Will a DAC improve my stereo sound

[Jonzilla25]

Hi Guys,

A question for everyone with experience using DACs.

I use my xbox360 as my movie/music source. I am planning on getting myself a stereo setup. I have no plans of an AVR yet.

My initial plan was to connect the xbox stereo output directly to the amplifier. My question is will there be a significant difference if I connect the optical out from the xbox to a DAC and then the amp. I haven't finalised on my stereo setup yet so am wondering if I need to factor in DAC to my budget. Also I am totally unaware of the different brands and pricing on DACs.

Looking forward to your suggestions.

Thanks,

John

 

[hemantwaghe]

What music you are playing? Are these Flac or Wav Files? What dac you are planning?

 

[Jonzilla25]

Hi Hemanth,

I will be playing audio CD's and MP3's. Music will mostly be rock from the Beatles, deep purple to modern rock. Not sure what DAC to buy yet, but the question is will the DAC make any difference in a stereo setup.

Thanks

John

 

[TaureanBull]

Not always and not necessarily. It all depends on the quality of DAC. A poor quality DAC may infact deprove your sound quality.

 

[Kittu]

Follow this thread , if not helping it will confuse you more

http://www.hifivision.com/cd-players/14941-dac-buy-not-buy-confused.html

 

[Jonzilla25]

Thanks Kittu.. that thread helped a lot.. should search more net time Considering I am on a budget I'll Keep my DAC buying for later.

 

[nosh]

I recently got the MF V-DAC, the rest of the kit being NAD 355 amp and Monitor GS10 speakers.

I'm use Foobar2000 on the PC, it recognized the device as a USB Audio DAC and that's what I'm using right now.

My question is should I install and use the ASIO component in Foobar instead? Which, if any, will give better sound?

 

[reignofchaos]

ASIO makes a massive difference especially if you are on Windows XP. If you are on Vista or 7, use WASAPI.

 

[nosh]

Wow! OK.

Just out of curiosity, since the point of using a DAC is to avoid any processing at the PC end, how is using ASIO different from sending the digital data directly to the DAC using the "USB Audio DAC" option. I'm assuming there's no processing involved with this option either... by the way I _am_ on XP.

 

[no1lives4ever]

Wow! OK.

Just out of curiosity, since the point of using a DAC is to avoid any processing at the PC end, how is using ASIO different from sending the digital data directly to the DAC using the "USB Audio DAC" option. I'm assuming there's no processing involved with this option either... by the way I _am_ on XP.

Actually when you output using the normal DirectX or waveout interfaces on windows, the digital audio passes through a digital mixer inside windows. This is what allows multiple applications to play sound at the same time. This mixer device introduces latency and will introduce additional digital signal processing. This essentially results in the original digital data being distorted and that does not always sound good.

When you use ASIO or WSASAPI, the digital output from your music player directly goes to the sound device w/o any intermediate processing. This is the only way to ensure that you get bit perfect transport between your application and the sound card driver. ASIO was originally developed by steinberg to interface studio audio hardware. The main advantage of ASIO is low latency. Traditionally ASIO needs to be offered as a separete interface by the sound card drivers. Over the years, Microsoft has introduced many different technologies which allows an application to bypass the kernel mixer and directly send the output from a program to the sound card. Kernel Streaming and WASAPI both allow you to do this.

Kernel Streaming is not supported directly by most programs. WASAPI is only supported on Vista and newer versions of Windows. Under Windows XP, it is best to use ASIO. If your sound card/dac drivers do not install ASIO support, then you can use ASIO4ALL (www.asio4all.com) to support ASIO with your existing sound card drivers.

-- no1lives4ever

 

[nosh]

Thanks no1lives4ever,
I've read about and am aware of the drop in quality when XP processes sound through kmixer.

Here's the thing I was not clear about: I assumed that when Foobar uses the "USB Audio DAC" option it bypassed kmixer and any other processing. It looks like that was a false assumption on my part. I'm going to take the ASIO route now. Thanks for the replies.

 

[magma]

@ROC/no one lives forver

umm now im a bit confused too
if hes connected that DAC via usb how does ASIO drivers on foobar/his SC affect anything?

ASIO would affect only if hes connected the SPDIF/digi out of the SC to the MF dac
isint that correct?

p.S : i run Foobar on a dedicated XP based music PC with ESI juli

seems i was under the wrong impression?

 

[no1lives4ever]

@ROC/no one lives forver

umm now im a bit confused too
if hes connected that DAC via usb how does ASIO drivers on foobar/his SC affect anything?

ASIO would affect only if hes connected the SPDIF/digi out of the SC to the MF dac
isint that correct?

p.S : i run Foobar on a dedicated XP based music PC with ESI juli

seems i was under the wrong impression?

Ok.. Here is the reason phrased differently:

When you play music using the regular interfaces provided by windows, i.e. waveout & directx, windows will pass the player's digital output through a software mixer. This software mixer will apply some digital processing to the stream. Also windows sounds are still available, so if an alert box is thrown, you get to hear the alert tone while listening.

Now using ASIO, Kernel Streaming or GASAPI interfaces will allow windows to pass directly the audio stream coming out of the player to the sound card and the output device. This is irrespective of the interface used, it can be hdmi, spdif, analog, usb, etc. These technologies also reduce the latency introduced by the software mixer built into windows.

I hope this answers your queries.

-- no1lives4ever

 

[nosh]

OK. Installed ASIO. There's just one problem. The only way I have of controlling volume from the PC when ASIO playback is in progress, is to adjust the volume in Foobar.
I'm used to Volumouse which lets you change the volume just by pressing a key and rolling the mouse wheel, this new development is really inconvenient.

I've created 2 Foobar installs, one configured to output to "DS: USB Device" for casual listening from the PC and the other outputting to ASIO. There is no master volume control for the DAC available in the volume control to map to Volumouse. Is there a way around this?

Edit: Duh! Just realised I could use Foobar's global hotkeys for this!

 

[Arahman]

If your using a v-dac from your laptop then it's not running a sync and it will not be as good. I suggest getting this and it will vastly improve performance as your v-dac will then be async over USB.

Musical Fidelity V-Link Asynchronous USB to Coaxial Converter

Hope this helps, it's my first post :yahoo:

 

[no1lives4ever]

OK. Installed ASIO. There's just one problem. The only way I have of controlling volume from the PC when ASIO playback is in progress, is to adjust the volume in Foobar.
I'm used to Volumouse which lets you change the volume just by pressing a key and rolling the mouse wheel, this new development is really inconvenient.

I've created 2 Foobar installs, one configured to output to "DS: USB Device" for casual listening from the PC and the other outputting to ASIO. There is no master volume control for the DAC available in the volume control to map to Volumouse. Is there a way around this?

Edit: Duh! Just realised I could use Foobar's global hotkeys for this!

Ideally you would want to control the volume from your pre amp when using ASIO. Also set the volume to 100% in the media playing software. This will ensure that the digital signal passes from your source media to the DAC without any changes and any volume controls, etc are applied in the analog domain post the DAC.

Now if you do not have any way to control volume outside the computer, or if you prefer to use the computer for volume control, then your media playing software's volume control is an acceptable alternative, but know that this will again result in processing of some sort in the software and it will also affect the output from your DAC.

In short, when you are using volume control in the digital domain, you are truncating the dynamic output of your DAC. This will increase the noise in your output. Most modern media players and systems will not try to bump up the volume in digital domain beyond the original source level as that introduces clipping which sounds bad and can be harmful for your tweeters. You will also loose one of the benefits of using ASIO when you do this.

-- no1lives4ever

 

[nosh]

After looking into the matter a bit more, I've decided to go with Directsound.

I'll admit I don't have the most discerning ears and haven't really run the system through any proper tests but it comes down to the fact that my ears can't tell the difference. I'm not saying there isn't any. The advantages of running via Directsound are however very real for me, smooth transitions when jumping up/down the same song, crossfades between songs, normal access to volume control. I do realise that the audio during these transitions will not be optimal. I'll switch to ASIO/bitperfect for dedicated listening sessions.

Found this post on kmixer that makes my decision not seem so bad.

Bitperfect - cmediadrivers - "Bitperfect" / "bit-exact" explained

Contrary to popular belief, the kmixer of Windows 2000 and XP doesn't modify the sound and is thus bitperfect if these four conditions are met:

1. The PCM/wave volume slider of the mixer (sndvol32.exe) must be set at its maximum. Some start-up applications modify the volume slider (e.g. hardware monitoring tools from Asus).
2. The player must be compiled for the same architecture that the OS was compiled for - e.g. 32 bit player on a 32 bit OS, 64 bit player on a 64 bit OS. This is the case for the vast majority of installations because the 64 bit version of Windows XP isn't very commonly used.
3. Applications other than the player mustn't play sounds, otherwise the two output streams will be potentially sample rate converted and mixed.
4. Applications which are using the soundcard for recording have to use the same sample rate as concurrently running applications which are playing sounds - the soundcard has only one clock generator and hence this limitation arises.

Seems like the only thing one has to watch out for to avoid _constant_ processing, is to ensure that the volume is on full, as pointed out by you, no1lives4ever.


BTW, just to clear a misconception, I've noticed that this post is often linked to in audiophile forums. The quote

If there are errors on the CD, you will hear them as there is not any correction being made to overcome the error.

seems false.

It has been pointed out to me that it's the CD firmware and not the sound drivers that handle CD error corrections.

 

[no1lives4ever]

Seems like the only thing one has to watch out for to avoid _constant_ processing, is to ensure that the volume is on full, as pointed out by you, no1lives4ever.

Make sure that volume is set to full in the following 3 places:
1. Master volume control in windows mixer.
2. Volume level for the wave out channel in windows mixer
3. Volume level in your media app. Some apps like winamp will control the volume level of the wave out channel in the mixer when you change volume in the mixer, while some others may not. Under Vista and newer windows OS, there is a application specific volume level also.

BTW, just to clear a misconception, I've noticed that this post is often linked to in audiophile forums. The quote
seems false.

It has been pointed out to me that it's the CD firmware and not the sound drivers that handle CD error corrections.

Actually this is again a grey area. If you are using the CD ROM drive to directly play the audio cd, then you are mostly using the analog output from the cdrom drive. This is perhaps the worst way to play CDs on a computer.

A large number of media players will try to rip the CD in real time and then play the digital data stream using the usual output chain. Newer versions of windows also do this automatically in some cases.

Now, I have done CD ripping for a long time and over the years I have noticed that unless you have a perfect scratchless CD, mostly the ripping process is not straightforward and in a lot of instances it happens in slower than realtime speed. Because of this I personally preffer to always rip my cds using EAC, convert them to FLAC and only play the FLAC files when playing on the computer. This also helps preserve the CDs a lot better.

If you have a reasonable quality media player with optical or coaxial spdif output and a computer capable of ripping CDs, then the combo of ripped CDs in WAV or FLAC format + a media player can replace any high end CD transport in your audio rig. You may also experience better quality sound if you are using damaged cds that EAC can read after spending a lot of time on error correction.

-- no1lives4ever

 

[ravi_d]

Wow! OK.

Just out of curiosity, since the point of using a DAC is to avoid any processing at the PC end, how is using ASIO different from sending the digital data directly to the DAC using the "USB Audio DAC" option. I'm assuming there's no processing involved with this option either... by the way I _am_ on XP.

Ok.. Here is the reason phrased differently:

When you play music using the regular interfaces provided by windows, i.e. waveout & directx, windows will pass the player's digital output through a software mixer. This software mixer will apply some digital processing to the stream. Also windows sounds are still available, so if an alert box is thrown, you get to hear the alert tone while listening.

Now using ASIO, Kernel Streaming or GASAPI interfaces will allow windows to pass directly the audio stream coming out of the player to the sound card and the output device. This is irrespective of the interface used, it can be hdmi, spdif, analog, usb, etc. These technologies also reduce the latency introduced by the software mixer built into windows.

I hope this answers your queries.

-- no1lives4ever

When I bought the CA DacMagic initially I used the USB out of the laptop connected to DAC. The onboard sound card got disabled and the DAC became the default sound device bypassing the sound card Digital to Analog processing. I could hear all other pc sounds along with the audio on my hi-fi.
I had a problem connecting the DAC via USB. When ever the computer hangs for a second and recovers continuous hissing noise was induced along with the other audio and I could nothing to get rid of that noise other than to re-start the DAC.

Later I started using the optical out on my laptop built into the headphone jack. I just selected the SPDIF Interface in the foobar2k output devices. Now the computer audio is still done using the sound card but the foobar2k uses the selected SPDIF interface.
If I want to hear audio on youtube or something I just remove the optical cable from the headphone jack and use the computer speakers for that. When I plug in the optical cable in the headphone jack the laptop speakers get disabled and the foobar digital out gets enabled blocking the computer sounds.
Now I installed WASAPI plug in for foobar2k and I choose WASAPI SPDIF Interface in the foobar2k output devices.

Irrespective of the WASAPI plug-in selecting SPDIF interface in foobar2k output devices also disabled the windows volume control. The windows volume control only works for all other applications except foobar2k.

May be I'm not sure if choosing optical out for foobar2k bypasses all the other windows processing.
But its nice to have such plug-ins and solutions without using professional sound cards as for laptops there are hardly any sound cards like we have for desktops.
At one point I wanted to buy a desktop with a good ASUS/ESI sound card just for music using it with the DAC. Now even I can use a laptop connected to DAC and still get bit perfect sound from it.:clapping:

 

[nosh]

Some more observations and doubts.

[NAD 255BEE amp + Monitor Audio GS10 spks]

Cable type: I've tried all 3 types of inputs that the V-DAC allows. USB, Coaxial/RCA and Optical/TOSLINK. The S/PDIF port used is from my Gibabyte mobo. The RCA connection was distinctly inferior to the USB and I abandoned the idea immediately. I picked up a TOSLINK yesterday (5 meters - Rs.550/=) since I'd been told that optical connections are superior to USB. The music definitely sounded fuller but it didn't sound quite as clean as with the USB (And I'm using a 10 meter USB extension cord that's plugged into another 2 meter cord). With the optical connection the bass sounded slightly muddy. That crystal clear quality that tells me I'm actually listening to decent equipment was missing. I played the same song and routed the output through optical and USB and flipped the source switch on the DAC to compare. USB was definitely the winner. It does sound a bit "thin" when compared to the optical but a bit of tone control adjustment on the amp and all is really well. The tone control works way better with the USB, with the TOSLINK it hardly seems to make a difference and even running the bass control all the way down doesn't stop it from sounding muddy. Perhaps the fault lies with the inferior quality of the S/PDIF outputs on the mobo...


Voltage Stabilizer: There is a regulated 12V power supply available on Amazon that people use in lieu of the adapter (wallwart) that come with the DAC. It apparently improves the sound quality by cleaning up the current. Having no such convenient option I went for a regular Voltage stabilizer. The wallwart is still part of the system but is now plugged into the stabilizer. There's no way I can A/B power sources for a test but I think it has cleaned up the bass a little. In any case it serves the purpose of protecting all the equipment so not a purchase to be regretted. I can hear the motor run for half a sec (it's quite soft and unobtrusive in case anyone's worried about disturbance) when the AC compressor kicks in and at some other times, so there is indeed some work for it to do.

Output Bit-depth in Foobar:
What I'm about to say is based on assumption. I'm putting it out there so someone more knowledgeable can correct me if required.

If you're using a non-ASIO output Foobar gives you the option of selecting 8, 16, 24 or 32 output bit depth. I've read that if you're using USB the output is limited to 16 bits. Also when I see the properties of FLAC files the bits per sample are 16 and not higher. I'm assuming that a regular MP3s doesn't get any higher either. The DAC will upsample everything to 24 bits in any case, so even if I were using an optical output, sending 24 bits would make no sense. Foobar would upsample the signal to 24 bits instead of the DAC, which should be better at the job.

A funny scenario comes to mind. If on a USB connection I send 24 bits output. Foobar will upsample a 16 bit signal to 24. The USB 16 bit limitation (if true) will somehow clip the signal to 16 bits and the DAC will then upsample it again to 24? There's probably some faulty assumption on my part in that chain of events but it does make for an interesting scenario if true. Clarifications appreciated!