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Hello again folks
I´d like to know if someone has ACTUALLY compared playing wav files from foobar through direct out with all windows sound drivers disabled to an USB audio dac,to the same, but out through ASIO instead of direct sound? (in foobar also)only actual comparison experiences are of interest to me right now.
thxs in advance
best
Follow Ups:
I found that K-Streaming is the only option which gives me completely clean SPDIF-out from my elderly M-audio DIO2448.Running high-freq test tones and sweeps(sines), I get occasional faint pops and clicks through ASIO and directsound, which while generally unnoticable during music, indicate something is not quite right.
Of the 2, ASIO seems slightly better, and it's the one I'd use if KS wasn't available.
Wavout, however, is unusable on my PC - crackles and pops like a worn out LP (via SPDIF).
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Setting foobar buffer to 63 has never worked to get rid of pops with ASIO. Do 2 things:1) Cont-alt-del to see all processes, then select Foobar, then set it to highest priority.
2) Download ASIOCaps utility, then set the ASIO buffer to maximum allowed by your hardware.
http://download.zol.com.cn/detail/11/104836.shtmlI can't conceive of situations where doing these 2 won't work!
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It bombs as soon as soon as I open the settings dialogue-box.
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Hi,I've tried modifying the process process before - again, no change.
I've dowloaded Asiocaps, might give it a go once I've worked out what the dialogue boxes are saying.
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In ASIO4all, try using hardware buffer instead. Set it to max. Hope it helps.
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Yes, and I suggest trying an ASIO driver designed for your sound card NOT ASIO4ALL. The point of ASIO afterall is have a direct path from your software to the sound card driver, not have software (like ASIO4ALL or Kmixer) in the middle.I have blind tested ASIO, Kernel Streaming, Directsound. My consistent preference is ASIO which has slightly fuller sound vs. KS. Directsound seems to loose some detail.
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Hi Lynn,While I have done no formal tests, I did conclude that Foobar didn't sound as good as the ACXO player I was/am using. After about a year, I realized that I had never installed ASIO in Foobar, but AXCO had it pre-loaded, and I had sellected asio...so I was listening to a player using ASIO, and one that wasn't.
That is one of the major differences, and my hunch as to why I liked AXCO much better.
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Hi,I don't know why KS vs ASIO would make a difference, or different players for that matter.
I have also blind tested Foobar ASIO vs. Mediamonkey ASIO vs. Mediacenter ASIO, and oddly I can reliably hear differences. I actually wanted one of the other players to sound best since they are easier to set up vs. Foobar, but Foobar sounds the best to me.
I realize that programmers have different choices to make in the way they process data, but in theory well written software should perform the same. I suspect the differences may be due to the way the data is handled internally in terms of word size (64 vs. 32 vs 24 vs 16 bits), how it is buffered for output (to avoid pops, clicks, and jitter), and how the program hands data off to the output device. These factors can come into play for the playback software, the hardware driver, and the operating system.
Most players were not written with audiophiles in mind. I wonder if anyone has found playback software that has made a diligent effort to provide bit perfect output? It should not be hard to do, but does require some care.
I have heard that Foobar output has been captured and compared to the input and it passes. However, when you add different Foobar plugins written in the public domain (faders, equalizers, etc) it raises a question in my mind about data integrity.
I will have to AXCO a try, but I have Foobar fine tuned with album covers and such so that I am hesitant to make a switch.
Let me know if you get a chance to try Foobar ASIO again, and tell me what you think. Keep in mind you have to install ASIO in the Foobar components directory, re-start Foobar, then turn it on in the main Output menu item, and then configure your soundcard in the ASIO plugin menu item.
Be sure to set buffer size for the ASIO plugin to at least 11 on a fairly fast PC, higher for slower PCs. The higher the buffer size the less responsive Foobar's volume and other controls will be. At a buffer size of 11 the lag is a second or so, at 64 the lag is several seconds.
I have been saying what you said about 'hidden' differences in the software, and differences in sound. However, I have been treated as nuts by those who say 'maths is maths' without recognising that the algorithm and model are important.
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I believe the discussion you might be referencing was about lossless audio which is more straightforward. It is actually very easy to prove that a lossless audio component is lossless (or not). Once this is proven, you can rely on the lossless component to send the same bit stream to the playback process as an uncompressed file. Hence, lossless is lossless.It is more difficult to determine how audio data is handled internally by playback software without the source code, and there is more room for error since the software can be more complex (especially if it offers advanced DSP functions, replay gain, etc.).
If the source code is not available, if you had a way to capture the output signal and compare it to the input you could determine if playback through the software was bit perfect using file compares.
'Math is Math' for both playback software and lossless compression. It is just that one easier to prove that it working properly vs. the other.
No, I was talking about exactly what I said in them post. Maths is often approximate. Programming to a set of equation also involve numerical approximations. Those familiar with engineering solutions to equations will know, particularly in terms of convergence and other criteria.
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Well, my position remains that if 2 programs are properly engineered to meet the same requirements, they will. 2 + 2 = 4 will come out the same way regardless if the program is written in C, Java, Pascal, VB or any other language assuming the programmer is competent.For lossless formats, the requirements are clear. You must be able to compress the data, then uncompress it, and the uncompressed data must exactly match the original. Easy to test to see if the programmer was able to competently meet these requirements.
For playback software, the requirements given the programmer can vary, and it is harder to test to see if the programmer has met those requirements (but not impossible).
It is not a given that approximations are required to meet the requirements of playing back music. For upsampling, yes. For simple playback of the 16 bit integer data in the bitstream, no.
Of course, if they are identically programmed with the right models. Unfortunately this does not happen and there is no transparency as to what is in the code.Different engineering programs give different results for the same set of equations depending on the numerical sches, assumptions, and convergent criteria as well as boudary conditions.
In our (older) days, we are taught to cross check and to use intuitation to see if software gives the right solutions. Nowadays sadly this does not happen with a lot of programmers.
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Just as I suspected... Which ASIO driver do you recommend?
First, the hardware driver supplied with soundcard must support ASIO.Then you must have an ASIO output plugin for your playback software. Assuming you are using foobar there is just 1 that I am aware of.
I tried to find the link for you, and strangely all of the old links to the Japanese sites that have historically been the source for the component are gone. I am reluctant to recommend a different source (virus and security issues).
Perhaps someone can provide a current link. If I find it, I will post it.
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Thanks for info. I have been searching: Maybe it is this ASIO link you were looking for:http://otachan.com/
Hello
Thxs for interesting "follow-up".
This is about the same conclusion I´m about to come to:try it
Best
Leif
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Tom and I will try the differences in Foobar this weekend.
Only way to find out
IMO
Best
Leif
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between direct sound and asio4all using a USB DAC (with drivers disables) I found that the asio4all is much more prone to pops and clicks which seem to be completely absent on the direct sound. As for differences in sound quality, nothing noticable to me.
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how about at 20% volume.....
ASIO4all seems to be more prone to pops and clicks than other ASIOs I used in the past and the sound quality may also differ between them. Anyway, until there's another option, I guess I'm limited to direct sound in my current computer/audio setup. Not bad, by any means.
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I just switched back and forth several times at about 20-25% volume and did not notice any difference. I tried this using really dynanic vocals (choral) and in some classic rock by Yes.
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Thxs interesting...may I ask in what kind of system?
and if you´right, we can assume that k-mixer doesn´t ruin things with regard to sound quality, OR?
best
Leif
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I believe k-mixer is disabled by having disabled all the drivers. This is probably the key factor.I'm using a 3GHz Intel, foobar, and Windows XP. I use this computer for work also, often using a lot of computer resources, so I'm more prone to pops and clicks than anybody having a more genuine music server.
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after having studied a lot of facts about k-mixer and the other audio features I´m afraid it´s not allowed to bypass the k-mixer.I believe all info is routed through it.this is what I´m trying to establish,but no one seem to know for a fact.just a lot of unqualified opinions!
IMO
best
Leif
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