SpDif confusion...

Hi
I'm a little confused - here's my problem.
I have a Bass Pod pro and want to use it for recording bass etc. and I want to re-amp some audio from Logic.
It's would be most convenient if I could select the output of the audio track I wish to re-amp to SpDif-L+R [out from MOTU to bass pod] and simply set the input of a record enabled track (software monitoring on) to SpDif-L+R [In to MOTU from Pod].
This works sometimes but I seem to be getting latency on the actual recording. Is there anything I can do to stop this?
Other times I seem to be getting a nasty loud hiss, usually on one side - any ideas?
Any help would be great as this is giving me a bit of a headache!
Cheers
Del

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Similar Messages

Whats the difference between SPDIF and digital connectio

OK, I've read around and I'm still confused, so I have some questions that hopefully you peeps can answer:
) SPDIF is a digital connection correct?
2) Can SPDIF carry 5. signals?
3) Whats better, optical or coax?
The next question is related to my speaker system...
4) I currently have the Creative Inspire 5. Digital 5500. Its currently using the SPDIF connection. Why is it that I'm only getting 4. and not 5. in the speaker (channel/noise) test? Why isnt the X-FI recognising my 5. speaker system?
Really need this sorted guys, so please help. Thanks.Message Edited by prankz on 0-22-2006 03:5 PM

. "Digital audio connection" is a concept - the sound is "transported" in some numeric form between 2 devices.
2. SPDIF is a particular hardware implementation (interface) of that concept (the most popular one). Was made by Sony and Philips to transport 2 channels of PCM encoded sound (wav files). Maximum bandwidth was enough for 2 audio channels encoded in PCM (uncompressed) at 24 bit/48kHz. Now it was extended to 24 bit/96kHz.
Phisically is done by a coaxial copper cable (RCA connectors) or via optical cable. The signal is the same in both cases so none is better than other.
3. Surround sound has more information that stereo so some companies invented a way to compress that 5. surround channels to fit into over the SPDIF connection. The most well known solutions are DolbyDigital and DTS. Those signals although digital need to be "decoded" at the receiving end - the PCM signal can be fed directly to DAC's (so no "decoding" ).
4. Sound in computer games is surround but is not compressed with any of the 2 methods. So if you feed that sound via SPDIF you will get only stereo PCM in the speakers. If you play in the PC a DVD that has the sound already encoded in DolbyDigital, that can be sended via SPDIF as such and be decoded in 5. signal in the speakers. Creative soundcards don't encode the surround sound from games in DolbyDigital or DTS (there are some other cards that ca do that but those don't have hardware EAX).
More about that: http://en.wikipedia.org/wiki/Digital_sound
Message Edited by SoNic2367 on 0-22-2006 0:44 PM

Analog sound out of SPDIF

Analog sound out of SPDIF' I have connected a digital coax cable from the digital out to a Harman Kardon stereo receiver. The cable has been run to another room in order to watch downloaded shows/movies in my living room.
I receive DTS and DD sound just fine using this connection however PC sounds/MP3s and other analog sounds do not work. Is there a way to get this to work without having to add cables? Being forced to manually switch the output signal on the computer and on the stereo receiver is not a very good option.
If this is not supported are there any other sound cards that do support this? I can't imagine I am the first person to want to be able to watch DTS movies and listen to MP3s on a stereo receiver without having to swtich inputs/use a different set of cables.

? I just wanted to what I have experienced based on my digital spdif connection:
After looking at the support site for stereo receivers, I see the following statement:
"If you are going to connect from the Digital Out on your Sound Blaster to the Digital Input on your receiver, you will only receive multi channel audio when your are playing DVD games, or DVD movies, or standalone AC3 files. You will only hear stereo (front left and right channels) when playing MP3, MID, AVI, WAC, or WMA et."
This statement is confusing bc the last sentence leads me to believe that I will hear MP3, and WMA audio from the digital out but it will only be in stereo sound, not surround sound. That is fine with me, I simply want to be able to hear sound from a single cable.
Can someone clear that statement up from the support site? Will any MP3, AVI, or WMA? sound play on a stereo using the digital coax input?

Using spdif input into logic to record..

hello, I am trying to record a track in logic 9 using spdif input..
It does not seem there is an option to choose a digital input, just input 1 and 2.
I am actually recording from a second mac running logic so i have two m-audio interfaces for each computer
so i am recording a stereo mix in logic on one machine goint to logic on my second machine.
There is a reason I am doing this!
help please!

But not the standard Fast Track, that doesn't have s/pdif.
The Master Computer in this case is the playback computer, you don't need to do anything as far as setting the clock goes. The Recording computer must be set to external sync in THE FAST TRACK software, don't mess with Logic or it will get confusing, this is all done in the Fast Track software applet. The important part is that the recording computer's audio interface is synced to the digital clock coming in on the s/pdif cable (along with the audio of course.
Don't try and use MIDI to sync the two, Logic only send MIDI clock it doesn't receive it, might be able to make it work using MIDI Machine Control but the start commands will come from the recording computer not the playback one. One thing at a time, the important thing is to get s/pdif working correctly.
What are you trying to do?

How do you Control SPDIF input levels in Logic Pro using Apogee Ensemble

Hello,
Question. I have an Apogee Ensemble audio interface hooked up to my Macbook Pro. I have a Motif XS8 that I record audio from through SPDIF.
I am able to get sound through to my audio track once setting the inputs in Logic, however the levels are not even close to halfway. I can not figure out how to fix adjust the SPDIF level to make it hotter b/c I know once I add virtual tracks and vocals through my preamp, the audio from my motif will likely not be hearable.
Please let me know.
Also, when I record an audio track is says about 15 minutes remaining to record. Does that mean I have 15 mins of audio for each track or for all my recording period???

Not owning an Apogee Ensemble, maybe someone else who does has more definitive info can jump in, but do not confuse the lower level of the SPDIF input with "lesser quality". It's a digital connection, with no analog conversion.
-18 dBFS on a digital scale is the same as 0dB on an analog scale.
I think what you're experiencing, is the fact that so many people think they need to get the meters in Logic closer to 0dB, which is actually NOT what you want (when staying ITB). This thinking is left over from the days of analogue recording, or the early days of 16 bit digital recording.
Turn your other sources in Logic down, and you'll reap the benefits of not overloading the 2 buss, allowing the plug-ins to do their computations without distorting, etc... Then you can bring the over all level of your mix back up, in mastering.
If the Motif is considerably lower in volume than that, you could always insert a gainer plug-in on the audio track.
As to the 15 minutes, that is what Logic has pre-allocated for recording to your hard drive. It "re-sets" each time you go into record, so it's not saying "you only have 15 minutes of record time available". It's saying, "you have 15 minutes of hard disk space allocated each time you go into record, based on your sample rate and bit depth settings". This can be changed in the Audio pathway, but it's recommended to leave this number as low as possible/necessary, to avoid disk fragmentation.

Analog Levels vs SPDIF Levels Input and Output in Logic Pro

Hello,
I ran a test last night for recording input and output levels from my Yamaha Motif XS8 through an Apogee Ensemble to compare Analog to SPDIF
I connected two TRS cables from the L and R outputs on the Motif XS into the Analog Inputs on the Apogee and also have the SPDIF connection from the Motif to the Apogee.
I put the master fader all the way up on the XS for the volume for analog.
The ensemble in Maestro has a +4 and -10 reference notional level option for analog inputs. i had it at +4 but changed to -10 and the analog got louder (i figured it would get louder for +10, confusing).
anyways, why is it that I can record louder levels for analog than the digital transfer?
I tracked both options at the same time then recorded vocals over it. the digital sound is too low. what's up with that?
and when I tried to bounce the recording to listen to it in ITunes, the volume levels were way lower than my cd playing through iTunes. Please enligthen me on these?
I use to record in a Roland 2480, and had similar results with loudness, but got a little louder through mastering but still....pro cds are way louder and still clear.

I think there is much confusion here!
In summary, you wont be able to control the recording level of S/PDIF.
The reason is that you don't want to!
You need to think of the SPDIF connection as being more like a file transfer method. You are copying the digital data at an output to your harddrive in effect. If I send you an MP3 via email you'd never imagine that your email software is capable of changing the gain of the MP3 I send you. This might sound daft but its a useful analogy. If you need to increase the "volume" of that MP3 then you'd need to ask the sender. Its the same with your set-up.
There could well be somewhere on your synth that adjusts the instruments level, other than the master (analogue) output control. For example, make sure the midi volume of the intrument being played is set to full - ie midi vol 128. Perhaps there is somesort of virtual mixer onboard to control all the muti-timbral parts so make sure your part has its virtual fader turned up.
This is what (basically) is going on in the chain...
Your synth creates sound in its "digital brain". This sound is sent to an "output stage" which will distribute the sound to various outputs. In the case of the S/PDIF it will just send the raw digital data untouched. For the analogue side the digital signal (same as the one sent to S/PDIF) will be converted to analogue and then sent to a amplifier to get it to an appropriate "line level". This final level could well be controlled by an anogue volume control which could be adding more gain (than you think) too.
When things go to your sound card/ daw...
The purpose of a analogue gain control is to set the i/p signal so that it suitably loud to beat any noise that exists in your input circuits - so that a good signal to noise ratio is achieved. Analogue signals need to work in the right loudness zone (so to speak) as the analogue electronics will be designed to handle signal levels of a particular range. the gain control is there to make sure the signal is in that range.
Digital signals are far more predictable though and there is no advantage to your recordings if the incoming digital signal gets an increase of level at the input stage. All you are doing here is effectively adding a few zeros to the binary digital data!
Lets face it the point of recording is to get a copy of the original sound, that is as similar to the original as possible. With S/PDIF you get a perfect copy of what's coming out of your synth - so job's a good un!
If, when you come to mix in logic, you find the level of the digital recording is indeed too low for mixing/mastering purposes then just boost it in logic via a fader or via the gain plugin.
Those referrence values of -10dBV and +4dBu refer to analogue voltage levels only. they have nothing to do with the digital domain. The -10/+4 switch will be only relevant to analogue inputs and outputs. Using an analogue VU meter you should find that a sine wave that peaks at 0dBVU (totally different to 0dBFS BTW) is the equivant of a digital sine wave peaking at -18dBFS.
The analogue headroom (how loud you can go before things distort) depends on the analogue electronics and varies with different design. Analogue stuff, like mixers) often has headroom of 24dB or more. So that digital stuff can interface with analogue properly we allow for that analogue headroom to be around 18dB (usually enough in practise!)... hence -18dBFS(digital)=0dBVU(analogue).
To make your digital and analogue input signals sound similar in level you will probably have to reduce the gain of the analogue input. If you set the incoming analogue signal to peak around -14dB (or less!) or so you will probably find things more equal. If you are working in 24 bit your analogue levels can be seemingly very low before sound quality is affected. Its quite safe to record at -20 or even -30dB as shown on logic's meters for eg.
I hope all this waffle helps LOL!

X-Fi SPDIF not working in Vis

I am using X-Fi platinum with DTT2500 (with external decoder).
Under the Creative Audio Console, i had already set the Decoder to use Bitstream Out, with SPDIF Bypass.
Under windows, I had set SPDIF as the default playback devices (rather that "Speakers"). However, my external decoder is still not receiving any digital signals.
To make things more confusing, I have no idea is my player running on a software decoding of the AC3 materials, or the X-Fi is doing the job.
Is there any way to verify this? As it seems that the current beta drivers does not include a on-screen notification of DD/DTS output.

I think you need to use Alchemy to sort of "convert" the game to recognise it.
The way sound is created in Vista is very different to how it was in XP and so older applications don't check the correct way to find it in Vista.
Alchemy is a free download for X-Fi users I think.

[SOLVED] Please help me with my asound.conf (2 soundcards/spdif)

well, i have an ESI juli@ card, that i want as my playback card and an intel onboard card which i want for the microphone part.
So what i need is to define those 2 default devices, but i cant for the sake of it get it to work.
My juli@ card is listed as:
> aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
sysdefault:CARD=Juli
ESI Juli@, ICE1724
Default Audio Device
front:CARD=Juli,DEV=0
ESI Juli@, ICE1724
Front speakers
surround40:CARD=Juli,DEV=0
ESI Juli@, ICE1724
4.0 Surround output to Front and Rear speakers
surround41:CARD=Juli,DEV=0
ESI Juli@, ICE1724
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=Juli,DEV=0
ESI Juli@, ICE1724
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=Juli,DEV=0
ESI Juli@, ICE1724
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=Juli,DEV=0
ESI Juli@, ICE1724
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=Juli,DEV=0
ESI Juli@, ICE1724
IEC958 (S/PDIF) Digital Audio Output
and my intel card as:
sysdefault:CARD=Intel
HDA Intel, VT1828S Analog
Default Audio Device
front:CARD=Intel,DEV=0
HDA Intel, VT1828S Analog
Front speakers
surround40:CARD=Intel,DEV=0
HDA Intel, VT1828S Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=Intel,DEV=0
HDA Intel, VT1828S Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=Intel,DEV=0
HDA Intel, VT1828S Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=Intel,DEV=0
HDA Intel, VT1828S Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=Intel,DEV=0
HDA Intel, VT1828S Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=Intel,DEV=0
HDA Intel, VT1828S Digital
IEC958 (S/PDIF) Digital Audio Output
My momentary (not working) asound.conf looks like this:
pcm.!default {
type plug
slave.pcm {
type dmix
ipc_key 1478
slave {
pcm "hw:0,0"
format S32_LE
period_time 0
period_size 1024 #useful if you ear scratch or have a crappy sound (try other values if it doesn't work)
buffer_size 8192 #useful if you ear scratch or have a crappy sound (try other values if it doesn't work)
rate 44100 #set to 44100 or other.
pcm.usb
type hw
card Intel
pcm.!default
type asym
playback.pcm
type plug
slave.pcm "dmix"
capture.pcm
type plug
slave.pcm "usb"
The microphone part seems to work just fine, but i cant manage to get the spdif device as default.
Anyone can help me?
Last edited by Rasi (2011-09-05 20:13:40)

That Juli@ card looks good! I really like how it "flips"... never seen that before.
Now to the problem(s). ALSA configuration is confusing, complex, and poorly documented. So I'm not too surprised you're having problems. I guess we should just be thankful that there's at least some documentation.
I think there should only be a maximum of one "pcm.!default" object and a maximum of one "ctl.!default" object in a configuration. Otherwise, which one is the default?
The microphone part seems to work just fine, but i cant manage to get the spdif device as default.
Confusion. Are you saying that you can't get the Juli@ card to be the default for playback, or you can't get the spdif ports on the Juli@ card to be the default for playback?
As a learning exercise for me, I took on your challenge by trying to get something similar to work on my box. It has an internal Intel HDA audio chip, and an external USB webcam (for the microphone part)... as shown here:
$ cat /proc/asound/cards
0 [Intel ]: HDA-Intel - HDA Intel
HDA Intel at 0x502c0000 irq 44
1 [J ]: USB-Audio - A4 TECH USB2.0 PC Camera J
A4 TECH A4 TECH USB2.0 PC Camera J at usb-0000:00:1d.7-4, high speed
Card 0 has the name "Intel" and the webcam on card 1 has the name "J".
And here are the devices on those cards:
$ cat /proc/asound/pcm
00-00: ALC662 rev1 Analog : ALC662 rev1 Analog : playback 1 : capture 1
00-01: ALC662 rev1 Digital : ALC662 rev1 Digital : playback 1
01-00: USB Audio : USB Audio : capture 1
So Card 0 (the card named "Intel), has both an analog playback and analog capture device ("00-00"), along with a digital playback device ("00-01"). In the ALSA world, these are known as "hw:0,0" and "hw:0,1" respectively.
While Card 1 (the card named "J") only has a capture device ("01-00"), known as "hw:1,0" in the ALSA world.
Next, I went to these web pages:
http://alsa.opensrc.org/.asoundrc
http://www.alsa-project.org/alsa-doc/al … ugins.html
and started playing around with the most simple "pcm.!default" object in ~/.asoundrc that would work when using the "type asym" plugin. At least in 2 channel stereo, since that's all I'm interested in. Then I kept adding to it (testing after each little change), until I got somewhat near to what I think you are looking for. But, as I'm sure you're aware, there are many ways to accomplish something in an ALSA configuration file. Anyway, I ended up with this:
pcm.!default { # declare the default PCM device
type asym # combine playback and capture devices
playback.pcm { ### Playback object ###
type plug # convert audio to mixer format
slave.pcm {
type dmix # mix multiple audio streams
ipc_key 6969 # shared memory key; must be unique
ipc_key_add_uid true # add UID to shared mem key
ipc_perm 0600 # shared mem file permissions
slave {
pcm {
type hw # define the playback hardware
card "Intel" # playback card name
device 0 # playback device on that card
format S16_LE # bits per sample format
rate 44100 # samples per second
channels 2 # number of audio channels
capture.pcm { ### Capture object ###
type plug # convert audio format from the hardware
slave {
pcm {
type hw # define the capture hardware
card "J" # capture card name
device 0 # capture device on that card
Obviously, you'd need to change the card names ("Intel" and "J" in my case), and the "device" numbers, to what's on your box. Along with anything else that fits what you are trying to accomplish. Including adding and deleting things; but perhaps you can use it as a starting "template". But it's well worth it to read those two confusing web pages at least 4 or 5 times.
OBTW: Not sure if you plan on doing any "professional" type of audio work with that Juli@ card, but be aware that when you use the "plug" and/or "dmix" plugins (among others), that ALSA may dither/resample/mix your audio in software before sending it to/from the audio hardware device. So the audio format you think you're using may not be what's actually being used.
Hope this helps.

Need help setting up 0404 PCIe for SPDIF in/out

Hello,
I had previously been using an M-Audio card but have moved to the Emu 0404 b/c M-Audio doesn't support PCIe yet only PCI.
Anyway, I am trying to make simple connnections and find the Patchmix very confusing. I don't want to use any of the DSP effects since I have excellent plugins in my DAW software.
All I want to do is have SPDIF L/R going from my Mytek Stereo 96 AD into my DAW and SPDIF L/R going out to my JBL monitors which have built in DA conversion.
So far I have not been able to get any sound at all from my Reaper projects, just trying to play them back at this point not even trying to record anything new yet.
Any suggestions? I've tried putting sends on the tracks etc but still can't get any sound to my monitors and the SPDIF outputs don't show up in my DAW software routing matrix...
Thanks

Well it looks like I've got more troubles. Reading the 0404 users manual last night there is a sentence there that says "The SPDIF inputs and outputs are usable at 44.1, 48 and 96khz." This is pretty much a dealbreaker for me since everything I've recorded for the past 2 years is at 24/88.2 and it's my preferred sample rate going forward.
Does anyone know of a PCIe interface with SPDIF in/out that supports the 88.2khz sampling rate?

Issues with SPDIF out Extreme Au

I have yet another issue with my Extreme Audio and Digital I/O, but I found my own work-around so I decided to post it.
I have the Digital I/O module and I just picked up a Z-5500 speaker and wanted to go digital on my HTPC. Nothing reported that I had a SPDIF out (PowerDVD, Nero, VLC). So I reinstalled the drivers (and did not reboot). It worked and all was good until I rebooted, then no more SPDIF out. Nothing I could do would re-able the SPDIF. Reinstalled the drivers (and didn't reboot). SPDIF working again. Hmmm, this is going to be a PITA.
Lightning Strikes!
I figured out that if I disable, then re-enable the card in Device Manager all is well (until I reboot). This is a 24/7 HTPC, so this isn't a major problem for me.

I have the Express configured now as ‘extend a wireless network’. Is this correct, or should it be ‘join a wireless network’ ? It presently shows the router mode as Off (bridge mode).
Router mode "off" is correct but "extend a wireless network" and the dotted line indicate that it is "wirelessly extending" your Extreme's network which is not what you intend.
If an Express detects a viable Internet connection from its WAN port, AirPort Utility should propose that connection. Since it did not, verify the Ethernet cable is intact by connecting a wired Mac or other device to it, ensuring its wireless is temporarily turned off to avoid confusion. When you are satisfied that the Ethernet connections are correct, reprogram the Express. Usually it is best to start with another "hard reset". AirPort Utility should configure it the way you propose. If an Express is already configured to connect wirelessly, it's difficult to convince it otherwise without a "hard reset".
It is possible that the complexity of your wired network could introduce an element of uncertainty. The noises you describe are unusual. An Express's internal power supply might make a barely detectable whine but an Extreme's power supply is separate and I can't think of anything within an Extreme that should make a noise that you describe.
Check each connection point and cable for connection integrity. If in doubt, check each connection point with an Ethernet cable known to be good, and a device capable of verifying its connection. If it's not a portable Mac, try the Xbox or similar device. An Ethernet network test appliance is another option. Cat 6 problems are usually found at connections, and they are not very forgiving of improper work.
There is nothing wrong with routing Ethernet cable alongside metal ductwork, as long as it's not within it.

No Front Panel Output/SPDIF input settin

I just frankentstein'd my X-FI Platinum out of an old 32-bit platform along with the software, network adapters, OS (Windows XP Pro-32 bit). I've got 2 gb ram, a total of .32 tb of HD space, a Q6600, an nForce 680i sli mobo, and a 850 watt psu. My speaker system is a Logitech Z5500 that was also hooked up to my last system. Aside from the mobo, ram, and processor, all things have remained the same. My problem is that I can't get the optical output of the front panel to work. I tried removing the card and placing it in the other PCI slot, but it still didn't work. I tried re-setting the cables to make sure that I didn't screw up when I placed them, still nothing. The red light on the other end of the cable is clearly visible. My Z5500 console still says "No Digital Data" when I set it to optical. Also, the SPDIF input settings in the Creative Audio Console is greyed out. I can't select it. What am I doing wrong here? It worked perfectly in the other system, and I didn't break it when I moved it.

Hi Dutu,
I agree there are 3 CDs in all. One is called the Install CD. The other 2 are called Software CD 1 of 2 and Software CD 2 of 2.
I did not use the Install CD at all. I started from from SW CD 1 of 2, because I wanted to dual boot with my old OS. I maybe wrong, but my impression was that if you start from the Install CD, it will take over the whole disk and destroy your previous OS. While the SW CD will give you the chance to do an interactive install. I agree it's confusing, beause Software CD gives you the impression that it contains only applications and not the OS. I don't know why SUN chose to call them S/W CDs, while they also contain the OS.
Regards
Yousef

975X - Realtek ALC882: SPDIF output locked to 48kHz??????

Hi,
finally everything works fine here with my Platinum 975x PowerUp. But I have a problem with the digital audio out.
With every driver I've checked, the output samplingrate on SPDIF is locked to 48 kHz, there are no options to change that. I don't like internal resampling, and the specs of the chip say that it has digital output with 44.1, 96 kHz etc. as well.
It is impossible to play DTS-encoded CDs, because you need bit-accurate transmission to the decoder in my AV-Receiver.
I've read somewhere that someone had success on Windows Vista, but not on XP.
Does anyone know more??
Thanks,
Martin

Quote from: Maesus on 06-February-07, 23:51:16
I think you're confused.
No, I'm not. I know the Realtekcard can't decode DTS. But a DTS-CD is an audio CD, where the PCM-stream is in fact a DTS-stream. A decoder (like an AV-Receiver) recognizes this and decodes the DTS stream. All you need is a bit-accurate soundcard - it doesn't have to understand what kind of "digital noise" it transmits.
DTS-CDs play on every CD-Player with a sample-accurate digital output. What you need is an external decoder. (On the analog output you hear just noise). They where a precursor of SACD and DVD-A, to play multichannel/audio-only content.
The only problem with my Realtek configuration is, that everything is converted 48 kHz. I've seen a 965 Platinum (with an ALC883), where you can select the output sample rate.
DTS was only one example which doesn't work. Samplerate conversion should be avoided whereever possible! Windows media player can decode HDCD-CDs. But what do I gain, when Realtek does on-the-fly sample-rate conversion afterwards without a reason??
Thanks,
martin

SPDIF "Bit accurate" recording.

Ok a little confused. I have Adobe Audition and Audgy 2 Platinum EX and need a high a quality true-to-original recording. A few questions:
1. Should I set "enable bit-accurate recording" and why if so
2. If I set Bit accurate recording and the Input via SPDIF is 16 bit @ 48 KHz should I set Adobe Audition to record at this also? Since adobe recommends always recording in 32 bit mode.
Message Edited by seiferalmasy on 08-20-2006 12:50 AM
Message Edited by seiferalmasy on 08-20-2006 12:50 AM

> Theodore, not sure if you will read this or not since its an old post but try using 50% input level. I know this is counterintuiti've but in my adventures with the Audigy 2 NX I have found that 50% input level and 00% output level resulted in a bit perfect loopback recording. Not sure if this will help in your case or not but try it out!

Confused: RS480M2-IL outputs on emachine T6212

Hello,
I'm looking into buying the emachines T6212 because it's a very cheap way to start a RS480M2-IL based PC.
I am confused about the RS480M2-IL motherboard options available on the emachines.
Various specifications for the RS480M2-IL differ in the inclusion of the S-video, composite video, and SPDIF out outputs.
The MSI site lists the S- and composite video outputs, but not the SPDIF out, and none show up on the motherboard picture
The egghead site lists video outputs and SPDIF outputs and include them on the motherboard picture
The emachines T6212 site does not list the video or SPDIF outputs and does not show them on the rear case picture.
Are these all different variants of the RS480M2-IL board? If so, my main question is whether there are at least SPDIF out header
pins on the emachines version of the motherboard?
Ideally I'd like to hear from a T6212 owner.
Thanks,
Colin

Quote
danno: You are saying you have a version of the board with no s-video or spdif? Is it the "non-video" version? If not, then I think it's safe to say there is a variant with integrated video and no s-video/spdif out.
No, I've got the version with the s-video and spdif ports on the rear, just like the MSI spec sheets and Quick User Guide show. You were asking if there were any board headers (I assume you meant pin headers) for add-on connection of same and mine doesn't. Usually, if the PCB incorporates a design for those, there's a place on the board (not just the spot for the rear connector). Since the s-video is "under" the Parallel port connector, it would tend to rule-out that location for a pin header (phyically impossible).
The only add-on connector capability is the COM port on mine. There's definitely a pin header for that.
I think you should look at the HP version. I believe that OEM board has the ports you want.
BTW, I know there are/were more versions of the board including one with a PCI riser slot. There's a space vacant on might for that. I thought the HP version might have that slot. When I get the chance, I'm going to take a peak at one.
Danno
Edit: When I was thinking about add-on connectors, I forgot there are also the 1394 and USB pin headers on the board as well as the front audio pin group. Sorry....

"SPDIF OUT" determination by LED connect

I have audigy 2 value and "4-pole mini-jack to 3 male RCA" cable.Can i find SPDIF OUT by connecting LED to RCA plugs ?

I think this is what we both need but i cant get anyone in creative labs to tell me whent its comming out in the states
I think there may be some confusion in this as we are introducing a new accessory with a similar name to the older Digital I/O 2 cards. There is a new I/O module that is being released that is what I believe you are referring to in the documentation. It has not been released in the US market as of yet, but is very similar to the breakout module that came with the original I/O 2 cards, and almost identical in appearance. This new product has update circuitry to work with the newer cards...
http://uk.europe.creative.com/shop/product.asp?prod=773

SpDif confusion...

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