192 kHz/24 Bit Audio Playback

Is there any DVD playback software which will provide 192 kHz/24 Bit Audio Playback, as very popular Power DVD provide only 96/24 and I can't use the full power of my sound card.

pushkir wrote:
Is there any DVD playback software which will provide 192 kHz/24 Bit Audio Playback, as very popular Power DVD provide only 96/24 and I can't use the full power of my sound card.
Iirc (did read on some review), the CyberLink Power DVD 6 supports 24-bit/192 kHz DVD-Audio. DVD-Audio @ 24-bit/192 kHz is mono/stereo only, as you propably already knew -->
DVD-Audio specifications
and the card you have there, propably supports 24-bit/192 kHz as mono/stereo only too.
jutapa
Message Edited by jutapa on 05-03-2006 01:23 PM

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FCP 4.5 24 bit Audio Playback

Hi,
I would like to know if FCP 4.5 can playback 24 bit audio.
While my files are correctly played in QT (7.1), in FCP I only got the image and a strong noise instead of the recorded audio.
Is it something I can do (except export the audio in 16 bit)?
Best regards,
Alex

pushkir wrote:
Is there any DVD playback software which will provide 192 kHz/24 Bit Audio Playback, as very popular Power DVD provide only 96/24 and I can't use the full power of my sound card.
Iirc (did read on some review), the CyberLink Power DVD 6 supports 24-bit/192 kHz DVD-Audio. DVD-Audio @ 24-bit/192 kHz is mono/stereo only, as you propably already knew -->
DVD-Audio specifications
and the card you have there, propably supports 24-bit/192 kHz as mono/stereo only too.
jutapa
Message Edited by jutapa on 05-03-2006 01:23 PM

Tape synchronization at high sample rates + recording at 192 Khz 24 bit

I was expecting at least that Apple would fix Emagix synchronization problem that occurs when Logic is slave to a tape recorder at sample rate other than 44.1 or 48Khz.
Further more I do get click , pops and digital noise when I try to master @ 192Khz 24 bit most of the time (not synchronized to any code).
For archiving tapes or going back and forth to tape I use Digi 002 , Rosentahl WIF (SMPTE to MTC) and Pro tools , as Logic is incapable for such a task , and for mastering and mix-down I use Lynx AES 16 with dCS904 AD and dCS954 DA with LogicPro.

Dear Mr. Logic 8 on Mac Intel.
I would like to thank you for you suggestions but :
1] The quality of LTC on my Otari 2inch/24Track is OK.
2] The LTC to MTC interface that I use with Logic 8 is Unitor 8 MK II.
3] My AD/DA convertors are considered to be la creme de la creme and I am pretty sure that they are OK.
4] The HD is an external la cie and has to be fast as I can bounce 8 tracks @ 96 Khz-24Bit using Pro Tools LE synchronized to tape.
5] The software is up to date and trust me Logic had serious sync issues several versions ago.
6] I checked and double checked even multichecked the prefs and settings of Logic .
It is interesting that Logic 8 locks @ 44.1/48 Khz but crashes @ any other rate.
I wonder if the guys of C-Lab (that left Emagic ages ago) are the responsible ones for the synch code and if so I wonder if they could help Apple with the synchronization.
Thank you again for even bothering to answer.
Any help would be welcome.

12 bit into 16 bit audio?

Hi there,
I have footage from a camera which, when capturing in FCP told me it was 12bit audio as opposed to the other footage I'd captured at 16 bit. On the FCP forum I was told that 48Khz was 16 bit and 32 Khz was 12 Bit...so I sent the audio from FCP into Sound track and highlighted all the audio and said "resample to 48KHz". I then relaid the track in FCP and I thought..happy days..but I'm looking along the time line in FCP and later on the mouths seem to be out of sync which makes me think it hasn't worked.
Does anyone know an actual setting to ensure that the audio is indeed set to 16 Bit from 12 bit?
Hope you can help (as always).
Thanks
Carol Ann

I have a similar problem. I got a tape from a client and the audio was in 12bit and not 16bit and FCP does not like it and then the audio drifts and becomes out of sync. It also appeared to be a huge pain even to figure out how to capture in FCP with 12 bit audio.
Here is my solution...
#1 - Confirm that the audio is in 12bit. Put tape in camera and play it back with it not connected to the camera. Turn the "display" on so it shows the details of the recording over the video. The camera should be able to show you the bit depth of the audio.
#2 - Capture the clips with 12 bit audio with Apple iMovie. Save out the project and quit out of iMovie
#3 - iMovie will save one file to your hard drive. This is actually a folder with the details of your iMovie and the raw footage. Right clip on your iMovie file and select "Show Package Contents". This will open up the folder and then in the "Media" folder you will see the raw clips captured.
#4 - open the clips in Quicktime Pro. Go to the "get info" and confirm the clips are captured with audio as "DV, Stereo, 32.000 kHz".
#5 - convert audio only to 16 bit 48 kHz You can do this with Quicktime Pro or with another sound utility.
a) with Quicktime Pro with your original video clip open select Window > Show Movie Properties. Then when that is open select the sound track and click Extract. This will open a new Quicktime window with only the audio file in it.
b) now "export" this new audio file as 48 kHz 16 bit stereo
c) open the new audio file and select all of the file with the play head then do a command "C" to copy it all
d) go back to the original movie file and with the properties window open select the audio file and click on the delete button at the top of the window.
e) position the playhead at the begining of the clip and select Edit > Add to Movie. This will paste your 48 kHz 16 bit audio in with the video track.
f) save this movie out and import this into FCP
Kind of a round about way of doing things but I found it pretty easy although cumbersome.

Does Airport Express support 24 Bit - 192 KHZ digital out ?

Hello.
I have Airport Express connected to a high end external audiophile DAC and would like to know if it supports 24 bit and higher clock frequencies like 96 KHz or 192 KHz.
Thanks for your thoughts.
Nuno

You may find the following article helpful in answering your question. (ref: http://www.stereophile.com/accessoryreviews/505apple/)

X-Fi Xtreme Audio - I/O Module - Mono in Bit-Matched playback m

I use the?X-Fi Xtreme Audio on my PC workstation. The soundcard is connected to my acti've speakers, and to this it'sis?working perfectly. But, my problem is this:?I also want to?use my PC to media purposes, Video playback, music playback from one room to the next. I baught the Digital I/O module, and connected my Denon Sound reciever to the Coax output. In the Creative Console Launcher, I switch to "Enable Bit-Matched Playback". It's working, but - the signal is in mono. What?is wrong?

Hi?I had the same problem, and here is the solution. ) Make sure u download the latest drivers for Vista. 2) After the installation u can goto the Control Panel> Sound Settings?You will see that there is 2devices listed. st is the Speakers configuration2nd there is the SPDIF configuration. Alter first the Speakers configuration to ur needs, enable the Digital out! > make sure u never select a frequency above 48khzAlter the SPDIF settings and test the sample rates that is given on the screen sothat u know what works with the card and not. If there are any problems leave the settings i gave exactly as instructed, and then goto urAudio Console and check there is a headphone settings and disable them. Try again thereafter!?Good Luck!Marvin?IT & Audio Specialist

X-Fi Xtreme Audio - I/O Module - "Mono" in "Bit-Matched playback" mode

I use the X-Fi Xtreme Audio on my PC workstation. The soundcard is connected to my active speakers, and to this it is working perfectly. But, my problem is this: I also want to use my PC to media purposes, Video playback, music playback from one room to the next. I baught the Digital I/O module, and connected my Denon Sound reciever to the Coax output. In the Creative Console Launcher, I switch to "Enable Bit-Matched Playback". It's working, but - the signal is in mono. What is wrong?

Hi?I had the same problem, and here is the solution. ) Make sure u download the latest drivers for Vista. 2) After the installation u can goto the Control Panel> Sound Settings?You will see that there is 2devices listed. st is the Speakers configuration2nd there is the SPDIF configuration. Alter first the Speakers configuration to ur needs, enable the Digital out! > make sure u never select a frequency above 48khzAlter the SPDIF settings and test the sample rates that is given on the screen sothat u know what works with the card and not. If there are any problems leave the settings i gave exactly as instructed, and then goto urAudio Console and check there is a headphone settings and disable them. Try again thereafter!?Good Luck!Marvin?IT & Audio Specialist

24/192 khz with DAC inside lighting to 30 pin

The DAC inside the lighting to 30 pin adapter is a wolfson series DAC that supports 24 bit / 192 khz playback. Since you can not upload these to your phone natively I was wondering if I got an app that supported uploads of these files to my iPhone 5 if it would play at the full 24/192 khz. I know this seems like a stupid question but a lot of apps say things that thy dont do, especially since I have no real way of telling.

Yes, we have DNS inspection turned on. I think what I need to work with the VCS server, is that my internal subnet on the inside interface of the ASA (192.168.225.0/24) need to access a NAT'd IP (1.2.3.0/29). The VCS server has an IP address which is in the outside interface IP range.
The internal clients are having issues connecting to 1.2.3.2 which is the VCS server.
192.168.225.x (Inside Interface Range)<=====>(Outside Interface Range)1.2.3.2/29
The internal hosts cannot connect to 1.2.3.2/29.
I didn't know if this was some security feature that didn't allow internal hosts to access the outside internface IP range or not. Currently all internal hosts are PAT'd to 1.2.3.1 (outside inteface IP)

Bugged audio playback? When I start working with more than 1 audio track.

Hi!
I have been using premiere elements 12 for editing FullHD videos.
Brand new for me just bought it like 2 weeks ago.
Now during the last couple weeks I have had this MAJOR proglem with editing the audio.
This problem is headache and turns my simple editing into almost impossibilites.
I have had this problem with all my projects. _ALL_
Usually the problem seems to appear when I want to start working with several audio tracks.
At somepoint my projects just get ruined, I simply cannot hear the audio tracks.
The audio starts to disapear randomly from all the audio tracks.
It is not only playback problem. When I exported the video, the audio was partly missing, same as the project.
I have tried to look help on forums and search all what I could find, but havent been able to solve this.
Small fix has slightly helped:
I export audio to single track and bring it back to project. This works but I still usually have more editing to do.
I simply cannot understand why the playback disapears, I fear it is a bug.
I found lot of earlier version problems on people with similar type of problems, but no help for this version of program.
To solve the matter I have changed my audio setting in PE12.
Tried Premiere Elements WDM Sound and ASIO4ALL V2, but same problem on both.
I have looked up that my audio playback device and PE12 matches, and are not muted, and volumes maxed.
I have deleted my C:\ appdata\roaming\adobe premiere elements & premiere elements 12 folder preferences.. not helping..
I press the update pretty much everyday hoping for new version.. But yes, program keeps telling me its the latest.
I usually only work with 3 audio track maxinum. And I am doing really the most basic stuff here..
I have set all my audios to foreground and it has given me best results, but it does not seem to take the problem away as at somepoint randomly my audio playback is missing again, randomly across the playback.
Please anyone? I am feeling hopeless here.
Info on my computer:
Operating system: Win7
Processor: AMD Phenom(tm) x4 955 processor 3.20 GHz
memory (RAM): 4GB
system: 64bit
<moved from downloading, installing, setting up forum by mod- kglad>
Message was edited by: kglad

First of all, thank you for replying my message!
I will go and try to answer best I can.
Source media properties:
Progressive frame rate, I record this from a game.
Recording program: dxtory
codec: Lagarith Lossless codec
Format: avi
Frame width: 1920
Frame height: 1080
Data rate: 1990656kbps
total bit rate 1992192kbps
Frame rate 30/frames/seccond
Audio of video properties:
Codec: PCM
Format: 16bit, stereo
Bit rate: 1536kbps
Channels: 2(stereo)
Audio sample rate 48 kHz
In addition to that I use wav files for music and narrate
Music files sizes 20-50mb
bit rate: ~1411kbps
In total sizes of audio involved is usually something like 300mb
What are you or the program setting as the project preset in the new project dialog. See Edit Menu/Project Settings/General and its Editing Mode, Timebase, and Frame Size readings.
I opened new project and this is what i got:
Editing mode: HD 1080i
Timebase: 25,00 frames/second
Frame size: 1920 x1080
Audio is 48000 Hz
(I dont know how to change these, btw. Program don't seem to let me change editing mode.
I also looked up my most recent project and found these settings:
Editing Mode: DSLR
Timebase: 30,00 frames/second
Frame size: 1920 x1080
..and same audio settings
Do the problems persist if you start a new project? Do you find the audio problems develop when the Timeline content is fairly well developed or can the problems occur when there a minimal amount of content of those tracks? Let us look at the size and duration of this project.
Yes, I have had this problem with all my projects. Sometimes it goes away tho, just enough for me to work bit more. My work progress usually starts by clipping the video material, like 1h material to 8minutes in clips. After that I want to work with audio tracks, and here problems start! When I start working with more than 1 track, so the problem is at later stage. To fix the problem a small fix I have used, I export audio like music and effects tracks together to 1 wav and bring it back. After that, I start working with narrate track. To describe the problem with audio:
Audio Randomly starts to disapear when I playback, and can also be gone if I export the video out.
My source videos total size are usually within 80Gb max limit and audio edit material 300mb total.
In my last project my biggest source video was 47GB and leingth of 1h 4m, in addition to that I had 3 smaller videos, sizes of 1GB, 1,54GB and 300MB.
Is there any evidence that you are accidentially moving the audio track rubberband to alter the audio volume. That rubberband is the orange line that runs horizontally through the middle of the audio clip when you are looking at the wave form display style of that audio track. The rubberband represents audio volume.
No, audio volumes might have been adjusted in my tracks but they are present. The problem is usually like ~2-5seconds, a random audio track can be heard (like music or narrate) or just no sounds at all. This makes it very difficult to work with the audio. Also I gotta say that this problem starts by simply having the 2-3 audio tracks.. In my recent project the only adjustment I used to edit the audio was fade in/out.
I don't have any other audio problems with any other programs.
I am more than happy to provide more information if needed. I hope this information helps.
In addition I can say that when I close Adobe Premiere Elements 12, it crashes. No idea why.
I have also looked up my firewall settings, anything involved adobe and premiere..
In addition some effects like HD text templates make "Adobe Media Core CS6" crash. I tend to just not use em, or if I need em, I add em at end of the edits.
I hope this helps, I would really like to solve this problem.
Thank you alot for taking the time to answer! I welcome anyone to help mee too! Please giev links to similar problems too. Thank you everybody!

Does iDVD support 24 bit audio?

I have a project that calls for impeccable audio quality. The client has supplied me with 24 bit aiff files.
The files import easily into Final Cut Express and when placed in the a 1080i60 timeline show no sign of needing a render.
Does this mean that I will retain the 24 bit quality all the way through export to authoring of the DVD in iDVD? Does iDVD compress the audio in such a way that it will no longer be 24 bit?
I can find little or no information about this particular question in the manuals or the discussions.
Thanks in advance.
Mike Pulcinella

Apple has never said much about the audio encoding in iDVD.
'Traditionally' it was just two channel uncompressed PCM audio; DVD SP (with Compressor) gives several other choices (including Dolby Digital).
Toast gives you the choice of uncompressed PCM or Dolby Digital as well.
Here is what the Toast manual has to say: Audio Format: Choose Dolby Digital or uncompressed PCM. In almost all cases, Dolby Digital is the best choice. It uses less disc space, so you can fit more video on the DVD, and the sound quality is almost the same as PCM.
Data Rate: Choose the bit rate to encode the Dolby Digital audio. Higher bit rates can produce slightly better sounding audio, but use more disc space. Toast uses 192 kbps by default.
Dynamic Range Compression: Enabling dynamic range compression reduces the range between loud and soft sounds in order to make dialogue more audible, especially when listening at low volume levels.
PCM audio should be used if you want to maintain 100% fidelity and only if the audio source is already uncompressed PCM, such as tracks on an audio CD or a high quality audio recording. If the audio files are already compressed (such as MP3 or AAC songs in your iTunes library) do not use PCM—it will not improve the sound quality.
Toast records PCM at 48 kHz / 16 bit levels or 48 KHz / 24 bit levels, which are higher than standard CD quality. Audio that has been recorded at 96 kHz / 24 bit is downsampled automatically unless you choose the 96 kHz / 24 bit option, which maintains the higher quality but significantly reduces disc space.
When you choose PCM, any existing Dolby Digital audio files will be “passed through” without reconverting to PCM.
Hope this helps.

Looking for 192 KHz

Alright so I have the K8N NVIDIA nForce4 Platinum/SLI socket 939 motherboard and it has the built in creative soundblaster and in the manual it states: 24-bit / 96 - 192 KHz audio quality. When I first hooked it up to my receiver I thought for sure that my receiver said I was getting 192 KHz through my coaxial connection. Now, no matter what the settings are on my PC, I can only get 96 KHz, whether I play an mp3 or an AC3 or anything. If anybody can help me get 192 khz or the maximum audio quality possible, that would be awesome.
Thanks a ton
Tim

I'm running an audigy2 pci card in my system and the max O/P to the SPDIF coax is 96 KHz.
I'm fairly certain that's the max you're going to get out of onboard SB (I'm not even sure that the new SB card will OP higher than that).
Still sounds pretty damn good running at 96 KHz though....

Downsampling to 96 KHz from 192 KHz

Hi all
I need to downsample am audio file from 192 KHz to 96 KHz. Ideally I would have to apply low pass filtering at 48 KHz, half of the Nyquist limit. However the lowpass filter in soundtrack pro limits at 2o KHz. Any ideas on how to overcome this?
Thanks

If you're looking for dithering, Apogee's UV22 algorithm is build into STP. It can only be selected from the file editor > Save as. It becomes available when choosing a lower bit rate.
http://www.apogeedigital.com/products/uv22hr.php

Do New iMacs support 24bit / 192 kHz sound?

I have a new 27 inch iMac from Oct 2010. I use optical out to a receiver that supports 24 bit / 192 kHz sound decoder. I want to know if I play a sound file that is 24 / 192 will the iMac be able to output this to be decoded by the receiver.

Ok, I don't know what that is. Normally it only goes up to 24 bit. Also, I just realized, how many channels and what bit streams like DD 5.1 etc does it support to get the audio out to the receiver?

Audio playback stops short

Has anyone had problems with audio playback in a previewed or published presentation where the playback stop short and advanced to the next slide?

Brian,
The fact that this mp3 is about the same size as a standard 16 bit/44.1k stereo file of a 3 or 4 minute song, indicates to me that it's an extremely long file.
I'm willing to bet you have run into the measure limit within a Logic song. It's 2000 something (can't remember it exactly).
The standard work around is to drop Logic's tempo to something really slow, thus allowing the larger file to be able to play within the Logic song limit.
See if that's it...

Does AppleTV(v2) output 24-bit audio through TOSLINK port?

I have some high-definition audio files (24-bit/44.1KHz and 24/96) in Apple Lossless format. I also have what I thought was a simple question: Does the AppleTV support bit-perfect playback of such files? In other words, will the AppleTV stream them without downsampling them? Posts on other forums suggest that the AppleTV downsamples everything to 16/48 (which would be absurd on Apple's part, as every other link in the audio chain, including iTunes, Mac, DAC, etc fully supports 24/96). Can anyone confirm either way? The technical specs are conspicuously silent on this issue...

Patrick Dubby wrote:
Aaaargh, I was thinking of getting one of these as a more convenient way of getting 24/96 sound out of my macbook than a long TOSLINK cable to my amplifier.
Looks like another wasted opportunity.
Yep
What I don't understand is that according to other reports, digital audio from a DVD is passed through no problems. Why would a 5.1 digital signal be untouched, but a similar signal from iTunes be screwed up?
Personally I think they don't see beyond the iTunes store purchase/rental model.
As you know you can't even buy true CD quality audio from iTunes let alone anything better.
When we're able to download/buy 4GB HD movies, I see no practical reason why we couldn't have CD quality or better encoded as Apple Lossless audio.
There must be some sort of workaround?
No. I ended up buying a Mac Mini.
The majority of other media boxes I looked at previously also were limited and even though some supported FLAC they couldn't output 24/48 ot 24/96.
I'm sure some of them must do it now, but it's a shame AppleTV can't - I doubt it's a hardware related issue, simply one of support.
And why, in 2011, is 24/96 still considered 'audiophile'? After 30 years of 16/44.1 you'd think we would have moved on somewhat to a new standard.
Truth being most people don't care. MP3/portable players have killed many people's appreciation of what good quality audio sounds like. I use these devices all the time on the go, but for home listening I do not want lossy compressed audio to play on the hi-fi, and in this day and age we should be able to ditch the silver discs easily.
AC

192 kHz/24 Bit Audio Playback

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