SPDIF Input Setti

My short question is "Is there any reason why the '' (in the SPDIF I/O tab)?in the Creative Audio Console would affect audio output generated by my PC?"?My problem is that my DD Optical Input (XBox 360) will only play smooth audio when the Input Settings is set to "SPDIF Bypass", but my PC won't output any audio in that setting. I don't understand why input settings should affect audio output. Is there any way?I can play back audio from my PC (connected via optical cable to an external DD receiver) AND from my XBox 360 (connected to the optical input on my Audigy ZS) without making selection changes to go back and forth?

SolusCado wrote:
I'm still not following. I understand why the SPDIF settings affect the XBox 360 signal - and it works as expected (though if I set the soundcard to NOT run the 'SPDIF Bypass', the 360 signal still seems to get output to my receiver, but with an extremely low center channel (or perhaps no center channel at all) - but only for a couple minutes, and then it stops altogether). What I DON'T understand is why SPDIF INPUT settings would affect the audio generated by the PC itself (such as Windows sounds, DVD playback, etc.). And I have a run of the mill Kenwood Dolby Digital Recei'ver... Not sure what exactly you're looking for...
Because Creative soundcards cannot CREATE sound in DolbyDigital. If you take a surround DolbyDigital signal from your XBox, decode it in the soundcard (so NO bypass), it will be outputed on the anlog outs as surround sound AND on the SPDIF out as stereo PCM. It cannot be reencoded back to surround Dolby Digital. All other sounds created by PC will be stereo PCM via SPDIF.

Similar Messages

Clipping problems using SPDIF input on SB Live

.Clipping problems using SPDIF input on SB Li've?Hello everyone
I'm new to this community and I'm banking on someone being able to help me!
I'm connecting a DAT player to the SPDIF input on my Soundblaster Li've card and recording using Audacity.
Within Creative Audio HQ I've got the SPDIF volume level set to 00% (for starters, perhaps someone can explain why there is a volume control when I thought this signal was staying in the digital realm?!) and when I record in Audacity the resulting material clips with high transient sound... a piano, for example, distorts on the louder notes.
I need to take the level down to about 70% (or the equivalent in Audacity would be 0.7 - the 2 'volume' controls are connected) to avoid any clipping.
I've had the sound card years and used to use SoundForge without any problems. Any ideas why this is happening in Audacity?
I'm using XP and?I have downloaded the latest drivers and turned off the MOBO onboard sound in my BIOS.
I would be grateful for any assistance anyone can give.
Best Wishes
Steve

paush wrote:
Hi, I am trying to sample my old vinyl record collection using my Sound Blaster Li've! PCI (think that was the model - it was the original (not Value) version of the card with the seperate header for the coaxial digital and MIDI I/O). Unfortunately, I get severe clipping regardless of the recording volume on the Volume Control panel. I know that my photo pre-amp is capable of high voltage outputs and therefore wondered if anyone was aware of the sensitivity of the analogue line input in volts of this particular model? Many thanks.
Regards, Paul.
If your pre-amp is a phono pre-amp (with RIAA in-build), I suppose it's not the culprit for clipping.
How have you set the recording and source (input) levels on Volume Control and Recording Control?
Try adjust the recording level by the 'hights' (loudest outputs) of the record (you know those recordings so you can easily find those places (or you can look on vinyls surface where the darkest areas are --> loudest outputs)).
jutapaMessage Edited by jutapa on 0-23-2006 :38 PM

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.

X-Fi I/O Drive SPDIF input stopped working - driver issue?

HI have an X-Fi Platinum card with the I/O Dri've module that I have been using for some time now to accept digital input via digital coaxial cable from my Alesis Multimix 6 music mixer to mix down and record MP3s from my hardware music synthesizers. I am running Windows XP SP3. The SPDIF input has always worked like a charm until I updated the driver via Creative AutoUpdate. The SPDIF input no longer registers any signal, but it still shows in both Creative Volume Panel and Windows Volume Control. Default sound input and output is set to X-Fi (D000). I tried uninstalling the new driver and reinstalling the original driver, and the SPDIF input is still dead.
Is this a known driver issue? Any idea on how I can get this working again?
Thanks, all.
Larry

I/O dri've now works with this software?http://forums.creative.com/creativel...essage.id=7369?which includes the 3/5/07 driver. Steve

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

XFI Xtrememusic Optical SPDIF input problem

Hi everyone,
I am unable to utilise the optical input of the XFI to decode Dolby Digital / DTS streams.
Setup:
PS3 to Creative Digital I/O dongle via optical cable.
Creative Digital I/O to XFI Xtrememusic via provided connector.
PS3 playing a DVD with both DD and DTS soundtracks.
Problem:
No sound at all, no matter whether I choose DD or DTS.
If I set my PS3 to output only PCM audio, there is stereo sound heard, so I can confirm the physical connections are good.
I am unable to select any "SPDIF Input Settings" options in either the Creative Audio Console's "SPDIF I/O" tab or the Creative Volume Panel's "Digital I/O" section.
. Audio Console's drop-down list is greyed out and not selectable.
2. Volume Panel's "SPDIF Input Settings" drop-down list is not even visible. (i.e. there is nothing displayed to the right of "Digital Output Sampling Rate")
Steps taken to try to resolve issue:
. Checked Creative Website for latest drivers and applications and installed them.
2. Tried to follow the electronic guide/manual but could not choose "SPDIF Bypass" due to the missing "SPDIF Input Settings" option.
The online flash tutorial (Beginner's Guide -> Digital I/O) shows these settings but it is just empty for me on my computer.
Can anyone please tell me how I can get my Xtrememusic to decode DD/DTS via optical input? I've searched everywhere and found no answers. Thanks in advance!

http://forums.creative.com/t5/Sound-Blaster/Cant-get-5--via-Digital-In-Windows-7-64-bit/td-p/552895

SPDIF Input on the S-Bracket (mod)?

Hey all, I'm back!
Is there a way to modify the S-Bracket so that the coaxial port becomes a SPDIF input instead of an output? I wish to run my Audiophile 2496's SPDIF output to it so I can use the onboard sound chip to give me a headphone mix of stuff I work with. It will save me buying a seperate headphone amp. Is there any way to do this at all, or perhaps make my own connecter that uses the SPDIF IN and GROUND pins on the motherboard? I'd like to modify the S-Bracket and I'm sure by sawping a few wires round it's possible - just which wires do I swap/remove? I assume the power ones are for the Optical connector, so I should remove them, and then swap the SPDIF output wire with the SPDIF input wire - it seems too simple to be true. It's silly that the motherboard has this feature but MSI don't give you access to it. No fair! Help!
Regards,
Neil.

Hey guys. There is a SPDIF IN pin sitting there and I also see an option to choose the SPDIF input in the recording mixer in windows and there is a level setting for SPDIF in the playback mixer (this is for the input aswell, not the output - the output would be controlled by the main volume control)! It's there. I'll be trying some jiggery pokery with the wires and I'll get back to you. As you may or may not know, the SPDIF connection only uses 2 pins - data and ground. Swap the data wire to the SPDIF IN pin on the board and that really should sort it. I won't be using the optical port so any problems it'll create for that won't be an issue. I just want input on the phono end. Have people tried this mod? You can't say it won't work unless you've tried it.
Regards,
Neil.

SPDIF input suddenly unlocking

Hi, I'm new to this forum having lurked with interest.
I have a problem with SPDIF input on a M-Audio Audiophile.
While trying to bounce in Logic, the digital signal from an XL-1 Turbo will cut out for just a second and ruin the pass. When I monitor through a Sony CDR W-33, I'll see the "Din Unlock" message corresponding to the brief drop-out. I've set external sync on the M-Audio driver software and wondered if something was missed in the Logic setup.
Still waiting on M-Audio's tech response.
Thanks in advance and yes, I'm saving for a better I/O.
Regards,
Jon

In the past it handled input from a dbx 386 with no problems but I don't remember if that was before and Audiophile driver update .
Might I ask if this problem could be specific to a devices (Emu XL-1) SPDIF output signal (it's manual yields no details on digital out)...

SoundStorm nforce2 SPDIF input

Hi all...
I'm wondering if anyone has had any experience with the SPDIF input on the Soundstorm on the MSI ILSIR nforce2 chipset.
I've used the SPDIF out before, and it works flawlessly.
However, I want to use the SPDIF in, to connect my second soundcard, an SBLive, to the SPDIF input on the S-Bracket pinheader.
The problem is, is that I'm using the S-Bracket for the 5.1 channel outs, so I can't remove the S-Bracket's connection to the motherboard (which I need to do, to connect the SBLive's pinout to the motherboard). The stock connector on the S-Bracket, as you all know, takes up all the pins on the pinheader, including the SPDIF in... but yet, it provides no discernable way for someone to connect anythign to the SPDIF in, unless you mangle the default plug, and solder or connect the SPDIF in wires some other way.
Is there some other way to use the SPDIF in, or am I going to be stuck with having to mangle the S-Bracket connector to do it? If I do this, am I goign to cause any problems with the terminals on the S-Bracket?
It's funny, because on the S-Bracket plug, there are clearly wires that connect to the pinheaders on the mobo, but cna't see where they connect to in the S-Bracket...
Thanks...

Oh, you were asking how I would get both soundcards working on the board! This is simple. You just plug the soundcard into the board. Windows recognizes it, as well as the SoundStorm APU. Then you set your "preferred device" in the Windows Sounds Control Panel to the Nvidia Soundstorm. Then, you connect your LINE OUT from the second soundcard to the LINE IN on the Soundstorm. The LINE IN on the Soundstorm then becomes the volume control for the second soundcard. I'll give you the reason I want to use the SPDIF in feature...
I use this online service called eatsleepmusic.com for streaming high quality Karaoke. Anyone who likes Karaoke should SERIOUSLY check it out, it ROCKS. Only problem is, the Soundstorm Mic-in SUCKS, so I have to use the SB Live for my Mic input, then connect the SB Live output to the Line IN on the Soundstorm. This ALSO allows me to control a separate EFFECT for the MIC on the SB Live compared to the EFFECT on the Karaoke song, which is processed as a WAVE stream on the Soundstorm. So I want to OUTPUT the SBLive MIC sounds with the SPDIF connector TO the Soundstorm instead of thru the LINE OUT / LINE IN method, to avoid background analog hiss.

PreSonus Firebox SPDIF input issues

Hello all,
I'm hoping you folks could help me on an issue I've been having with getting the input chain correct to record a SPDIF input signal. I've searched this forum with no other postings found. I've also emailed PreSonus tech support without much luck. The emails I sent are posted below. I'll refrain from posting the replies as they are the property of PreSonus. I'm not looking to bash them. I just want to get on with recording. Please read through them if you feel you could be of help to me.
Thank you,
1st email...
Hello,
I'm an running the following equipment;
PreSonus Firebox
Logic Pro 7.1
Apple 12 inch powerbook running OSX Tiger
Focusrite Voice Master Pro
I recently purchased the digital A/D card with SPDIF for the Focusrite Voice Master Pro (preamp / channel strip). For the life of me, I have not been able to figure out the recording chain for the digital SPDIF input coming from the Focusrite Voice Master Pro. The Firebox manual shows a selection is needed for Windows XP within the Control application. However, non exists for OSX. I'm not a new user to DAW systems. I recently upgraded from a MOTU setup and used SPDIF to / from a DAT recorder without issue. Considering how simple it was to use digital recording with that set up, I know I must be just missing something here. But, after a week of reading the manuals and reading the Logic online forums, I'm still stumped. Please advise and help.
Thank you,
In the reply, I was told I had to download the control app for OSX and select inputs 5-6 within the logic channel strip. I did so and still no go.
2nd email...
Thanks for the reply. I did download the app and tried to use 5-6 input without luck. No signal, bummer. I guess I have more than one issue? I'll have to do some testing by changing cables and running the signal from my DAT player instead of the Focusrite. Maybe I have a bad A/D card on the Focusrite. I'll try running the Focusrite into the DAT as well. Is there any way to test and/or monitor the SPDIF on the Firebox (without Logic) for sync lock and signal? I tried installing the Cubase software included as a test, but I get an error message. So, no go there.
In this reply, I was told I should download the Firebox Mixer and a patch for Cubase. The mixer came with the unit so I really don't see the need to download it unless there's an update. I really don't have a need for Cubase. I just wanted to install it in order to use it as a test.
3rd email...
Hi,
Good news... I was able to send output and record to my DAT player via SPDIF. And, the Focusrite also recorded to my DAT player via SPDIF.
Bad news... No input via SPDIF via the Firebox.
I see that there are many options to route the output signal though the mixer, but no "real" input chain settings. There's a lack meter or peak monitoring on the mixer or the unit. There's no way to be certain the correct sample rate is set and/or received so that I can be sure a digital lock is established. I set Logic to input 5-6. Nothing. I can only assume that when a SPDIF signal is received, it's automatically set to 5-6. Is this correct? If so, something must be wrong with the unit. Perhaps you can tell me (or send me a screen shot of) the correct mixer setting I should be using so that I may test the Firebox.
To be honest, I'm about to trash this unit. I've spent several hours trying to figure what's wrong here. I find the documentation woefully inadequate. It's integration lacking. The fact I had to download the undocumented control software to get the SPDIF capability that was to already to be included with the unit as advertised was not what I was expecting. Partially against other manufactures products I've used. I do appreciate your help in this matter. At the same time, if I have to spend much more effort on this, I'm more likely find myself buying a better box. It will not be a PreSonus.
I sent that email 2 days ago and never got a reply.
I really don't want to buy something else when I'm likely to be just missing something. Please help.
Thanks,
Tim
G4 12inch Powerbook Mac OS X (10.4.4)
G4 12 inch Powerbook Mac OS X (10.4.4)

Well.... I did download that control app and I finally got the SPDIF input working. That's a different version than the 1st control app I was told to download just days before. WHATEVER. I'm a happy guy now. Wheee!!! The funny is/was that's the same download the presonus tech told me was the the mixer app. That's why I didn't download it right away. The 1st couple of times I used it caused some audio shuttering, but then that has gone away. Hopefully, that's not a new issue.
Good luck to all.
Tim

NI SPDIF Input Module

Has anyone tried to use the NI SPDIF Input Module? I already checked the document, but still, I am having troubles making it work. It doesn't read a 48 KHz signal with a 7831R board.
Thanks for any suggestions.

I have two suggestions to try.
In regards to developing your own SPDIF reader; when you only measure the pulse lengths you get the proper readings, but as you add more code to the loop the readings changes and become incorrect. This is most likely a result of the extra code slowing down the loop and causing it to not run fast enough to read each pulse. Using a regular While loop in LV FPGA, each function block you use takes up at least one FPGA clock cycle. In addition the While loop has an overhead of two FPGA clock cycles. So assuming you have a While loop with 5 consecutive functions it will take 7 clock cycles to run each iteration of the loop. If you were to use this loop to measure the pulse length it would give you a measurment with a resolution of 7 ticks, i.e. the pulse length would be 7, 14, 21, 28, etc. As you add more code to this loop the length of each iteration increases and the resolution of the pulse width decreases. In order for us to adequately measure the width of the pulses in the SPDIF signal we need to have a loop iteration time of less than 6 or 7 clock cycles (about half of the shortest pulse we want to measure). This is the reason why the original SPDIF code is set up in three different loops and the first loop that measures the pulse width is kept very simple. It also uses pipelining to further reduce the time of each iteration and we compile the FPGA for a 80 MHz clock rate to get more clock cycles per pulse. I think in your code as you add the Case structure and additional code, you are increasing the loop time above the necessary limit.
One 'easy' solution to this problem is to use the Single Cycle Timed Loop (SCTL) instead of the While loop. This loop is processed differently by the compiler so that all of the code inside the SCTL is executed in one clock cycle. (The SCTL was not available in LV FPGA when the original SPDIF code was developed, otherwise we would have used it then.) So in the updated SPDIF code we use the SCTL and can include all of the SPDIF processing in one loop. There are some limitations on the code that can be included in a SCTL. Information about the SCTL and optimizing FPGA code can be found in this prsentation (Advanced FPGA Programming). I think if you convert your code to the SCTL (right click on the While loop and chose Replace with Timed Loop) you should be able to measure your signal correctly.
Getting back to the SPDIF code I included on my last post. This code does work correctly on my end with a SPDIF signal coming from a Soundblaster sound card. This signal has two raw audio channels with 20-bit data per channel. The 4 AUX bits in the SPDIF packet are not used. I have logged the data to file and can load the file using the Reader VI included in my last post. To play the sound back on a regular sound card in LabVIEW I have to resample the data to 44.1 kHz and then use the upper 16-bits of the 20-bit audio signal. I'm attaching a sample file of logged data on this post that you can open and play back using Read SPDIF Data File.vi.
Is there a chance that the audio data in your signal is formatted differently? Could it be using 5.1 or 6.1 encoding with DTS or Dolby?
Christian Loew, CLA
Principal Systems Engineer, National Instruments
Please tip your answer providers with kudos.
Any attached Code is provided As Is. It has not been tested or validated as a product, for use in a deployed application or system,
or for use in hazardous environments. You assume all risks for use of the Code and use of the Code is subject
to the Sample Code License Terms which can be found at: http://ni.com/samplecodelicense
Attachments:
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Delta R-Bus spDIF input no longer works under Tiger?

The spdif input on my M-Audio Delta R-Bus card works flawlessly in Logic 4.8 under OS9, but does not work in Logic 7.2 under OSX Tiger.
I spoke with M-Audio Support but they said that the Delta R-Bus is mothballed and I should get something newer They didn't even want to bother to test it.
Any other Delta R-Bus users out there who could confirm or deny this fault?

Thought you could use this info about getting FireWire to work!
First, you should always Repair Permissions in Disk Utility before and after any update; I would do that now. If you installed your update with FireWire plugged in, your Mac may not recognize it anymore.
Try resetting your FireWire ports by disconnecting all FW devices, shut down your Mac and unplug it from AC power (wall socket) for 30 minutes. plug it back in and try FW.
If that didn't work, download the combo update from this site (yours may be corrupt), not Software Update, disconnect all firewire + USB devices except keyboard + mouse, Repair Permissions, re-install update, Repair Permissions again + try.
If that still didn't get it Zap the PRAM. While restarting Mac hold down the following key combo Command-Option-P-R. Keep holding those keys down till you hear the second start-up chime, then release.
For those of you with Macs that are not PowerBooks or iBooks, a bad internal PRAM battery can cause FireWire to not be recognized, so make sure it's good! Also, here is Apple KB HT1317 on getting FireWire to work.
Here's another Fw Faq.
DALE

Dictation will not respond to selected input setting

Dictation will not respond to selected input setting. It was working fine until the last update now this problem has arisen. I have to do a workaround through sound and select internal mic directly from there which I never had to do before.

If you updated to 6.1 & the problems began then - you might have Glims installed...
How do I uninstall Glims? | www.MacHangout.com
If not, test briefly in a new or Guest user account on your computer.

Audigy 2 zs spdif input bracke

Hi all,
I have a standard audigy 2zs pci card with the extra pci bracket that has digital din, midi in and out and coaxial spdif in and out. Im now in need of an optical spdif input and was wondering if I can get a different pci slot bracket with this input ?
Really hoping I dont have to buy a whole new soundcard followed by the drivers - drivers - drivers routine !
Any help would be great,
Mick

you've on every standard audigy2(zs) card a sp/dif digital in on the card itselves.
It's the CD-IN digital(or CD-sp/dif) input on the card(internal). So you only need a wire.
Get a rca panel connector, mount it on the front of your compu-case and connect(solder, so remove connector on one side of the cable and solder it on the rca panel connector(outside=- inside=+) a 2-pole internal sp/dif audio cable to it. Connect the other side of the cabe (with original 2 pole sp/dif internal audio connector) to the cd-sp/dif connector on the card itselves and you're done for say 6 euro(rca panelconnector+2-pole sp/dif internal audio wire)
then you've a sp/dif digital audio input on the case of your pc, and you can adjust it in the surround mixer with the CD-Digital volume slider...

SPDIF Input Setti

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