Keeping high audio quality, despite Pr Pro Audio Conforming

I'm working on a film project, which will be edited in Premiere Pro.  I am recording most of the audio at 24/96 .wav files.  There will however be some audio files recorded at 16/48 compressed, coming from different sources.  This is unfortunately unavoidable.  SO, upon digging in the Adobe help sections, I found that if I start my project and set the sequence settings for audio to a sample rate of 96 (or 96,000), Prem Pro WILL NOT CONFORM my 24/96 .wav files.  (Pr Pro will not conform uncompressed audio of a supported format if it is brought into a sequnce where the sample rate of the sequence is the same as the clip.) This is great.  My 24/96 files will hold their quality all the way through the process.  It will however conform the other files such as the 16/48 compressed files.
So here's my question.  (Keep in mind that my final product will be exported to 24/96.) I can take all my 16/48 audio files and convert them to uncompressed 24/96 before bringing into Prem Pro and then Prem Pro won't conform them at all.  So what will yield the best audio quality - a 16/48 audio file that has been conformed by Prem Pro and then during the export done by Prem Pro converted to 24/96, or will the audio quality be higher if I convert that 16/48 audio file first to 24/96 and then bring it into Prem Pro, where it won't be conformed by Prem Pro at all?  In other words, for best audio quality, do I let Prem Pro conform my 16/48 files and then convert them to 24/96 during export, or do I do the 16/48 converstion to 24/96 in another program like Audition?
(((And yes, the higher audio quality matters.)))
Thank you

matt_gh2 wrote:
I'm working on a film project, which will be edited in Premiere Pro.  I am recording most of the audio at 24/96 .wav files.  There will however be some audio files recorded at 16/48 compressed,
Thank you
If these files are .wav files then they aren't compressed just a lower sample rate and bit depth. There shouldn't be any difference in quality between the Premiere up converted or the Audition up converted files, although I would trust Audition a bit more than Premiere to do an accurate conversion.
Not really sure why you need to work at 96K, it just uses more resources but it is worth keeping the sored files as at least 24bit. Since Audition works at 32bit floating point bit depth as it's native format it may be that your saved files from Audition are 32bit and that is what Premiere is choking on.

Similar Messages

  • Audio quality in DVD pro

    I just brought a file from FCP through Compressor into DVD Studio Pro. I have a music track for Menu 1 as well at the audio which came out of Compressor from my original file. The music for Menu 1 sounds fine, but the music of the movie when played back in Simulator is weak and muted. It sounded much better in FCP; I have to raise the volume substantially just to hear it in DVD SP.
    Thoughts?
    Thanks
    Shelley

    This will almost certainly be down to the way you encoded the audio. Go back into Compressor and check the settings...
    When you go into Compressor the encoding profile for the audio (AC3) will probably have a 'dialog normalization' setting of -27db by default. This will attenuate your audio and cause it to sound more quiet than in FCP. To overcome this, change that setting to -31db, which will ensure there is no attenuation and the sound coming out should then be the same as the sound going in.
    Then flick through to the preprocessing tab where you may see the profile set to film standard compression - change that to be 'none'.
    These two changes should help ensure the sound coming out is nearer in volume to the sound going in.

  • Melodyne "Audio to Midi" versus Pro "Audio to Score"

    Would anyone be able to comment on how these two programs compare? I am looking to use audio (monophonic) recordings of a marimba (percussive, tuned instrument) to create a Score and/or Midi tracks. I understand that there will be editing, but wonder if anyone has the experience to tell whether one program works better; is more accurate; easier to use/edit than the other.
    I also do a lot of compositions in 12/8 and 6/4 and wonder about whether I'll run into problems as a result?
    New to this Message Board and thankful for so much help already!
    gpeckgolf

    Hi!
    I would also like to know!
    Also, I found this app. which has recently released an AU mac version, so I'm going to try it out.
    http://www.widisoft.com/english/mp3-midi-products.html
    I find Audio to Score a bit hit & miss as far as velocity is concerned- I'm converting live drums to BFD.
    And Melodyne...I need the studio version for multitrack drums.... I would like it to be cheaper!
    As usual, I find myself going back to Beat Detective to quantise, so if I could find a quantise and convert app.
    ... perfect!
    Macbook Pro 2.16, G5 iMac   Mac OS X (10.4.8)   Logic Pro, Final Cut Studio

  • Adobe Premiere Pro audio distortion

    Hi,
    I have Adobe Premiere CS2 and some audio distortion problems. I captured a footage from my Sony DV Digital-8 camcorder. It's a standard DV footage with 16-bit PCM audio. When audio is nearly-peaking (not peaking), it gets distorted in Premiere, but doesn't in other applications (VirtualDub, Media Player Classic, Windows Media Player, mplayer). I'm sure this is because an internal conversion (called conforming) from 16-bit PCM to 32-bit floating point PCM. Project Setting of the Audio Track is 48000Hz 32bit floating point - Stereo for a DV project, despites DV standard supports 16 or 12 bit Audio tracks. In user manual they write the following:
    "For maximum editing performance and audio quality, Adobe Premiere Pro processes each audio channel, including audio channels in video clips, as 32-bit floating-point data at the project’s sample rate. To do this it must conform certain types of audio to match the 32-bit format and the project sample rate."
    Well, if I'm a professional video editor, I can (and have the right to) decide whether which sample rate and bit-depth are the best for me, but they don't let me set audio other than the default, there is no option like bit-depth available not even when creating a NEW project. No problem if Adobe's genius developers thought 32-bit mixing provides much more quality, but if they forget to implement 16-bit to 32-bit internal conversion correctly, please let me decide to use at least a fallback method.
    Thanks for your help
    stringZ

    Just looking at the waveform confirmed my initial reaction: Way, way
    oversteered. Everything is clipped and your best approach is to
    reshoot. That some players can handle this seriously mistreated audio
    may be caused by audio limiters. They should, otherwise you would blow
    up your speakers.
    Well, you can say that and if this sample was a music maximized with compressors, it would distort too. Sony Digital-8 camcorder has an internal limiter, so only soft-peaking can occur, but no serious distortion. Media Players (eg. mplayer) don't have limiters unless you specify an external limiter filter. I've already told this isn't the clip you want me to re-shoot. This is a sample I created to show you the problem. I sent this to one of my friends who has Ulead MediaStudio Pro 7.0. It doesn't distort there either. Have you tried loading it to an audio editor like Audacity or Adobe Audition? You can see no hard-peaking or distortion if you zoom in the waveform.
    Waveform at 00:00:00:21. No distortion, only high levels can be seen on this waveform.
    Waveform after 00:00:00:21. Soft peaking in right (bottom) channel.
    I'd appreciate if someone took this seriously instead of repeating "reshoot it" and "high levels".

  • What settings give the BEST audio quality (original audio CD to iTunes)

    Further to my last post, can ya'll PLEASE help me get the bottom of this? I have heard so many different opinions & need to resolve ASAP....
    What pref' settings will give me the VERY BEST audio quality, when importing original audio CD's into iTunes 4?
    File size is of no issue (I have TONS of storage). I aim to import all at the highest quality possible (files need to be "broadcast quality" - for playing through a big PA system). I have been advised to either.....
    1. Import using the AAC encoder at a stereo bit rate of 320 kbps with a sample rate of 48.000 kHz.
    2. Import using AIFF (lossless uncompressed).
    3. Import using Apple Lossless (lossless compressed).
    WHICH ONE SHOULD I USE? Bearing in mind that the files will be played LOUD through a professional PA system & I'd prefer the crowd not to notice a significant diference between iTune files & orginal audio CD's (I will be spining both, side by side).
    Cheers - Sweetamix.

    You are finding out that the answer to this question tough. Everyone has their own opinion on this one.
    I saved the responses to a topic I opened six months ago on this same subject. I saved them and cut and pasted them into one document. Here were the answers I received. Good luck. bob
    Subject: Re: Classical Music & Bitrate
    Date: Sunday, June 26, 2005 11:12 PM
    To: <[email protected]>
    RE: Classical Music & Bitrate
    My experience with classical music (and music in general) is that the higher the bitrate, the better the quality and richness of the music. On the other hand, there is a limit to how much distinction your ears can make, and I think that anything above 192 kbps (mp3 format) is not significantly better (and not worth the larger file size). And in terms of volume adjustment, I personally have not found any need for it.
    RE: Classical Music & Bitrate
    I copied the same CD to my computer in 128, 160, 192, and 320 bitrates, and I couldn't tell a difference between any of 'em. I was listening to them with Grado SR60 headphones too.
    RE: Classical Music & Bitrate
    I’m sure you will get several if not many responses. You may also find they vary with the preference of the individual. However there are some rules of thumb. Generally speaking I think you will indeed find the higher the bit rate brings you closer to the CD sound.
    Certainly, 32-bit rate does not offer very good sound quality. Actually going from 32 to 320 should have been noticeably better, if not…. Then you’re in good shape. Because what you don’t know is there, can’t be missed.
    Also, the bit rate is only part of the equation. What format codec <http://docs.info.apple.com/article.html?artnum=51910> are your songs ripped to.
    Consider the following - iPod: About compatible song formats
    <http://docs.info.apple.com/article.html?artnum=61476>
    MP3 (from 32 Kbps to 320 Kbps)
    MP3 Variable Bit Rate (VBR)
    AIFF
    WAV
    M4A AAC
    Apple Lossless Encoder
    It sounds to me as though your songs may be in MP3 format. As a suggestion you might try ripping your CD’s to 128 AAC format. The general consensus is that AAC is better than MP3. Give it a shot…for the heck of it, and see what you think.
    Finally in the end…..it matters little what we all might think….If you think your sounds are better ripped at a MP3 320 bit rate……Great. If not…experiment, and play around till you find what sounds wonderful to you!
    RE: Classical Music & Bitrate
    Sorry for taking so long to get back to this thread: had two twelve-hour work days in a row.
    I did a Get Info on a typical piece in my iTunes and got this
    Kind: AAC Audio File
    Bit Rate: 128kb (have some at 192 & 2 or 3 at 320)
    Sample Rate: 44.100 kHz
    Profile: Low Complexity
    Channels: Stereo
    Volume: +2.3
    I was troubled that one responder couldn't tell the difference among 128, 160, 192, and 320 bitrates even using Grado SR60 headphones!
    And another seems to be of a similar opinion when he says that I think that anything above 192 kbps (mp3 format) is not significantly better (and not worth the larger file size). Does this apply to AAC format, too?
    However, I am going to continue to add tunes at the 320 bitrate - for a while - because I swear that I can tell a difference when I play them over my car radio via the cassette adapter.

  • How do I keep 24/96 audio quality in CS6 Premiere Pro?

    Another rook question here:  I record audio in 24/96.  How do I make sure to maximize the quality and keep the 24/96 when I import audio into Premiere Pro for editing, and also when I export (assuming I'm exporting to a high data rate 444 codec)?

    I guess that what you do need to know is that Premiere can actually Export Audio at 24/96.
    It can also do that with various "movie exports" as well. (Video/ audio streams)
    Check them out in Adobe Media Encoder settings.
    What you need to decide is if these "exports  " fit your workflow plan as regards Mastering ..then distribution / display.
    You should be running tests on this anyway - especially seeing you are running dble system sound system and have a major synching exercise ahead of you as well
    If  you mix a master 24/96 audio outside of Premiere...you should be able to use that in a sequence for mastering from but I think it will conform for previewing. 
    Plan...test...test..test
    If you want to really know what i think.  ....24/96 is over kill and you wont here the difference considering your intentions  ( but does not hurt to have up your sleeve and the audio guys may be grateful if they have a heapof work to do on it)
    Are your mics up to it..considering you are shooting on a DSLR???

  • How do I get audio from my macbook pro to my receiver with higher quality than the headphone jack?

    I recently got a receiver and surround sound speaker system.  The quality of the audio from the CD player connected to it is far beyond the quality of the audio that comes from the MacBook pro via headphone jack to RCA connection.  This was confirmed by burning a CD from the MacBook, placing it in the CD player then testing the two against each other  -  the same audio piece from the CD player or the MacBook, both connected to the same receiver/speaker set up.  What other options might get better audio quality from the MacBook to the receiver? 

    The Mac's audio output doubles as a optical Toslink stereo mini jack, just use a adapter to regular Toslink and plug the optical cable to your surround sound system. Fiddle with the sound settings to enable optical output.
    Mac's pass surround sound through, they don't do any processing.

  • Losing audio quality when exporting from premiere pro

    I have now had this happen with two different videos. I play the video in premiere pro and the audio is fine. I export the video and lose most of the quality of the audio portion. One project is a music session when exported I only get the main vocal the drums aren't audible only a buzz sound is made where the kick drum would usually come in. On another project is a thank you video for a family member and the audio is distorted after being exported. The audio sounds fine inside of the session. Once exported the audio will still work fine on my computer and my iphone 5s as long as I'm wearing headphones, once I try to play the video through my iphone 5s speakers the audio quality is gone, making the noises as previously stated. My iphone can still play other tracks fine but videos and audio exported from Premiere Pro are distorted. Any advice on how I can export the video so I wont lose audio quality when playing the video on an iphone? I know it's not my iphone speakers, they play other songs just fine.

    The source files as far as the audio goes are wav files that I've pulled off of my desktop. My sequence settings are 1920 by 1080 at 29.97 frames per second. The export settings are H.264 with the preset YouTube (HD 1080 29.97 FPS) and I've also tried HD video 1080p 29.97 FPS. If this isn't enough information please let me know I'm in the learning phase. Audio wise the quality is set to high and I'm exporting at 4800 kh and a bit rate of 190. I've also tried a bit rate of 320 and 264 I think it is and still had the same problems.  Let me know if I need to do screen shots I really want to figure this out.
    Thanks for your help!

  • When playing any audio the audio quality on my Thunderbolt display steadily worsens over an hour period and becomes garbled. Restarting the media player fixes it. Does not occur when MacBook Pro is undocked.

    When playing audio/video on my MacBook Pro docked to Thunderbolt display the audio quality steadily worsens until it is garbled beyond understanding. I don't know if it occurs when using the MacBook undocked. It takes about an hour to become noticeable and steadily worsens. Simultaneously when the computer speaks the time this is also gargled. As the problem develops it sounds like there is a static halo around each spoken word.

    Hi GeoChester,
    Welcome to the Apple Support Communities!
    I understand that you are experiencing some sound issues on your Mac that worsens over time. In this situation, the first troubleshooting step that I would recommend would be to reset your computer’s PRAM. To do this, please follow the instructions in the following attached article. 
    OS X Yosemite: Reset your computer’s PRAM
    Best regards,
    Joe

  • Retina Macbook pro audio quality is bad in windows 8.1

    Like the title states, my Retina Macbook pro's (2012 model) audio quality is very bad in windows 8.1. It is unbearable to listen to music on it. I am using the latest bootcamp drivers, 5.1. I believe this is a driver problem, as it sounds fine in OS X. I tried to find the proper driver myself with no luck. I feel this is unacceptable on a computer I spent so much money on. Any help would be greatly appreciated.

    Yes...
    I also just recently purchased a MacBook Pro retina 15 (mid-2014) with bootcamp (windows 8.1) installed.  While using my windows partition, the MBP seems to be running a bit hot, at least hotter than the OS X partition.
    As for comparison, I have the 2012 MacBook Pro retina 13 with bootcamp (windows 8.1).  It does not get as hot as the 15, and in fact, I don't think it runs hotter than OS X...
    Any thoughts?  Is it perhaps because I have the NVIVIDA dedicated graphics and in the windows partition, IRIS PRO isn't running?

  • Ableton Live 6 vs Logic Pro 8 CPU usage/audio quality.

    Hi,
    Any user's have feedback regarding CPU usage between Abelton Live 6 vs Logic Pro 8. I'm currently running Live 6 on an MBP 2.0ghz (CD) and am constantly running into audio drop outs etc etc.I am reading throughout the net that Logic Pro 8 is more CPU friendly than LIve 6... is this true, overall? Also any users ever do a mixing test between the two apps to see if there was an audio difference when rendering your final project to aiff or wav? I am thinking on using Live 6 as a stratch pad to "create" as I enjoy it's flexibility then transfer the project over to Logic, however that depends if there really is a difference in CPU usage and Audio quality between the two DAW's.
    Thanks.
    guess i should also state that i will be running on the new imac's come tax return time. lol

    Logic 8 is very CPU friendly, I can say that much. I never found Ableton Live 5 to be very taxing either, though. Pro Tools is.
    But like you I use Ableton as a scratchpad, I love how it works, and Logic for the finishing touches. But that's just because I'm not good at Logic yet. Not fast at it.
    Get them both! You'll be glad you did.

  • Can I modify the Export to... iTunes settings for better audio quality and compression?

    I am working on some Music Videos to DJ at a party with, I notice that when I export them as .mov files with Final Cut HD Express, they have 320 kbps and 44.100 kHz in the audio compression settings. Once I export them in Quicktime 10.2, using the File / Export to... / iTunes drop down menu, the Second, (ipad, iphone 4, & AppleTV) setting (not using HD, so the Mac and PC tab is not an option.) The final exported file, that plays great in itunes, appletv, ipad, etc. has a reduced quality audio of Bit Rate of 159 kbps (instead of 320 kbps) and Sample Rate of 32.000 kHz (instead of 44.100 kHz). Is there anyway to change these preset settings somehow?
    The Formats on the File / Export / menu are even less dinamic?
    I also noticed that some of the Music videos I purchased from the itunes store some years ago, have a 256 Kbps and 44.100 kHz setting. Which is better quality than the ones exported from Quicktime 10.
    I thought maybe with Quicktime 7 Pro, I could fine tune all the presets, but it got too complicated for me, might this be an option? What would the settings be?
    Using Widescreen 853 x 480 at full quality .mov file.
    thanks.

    I am working on some Music Videos to DJ at a party with, I notice that when I export them as .mov files with Final Cut HD Express, they have 320 kbps and 44.100 kHz in the audio compression settings. Once I export them in Quicktime 10.2, using the File / Export to... / iTunes drop down menu, the Second, (ipad, iphone 4, & AppleTV) setting (not using HD, so the Mac and PC tab is not an option.) The final exported file, that plays great in itunes, appletv, ipad, etc. has a reduced quality audio of Bit Rate of 159 kbps (instead of 320 kbps) and Sample Rate of 32.000 kHz (instead of 44.100 kHz). Is there anyway to change these preset settings somehow?
    Your question basically has too facets. The first is the quality of the source audio in terms of sampling rate and data rate. Under normal circumstances you don't want to export the audio using a sample rate or per channel data rate that is lower than your original source if you are trying to maintain the quality as high as possible and, on the other hand, you don't want to use settings greater than the source values since they increase the final file size without improving the quality of the audio.
    The second is the workflow. Since you indicate you are using FCE HD for editing, then using QT X to re-export the data seems somewhat redundant. While it has been many, many years since I last used FCE HD, if I remember correctly, you have both "Export QT Movie" and "Export using QT Conversion" options. The first option takes the audio and video format in which you are editing and merely copies the edited data in that format to a new MOV file container. On the other hand, the second option allows you export the edited data format directly to a user selected compression format using user selected options (but does assume the user knows how to best use these options). These are, BTW, the same export options available in QT 7 Pro.
    I thought maybe with Quicktime 7 Pro, I could fine tune all the presets, but it got too complicated for me, might this be an option? What would the settings be?
    What is so difficult here? If the source is DVD stereo quality (usually 48.0 KHz @ 192 Kbps then set your output for stereo 48.0 KHz @ 192 Kbps. If you wish to save a bit of file space, the you can use 48.0 KHz @ 160 Kbps but that is normally as low as I go. As to setting combinations in QT 7 Pro, once you set the sampling rate, the application becomes cuntextually adaptive and will only allow you to select data rates that are "standards" compliant and non-compliant ones are automatically greyed out. Rendering quality settings are self-explanatory. These are the basics if you are using a preset option like the "Movie to MPEG-4" export option. If using more complex options like the "Movie to QT Movie" option, then you will need to also consider how you plan to use your file to select the correct encoding strategy. For instance, a constant bit rate is good for optical media from the standpoint of providing a constant flow of data to the player. Unfortunately, this is not very efficient since it forces the quality to vary as it maintains the constant data rate. A variable but contarined data rate is also good for optical media or fast start online files but allows the data rate to vary between user selected data rate excursions and thus provides improved quality over the constant data rate strategy. The variable data rate allows the user to target a specific level of audio quality and will then allow the data rate to vary as needed to maintain a constant level of quality. This option is better suited for playback on a computer since the bit rate can vary greatly in magnitude. The average setting allows the user to target a specific average target data rate but allows it fluctuate as the encoder applies predictor-corrector adjustments. This option, like the constant data rate option, is good for average complexity content when creating a target file of predictable size but can somtimes have problems towards the end of the file if the content suddenly becomes more complex than predicted.
    Using Widescreen 853 x 480 at full quality .mov file.
    I would rarely recommend using the full quality slider setting. This is more of a redundancy setting forcing the encoder to recycle through certain encoding routines in an effort to improve quality without increasing the data rate. You would like do better to increase the video data rate limit and decrease the quality slider to something in the 50 to 80% range. You also indicate the files are 853x480. Such files are frequently the result of anamorphic widescreen MPEG-2 or DV source files. If so, I would normally recommend retaining this anamorphic strategy by encoding the H.264 video at the same 720x480 matrix dimensions and allowing the player to properly display the files at their targeted aspect ratio. This strategy will either allow for slightly better quality at the current video data rate setting or allow the user to retain the current level of quality using a slightly lower data rate which, in turn, means a slightly smaller final file size.

  • Does Encoding AAC Files with VBR on Improve Audio Quality ?

    Does Encoding AAC Files with VBR on Improve Audio Quality ?
    Are there Disadvantages ?
    Bonus Question 1:
    Usually, an 'auto' type encoding tool will screw up the natural flow of the music and miss subtle changes in the energy, volume, etc.. -stuff that is just to subtle for it to catch -Generally I wouldn't trust an 'auto' type setting to pick up this subtle stuff. Yet people say using VBR improves sound quality (and NOT file size). Why and how ?
    Bonus Question 2:
    What is the max bit rate VBR uses ? If I set my AAC encoder to 320 kbps and turned VBR off, wouldn't the sound quality be superior to encoding with VBR on (simply because the kbps are set at 320 the whole time...) ? Sure, the files encoded with VBR off would be larger, but wouldn't the quality be better ?
    Bonus Question 3:
    Lastly, I did a little test and encoded one song with VBR off and one with VBR on. The VBR song was 1.5 MBs bigger - Huh, I thought, is that increased file size the result of improved resolution throughout the whole song, or just one little section (haven't had time to listen to them)?
    Message was edited by: temptemp9

    These are some pretty tough questions. I like VBR encoding in general, as it does allow for more complex passages to take advantage of higher bit rates while less complex passages fall back to a lower bit rate, while keeping within fairly consistent file size parameters. Whether VBR yields better results at different target/average bit rates is another matter, and really requires that you encode and properly test output files before deciding on anything but "recommended" settings.
    At maximum lossy bit rates of 320 kbps, it doesn't make sense to use VBR. If I recall, the LAME MP3 encoder presets actually preclude VBR encoding for the highest quality files, and the same may be true for iTunes AAC.
    I have to admit that to speak with any certainty or confidence about the issues surrounding your various questions, one would really need to encode a bunch of files with iTunes from lossless sources and test them in software such as foobar2000. I could investigate this easily on my own PC, but I don't encode music at 320 kbps AAC, since I have enough disk space to just listen to lossless files. iTunes doesn't display dynamic bit rate changes as foobar2000 does, and I haven't come across any Mac software that does as good a job of allowing for proper testing as foobar2000 for Windows does.
    If you really need definitive answers to these three questions, then I think your best bet would be to do as I've hinted and use iTunes and foobar2000 if you've got access to a Windows computer to do this testing and evaluation for yourself, or sign up for a user account at Hydrogenaudio and post the same message there.

  • "audio quality" Audigy 2 vs. Nforce onboard

    5"audio quality" Audigy 2 vs. Nforce onboardS hi everyone,
    i bought myself an old Audigy 2 from Ebay a week ago, hoping i can increase my fps ingame on my old 2200+ computer.
    i really felt the increase, especially in Doom 3. But today i threw out the Audigy again.
    the bass from my subwoofer (teufel concept e mpe) is sooooo weak (in games and winamp) and sounds like a cheap passi've subwoofer.
    i tested every single setting, hoping to get a better sound out of it, but it didn't work. the bass just feels "weak".
    i'm pretty surprised, because i thought i'd get at least the same sound quality.
    does anyone have experiences with that? i can't believe the onboard sound is better than the Audigy's!
    i set the frequency for subwoofer to 20hz, increased the volume of the subwoofer, changed the equalizer.
    but it just sounds like a passi've sub and "cheap". (sry my english isn't good enough to find fitting adjecti'ves)
    can anyone help me with that? does anyone have the same problem?
    thanks in advance.

    Meocene wrote:
    If you used the auto update function you're probably already using the latest drivers. However if you didn't then definately install Dan K's pack - they're the best drivers currntly available for the Audigy 2. You'll want to uninstall the previous drivers, then do a clean up using Driver Sweeper beforehand though - Creative's drivers can be problematic
    The problem with the Audigy 2 is that its analog output quality isn't especially good anymore, when compared to newer devices - even onboard. This is demonstrated when and if you swicth over to use Dolby Digital Li've.
    I'm assuming the sound quality improvment I got has got to do with the fact that when using DDL, the Audigy's DACs and other analog components are skipped because the audio signal is being sent straight out, digitally, to my reciever.
    The audio quality improvement I got was huge, to the extent that, eax/gaming aside, I can't actually hear the difference between an X-Fi and my Audigy anymore. Before using DDL li've the difference was obvious.
    It may simply be that your nForce's analog output is of a higher quality.
    Analog quality aside though, the Audigy 2 is still the better device, or at least the more capable. It really depends on whether or not you're a gamer I suppose.
    If you've got a digital reciever then try the DDL pack and connect your Audigy to your reciever using the digital output.
    If not, and you're not a gamer, then I'd stick with your nForce if that sounds better? < - this being advice I'd imagine you must find rather anoying.
    Message Edited by Meocene on 08-0-2009 03:5 AM
    thanks for these informations.
    unfortunately i can't use DDL, because i don't own a receiver.
    music is at the moment the more important thing, i'm gaming about 4 hours a week.
    i'll keep the audigy though, maybe i'll rebuild an old 3000+ system i still have. (the sound THERE really sounds crappy)
    thanks for all answers and your help!
    EDIT: does the usual audigy 2 driver panel look like this: http://www.hartware.de/showpic.php?type=review&id=606&path=/media/reviews/606/xfi_screen_thx_big.jpg
    it's a screenshot from an x-fi card.
    mine looks different!
    Message Edited by chackachacka on 08-0-2009 0:33 [email protected]

  • CC 2014 - Audio quality reduced from original video

    I used to use Adobe Premiere Pro CC WITHOUT THE 2014 version FYI
    Let me explain:
    -Back then i've been using Adobe Premire Pro CC peacefully using the GPU Mercury Playback thing activated. No problems AT ALL.
    -After recording a TV program and getting the AVI File when finished recording, i tested that RAW file to hear it's audio and see the vid with Windows Media Player. Audio quality was perfect,clear and high as it should.
    -Whenever i placed that AVI file into Premiere Pro CC (again, NO 2014 version), the audio and video played as it did in WMP.And when rendered/exported to a new file after some edits, it still kept the quality and all that stuff!
    .....NOW, with Adobe Premiere CC 2014, there's a BIG PROBLEM....Whenever i place THAT SAME RAW FILE in CC 2014....The video played nicely,yeah...framerate stable etc...The problem is...that the Vid's AUDIO quality has been REDUCED and it's NOT CLEAR ANYMORE,and it happens to ALL THE PREVIOUS AVI FILES I HAVE!!!
    What have adobe done?...Don't break what's working correctly!!!...Now i can't go back to Adobe Premiere CC WITHOUT THE 2014 version because of the new version STAYING FOREVER!
    Please,HELP ME!!!. I can do a video of what i'm saying so you guys understand the situation.
    DV

    To be honest,i have not a single clue about it...Never touched the Audio and Audio Hardware Settings. I tried searching online on google  for a solution and i think found a solution (This one had a similar? problem as mine) . I had to change an option in Settings->Audio Hardware-> ASIO Settings then something above 512 of the Buffer, aka 1024, though i've touched "default" and it changed to 2048. The problem disappeard.
    But...i think i know why it might've been the problem after all..i think... What i did to begin with  was to put a "Bandicam" (It wasn't a TV show in the end,SORRY!) recording (in it's default settings) into CC 2014 and it's audio (in CC 2014) has been like muted ,though whenever you skip the timeline forward,you can hear "a huge distortion" that doesn't matches the recording's audio AT ALL (D**n,i played a music in the background while recording and,once finished recording i've played the vid i've made and the audio was there,perfect quality)... After that i wanted to "Mute the Bandicam's audio track" and put the song that should've played over it.. Then whenever i've played the vid to see how it looked like,that song i've placed over the "broken audio" GOT AFFECTED TOO... Perhaps it was just THE BANDICAM VIDEO'S AUDIO FAULT!!
    (Though somehow i got lucky to ,from a random chance, to have that audio i've inserted manually to play and also render/export in it's correct quality...Weird!)
    ...Then i tought "Did Adobe screwed up?...". I've closed CC2014 100%,restarted the PC,opened CC 2014 , grabbed a DIFFERENT Video file ,this time from a TV Recording (this time,YES,TV,sorry)  and whenever i click the Play button in the preview...The audio played exactly as the original video did...I even tried placing that same song over it to see if it did distorted and didn't
    ...Perhaps i am doing something wrong now...or are some Adobe settings being changed like crazy without my aknowledge (or something) ?
    My PC is:
    Intel i5 2500K
    660GTX Ti OC 2GB (GIGABYTE/Nvidia)
    8GB DDR3 1333Mhz
    Windows 7 64bits Home Premium
    250GB SSD
    1TB HDD
    If there's something i should try out to see if the problem could be vanished FOREVER please let me know X_X;

Maybe you are looking for