply to: 23184
Looking for a good external Power Supply
I am looking for a good external power supply for ATX motherboard and Sotm pci to usb card.
i need 12v, 3.3v , 5v, -12v , 9v


sometimes i think if i use CEC TL0-X power supply it may help but i did not measured it's output voltage.
CEC TL0-X has 4 transformer with 7-pin dc output.

some battery power supply are in the market like acoustic revive and red wine audio but i think battery is a little dead at lower frequency harmonics.
very few companies sell external power supply in the market.

is there any idea?



www.amiraudio.com, www.hifi.ir
04-27-2017 Post does not mapped to Knowledge Tree
martinshorn
Germany
Posts 114
Joined on 04-14-2017

Post #: 155
Post ID: 23198
Reply to: 23197
Simple and cheap
hi Amir
Ive been experimenting for long with this till i discovered it can be so easy and cheap. 
Just get a isolation transformator and an HF blocker. In this order respectively, for the pc only. Available for small money from industrial purpose, 2nd hand. 
Got plenty of DC and scam in my industry city as well as routers and chinese switching supplies at home. Running the worst and the most sensitive devices isolated like above, its 100% clean (in relevant dimensions). 
CheersJosh 
04-27-2017 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 361
Joined on 02-11-2009

Post #: 156
Post ID: 23199
Reply to: 23198
Purepower 3000 is on the way
 martinshorn wrote:
hi Amir
Ive been experimenting for long with this till i discovered it can be so easy and cheap. 
Just get a isolation transformator and an HF blocker. In this order respectively, for the pc only. Available for small money from industrial purpose, 2nd hand. 
Got plenty of DC and scam in my industry city as well as routers and chinese switching supplies at home. Running the worst and the most sensitive devices isolated like above, its 100% clean (in relevant dimensions). 
CheersJosh 


Armen my friend in iran is importing PurePower 3000. i will use PP3000 and i guess it will sound good.
about isolation i should test some stuff but now my problem is finding a good PSU for my PC .



www.amiraudio.com, www.hifi.ir
04-27-2017 Post does not mapped to Knowledge Tree
martinshorn
Germany
Posts 114
Joined on 04-14-2017

Post #: 157
Post ID: 23200
Reply to: 23199
Isolation is everything
There is no problem if you isolate it. Theyre all crap so take any, isolate the AC, go out via Toslink or Usb transformator to cut it off galvanic. 
04-10-2018 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 361
Joined on 02-11-2009

Post #: 158
Post ID: 24797
Reply to: 23200
PC Audio
I spend my time/effort/money to get a good sound from computer or MacBook.my trial and error proved me that I could not attack CEC TL0-X by Computer Audio.I also lost my money 4500$ in ordering external PSU to Paul Hynes. Paul is not trusted and do not pay him if you need external psu.
check computer audio forums like "computeraudiophile" and you will find every day news to get better sound but I think it is hype.the money you pay for PC Audio (motherboard/external clock/pcie to usb cards/cables/external psu/shield/chassis/ram/cpu/ground/network/software/dual setup/optimise codes/usb purifier/network streamer) maybe even more than a good transport.
after those trial and errors you will find CEC TL0-X sounds better than PC Audio at lower price :-))
I also paid for CAD GC3 Ground Control. GC3 is designed by Computer audio design from UK. GC3 claims to remove noise of ground line but in my system Audio Note/2way Living Voice the GC3 was not good in my system. it kills micro dynamics and I do not know why it's sound is bad.
CAD_GC3-GC1.jpg
I think Computer Audio will bother you if you think you can beat CEC TL0-X. 
my orders:macbook pro 1500$USB Cable 2000$Roon software 500$audirvana 140$Paul PSU 4500$GC3 Ground 3200$motherboard cpu ram 1500$berkeley alpha usb 2000$chassis 300$external psu 1500$optimise code and play 200$shipments 1000$




www.amiraudio.com, www.hifi.ir
05-02-2018 Post does not mapped to Knowledge Tree
Max Shatsky
Posts 19
Joined on 01-03-2005

Post #: 159
Post ID: 24828
Reply to: 24797
Toslink might help
Amir,
You can try to connect Toslink cable to your MacBook headphone jack, if your model supports this. In my case, it made a huge difference vs the USB interface. And then you can compare it to Toslink output from your CEC. Unfortunately, different software players sound different. 
Max.
05-02-2018 Post does not mapped to Knowledge Tree
Amir
Iran Tehran
Posts 361
Joined on 02-11-2009

Post #: 160
Post ID: 24829
Reply to: 24828
I dont think about it
 Max Shatsky wrote:
Amir,
You can try to connect Toslink cable to your MacBook headphone jack, if your model supports this. In my case, it made a huge difference vs the USB interface. And then you can compare it to Toslink output from your CEC. Unfortunately, different software players sound different. 
Max.

Max I have paid for all types of pc audio like dual PC setup , Single PC setup, linux on alix boards, old macbook , new macbook , optical , usb ,high speed cpu , low power cpu , jplay, windows , mac ,roon , async adaptive , all you find in computer audio have been tested.You can get decent sound from computer even better than many transports but it is impossible to attack CEC TL0-X.If you would like to have best sound i recommend you Do not waste your money and go for CEC TL0-X.CEC TL0-X is also very different to other transports and you get so much low level dynamics information from it .Don’t forget your system should be very transparent and linear bass response to compare CEC to your pc audio.
If your system is not so transparent then you could not get full potential of CEC.We could not stop computer audio and CDs will be replaced By files but i will not pay more on pc audio.


www.amiraudio.com, www.hifi.ir
Page 8 of 9 (170 items) Select Pages:  « First ... « 5 6 7 8 9 »
   Target    Threads for related reading   Most recent post in related threads   Forum  Replies   Views   Started 
  »  New  The commercial music servers...  Touch screen remote...  Didital Things  Forum     37  350951  01-10-2008
  »  New  About the DAW playback software...  Best hardware with best software...  Didital Things  Forum     11  107352  03-22-2008
  »  New  Recording/Playback software..  Free stuff...  Didital Things  Forum     14  106480  08-24-2008
  »  New  To Rip or not to Rip...  Ripping with better playback...  Didital Things  Forum     2  34803  03-18-2009
  »  New  The contra-ridicules solution for a good DAW?..  Happy to see this thread...  Didital Things  Forum     1  33581  06-18-2009
  »  New  DAW drives configuration and backup strategies...  Not expensive to recover DATA, avoid Corporate Recovery...  Didital Things  Forum     3  40802  10-05-2009
  »  New  Weiss Engineering DAC202..  Attenuation...  Didital Things  Forum     5  54555  06-21-2010
  »  New  Pacific Microsonics Model Two: What Platform, Software ..  XLR to RCA adaptor. Watch out...  Didital Things  Forum     1  28650  03-17-2011
  »  New  Windows Based Transport: A quiet and capable Source?..  DAE Firmware quality...  Didital Things  Forum     47  307094  11-01-2011
  »  New  Memory Player Box?..  Maybe I will not order the Pure Teflon capacitors after...  Didital Things  Forum     2  48876  11-03-2011
  »  New  Why I hate computer playback...  Higher power cpu...  Didital Things  Forum     17  132939  04-16-2012
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