Guust-Fi
100+ Head-Fier
- Joined
- Sep 9, 2004
- Posts
- 372
- Likes
- 12
Here it is: The revised Chaintech AV-710 setup guide
When I kept seeing a dozen Chaintech posts a day, I thought it was time to do my part. Head-fi member "Mr. Radar" already wrote a nice graphical Chaintech setup guide but it seems like some issues weren't clear enough so I decided to write a more detailed guide. I used a lot of information from the internet and the head-fi forums. Thanks to all of you who helped solving some problems relating to 'THE' Chaintech card.
I am still trying to comprehend how computer audio (and digital audio in general) exactly works. Hence, more information and perhaps a guide to pc audio in general will be added in the future.
By the way, Mr. Radar did the proofreading and corrected all my mistakes so if there are any unanswered questions you should contact him. He really knows what he's talking about.
Hopefully this will contribute to the true enjoyment of pc audio. To me, it's the future! Troubleshooting can be frustrating though...
The Chaintech Guide
Installation
Well, you've finally received the card. Don't worry if you have the “s” version of the av-710. It is still exactly the same card. It's just a European manufacturing code.
To start with, unplug your computer from the wall and remove your computer's case cover so that you have plenty of room to work. More on this here. Also have your anti-static wristband on or at least touch something grounded made out of iron, like a radiator e.g. because PCI cards are VERY sensitive to static electricity. [Note from Mr. Radar: I've installed many components in my computer, and even built one while, wearing socks on dry Minnesota winter days, and I have to this date had no issues with static electricity, though before and during any computer hardware installation you should touch a grounded metal object, such as a water pipe or your computer's power supply while it's plugged in (be sure it's unplugged while installing hardware.)] When physically installing the AV-710 insert it into the lowest available PCI slot to help keep it away from electrical noise caused by components such as the CPU and Video card. (Would isolating with lead or other material make a difference?).
Locate your PCI card slots (labeled A) on your motherboard. Then use a screwdriver to remove the screw (labeled B) holding the PCI slot cover. Once removed, set aside the screw, you'll need that later. Align your PCI card with the slots on the motherboard and make sure the hole in the face of the PCI card lines up over the hole which you removed the screw. Now to install the PCI slot card, firmly (carefully) press down on it until it is in position.
Finally, replace the screw that you removed before to secure the PCI card into place. Now close your case and connect your speakers to the “rear-out” or the digital out. Plug your computer back in and turn it on. Your computer should detect the new card.
Drivers
The 1.43d drivers are good because they do not have the volume bug. Starting from 3.10 and up to the 4.32b drivers (released 24 October – which I use) all firmware has this little bug that causes the sound level to change only on one channel after adjustment of the main volume control sliders in the via control panel. If you leave the sliders at the top and only adjust the volume on your amp or speakers this shouldn't be a problem, however if you do need to adjust the volume for some reason, you should simply adjust the rear channel volume only. By switching to normal playback, putting the sliders back at the top and switching back to high resolution mode, you can re-level sound levels too. Setting the card volume as high as possible is best on all soundcards because your speakers or amp don't have to amp the sound that much. The use of the Audiotrak Prodigy (or other) soundcard drivers will be discussed later in the “Bit-perfect” section.
I recently discovered another problem: When your pc wakes up from stand-by mode, sound won't work anymore. You can solve this by either restarting your pc or by typing “services.msc” in the run window in the start menu and pressing “enter”. Locate “Windows Audio” and right click the item. Click: Restart and when restarting is completed, close the services window. Do not touch all the other processes unless you know what you're doing. I noticed sound comes back with Foobar but not with iTunes. For some reason sometimes it doesn't work at all and you need to restart. Disabling standby might be another solution if you don't really care about it.
Sample Rates - kMixer
When High sample rate mode is enabled (24bit - 96kHz), sound is processed by the Wolfson DAC (Digital to Analog Converter) which would only process the rear channels in normal playback mode. The other DAC, a VIA 1616 AC'97 codec, operates the first six channels. Only the rear-out (now going to be your line-out) (and the digital outputs with the 1.43d driver) are enabled in high resolution mode so you should connect your speakers/headphones to the rear out instead of the front out.
kMixer takes care of all the sound mixing in windows unless you use ASIO or Kernel Streaming ( already included in the Foobar package ) which bypasses the default windows mixer. In Windows 2000, kMixer only allows sample rates up to 96kHz, Windows XP on the other hand allows sample rates up to 192kHz. kMixer resamples the sound to the samplerate your card is set to and degrades the sound by doing this, though without that feature it would be unable to mix the audio properly. Even if source and destination format are the same it will strip the last 1 or 2 bits resulting in inferior dynamic range (by a few dB). Because in 24bit mode the theoretical dynamic range is so high, some people might not even be able to tell the difference with non high-end equipment.
(Head-Fi member Glassman wrote you could replace the XP kMixer file with the Windows2000 kmixer.sys file to solve a certain problem, haven't heard anything about it yet…)
When you using Kernel Streaming or ASIO with this card, all audio sent to the soundcard has to be in a format natively supported by the soundcard. ASIO can handle resampling by itself but not in this case ( Check the ASIO section ) With either ASIO or KS other sounds in windows will cause playback to stop in Foobar. When this happens, Foobar will stop producing sound and you will have to restart the track. You can disable all windows sound in the “Sound” tab under “Sounds and Audio Devices” in the Windows control panel. Another solution is to set your onboard audio as the default sound device and connect those to some low-end speakers, so all the “Windows sounds” will not be sent to the AV-710. You could even use to seperate sound systems with two different soundcards to do this if you really want to.
Upsampling, or converting from a low sample rate (such as the 44.1 KHz used by CDs) to a high sample rate (such as the 192 KHz used by DVD-Audio discs), can result in a different sound. Whether or not to upsample and whether the effects of upsampling are good is debatable. Whenever possible you should upsample to a multiple of the original rate. In this case we have to upsample to 96KHz in order to use Kernel Streaming or ASIO with the AV-710.
In Foobar, you have the choice between the SSRC and the PPHS resampler (also included in the “Special” and “Full” Foobar2000 packages) which is much faster (>10 % CPU usage difference) but slightly lower in terms of quality. Just place the “.dll” file in your Foobar components folder and restart the program. The resampler DSP can be placed anywhere in the DSP chain, but it should be placed either at the top or bottom depending on your needs. You might prefer using the advanced limiter at the very bottem of the dsp chain to control clipping. If you place the resampler (either SSRC or PPHS) at the top, you could get overall sound quality because all the DSP's under it will be running at 96KHz, however if you place it at the bottom you may see lower CPU usage because the DSP's above it will all be running at the audio's native rate.
Set the Target sample rate to 96000 Hz and enable slow mode. If Foobar2000 uses too much of your CPU in slow mode then you can disable it. You'll loose a bit of sound quality, but FB2k will use much less CPU. This shouldn't be an issue on Intel CPU's over 2.0 GHz and AMD CPU's over 2000+ (1.6 GHz). For slower CPUs where even SSRC's normal mode uses too much CPU, you should use the PPHS resampler to lower the processing power needed to run Foobar.
Bit rates
FYI: 16-bits provide a dynamic range of 96dB, while 24-bit offers a dynamic range of 144dB. Generally speaking this means 24bit sounds better because silent sounds will come out more clearly, the difference in sound levels will be better defined. The recording will not sound louder as if it were 16 bit though.
The AV-710 will only accept 32-bit data so set the Output data format to “24bit fixed-point padded to 32bit” in Foobar under the Playback preferences. The card can not handle 32 bit properly and will discard the padded additional 8 bits. Due to the way the preferences in Foobar display, if you close Foobar and go back to the playback preferences, the Output data format box may be blank but it should still be set at “24bit fixed-point padded to 32bit.” You can verify this by going to the console in Foobar's components list. It should list: “Created stream: 96000Hz 32bps 2ch”
Note: In the same menu you should not enable dithering. Dithering is used to make 16-bit audio sound more like 24-bit audio and since the AV-710 natively uses 24-bit there's no point in dithering, you won't hear any difference (or, because dithering reduces the SNR, you might hear a reduction in sound quality).
Kernel Streaming
Since you will bypassing kMixer with KS (Kernel Streaming), which normally resamples to a native format, resampling to 96kHz is necessary because when you set the AV710 to hi-res mode ( to make use of the Wolfson DAC ), it only supports a native format of 96KHz/24-bit.
Just download it (the “Full” and “Special” installers should come with it as an option) and put it in Foobar's components folder. Then select “Envy24 Family Audio” as the Device in the Kernel Streaming settings, and set “Kernel Streaming” as the “Output method” in the Output settings.
ASIO
ASIO, like Kernel Streaming, is an audio input/output method that bypasses kMixer. Because the Via drivers do not support ASIO natively, you need to use ASIO4All to convert the ASIO to Kernel Streaming. IMHO, there is no point to use ASIO with Foobar2000 when it already has a Kernel Streaming option. You gain nothing by using ASIO and while also having to tweak lots of settings, it might be difficult to get it to work right. To get ASIO to work, you need the appropriate “.dll” and ASIO4All.
To install the ASIO output in Foobar, download the .zip (~80K). If your CPU is a P4 then you must use the SSE3 version in the bin folder. If your CPU is an Athlon64 then you must use the SSE2 version. If it is an AthlonXP or a P3 then you must use the SSE version. If it is a regular Athlon or lower, or a P2 or lower then you should use the Normal version.
Drop the .dll into Foobar's components folder.
You can download ASIO4All here. Installation shouldn't be a problem.
Choose ASIO as Output method in Foobar's Output settings. Also of note, it's important to make sure Wuschel's ASIO4All (=asio4all 1.8) is the method selected. Then, open up the external ASIO4All control panel and start playing. You'll first want to try the Direct DMA Buffer I/O method, but if that fails, disable it, and start playing with buffer sizes. Set the buffer size in Foobar to 0 and let ASIO4All deal with that. The buffer size needed is likely going to vary widely depending on your system configuration. I'm running 1024/2, but that's just because I got tired of occasional pops while launching an application. You can have it down to around 500 if all you're doing is listening to music or browsing the internet, but launching anything causes pops and crackles. Normally ASIO does not require the resampler and can handle this by itself but in this case the card doesn't have full asio support. Therefore, you will need the resampler like with KS.
DirectSound – WaveOut
If you're on Windows ME or 98 then you'll need to use WaveOut or DirectSound. You can also use it on Windows2000 or XP, unlike ASIO or KS, playback doesn't stop when another windows sound is played. On Widows 9x & ME WaveOut might be more stable, and on Windows2000 & XP you should use DirectSound because WaveOut is depreciated. You should set it up as described in sections 2 & 3 of this guide. Using either of these will route the audio through kMixer which will alter the sound anyway despite the fact that source and destination samplerate are the same. Again, that's why KS is the prefered output method.
Jumpers
On the card itself there are 2 jumpers named Line-Out and Speaker-Out, I guess one jumper for each channel. For us, using high resolution mode it doesn't have any use because by moving the jumpers to the left, it only makes the soundcard treat the front audio output (green) as a line out instead of a speaker out.
In default position sound will be amplified a little bit before being sent to the front output. This doesn't have any use because the internal amplifier of your speakers will most likely amp the sound too. Functioning as a line out, the sound output would be cleaner, but still not as good as the rear out which uses the Wolfson DAC. You can still move the jumpers to the left in high resolution mode but it won't make any difference.
Bit perfect – External DAC
The digital SPDIF optical output of the AV-710 is bit-perfect when used with Kernel Streaming, however if you wish to get bit-perfect output from DirectSound applications, like iTunes, you'll need to flash its firmware to that of the Audiotrack Prodigy soundcard or other envy based soundcard. By doing this you can output your PC sound to your external DAC or receiver and it will do an even better job on converting the digital signal from your PC to analog sound. Some DAC's even support upsampling by themselves, so you do not need to waste your CPU power if you wish to upsample. Not all DAC's support all sample rates, so be sure to check which samplerates it supports. All the analog connections will not function anymore unless you flash your card back to the original Chaintech firmware. If you use any DSP's (even Volume Control) the signal will no longer be bit-perfect, though for PCM audio (the format CD's use) this should not be an issue. Iit would be an issue if you were trying to play back a raw DTS-WAV file for example.
When an output is bit-perfect, a DTS-WAV file would playback correctly if you you would connect the digital out from your soundcard to an external receiver for example. If the output is not bit-perfect the sound you'll get from the receiver will be random noise. Hence, this is an easy way to test if your soundcard is bit-perfect.
Here are all the files you need to flash it. Read the readme first!
You will need to reboot into DOS mode to do the flashing. After flashing uninstall the Chaintech drivers completely and install the latest Prodigy drivers from here. Because you flashed the firmware. Your computer will recognise your soundcard as an Audiotrak Prodigy, otherwise the Prodigy drivers wouldn't work.
The Prodigy soundcard is actually based on Envy24HT, but Chaintech uses slightly different version of that chip called Envy24HT-S. I'm sure their internals and registers are pretty similar. The S/PDIF ( a digital audio transfer format developed by Sony and Phillips usually carried by a coaxial or optical cable ) encoder, in particular, is the same on both, so your card has even better drivers and will work fine as an apparent Prodigy card now. These drivers should be alot more stable as well. That's why the Chaintech is considered to be a very decent transport (after flashing): It is bit-perfect, cheap and uses good drivers that support ASIO. There are cards with a better digital out available (Such as the EMU 0404). Paired with a very good practically jitter immune DAC such as a Benchmark DAC1, you wouldn't be able to hear the difference anyway.
Note: Jitter is the very small timing error that occurs in the digital stage which produces noise after conversion to analog sound.
To visualise this: The same signal below with timing difference.
0 10 1 00 1 0
01 0 1 0 0 1 0
0101 0 0 10
It is unavoidable due to the fact that the S/P DIF standard sends the clock signal together with the sound signal which is considered to be a mistake in the standard, especially to audiophile norms.
There is no need of the Wolfson DAC anymore because you will be using an external DAC so you can set your card to normal mode. You won't need resampling anymore unless you like the way it sounds, or if your DAC cannot handle the sample rate of a file you're trying to play. [Note: Upsampling will result in non-bit perfect output.]
If all you're playing is 44.1KHz/16-bit audio losslessly ripped from CD's, you should go with the 16-bit fixed output and 44.1Khz in Foobar (with no DSP's, ReplayGain, or dithering) for real bit perfect output. If you are using ReplayGain, DSPs, or if you have a large collection of music in lossy formats (such as MP3, WMA, Ogg Vorbis, Musepack, etc.) then “24bit padded to 32bit” is the best you can do. If you want to use ASIO instead of KS you should 'shift output channels' in the configuration tab of Foobar's ASIO plug-in by 8, this way you skip the eight analog outputs Prodigy has (the firmware makes your pc think it is dealing with a Prodigy card) and get to the output 9/10, which is routed to S/PDIF transmitter on the Prodigy.
Qsound & Sensaura
Qsound & Sensaura processes mono, stereo or even surround sound to create a simulated expanded 3D stereo environment with headphones or stereo speakers. You can find some demos of QSound here.
Enable QSound or Sensaura in the Via Control Panel if you wish. By doing this Kernel Streaming and ASIO will not work anymore.
There are superior technologies such as Dolby Headphone and in the future SVS that do a better, respectively superb job creating real surround sound experience through headphones in particular
Linux
These instructions on getting the high sample rate mode to work under Linux with ALSA were originally posted by Head-Fi member ADS at www.vandemar.org
Originally Posted by ADS:
In order to get the Chaintech AV-710 to run in high-res mode and use the superior Wolfson DAC, download the asound.state file here(mirror).
Copy this to your /etc/ directory as root, and run ‘alsactl restore'. This will enable the high-resolution jack. Notice that this will not mute the volume on your speaker outputs (unlike the windows version), so you can listen to your headphones and speakers at the same time. It's a fun effect, but you'll probably want to disable it. To do this, open up alsamixer and mute the ‘Master' and ‘Master Mono' controls. If you want your speakers muted by default, then after alsamixer run ‘alsactl store'. If still haven't figured out how to get the card into 96000 Hz sampling mode without problems, so if someone could get that to work I'd appreciate it. I know it has something to do with the ‘Multi Track Internal Clock' setting. To play around with it you can use ‘amixer cset numid=43′.
This document has been proofread by Tyler 'Mr.Radar' Knott.
Previous udated: December 24, 2004: Link broken; People still keep asking questions that have been answered already...
Last updated: January 20, 2006: Minor adjustments, additional information on jitter, bit-perfect output
When I kept seeing a dozen Chaintech posts a day, I thought it was time to do my part. Head-fi member "Mr. Radar" already wrote a nice graphical Chaintech setup guide but it seems like some issues weren't clear enough so I decided to write a more detailed guide. I used a lot of information from the internet and the head-fi forums. Thanks to all of you who helped solving some problems relating to 'THE' Chaintech card.
I am still trying to comprehend how computer audio (and digital audio in general) exactly works. Hence, more information and perhaps a guide to pc audio in general will be added in the future.
By the way, Mr. Radar did the proofreading and corrected all my mistakes so if there are any unanswered questions you should contact him. He really knows what he's talking about.
Hopefully this will contribute to the true enjoyment of pc audio. To me, it's the future! Troubleshooting can be frustrating though...
The Chaintech Guide
Installation
Well, you've finally received the card. Don't worry if you have the “s” version of the av-710. It is still exactly the same card. It's just a European manufacturing code.
To start with, unplug your computer from the wall and remove your computer's case cover so that you have plenty of room to work. More on this here. Also have your anti-static wristband on or at least touch something grounded made out of iron, like a radiator e.g. because PCI cards are VERY sensitive to static electricity. [Note from Mr. Radar: I've installed many components in my computer, and even built one while, wearing socks on dry Minnesota winter days, and I have to this date had no issues with static electricity, though before and during any computer hardware installation you should touch a grounded metal object, such as a water pipe or your computer's power supply while it's plugged in (be sure it's unplugged while installing hardware.)] When physically installing the AV-710 insert it into the lowest available PCI slot to help keep it away from electrical noise caused by components such as the CPU and Video card. (Would isolating with lead or other material make a difference?).
Locate your PCI card slots (labeled A) on your motherboard. Then use a screwdriver to remove the screw (labeled B) holding the PCI slot cover. Once removed, set aside the screw, you'll need that later. Align your PCI card with the slots on the motherboard and make sure the hole in the face of the PCI card lines up over the hole which you removed the screw. Now to install the PCI slot card, firmly (carefully) press down on it until it is in position.
Finally, replace the screw that you removed before to secure the PCI card into place. Now close your case and connect your speakers to the “rear-out” or the digital out. Plug your computer back in and turn it on. Your computer should detect the new card.
Drivers
The 1.43d drivers are good because they do not have the volume bug. Starting from 3.10 and up to the 4.32b drivers (released 24 October – which I use) all firmware has this little bug that causes the sound level to change only on one channel after adjustment of the main volume control sliders in the via control panel. If you leave the sliders at the top and only adjust the volume on your amp or speakers this shouldn't be a problem, however if you do need to adjust the volume for some reason, you should simply adjust the rear channel volume only. By switching to normal playback, putting the sliders back at the top and switching back to high resolution mode, you can re-level sound levels too. Setting the card volume as high as possible is best on all soundcards because your speakers or amp don't have to amp the sound that much. The use of the Audiotrak Prodigy (or other) soundcard drivers will be discussed later in the “Bit-perfect” section.
I recently discovered another problem: When your pc wakes up from stand-by mode, sound won't work anymore. You can solve this by either restarting your pc or by typing “services.msc” in the run window in the start menu and pressing “enter”. Locate “Windows Audio” and right click the item. Click: Restart and when restarting is completed, close the services window. Do not touch all the other processes unless you know what you're doing. I noticed sound comes back with Foobar but not with iTunes. For some reason sometimes it doesn't work at all and you need to restart. Disabling standby might be another solution if you don't really care about it.
Sample Rates - kMixer
When High sample rate mode is enabled (24bit - 96kHz), sound is processed by the Wolfson DAC (Digital to Analog Converter) which would only process the rear channels in normal playback mode. The other DAC, a VIA 1616 AC'97 codec, operates the first six channels. Only the rear-out (now going to be your line-out) (and the digital outputs with the 1.43d driver) are enabled in high resolution mode so you should connect your speakers/headphones to the rear out instead of the front out.
kMixer takes care of all the sound mixing in windows unless you use ASIO or Kernel Streaming ( already included in the Foobar package ) which bypasses the default windows mixer. In Windows 2000, kMixer only allows sample rates up to 96kHz, Windows XP on the other hand allows sample rates up to 192kHz. kMixer resamples the sound to the samplerate your card is set to and degrades the sound by doing this, though without that feature it would be unable to mix the audio properly. Even if source and destination format are the same it will strip the last 1 or 2 bits resulting in inferior dynamic range (by a few dB). Because in 24bit mode the theoretical dynamic range is so high, some people might not even be able to tell the difference with non high-end equipment.
(Head-Fi member Glassman wrote you could replace the XP kMixer file with the Windows2000 kmixer.sys file to solve a certain problem, haven't heard anything about it yet…)
When you using Kernel Streaming or ASIO with this card, all audio sent to the soundcard has to be in a format natively supported by the soundcard. ASIO can handle resampling by itself but not in this case ( Check the ASIO section ) With either ASIO or KS other sounds in windows will cause playback to stop in Foobar. When this happens, Foobar will stop producing sound and you will have to restart the track. You can disable all windows sound in the “Sound” tab under “Sounds and Audio Devices” in the Windows control panel. Another solution is to set your onboard audio as the default sound device and connect those to some low-end speakers, so all the “Windows sounds” will not be sent to the AV-710. You could even use to seperate sound systems with two different soundcards to do this if you really want to.
Upsampling, or converting from a low sample rate (such as the 44.1 KHz used by CDs) to a high sample rate (such as the 192 KHz used by DVD-Audio discs), can result in a different sound. Whether or not to upsample and whether the effects of upsampling are good is debatable. Whenever possible you should upsample to a multiple of the original rate. In this case we have to upsample to 96KHz in order to use Kernel Streaming or ASIO with the AV-710.
In Foobar, you have the choice between the SSRC and the PPHS resampler (also included in the “Special” and “Full” Foobar2000 packages) which is much faster (>10 % CPU usage difference) but slightly lower in terms of quality. Just place the “.dll” file in your Foobar components folder and restart the program. The resampler DSP can be placed anywhere in the DSP chain, but it should be placed either at the top or bottom depending on your needs. You might prefer using the advanced limiter at the very bottem of the dsp chain to control clipping. If you place the resampler (either SSRC or PPHS) at the top, you could get overall sound quality because all the DSP's under it will be running at 96KHz, however if you place it at the bottom you may see lower CPU usage because the DSP's above it will all be running at the audio's native rate.
Set the Target sample rate to 96000 Hz and enable slow mode. If Foobar2000 uses too much of your CPU in slow mode then you can disable it. You'll loose a bit of sound quality, but FB2k will use much less CPU. This shouldn't be an issue on Intel CPU's over 2.0 GHz and AMD CPU's over 2000+ (1.6 GHz). For slower CPUs where even SSRC's normal mode uses too much CPU, you should use the PPHS resampler to lower the processing power needed to run Foobar.
Bit rates
FYI: 16-bits provide a dynamic range of 96dB, while 24-bit offers a dynamic range of 144dB. Generally speaking this means 24bit sounds better because silent sounds will come out more clearly, the difference in sound levels will be better defined. The recording will not sound louder as if it were 16 bit though.
The AV-710 will only accept 32-bit data so set the Output data format to “24bit fixed-point padded to 32bit” in Foobar under the Playback preferences. The card can not handle 32 bit properly and will discard the padded additional 8 bits. Due to the way the preferences in Foobar display, if you close Foobar and go back to the playback preferences, the Output data format box may be blank but it should still be set at “24bit fixed-point padded to 32bit.” You can verify this by going to the console in Foobar's components list. It should list: “Created stream: 96000Hz 32bps 2ch”
Note: In the same menu you should not enable dithering. Dithering is used to make 16-bit audio sound more like 24-bit audio and since the AV-710 natively uses 24-bit there's no point in dithering, you won't hear any difference (or, because dithering reduces the SNR, you might hear a reduction in sound quality).
Kernel Streaming
Since you will bypassing kMixer with KS (Kernel Streaming), which normally resamples to a native format, resampling to 96kHz is necessary because when you set the AV710 to hi-res mode ( to make use of the Wolfson DAC ), it only supports a native format of 96KHz/24-bit.
Just download it (the “Full” and “Special” installers should come with it as an option) and put it in Foobar's components folder. Then select “Envy24 Family Audio” as the Device in the Kernel Streaming settings, and set “Kernel Streaming” as the “Output method” in the Output settings.
ASIO
ASIO, like Kernel Streaming, is an audio input/output method that bypasses kMixer. Because the Via drivers do not support ASIO natively, you need to use ASIO4All to convert the ASIO to Kernel Streaming. IMHO, there is no point to use ASIO with Foobar2000 when it already has a Kernel Streaming option. You gain nothing by using ASIO and while also having to tweak lots of settings, it might be difficult to get it to work right. To get ASIO to work, you need the appropriate “.dll” and ASIO4All.
To install the ASIO output in Foobar, download the .zip (~80K). If your CPU is a P4 then you must use the SSE3 version in the bin folder. If your CPU is an Athlon64 then you must use the SSE2 version. If it is an AthlonXP or a P3 then you must use the SSE version. If it is a regular Athlon or lower, or a P2 or lower then you should use the Normal version.
Drop the .dll into Foobar's components folder.
You can download ASIO4All here. Installation shouldn't be a problem.
Choose ASIO as Output method in Foobar's Output settings. Also of note, it's important to make sure Wuschel's ASIO4All (=asio4all 1.8) is the method selected. Then, open up the external ASIO4All control panel and start playing. You'll first want to try the Direct DMA Buffer I/O method, but if that fails, disable it, and start playing with buffer sizes. Set the buffer size in Foobar to 0 and let ASIO4All deal with that. The buffer size needed is likely going to vary widely depending on your system configuration. I'm running 1024/2, but that's just because I got tired of occasional pops while launching an application. You can have it down to around 500 if all you're doing is listening to music or browsing the internet, but launching anything causes pops and crackles. Normally ASIO does not require the resampler and can handle this by itself but in this case the card doesn't have full asio support. Therefore, you will need the resampler like with KS.
DirectSound – WaveOut
If you're on Windows ME or 98 then you'll need to use WaveOut or DirectSound. You can also use it on Windows2000 or XP, unlike ASIO or KS, playback doesn't stop when another windows sound is played. On Widows 9x & ME WaveOut might be more stable, and on Windows2000 & XP you should use DirectSound because WaveOut is depreciated. You should set it up as described in sections 2 & 3 of this guide. Using either of these will route the audio through kMixer which will alter the sound anyway despite the fact that source and destination samplerate are the same. Again, that's why KS is the prefered output method.
Jumpers
On the card itself there are 2 jumpers named Line-Out and Speaker-Out, I guess one jumper for each channel. For us, using high resolution mode it doesn't have any use because by moving the jumpers to the left, it only makes the soundcard treat the front audio output (green) as a line out instead of a speaker out.
In default position sound will be amplified a little bit before being sent to the front output. This doesn't have any use because the internal amplifier of your speakers will most likely amp the sound too. Functioning as a line out, the sound output would be cleaner, but still not as good as the rear out which uses the Wolfson DAC. You can still move the jumpers to the left in high resolution mode but it won't make any difference.
Bit perfect – External DAC
The digital SPDIF optical output of the AV-710 is bit-perfect when used with Kernel Streaming, however if you wish to get bit-perfect output from DirectSound applications, like iTunes, you'll need to flash its firmware to that of the Audiotrack Prodigy soundcard or other envy based soundcard. By doing this you can output your PC sound to your external DAC or receiver and it will do an even better job on converting the digital signal from your PC to analog sound. Some DAC's even support upsampling by themselves, so you do not need to waste your CPU power if you wish to upsample. Not all DAC's support all sample rates, so be sure to check which samplerates it supports. All the analog connections will not function anymore unless you flash your card back to the original Chaintech firmware. If you use any DSP's (even Volume Control) the signal will no longer be bit-perfect, though for PCM audio (the format CD's use) this should not be an issue. Iit would be an issue if you were trying to play back a raw DTS-WAV file for example.
When an output is bit-perfect, a DTS-WAV file would playback correctly if you you would connect the digital out from your soundcard to an external receiver for example. If the output is not bit-perfect the sound you'll get from the receiver will be random noise. Hence, this is an easy way to test if your soundcard is bit-perfect.
Here are all the files you need to flash it. Read the readme first!
You will need to reboot into DOS mode to do the flashing. After flashing uninstall the Chaintech drivers completely and install the latest Prodigy drivers from here. Because you flashed the firmware. Your computer will recognise your soundcard as an Audiotrak Prodigy, otherwise the Prodigy drivers wouldn't work.
The Prodigy soundcard is actually based on Envy24HT, but Chaintech uses slightly different version of that chip called Envy24HT-S. I'm sure their internals and registers are pretty similar. The S/PDIF ( a digital audio transfer format developed by Sony and Phillips usually carried by a coaxial or optical cable ) encoder, in particular, is the same on both, so your card has even better drivers and will work fine as an apparent Prodigy card now. These drivers should be alot more stable as well. That's why the Chaintech is considered to be a very decent transport (after flashing): It is bit-perfect, cheap and uses good drivers that support ASIO. There are cards with a better digital out available (Such as the EMU 0404). Paired with a very good practically jitter immune DAC such as a Benchmark DAC1, you wouldn't be able to hear the difference anyway.
Note: Jitter is the very small timing error that occurs in the digital stage which produces noise after conversion to analog sound.
To visualise this: The same signal below with timing difference.
0 10 1 00 1 0
01 0 1 0 0 1 0
0101 0 0 10
It is unavoidable due to the fact that the S/P DIF standard sends the clock signal together with the sound signal which is considered to be a mistake in the standard, especially to audiophile norms.
There is no need of the Wolfson DAC anymore because you will be using an external DAC so you can set your card to normal mode. You won't need resampling anymore unless you like the way it sounds, or if your DAC cannot handle the sample rate of a file you're trying to play. [Note: Upsampling will result in non-bit perfect output.]
If all you're playing is 44.1KHz/16-bit audio losslessly ripped from CD's, you should go with the 16-bit fixed output and 44.1Khz in Foobar (with no DSP's, ReplayGain, or dithering) for real bit perfect output. If you are using ReplayGain, DSPs, or if you have a large collection of music in lossy formats (such as MP3, WMA, Ogg Vorbis, Musepack, etc.) then “24bit padded to 32bit” is the best you can do. If you want to use ASIO instead of KS you should 'shift output channels' in the configuration tab of Foobar's ASIO plug-in by 8, this way you skip the eight analog outputs Prodigy has (the firmware makes your pc think it is dealing with a Prodigy card) and get to the output 9/10, which is routed to S/PDIF transmitter on the Prodigy.
Qsound & Sensaura
Qsound & Sensaura processes mono, stereo or even surround sound to create a simulated expanded 3D stereo environment with headphones or stereo speakers. You can find some demos of QSound here.
Enable QSound or Sensaura in the Via Control Panel if you wish. By doing this Kernel Streaming and ASIO will not work anymore.
There are superior technologies such as Dolby Headphone and in the future SVS that do a better, respectively superb job creating real surround sound experience through headphones in particular
Linux
These instructions on getting the high sample rate mode to work under Linux with ALSA were originally posted by Head-Fi member ADS at www.vandemar.org
Originally Posted by ADS:
In order to get the Chaintech AV-710 to run in high-res mode and use the superior Wolfson DAC, download the asound.state file here(mirror).
Copy this to your /etc/ directory as root, and run ‘alsactl restore'. This will enable the high-resolution jack. Notice that this will not mute the volume on your speaker outputs (unlike the windows version), so you can listen to your headphones and speakers at the same time. It's a fun effect, but you'll probably want to disable it. To do this, open up alsamixer and mute the ‘Master' and ‘Master Mono' controls. If you want your speakers muted by default, then after alsamixer run ‘alsactl store'. If still haven't figured out how to get the card into 96000 Hz sampling mode without problems, so if someone could get that to work I'd appreciate it. I know it has something to do with the ‘Multi Track Internal Clock' setting. To play around with it you can use ‘amixer cset numid=43′.
This document has been proofread by Tyler 'Mr.Radar' Knott.
Previous udated: December 24, 2004: Link broken; People still keep asking questions that have been answered already...
Last updated: January 20, 2006: Minor adjustments, additional information on jitter, bit-perfect output
