INTERACT FORUM
Devices => Sound Cards, DAC's, Receivers, Speakers, and Headphones => Topic started by: Matt on January 12, 2010, 10:14:41 am
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Hi all,
I'm looking for a way to output 7.1 high-quality analog sound. SPDIF and HDMI aren't an option.
I have a 4-channel M-Audio card I like. M-Audio has a 10-channel card Delta 1010LT that looks alright.
But I see Creative, Asus, and others also offer 7.1 analog cards, some of which are considerably cheaper.
They all rate their S/N ratio is some make-believe way so it's hard to know what cards will really deliver clean output.
Any advice?
Thanks.
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I really like the sound from my Auzentech X-Meridian. The only complaint I have is that I can't seem to control the volume of the Digital input on it. I would like to be able to use the digital out from my cable box to provide surround sound for TV watching (my pc is my surround source).
I have also had good results with my Terratec Aureon Firewire.
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...I have also had good results with my Terratec Aureon Firewire.
I immediately thought about the excellent Terrratec Aureon 7.1 Firewire device, however it has been discontinued and driver support may not continue. Terratec released XP/Vista 32 & 64-bit drivers, but probably they don't develope new drivers anymore. AFAIK, it was also not easy to find a dealer in the US even when it was a current product. In addition, you would need to have a Firewire port. The new external model Terratec DMX6 fire USB is also very nice, but it is only 5.1. Probably they think that digital connections are what most people want for HTPC use and 5.1 is enough for analog desktop speakers.
Terratec had also 7.1 PCI cards in the Aureon series. They were very good, but they too are discontinued.
I viable option is Audiotrak Prodigy 7.1 Hifi if you have an available conventional PCI slot.
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Thanks for the feedback.
I'm considering getting a block amplifier with no volume and I'm testing the idea by just using a receiver with analog input and the volume at 100%.
I've been trying a Prodigy 7.1 soundcard (older one, not hifi) because we have one at work. It's a whole lot better than my motherboard analog outputs, which isn't much of a surprise.
But there's still some noise on the line that's audible if the amplifier is maxed, so I'd think with a power amp it'd be even more audible since there's more power. I'm not sure what to do about it. Any ideas?
Also, when rebooting the computer, the soundcard puts out a tick when it powers up that about blows me out of the room. I don't know if any cards have a soft power-up?
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I'm considering getting a block amplifier with no volume and I'm testing the idea by just using a receiver with analog input and the volume at 100%.
Due to the economies of scale you will end up spending more $ on a good analog PC card + Power amp than what you were looking at for a new Receiver at the same quality level....Then again I've been trying to justify this one...http://www.axiomaudio.com/a1400-8.html
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Matt,
I have an m-audio Delta1010LT which is a great little PCI card. Lots of inputs and outputs and no issues with sound quality.
I had no issues with it under XP Pro, but did get some dropout issues using SPD/IF and XP MCE (m-audio state they don't support MCE).
I don't currently have it installed in my Win7 box so don't know how it performs wth that.
I could probably loan it to you for a month or two if you wanted to test it out.
Richard
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The Prodigy 7.1 I'm playing with has a bit too much noise when amplifying the signal a lot. My M-Audio Audiophile card is better, but it doesn't have enough outputs.
I read some favorable benchmarks about the Asus Xonar DX and found one for $40, so I'm going to give it a shot.
I'll report back in a week or so.
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The Prodigy 7.1 I'm playing with has a bit too much noise when amplifying the signal a lot.
My biggest concern running a card directly into a power amp(s) runningat max volume is when Windows or MC have a hickup and dump noise out!
Last night I have TV playback do something odd and I got LOUD noise out all my speakers... if my amp was at max, I think I would have been scraping the speaker cones off the back wall!
Richard
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My biggest concern running a card directly into a power amp(s) runningat max volume is when Windows or MC have a hickup and dump noise out!
Last night I have TV playback do something odd and I got LOUD noise out all my speakers... if my amp was at max, I think I would have been scraping the speaker cones off the back wall!
Richard
Yes, this is a real problem.
If I put the machine to sleep or reboot it, at least with the Prodigy card, it's scary.
I'm going to try a few cards to see if any do a better job of this.
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I'm running an Asus Xonar DX at the moment, before that I was using the M-Audio Revolution. The Xonar is better than the Revo was, more detail. Although the M-Audio driver was way better. You can't even set distance in the Asus driver, although with the new improvements in MC DSP this doesn't matter much anymore. :) I'm eying the Xonar ST though, as this seems to be the best at the moment. RCA jacks for 7.1 channels. Auzentech Bravura is a new card which seems to be targeted at the Xonar ST, not many reviews yet though.
I'm driving 2 monoblocks, PassLabs Aleph5, been doing this for years and haven't had any problems with this setup.
My biggest concern running a card directly into a power amp(s) runningat max volume is when Windows or MC have a hickup and dump noise out!
Last night I have TV playback do something odd and I got LOUD noise out all my speakers... if my amp was at max, I think I would have been scraping the speaker cones off the back wall!
Richard
I always set the internal soundcard as the primary soundcard in windows. This way any systemsounds etc. isn't routed to my main speakers.
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I neat portable device that I've read good reviews on is this thing: http://www.esi-audio.com/products/gigaporthd/
No inputs unfortunately, but it gives you 4 stereo or 7.1 output.
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I wanted to follow-up about the Xonar.
The hardware itself is great. It's got a wonderful S/N ratio so there's no audible noise in a silent room with an ear right next to the speaker even with a huge gain on the amplifier. And it doesn't snap or pop too much on computer sleep, reboot, etc. which helps in an always high-volume situation. It gets power directly from the computer power supply, and maybe this helps with both of these.
However, the driver on Windows 7 is terrible. There's all sorts of cryptic processing and it's just not clear what's actually being delivered to the DAC.
It requires selecting the input channels and sample rate on the driver panel instead of letting the software using it control it. I can't figure out what happens if I set the card to 48, 96, etc. and send it 44.1 data. It's not clear if it changes its clock (like most good cards) or if it resamples everything. The driver panel keeps showing the other number, so I suppose it's resampling. But that's terrible since I want to mix 44.1 (CDs), 48 (TV, DVD), and high-res content.
And the channel routing is totally confusing. It really wants to clone channels and do other magic. So if you have 5.1 output configured correctly, then pick 7.1 output, each time you tell it to test the right speaker, the sound will come out a different speaker! It's just ridiculous.
So the search continues for good analog outputs.
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I have been using 7.1 directly to my amplifier for about 6 years now. I started with the M-Audio Revolution 7.1, upgraded to the Creative X-Fi Elite about 4 years ago and purchased the Asus Essence ST in November of 2009. The Xonar HDAV H6 add-on card is not yet available in the US, but I purchased one from someone that had the HDAV 1.3 Deluxe. I now have 7 channels with RCA jacks and 21 opamps that can be swapped out (3 per channel). I haven't had any problems with the drivers. However, they are releasing new drivers soon that offer automatic bit-matching and support for 88.2 and 176.4. If you do use the new drivers in bit-matched mode, you will not be able to use the master volume control. This is how it is on the X-Fi, too.
I am using Windows 7. In the MC Output Format DSP I have Channels set to 5.1, Surround mixing to No Mixing, Bitdepth to 24-bit, and Sample Rate to Source Sample Rate. This sends only the channels that are being played to the Asus drivers, but at 24-bits. In the Room Correction DSP I have set my distance levels for all speakers and subs. I have two subs, but they are the same distance from the listening position. If the Output Format was set to source number of channels, the soundcard's drivers always need to be changed to match the input channels.
In the Asus drivers, I currently have to manually change the bit rate to match my source in order to avoid resampling. The card itself only has a 24.576 mHz clock and I have read that it is best to let the card resample to 48, 96, or 192 kHz for the best quality. I'm still trying to decide which way I like best.
In the drivers I have input channels set to 5.1 and output channels set to 7.1. I also use the bass management in the drivers to set the crossover for the subwoofers. I currently have it set at 80 Hz. So far, all DVD's and Blu-rays I have watched have a 5.1 soundtrack. If I have something with a 7.1 soundtrack, I will change the input channels to 7.1. By leaving the input at 5.1, I can use the driver's Dolby Prologic IIx to matrix the rear surrounds for movies. I leave Dolby Prologic IIx off for music.
This soundcard really sounds great with excellent detail, dynamics, and soundstage. I like it that the audio section is powered directly from the PSU. I plan to keep this in my system for probably the next five years. I think that because each channel has its own I/V opamp and buffer opamps, the sound quality of the surround channels and subwoofer channel are outstanding. These were a bigger improvement over the previous cards than even the main channels.
The card does not support PAP for Blu-ray so if you are playing a a Blu-ray directly using commercial software, the lossless audio will be downsampled. However, I can rip the Blu-rays using Another EAC3to GUI-Plus (http://www.curtpalme.com/forum/viewtopic.php?p=198879) to rip the movie to an MKV with lossless multichannel FLAC audio. It takes about 45 minutes, but it can then be played in MC with full resolution lossless audio.
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However, the driver on Windows 7 is terrible. There's all sorts of cryptic processing and it's just not clear what's actually being delivered to the DAC.
It requires selecting the input channels and sample rate on the driver panel instead of letting the software using it control it. I can't figure out what happens if I set the card to 48, 96, etc. and send it 44.1 data. It's not clear if it changes its clock (like most good cards) or if it resamples everything. The driver panel keeps showing the other number, so I suppose it's resampling. But that's terrible since I want to mix 44.1 (CDs), 48 (TV, DVD), and high-res content.
And the channel routing is totally confusing. It really wants to clone channels and do other magic. So if you have 5.1 output configured correctly, then pick 7.1 output, each time you tell it to test the right speaker, the sound will come out a different speaker! It's just ridiculous.
All the Creative and Auzentech cards resample to 48 kHz unless you are in audio creation mode, have ticked Bit-matched in the drivers, and are using Asio. When in this mode, the master volume is disabled and you get full volume out to your receiver or amplifier. The Asus drivers will always resample to the rate that you have set. There are bit-matched drivers that are currently being worked on by Asus. You can change the sample rate on the fly in the drivers to see of you can tell an audible difference. If you can't, then just set it at the highest rate you may use and let it resample.
It does want to clone channels. That is why I use 5.1 in MC with unused channels silent and 5.1 as my input in the Asus drivers. By sending silent channels to the Asus it prevents it from filling all speakers with sound and keeps me from having to manually change the drivers all the time.
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I gave up on the buggy Xonar DX Windows 7 drivers. Lots of people have the random channel swapping bug. It cripples the card, and there was no indication at the Asus forums it would be fixed.
Today a Soundblaster X-Fi Titanium Fatal1ty (silly name) arrived. I've only had a few hours with it, but it seems like the best card so far.
It has the least noise on the line, highest available analog output volume, very little pop on boot and shutdown, and so far Windows 7 x64 drivers that work well.
One question, to mojave's point, is that it's not clear if WASAPI exclusive output in audio creation mode with bit matching enabled is resampling. I can't find anywhere the card reports its current clock rate.
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One question, to mojave's point, is that it's not clear if WASAPI exclusive output in audio creation mode with bit matching enabled is resampling. I can't find anywhere the card reports its current clock rate.
In the bottom left hand corner of Audio Creation Mode right above Settings it shows the sample rate. If you have enabled bit-matched playback in Audio Creation Mode's Settings and are using Asio, then the sample rate will change based on the Output selection settings in MC (only up to 96 kHz is supported via Asio on Creative's cards). When I use WASAPI the sample rate does not change when it is changed in MC. Either their indicator doesn't work for WASAPI or it is being set to the Master Sampling Rate used in the driver's settings and not bit-matching. I believe the latter is happening.
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In the bottom left hand corner of Audio Creation Mode right above Settings it shows the sample rate. If you have enabled bit-matched playback in Audio Creation Mode's Settings and are using Asio, then the sample rate will change based on the Output selection settings in MC (only up to 96 kHz is supported via Asio on Creative's cards). When I use WASAPI the sample rate does not change when it is changed in MC. Either their indicator doesn't work for WASAPI or it is being set to the Master Sampling Rate used in the driver's settings and not bit-matching. I believe the latter is happening.
Thanks.
I must not have everything installed, because all I get with Windows 7 x64 after installing the latest driver is a "Settings..." button on the sound page that shows a popup with a few tabs. It allows changing the audio creation mode, etc. but shows no status.
I haven't seen anything like this:
(http://blog.activeservers.com/content/binary/X-FIcreate.jpg)
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That is one thing I don't like about Creative. You can't just download the latest drivers. You also need to download all the latest software. You need to install the Creative Console Launcher 2.61.35. If you go to Creative's download area (http://support.creative.com/downloads/welcome.aspx) and select your card and operating system you should be able to download it.
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Thanks mojave. The console application shows the sample rate.
I'm disappointed to see that WASAPI exclusive doesn't change the clock, but instead resamples.
If I use ASIO instead, it changes the clock to 44.1 and works with 5.1 output. That's good.
But then the clock stays at 44.1 if I play television at 48 KHz, so it resamples that.
Argh. I just want the card to run at the same frequency as the input.
All the high-end cards from M-Audio, E-MU, etc. support this. Why not Creative or Asus?
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I wonder if an option for video playback would be useful to anyone else:
- Set sound device's master sampling rate using ASIO when starting a video
You'd have to be able to pick the ASIO device name since the ASIO name 'Creative ASIO' maps to the video playback device name 'Speakers (Creative SB X-Fi)'.
Any other ideas?
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Any other ideas?
Matt, Have you had a play with Reclock to see if it changes the sound device's master sampling rate? It may be worth having a look at the Reclock code (it's open source, published etc) as it already lets us set a big range of options including:
- Choice of Renderer (including WASAPI)
- Bit Depth (including Output = Input + 16, 24, 24 padded into 32, 32 bit integer, 32 bit float)
- Sample Rate (including Output = Input + 32, 44.1, 48, 88.2, 96, 176.4, 192khz)
- Configure Speaker Output = # of Input Channels
Many of us already use this filter to do this plus more (primarily to keep audio and video in sync, + some for PAL speedup removal) already for all our Video sources so I'd hate to see JR reinvent the wheel or add another layer. There is an active forum for changes, improvements etc, so if it does not do exactly what you want they listen to the proposals and readily include them to make it better. They added WASAPI and Configure Speaker Output after some of my enquiries, and currently they are comparing the merits of various resampling engines and algorithms which I'm sure would be right up your alley and make it better for us all!! ... http://forum.slysoft.com/forumdisplay.php?f=85
Thanks
Nathan
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Matt, Have you had a play with Reclock to see if it changes the sound device's master sampling rate? It may be worth having a look at the Reclock code (it's open source, published etc) as it already lets us set a big range of options including:
- Choice of Renderer (including WASAPI)
- Bit Depth (including Output = Input + 16, 24, 24 padded into 32, 32 bit integer, 32 bit float)
- Sample Rate (including Output = Input + 32, 44.1, 48, 88.2, 96, 176.4, 192khz)
- Configure Speaker Output = # of Input Channels
Many of us already use this filter to do this plus more (primarily to keep audio and video in sync, + some for PAL speedup removal) already for all our Video sources so I'd hate to see JR reinvent the wheel or add another layer. There is an active forum for changes, improvements etc, so if it does not do exactly what you want they listen to the proposals and readily include them to make it better. They added WASAPI and Configure Speaker Output after some of my enquiries, and currently they are comparing the merits of various resampling engines and algorithms which I'm sure would be right up your alley and make it better for us all!! ... http://forum.slysoft.com/forumdisplay.php?f=85
Thanks
Nathan
I'll take a look. Another option would be for us to be able to use our existing audio output plugins (ASIO, WASAPI, etc.) during video playback.
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I asked about using our output plugins during video playback here:
http://yabb.jriver.com/interact/index.php?topic=56125
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You can run your output to an external DAC w/ volume control; great sound when going direct from DAC > power amp.
But you guys seem to be doing multi-channel stuff so....
My biggest concern running a card directly into a power amp(s) runningat max volume is when Windows or MC have a hickup and dump noise out!
Last night I have TV playback do something odd and I got LOUD noise out all my speakers... if my amp was at max, I think I would have been scraping the speaker cones off the back wall!
Richard
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Argh. I just want the card to run at the same frequency as the input.
All the high-end cards from M-Audio, E-MU, etc. support this. Why not Creative or Asus?
Most consumer soundcards are designed to work all the time regardless of what is being played. When I use WASAPI exclusive mode in MC, pause playback, and then view a YouTube video in a browser, I get no sound. I know why I'm not getting sound, but most people don't. They just want it to play. Having auto-sample rate matching is great, but only if you are playing from one application at a time and don't need any other Windows sounds/beeps.
Here is a good quote that summarizes it fairly well:
Here is the problem the operating system is trying to solve. You have one audio device which expects one stream of audio to play. Yet, you are free to run many applications on your PC which want to play music or some sound. Outlook wants to beep on new email, and your favorite music application is playing music. Worse yet, the sampling rate is different as each app is free to pick anything from 8Khz to 192 KHz. The sound card must be programmed to one sampling rate only.
But wait, there is more! Apps also want to change volume and do so independently of other sources. Again, we have one volume control in the sound card.
To make all of this work you need:
1. A mixer to combine the outputs from all the applications into one. This is harder than it seems since you must avoid clipping. Imagine two apps playing at 100%.
2. Resampling. Since sampling rates are different, you must make the sampling rate common to all the streams before mixing.
The main problem is that developers assume they know what is best for their customers. ;) Having the bit-matched playback would still be nice for those that understand its limitations. According to this post at Guru3d (http://forums.guru3d.com/showthread.php?t=315310), Asus is working on bit-matched drivers. Asus is closed from February 13-20 for observance of Chinese New Year. Hopefully these will be released soon.
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Bit matched audio playback and windows sounds
With my below setting i play windows sounds with small speakers that are usually left off,
while at the same time i play my music's and my video's sounds with my Creative Titanium sound card.
This way i can listen to all my gloriously bit matched music on my stereo, while all other
sounds like error beeps, im pings, and youtube videos are sent to the tiny speakers.
Settings:
I use windows 7, and in windows sound settings I leave my onboard soundcard set as the default sound device.
There is a cheap set of tiny speakers plugged into it at all times.
For my J River audio, I use ASIO and bit matched playback with a Creative Titanium Fatal1ty(lame name) sound card.
I leave my subjectively nice 5.1 system plugged into it at all times.
For my J River video i set the "Audio for Video" to play through my Creative titanium card
In Arcsoft Total Media Theater, while both soundcards have speakers connected, there is an option under audio to choose from both sound cards. Here i choose the Creative titanium card. Note that this setting gets messed up if you open TMT with out both soundcard having speakers connected. This is why i always leave 2 sets of speakers connected.
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I'm bumping this because I'm curious if there have been any hardware developments in the last couple years to improve the state of 8-channel (or more) analog output from a computer.
I'm reasonable happy with my X-Fi Fata1ty, although I would give the driver a B-. The control panel is overly complicated (I feel sorry for a new user trying to get channel routing right), and it loses its settings sometimes if the machine doesn't reboot gracefully.
Does anyone have first hand experience between an X-Fi and something like the Xonar HDAV1.3 Deluxe? Has Asus gotten it together with their drivers, or are the alternate drivers really solid?
Are there any audiophile-grade USB DACs that do 8 channels? I keep asking Gordon Rankin for this, but I think audiophile business must be predominately in the 2-channel camp.
Any other hardware worth checking out?
Thanks.
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Are there any audiophile-grade USB DACs that do 8 channels? Thanks.
I'm using the 16 channel model and it works great. It uses Lynx Studio Technology’s proprietary SynchroLock™ technology to virtually eliminate USB bus jitter when slaving the Aurora to USB. It sound's fantastic (meaning I don't hear it at all) with plenty of output voltage to drive power amps directly.
http://www.lynxstudio.com/product_detail.asp?i=2 (http://www.lynxstudio.com/product_detail.asp?i=2)
http://www.lynxstudio.com/product_detail.asp?i=58 (http://www.lynxstudio.com/product_detail.asp?i=58)
(http://www.lynxstudio.com/nav/getFile.asp?t=productimage&i=25)
(http://www.lynxstudio.com/nav/getFile.asp?i=83&t=productimage)
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Echo audio's drivers get good reports from pro audio users.
http://www.echoaudio.com/
I have no experience using their products but I did some reserarch. It appears that their drivers allow the player to change the sample rate/size.
They make PCI and firewire audio interfaces.
Bill
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I still have the M-Audio Revolution 7.1, Creative X-Fi Elite, and Asus Xonar ST with H6 daughtercard. With the Unified Xonar Drivers, the Asus works very well and has better ASIO drivers than the Creative.
I also have the Steinberg MR816x (Firewire) and UR824 (USB). Both are 8 channels and sound great. The MR816x is most like a regular soundcard in that it has proper Windows Direct Sound 8-channel drivers so you can use it with games. Many pro audio devices tend to separate the channels into 4 stereo pairs when using WASAPI. The UR824 doesn't have proper Windows Direct Sound drivers, but I can now work around that with the rerouting of audio through JRiver.
I guess I have 64 channels of audio with these five 8-channel audio devices plus three 8-channel motherboard soundcards! :o
One of the Interact members has the new exaSound Audio Design e18 8-channel DAC (http://www.exasound.com/e18DAC/Overview.aspx). It has RCA outputs and is the DAC that seems best suited for home use.
Digital Audio Denmark makes the AX24 but it is very expensive.
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mojave, you're rich in the way of audio hardware!
Given your experience, what would you recommend someone buy? Is the MR816x better in some way than using a cheaper device like the Xonar, X-Fi, or M-Audio?
Is the M-Audio, X-Fi, or Xonar + H6 the best buy? Do they all suppress D/C pop on boot / resume?
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The M-Audio caused a pop to the amp. This is suppressed with the Creative and Asus soundcards.
The Creative X-Fi Elite can decode DD/DTS on its SPDIF input so you can use with an external source such as cable/satellite.
The M-Audio and Creative use 3.5 outputs, the Asus Essence ST + H6 uses RCA outputs.
The Asus had the best sound quality and best drivers if using the Unified Xonar drivers.
From just checking on the internet, it seems the H6 isn't available anymore. That means that the Asus Essence ST is the only solution still available out of the three, but it is just 2 channels. The HT Omega Claro Halo XT (http://www.htomega.com/) is a similar 8 channel soundcard that I thought about getting a couple years ago. Auzentech (http://www.auzentech.com/site/products/cards.php) also makes some nice multi-channel cards. Both the HT Omega and Auzentech cards suppress any pops on boot/resume.
To use a measurement microphone you need a preamp with phantom power. The MR816x provides 8 channels of outputs plus 8 channels of inputs with phantom power. This helps provide more value for use with REW or other software. You could have several mics setup in a room and take measurements from each mic and average the response in REW without having to move one mic to other locations. The MR816x also allows two more MR816x's to be daisy chained and it will look like one 16 or 24 channel DAC for active crossovers or multiple subwoofers. The MR816x is made in Japan by Yamaha and seems to be very high quality physically. It has excellent D/A converters and I like it much better than the internal soundcards. You can read (or reread) my little review (http://yabb.jriver.com/interact/index.php?topic=65292.0) of the MR816x for more details.
There is a slippery slope of trying to find improvements in sound quality. You can make changes to speakers, equipment, room treatments or EQ/Room Correction to improve things. They are interdependent and sometimes you need better speakers to hear the changes from new equipment or you need more (or some if you don't have any) room treatments to hear the improvements from new speakers.
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I've been using an Asus Essence ST with H6 Daughter card in our 5.1 system for about a year and a half. I have been very pleased with it. Have also been using the Unified drivers for quite awhile and they are working without problem. I am running with the original op amps but I know Mojave changed his out. On our Klipsch Heritage speakers I don't find it harsh as I think he did with the original op amps.
In case you don't go for some of the big buck DACs mentioned here you might consider trying to find a used Essence ST with H6. If you could find one, and it would probably be cheaper than a new one (when the H6 was available), you would sure get a lot of bang for the buck.
Good luck.
Rod
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mwboater do you use a preamplifier or your card is directly connected to the power amplifier? I know that many people here use this solution, and is very attracting...but still scares me to have a 300watt power amplifier and rely on digital volume control hoping that windows doesn't freeze during iron man cave scene
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mwboater do you use a preamplifier or your card is directly connected to the power amplifier? I know that many people here use this solution, and is very attracting...but still scares me to have a 300watt power amplifier and rely on digital volume control hoping that windows doesn't freeze during iron man cave scene
No preamp in the system. I have never had a problem but am a tiny bit concerned about that possibility. I'm not sure how a preamp would help, except there would be a handy volume control or power switch to shut things down.
I think in reality if windows or MC froze that any audio output would go to zero. But would it in all cases?
In our application we almost always use Theater View controlled by a Harmony Remote. We don't have a means to kill all amps, or their inputs rapidly. Coming from the industrial control world I think of a big red 'Emergency Stop' switch. Probably something that would transfer all amp inputs away from the PC.
Anybody else concerned about this potential speaker disaster?
Rod
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@8-channel analog hardware
It's too bad that Asus discontinued the H6, since that gets high marks.
Anyone else, please keep posting other hardware suggestions for analog output. It's a valuable resource for users trying to create setups like this.
Someday it might be fun to get several of the most recommended devices and do a proper comparison.
@Safety of connecting straight to a power amplifier
We added 'Volume Protection' (makes it hard to unintentionally get loud sounds) and 'Protect Mode' (a digital fuse) to support an installation like this nicely. It's how I use the program personally.
If some hardware thing froze during playback, the worst that would happen is that the audio device buffers would loop. It would be no louder than what you were already listening to.
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My computer has been my source for music since 1996 and I've always connected directly to amps. I never had a problem with any internal soundcards.
Recently I've started using external pro audio devices as the soundcard. The only issue I've had is when I put the system to sleep while playing music. When it came awake again the music tried to play but the soundcard hadn't synced yet or something. It made an irritating buzz and I had to stop and restart playback. It wasn't loud and was no cause for alarm.
I have inadvertently had the volume turned up all the way and played something, but that is user error and I fixed it by using the WASAPI loopback feature for any external sources. This way JRiver is always in control of the volume. This is very safe since the default audio device actually has no output and anything Windows tries to play or do has no affect.
I use Room Equalization Wizard and when using the pro audio device for both output and input I have had settings wrong and played a test tone or frequency sweep too loud. This was because I had the routing setup incorrectly in the drivers. One nice thing about the pro device is that is has a big volume control and an on/off button. I can quickly turn things off if I make a mistake.
In conclusion, I've never had an issue that was Windows fault (from Windows 95 to Windows 7) and, as Matt mentioned, playback/volume is extremely safe when it is within JRiver.
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Anyone else, please keep posting other hardware suggestions for analog output. It's a valuable resource for users trying to create setups like this.
Echo audio's drivers get good reports from pro audio users.
http://www.echoaudio.com/
I have no experience using their products but I did some reserarch. It appears that their drivers allow the player to change the sample rate/size.
They make PCI and firewire audio interfaces.
Bill
Matt - Have you looked into the Echo products? Specifically the Audiofire 8 (Firewire Interface) and the Layla 3G. I believe that mojave auditioned an Audiofire 8 and went with the Asus Essence ST instead.
Rod
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@8-channel analog hardware
Anyone else, please keep posting other hardware suggestions for analog output. It's a valuable resource for users trying to create setups like this.
Someday it might be fun to get several of the most recommended devices and do a proper comparison.
I would suggest M-Audio 610 very nice card.
@Safety of connecting straight to a power amplifier
We added 'Volume Protection' (makes it hard to unintentionally get loud sounds) and 'Protect Mode' (a digital fuse) to support an installation like this nicely. It's how I use the program personally.
If some hardware thing froze during playback, the worst that would happen is that the audio device buffers would loop. It would be no louder than what you were already listening to.
Good to know! You guys are great! Imo it's a very important selling point
I'm not sure how a preamp would help, except there would be a handy volume control or power switch to shut things down.
It helps simply because it would totally manage the volume control: in this scenario, software volume has to be set on maximum level.
We don't have a means to kill all amps, or their inputs rapidly. Coming from the industrial control world I think of a big red 'Emergency Stop' switch. Probably something that would transfer all amp inputs away from the PC.
We could use as "Emergency Stop" a remote controlled socket to power on/off the power amplifier like this http://www.belkin.com/IWCatProductPage.process?Product_Id=459507
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@Safety of connecting straight to a power amplifier
We added 'Volume Protection' (makes it hard to unintentionally get loud sounds) and 'Protect Mode' (a digital fuse) to support an installation like this nicely.
Volume Protection can be enabled from the little speaker icon to the left of the volume Control. I can't find anything for Protect Mode. Is that just a feature with no settings?
Rod
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We could use as "Emergency Stop" a remote controlled socket to power on/off the power amplifier like this http://www.belkin.com/IWCatProductPage.process?Product_Id=459507
I actually use a very similar 'Belkin Conserv' device. It senses when we turn the TV on (normally from the Harmony remote) and turns on our amps. Likewise when the TV goes off it shuts off the amps.
Rod
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Volume Protection can be enabled from the little speaker icon to the left of the volume Control. I can't find anything for Protect Mode. Is that just a feature with no settings?
Rod
Protect Mode was added as part of Clip Protection in the DSP.
16.0.38 (2/10/2011)
3. NEW: Added Protect Mode to audio engine that engages if a huge spike (due to a corrupt file, misbehaved DSP, etc.) comes through the audio chain and switches to silence for a few seconds (part of 'Clip protection' mode selected at bottom of DSP Studio window).
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I guess it's built in:
Audio Protect
When a large signal spike comes through the audio chain, the program will enter protect mode. Protect mode suppress the signal spike and plays silence instead. The 'internal volume' system has also been improved so that loud sounds, from a corrupt file or misbehaving DSP, can never play louder than the set volume. These changes combined make Media Center 16 better than ever when using a power amplifier or other high-end equipment.
http://www.jriver.com/v16.html
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I guess it's built in:
Audio Protect
When a large signal spike comes through the audio chain, the program will enter protect mode. Protect mode suppress the signal spike and plays silence instead. The 'internal volume' system has also been improved so that loud sounds, from a corrupt file or misbehaving DSP, can never play louder than the set volume. These changes combined make Media Center 16 better than ever when using a power amplifier or other high-end equipment.
http://www.jriver.com/v16.html
A couple weeks ago I was playing a music file mixed by an "Electronic Musician". When he was writing out his mix file he got a clipping error. Being in a hurry he just lowered the overall level until it went away. So I'm playing back this Masterpiece of his with the volume cranked due to the lowered level and then along comes the spike (I have 4kW of power amps) :o Best M-80 in the room simulation you could imagine.
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Pardon me i didn't understood...
Is M80 a speaker? Have digital fuse worked?
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M80 is a firecracker. It makes a loud noise.
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Hi !
Can someone enlighten me on this situation, please ?
So that i can convolver every sound that windows and primary games (JRiver takes cares of music and video, need solution to games) 5.1 output I'm searching on the best approach to this.
So far I've learn and used VAC and Vsthost to do it, i just have to be careful with the filters because of latency. So far I've been more or less successful, my main problem was latency but I have learned the hard way that I cannot use TTD with this solution since it will add lag because of the nature of the filter. Ok, only frequency correction for games. I'm ok with that. What I'm not ok is with the sound card selection for the task. I have a Asus xonar d2x but it has no internal routing...
I need a card with internal routing capabilities. I get that one. My problem is that i need 5.1 audio, not stereo. I see that the prodigy 7.1 hifi control panel has directwire from wdm to asio with at least 6 channels http://imageshack.us/photo/my-images/256/directwirenu5.jpg/ (http://imageshack.us/photo/my-images/256/directwirenu5.jpg/) so that would be ok for the task ?? BUT i read that windows 7 drivers are really bad... or is it just in peoples head and it is good to go ??
OR m-audio 1010 lt ? Also has internal routing but i only see "monitor L and monitor R" = stereo ? only ??
Did I manage to explain what i need ??
5.1 wdm -> internal routing -> asio in - vst host - "convolver plugin" - 5.1 asio out -> 6 analog channels -> amp ? Does this sound correct ?
so, what sound card do you guys reckon would do the best job ?
Thank you so much !
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I'm using the 16 channel model and it works great. It uses Lynx Studio Technology’s proprietary SynchroLock™ technology to virtually eliminate USB bus jitter when slaving the Aurora to USB. It sound's fantastic (meaning I don't hear it at all) with plenty of output voltage to drive power amps directly.
http://www.lynxstudio.com/product_detail.asp?i=2 (http://www.lynxstudio.com/product_detail.asp?i=2)
http://www.lynxstudio.com/product_detail.asp?i=58 (http://www.lynxstudio.com/product_detail.asp?i=58)
I couldn't tell from reading the data sheets, etc., on these devices whether they can decode DSD. Have you had any experience bit streaming MC DSD to your Aurora converter?
Also, do you happen to know what DAC chip is used?
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I need a card with internal routing capabilities. I get that one.
JRiver provides internal routing with its WASAPI loopback feature (http://yabb.jriver.com/interact/index.php?topic=70242.0). I route everything through JRiver. You can then get any 5.1 or 7.1 audio device and apply convolution to all PC sources.
It looks like this:
5.1 WDM > JRiver Loopback > JRiver's 64 bit Convolution Engine > ASIO output to audio device > amp(s)
or you could do this:
2.0 WDM > JRiver Loopback > JRSS (JRiver upmixing) to 5.1 > JRiver's 64 bit Covolution Engine > ASIO output to audio device > amp(s)
If you have a 7.1 channel audio device, but a 5.1 channel system, you can use the outputs on the soundcard for extra zones or for multiple subwoofers. JRiver is flexible enough that you can do just about anything you can dream up.
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Just to report that after allot of poking around the web for info I have bought and install the 1616m on windows 7 x64 and it works has I wanted it to. I can internally route 6 wdm channels into 6 asio IN channels.
mojave , JRiver loopback didn't work correctly here. I had allot of cracks and skips.
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I've been trying to contact many of the manufacturers listed here.
I don't think there's a great resource for picking this hardware, and I think JRiver could help a little.
If things work out, I'll try to slowly post reviews and start owner threads for various pieces of popular hardware.
If any manufacturer happens to stumble onto this thread (or if you have good contacts at a manufacturer), please contact me at matt at jriver dot com.
Thanks.
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There is also the very viable DIY high-end 8-ch DAC option by using these together. Ive just built one:
http://www.exadevices.com/Home.aspx
http://www.twistedpearaudio.com/digital/buffalo.aspx
Here are forums:
http://www.diyaudio.com/forums/twisted-pear/188941-buffalo-iii-flexibility-without-compromise.html
http://www.diyaudio.com/forums/exadevices/200285-exau2i-32bit-384khz-multi-channel-asynchronous-usb-i2s-interface.html
http://www.diyaudio.com/forums/exadevices/183374-exau2i-multi-channel-asynchronous-usb-i2s-interface.html
http://www.twistedpearaudio.com/forum/yaf_topics19_Buffalo-DAC.aspx
Have fun
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My setup:
Saffire Pro 24 (only 6 analogue outputs)
http://www.focusrite.com/products/audio_interfaces/saffire_pro_24/
(http://c0000390.cdn3.cloudfiles.rackspacecloud.com/saffire_pro_24237.jpg)
AU$330
Behringer A500 (for mains)
http://www.behringer.com/en/Products/A500.aspx
AU$330
(http://www.behringer.com/assets/A500_P0217_Reflective_web.png)
Behringer EPX2800 (for the sub)
AU$530
http://www.behringer.com/en/Products/EPX2800.aspx
(http://www.behringer.com/assets/EPX2800_P0ABA_Reflective_web.png)
All balanced interconnects, so no noise problems, and can run interconnects as long as I like. I've tried using straight power amps, but didn't like it due to the problem described with overpowering them, and also I prefer to feed the preamps with as hot a signal as possible and match levels with the knobs on the amps.
Don't bother looking too hard for sonic differences between audio interfaces, my experience is I can't tell the difference between anything modern in blind testing. I just prefer the form factor and ruggedness of the pro gear.
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I just got an Echo AudioFire 12 today, and so far I'm quite happy with it. Time will tell. Sound quality is quite good, though my only comparison is to the D/As in the 2x4 miniDSP and some Chinese stuff a while back.....I won't mention my PC's onboard sound ::).
Subjectively (and it's been a while since I heard the mini's) I'd say it's better. Build quality looks top notch, and the software is super easy to use. I had my 4-way active speakers playing music in a half hour from opening the box. Initially, it didn't sound right, almost harsh. For whatever reason it always takes me a couple minutes to adjust to new gear/speakers. That, and I was getting used to my hazy sounding mobo outs. Much more defined detail now, and a smoother presentation. That's about as far as I go with audiophile descriptions :P
I was considering higher end soundcards like the Asus ST + H6 and the Omega Claro stuff, but couldn't get past having the analog section in the PC. That, and eventually I'll need more than 8 channels. Figured I'd just bite the bullet now and hopefully not have to upgrade for a few years (yeah right!).
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Nate, that interface looks pretty awesome. And for US$600 that's $50 per output! At 24/192 that seems excellent value.
It also has a beefy DSP. Could Media Center take advantage of the DSP's included in many of the audio interfaces on the market?
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DSP, yeah........but it's basically just a digital mixer. AFAIK it can't do any onboard processing other than routing and mixing channels. Some of their other models might.
I think a computer is a much more powerful platform for DSP than an interface though.....I'm half tempted to build a PC just for audio processing! It doesn't seem that J River could really work with the onboard DSP of any of these devices, they all use different software for configuration and JR would probably have to work with the manufacturers individually as none of this stuff is open source AFAIK.
Gotta say I'm liking this thing even more today!
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Im using Xonar HDAV1.3 Deluxe connected to my Arcam AVR 280 Receiver through the DVD-A/SACD analog inputs. So far no problems! I resample everything in JRiver to 192kHz/24bit.
I use the included Xonar software for subwoofer and speaker-level settings and have no problems listening to music or watching blueray movies, its awesome soundquality! ;D ;D ;D
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I think the new Mytek's will even do multi-channel DSD.
I'm bumping this because I'm curious if there have been any hardware developments in the last couple years to improve the state of 8-channel (or more) analog output from a computer.
I'm reasonable happy with my X-Fi Fata1ty, although I would give the driver a B-. The control panel is overly complicated (I feel sorry for a new user trying to get channel routing right), and it loses its settings sometimes if the machine doesn't reboot gracefully.
Does anyone have first hand experience between an X-Fi and something like the Xonar HDAV1.3 Deluxe? Has Asus gotten it together with their drivers, or are the alternate drivers really solid?
Are there any audiophile-grade USB DACs that do 8 channels? I keep asking Gordon Rankin for this, but I think audiophile business must be predominately in the 2-channel camp.
Any other hardware worth checking out?
Thanks.
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Hi Matt,
The threads a little dated but i just read it so hope my prespective helps.
I use a M-Audio Profire 610 which has 8 analog outs and 2 digital. It also has 2 mic pramps with phantom power which is great if you want to measure your speakers with say REW. The outs are balanced and 1/4 inch phono (TRS) so you can DIY HQ cables for direct connection to a mulitchannel poweramp. This is a firewire device so you need a MOBO with firewire or card to support. You can pick these up for $200, sometimes $150 off ebay so is great value. Unit also has a master volume control.
They perform great and audio is crystal clear. The response is extremely flat (i've calibrated my card), work great with JR with ASIO drivers with fastactic DAC's and dynamic range doing 10 channels at 24/196 if required. The only downside is (IMO) M-audio DACs are a bit on the sterile side, but accurate (compared to more expensive pro audio devices which can be warmer (i.e. presonus). Unless you are doing pro audio,for a HTPC they are great.
In terms of poweramp pop on boot, get an amp with 12v dc triggers (moves the problem away from the audio card).
Thanks,
bassmann
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Did you consider the RME Fireface (http://www.rme-audio.de/products_fireface_uc.php) family, professional "sounndcards"?
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I have the Echo AudioFire 12 and really like it.
John.
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I'm using the 16 channel model and it works great. It uses Lynx Studio Technology’s proprietary SynchroLock™ technology to virtually eliminate USB bus jitter when slaving the Aurora to USB. It sound's fantastic (meaning I don't hear it at all) with plenty of output voltage to drive power amps directly.
http://www.lynxstudio.com/product_detail.asp?i=2 (http://www.lynxstudio.com/product_detail.asp?i=2)
http://www.lynxstudio.com/product_detail.asp?i=58 (http://www.lynxstudio.com/product_detail.asp?i=58)
(http://www.lynxstudio.com/nav/getFile.asp?t=productimage&i=25)
(http://www.lynxstudio.com/nav/getFile.asp?i=83&t=productimage)
Nice piece of gear. Have you compared it to the Echo AudioFire 12? just looking at the spec sheets and other than the 4 additional channels and 113db vs 117 db S/N, I'm not sure I see $2K worth of differences. Not that it isn't there, it just isn't obvious to me.
thanks,
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That echo unit looks good. I was not aware of it. I could have bought two of those as I need 16 channels for my set-up. I did buy the Aurora model with variable analog gain on all channels which is nice to match gain of different speakers.
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Nice piece of gear. Have you compared it to the Echo AudioFire 12? just looking at the spec sheets and other than the 4 additional channels and 113db vs 117 db S/N, I'm not sure I see $2K worth of differences. Not that it isn't there, it just isn't obvious to me.
thanks,
There's more to it than just the spec sheet. You could have the latest, greatest, most expensive DAC chips but if the analog implementation of the conversion is crap then it doesn't really matter. Power supplies are a biggie too. The quality/tolerance of the components used can make a huge difference as well. That extra $2k can mean a lot if it's actually done with quality in mind and not dollar signs.......
Glad you like your Echo! I've had mine for about 4 months now and it's solid. No problems whatever and the sq is excellent.
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Nate,
I wasn't trying to diss anything. I was just curious. I know there is more to it than just the spec sheet, thus why I asked the question. When I was looking at the Echo, I also considered the MOTU HD192. Both 12 ports, both 192/24 capable. I went with the Echo due to cost (ease of getting into this type setup). However, in a couple years when my Echo stops working for some reason (I bought it used), then I'll reconsider MOTU, Lynx and others (I wasn't aware of Lynx before this thread).
Thanks,
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I'm using the 16 channel model and it works great. It uses Lynx Studio Technology’s proprietary SynchroLock™ technology to virtually eliminate USB bus jitter when slaving the Aurora to USB. It sound's fantastic (meaning I don't hear it at all) with plenty of output voltage to drive power amps directly.
http://www.lynxstudio.com/product_detail.asp?i=2 (http://www.lynxstudio.com/product_detail.asp?i=2)
http://www.lynxstudio.com/product_detail.asp?i=58 (http://www.lynxstudio.com/product_detail.asp?i=58)
+1 on the Lynx. While I don't do multi-channel, I have used Lynx Aurora products for multi-channel recording in the studio and at home I use the Lynx Hilo. Lynx is one of the best sounding AD/DA's you can purchase today, but a bit pricier than most internal sound cards.
Another audiophile grade 8 channel solution is from Exasound here in Canada: http://www.exasound.com/e18DAC/e18DACOverview.aspx
Cheers,
Mitch
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I'm cross-linking in this Lynx vs Steinberg vs Creative comparison since it directly addresses the topic of this thread:
http://yabb.jriver.com/interact/index.php?topic=77508.0
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Im using Xonar HDAV1.3 Deluxe connected to my Arcam AVR 280 Receiver through the DVD-A/SACD analog inputs. So far no problems! I resample everything in JRiver to 192kHz/24bit.
I use the included Xonar software for subwoofer and speaker-level settings and have no problems listening to music or watching blueray movies, its awesome soundquality! ;D ;D ;D
I have been using the Xonar 1.3 deluxe for about 2 years. I love it. Before that and still being used for other recording an Echo Mona it had 6 analog outputs and sounded great. I had to send the Mona in for service and demoed the Echo Firewire model and it did not integrate well with my older laptop I already had a Behringer Adatthat gave me a total of 12 24/48 channels for recording. I was happy when Echo was able to repair my Mona and returned the firewire model.
Comparing both each had things I liked. The Mona sounds better for music playback and the Xonar was better for playback of movies. It is spec-ed to pass 24/192 audio to the analogs and will decode DTHD and DTSMA. The drivers work for me but could be better. They need a channel delay and they would be A- in rating. Echo on the other hand has been great about driver updates. For my Mona their last W7/W8 update was 12/19/11 not to bad. I’m still waiting for the Asus updates. Go on the Asus forum and see how happy they are.
If it were me, and I only know of these two manufactures I would try to see if Echo would work their customer service is much better.
I don’t know if my comments are too late but it was kind of nice to see I not the only one using analog inputs still.
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Just throwing this out there, but back a few years ago when I was really into music production, all the technicans I talked to raved about RME cards. Impeccable sound quality and some of the lowest latencies around.
http://www.rme-audio.de/en_index.php
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Just throwing this out there, but back a few years ago when I was really into music production, all the technicans I talked to raved about RME cards. Impeccable sound quality and some of the lowest latencies around.
http://www.rme-audio.de/en_index.php
I've communicated with the owner of RME a few times. He's a nice guy.
I was trying to see if he would be interested in marketing a little more to the home audio crowd. I was also hoping they would like to participate in a hardware review.
But last we spoke, they weren't too interested in either. Someday I might try them again.
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I'm cross-linking in this Lynx vs Steinberg vs Creative comparison since it directly addresses the topic of this thread:
http://yabb.jriver.com/interact/index.php?topic=77508.0
Good review..pls...have you controlled if these interfaces makes loud pops during the auto change of sample rate during playing of audio tracks?
Thx.
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I am using with MC18 a RME Fireface UFX on USB with ASIO. Computer: Win7 64-Bit
Changing sample rate, you hear a slight click, nothing bad but its not silent.
Otherwise the audio-quality is great.
You get 12 analog out.
Peter
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ok....thx
and with other your interface motu 828 Mk3 Hybrid? you've never tried for problem on auto change sample rate or you've tried on other software like foobar?