Computer based Hifi

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I agree with everything justblair said. Vista got rid of Kmixer and DirectSound and is using WASAPI instead. There are two modes for WASPI and I guess one of them is supposed to work more like ASIO where it just passes the signal along unaltered directly to the output.

But since I use ASIO I have actually had a decrease in audio quality on Vista 64. I do believe this is related to the differences of the OS itself and it's DPC Latency, which is much higher on Vista than XP when running identical hardware.

The drivers for various hardware can come into play as well making the situation worse. On Vista which I am running for my internet connection I can get interruptions from my Network Interface Card if I am trying to playback in 192k 24bit in foobar ASIO at the same time as surfing the web. XP has a lower and more stable DPC Latency so I use that for my recording and mixing.

ChrisA's advice is inline with mine. If you want good sound quality I would use something with firewire or usb that coverts outside of the computer case.

EDIT: Also I would run a search for RMAA test results on the specific cards you are thinking of buying. In my experience these measurements are a great indicator
of how clear the soundcard will perform.
 
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On Vista which I am running for my internet connection I can get interruptions from my Network Interface Card if I am trying to playback in 192k 24bit in foobar ASIO at the same time as surfing the web
Have you tried increasing the ASIO buffer size. I too noticed interference when I was browsing, I increased the buffers to 1024 from 512 and no longer have this issue. This was only happening on my laptop.

I too agree with the external device recommendation.
 
Actually the best improvement from when I first started out was updating Foobar's ASIO component. At that point it brought me to where I am now where there might be an occasional clicking when playing large bandwidth files like 192k 24-bit. I usually have the ASIO buffer set to full 2048 because of multitrack mixing demands.
 
Look for an audio interface that is marketed to recording studios not PC gammers or home theater. Most of the interfaces sold to musicians and recording engineers sound good, they have to because these buyers have well educated ears. You will find features like balanced outputs and SPDIF but most importently _analog_ volume controls. SPDIF offers another option because then you can send the optical signal to an outbound D/A converter.

They make both USB and firewire. For such a simple task as outputing two channel audio USB is "good enough" Firewire works better for 8, 16 or 24 channel work.

These brands are worth looking at, all price about about $100
E-MU, Lexicon, PreSonus, Focusrite. With these prices there is no real reason to stay with the consumer/entertainment grade stuff.
Some resellers to look at: www.sweetwater.com, samash.com

You pay a bit extra because these all have good inputs, decent preamps ad so on. But maybe some day you will want to record something or if not measure a speaker with a measurement mic. Or maybe you want to rip from vinyl? Then you will need TWO input channels for stereo. The E-MU "0202" looks to be pretty good. http://www.emu.com/products/product.asp?category=610&subcategory=611&product=15186

I have an Apple "Airport Express". It is a small little 3" box that will receive digital audio over wifi and send those signals out it built-in optical S/PDIF jack. Cost about $80. I put this into a conventional 74 WPC stereo amp and then to some vintage 1970's Infinity speakers.

x2 on pro stuff :)

Lots of deals can be had on Craigslist for these, too...with bands and small studios constantly changing gear.
 
SPDIF offers another option because then you can send the optical signal to an outbound D/A converter.

They make both USB and firewire. For such a simple task as outputing two channel audio USB is "good enough" Firewire works better for 8, 16 or 24 channel work.

ChrisA, I was looking over my mother board and didn't find a SPDIF connection and also the SB0200 card doesn't have one either....after further investigation I found the CT4780 SB Live card does have a SPDIF output. I can get these cards on the cheep. At least its a connection point for firewire. I'd rather not take up a USB port for this but I could if need be. Thanks for the tip on the SPDIF hook up....opens options.
 
Music PC

I have an ASUS P5WD2 mobo with a Realtek a/d section capable of 24/192 processing. This mobo also has a SPDIF port that I prefer to use. I feed the SPDIF signal into a Behringer DEQ 2496 and this combo sounds just great. The Behringer has been set up for room correction and it's el-cheapo DAC is actually quite good. My only issue is keeping the Dual Code processor cool without blowing up a sonic storm. My next system upgrade will be a fanless cooler.

For software, I have finally settled on Media Monkey for playback and EAC for ripping. I rip to wav and the results are generally excellent.
 
I have now read several posts in this thread and are getting a little bit confused - maybe I´m on the wrong thread. . .

I don´t follow the discussion as it turns to a discusion about how much different OS allows one to have hardware control. . . From my point of view there must a lot of other things to discuss sounds from PC before hardware cnotrol. I might be wrong about this.

For me I can e other things, like:

Different media players sound different on all my PCs. Media monkey that I like does sound awful while Foobar2000 sounds geat. Why is that and is it possible to get the Foobar sound even better?

Sound card sounds very different to eachother. I thought the X-fi cards could be good as the press had spoken so good about them. But what a joke. I bought the best one available (I really wanted the 7-channels) at the time and dumped it after i week. Instead I bought the EMU 1616M which is a 2 channel studio card - but YES hoe it sounds. Absolutely great. But is it possible to get it to sound even better. Does anyone know?
 
Better is always so subjective. Foobar 2000 is my preferred player too. I like it because it allows me to use ASIO4all (there are other reasons besides but this is about the sound). I think that with my equipment and the fact that I use 3-4 different DACs it offers the best sound and easiest control. I do not use a sound card, I use spdif or USB out to one of my DACs. I guess what I am rambling about is try an external DAC, for me it was a nice improvement. There are several nice DIY ones out there, Bantam, Gamma 1, etc.
 
Hi, and thanks for the quick reply.

ASIO4all is what? Is it used to get the signal out without any ound card, or...?

I´ve had external DACs both after and before the era of PC-sound and find most of them less good than the E-MU 1616M card. I´m at the moment connected with balanced home made 4-twinned (actually 4+1) cables to a NCD1 classD power amp. A combination that outperformes anything else that I´ve had (which includes multi$ setups). The speakers are since 5 years a couple of seriously built heavy 3-way Thiel floorstands connected with single wire cotton insulated home made 4 twinned pure copper cables (tha is outperforming lots of cables I´ve had and listened to).

Is it possilbe to get more out of Foobar with ome tweak?
 
It seems that my keyboard is running out of "S"s . . .

Billyk, "Better" to me is a clear and crisp sound with control without distortion. My whole set up is about that, which some people would say is boring. Thats why f ex the Creative X-fi didn´t aply to me as it was to much of the sounds of the old Getto balters in the 80s. Lot of sharp treble and bumpy base.

Most of the sound cards I listen to seems to loose control and adds lots of distortion to the music as soon as it gets a little bit complex signal wise (read: lots of different intruments and high noise). Is it any change to adjust f ex Foobar to be even better in that respect?
 
Sharky you have to download the ASIO component for foobar2000. The E-MU should already have it's own 1st party ASIO drivers so you wouldn't need ASIO4All unless you have a problem with EMU's drivers.

ASIO is basically a strict platform which does not change the signal (Samplerate, bit-depth) without your knowledge. For this reason (and others) it is one of the more trusted ways to interface a PC for people who are mixing or doing production. It just sends the signal straight to your hardware as opposed to anything Microsoft based which will resample on the fly - depending on which version of windows you are using and how you have your PC setup.

Using ASIO will give you slightly better quality in most cases but it will usually only give you much better quality if your previous playback was resampling on the fly to an uneven multiple of the sources samplerate mangling the files on the fly.
 
From Wikipedia:
ASIO bypasses the normal audio path from the user application through layers of intermediary Windows operating system software, so that the application connects directly to the soundcard hardware. Each layer that is bypassed means a reduction in latency, the delay between an application sending sound information and it being reproduced by the soundcard, or input signals from the soundcard being available to the application. In this way ASIO offers a relatively simple way of accessing multiple audio inputs and outputs independently. Its main strength lies in its method of bypassing the inherently high latency of Windows audio mixing kernels (KMixer), allowing direct, high speed communication with audio hardware. Unlike KMixer, an unmixed ASIO output is "bit identical", that is, the bits sent to the sound card are identical to those of the original WAV file, thus having higher audio fidelity.

It is worth a try, I found it a bit counter intuitive at first but am used to it now.

I am surprised that the DAC was not a better option, I will have to try my MAudios again, but I found that even the Bantam (PCM2702 based) sounded better. I guess YMMV hence my statement about better being subjective, it can always be better, that's why my walllet hurts so much!
 
Eh with soundcards I don't believe in subjectivity all that much. Generally the cards that sound the best will measure the best with Right Mark Audio Analyser in a loopback test. Of course this only shows you your entire DAC>ADC loopback so say if you have a problem on one side of the equation (lets say the output of the DAC only but not the ADC) then you might get artificially bad results from the test. My current card has a little bit of "ripple" in the frequency response on the output of the DAC all of the settings except 192kHz but the ADC is perfectly flat in any samplerate - I had to use multiple soundcards to deduce this problem.
 
Great posts guys ! ! ! ! !

It´s the same thinking that the EAC CD-ripper has, that it reads the CD over and over again until it gets a bit identical copy of the CD which any CD player regardless of price does.

As my music is stored both as WAV and MP3 (for Ipods etc) I guess I could get a bit identical signal path all the way from the CD (now stored in a drawer) through into Foobar2000 and further on to the E-MU 1616M hardware and it´s internal 192bit DAC?

Is that so?
 
Well the reason EAC is so paranoid is because you are doing a format conversion. And I guess there is a lot that can go wrong when converting from Red Book CD audio to the hard drive. Most of these things that can go wrong wont even effect the sound quality but since technically you will not be getting a 1:1 copy of a disc it is considered an error by those who want "perfect" copies.

If you want a bit-identical library of your CDs I would recommend ripping to a lossless format like FLAC. You'll need to download the latest FLAC.exe and place it in your EAC program folder. You then need to enter a command line into EAC so that it automatically converts and tags your CDs to FLAC. From there you can convert directly from FLAC to mp3 within foobar - which is one of the fastest and more intuitive tools to use for batch conversion.

You do NOT want to use .wav. Wav has no tags so your files will not be easy to organize or identify and you will only be wasting a lot of space ripping to .wav instead of a *lossless* compression format.
 
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Key,
As i´ve been told WAV i the most "loseless" you can get, and sort of speak the "original" format on the CD which means no convertion at all. FLAC is a convertion from one thing to another whick WAV wouldn´t be in that case. Right?


EAC makes the MP3s on the fly at the same time as the WAV files whith a Lime plug in (which is known to be the best on the net as I´ve read it).
 
Measurements are good and the way to campare devices, but what sounds best to you is still...

I agree with flac as being the way to store your files. I would like to sugest dbPoweramp as the app to use for ripping and conversions. It is by far the best tool I have ever used for ripping and converting. I have a few TB of music, mostly live and am converting to MP3 and flac from flac and SHN as well as ripping and find nothing compares for speed, accuracy and ease of use.
Forgot to mention dbPoweramp uses a huge database of hash files for comparison to insure you rip is accurate. Makes for very fats rips and only slows when the hash disagrees with the database. Pretty cool stuff.

Yes it would be a fair assumption to say that. The path through to the card using foobar and ASIO4all would be a close path, your CD playing in the PC would depend on how it was connected, to the digital spdif out or the audio out. Audio out would mean it is using the drives DAC and circuits befor the sound card got the signal and prolly the worst sounding way to play. Ripping, foobar, asio, emu is prolly the best.
 
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"Doesn´t" should be the last word in the first sentence in my last post. No CD player reads the CD with 100% accuracy only differerent levels of "almost".

Well in theory this is probably still true as soon as any audio signal hits your DAC output. Just in general I think Hard Drive based playback is more reliable than optical based. It's hard to say exactly why that is without knowing how exactly a specific hardware CD player reads a CD.

Look for a tutorial on how to setup "Secure Rips" from EAC. You will notice there are a lot of different settings within the CD-ROM that you should probably turn off. Who knows how hardware based players use some of this functionallity? There could be some sort of error corrections going on to prevent skipping that would be present in real time that wouldn't be when playing back from a hard drive utilizing ASIO. The process is just more open to viewing and controlling by the user.
 
Key,
As i´ve been told WAV i the most "loseless" you can get, and sort of speak the "original" format on the CD which means no convertion at all. FLAC is a convertion from one thing to another whick WAV wouldn´t be in that case. Right?


EAC makes the MP3s on the fly at the same time as the WAV files whith a Lime plug in (which is known to be the best on the net as I´ve read it).

There is no such thing as "more lossless" or "less lossless". Lossless is Lossless is lossless. By definition there is NO LOSS. If there was a file format that was "less lossless" then by definition it would become "lossy". You accomplish nothing by ripping to .wav except wasting twice as much hard drive space and using a file format which does not let you lable the files.

Also use windows explorer to look at a CD before it's ripped. Do you see any .wav files? Nope. Wav is not the native format of a CD. It is a container for PCM.
 
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