Audio Quality Changes

Why does a perfectly good audio changes to a bad one the minute I place it in my timeline? I've tried in formats mp3 wma wav and the audio sounds good in the prevew but when i place it in the timeline is like it looses its qualiti and sound horrible
how can i fix it?

I am having the same problem... I am using Adobe Premiere CC. I have also tried different sample rates to no avail. I've tried Asio4all and it sounds the same...
The weird thing is that the audio sounds as it should in the source monitor but once dragged onto the timeline it sounds muffled and cruddy...
Any help would be appreciated =)
Thanks,
Rob
EDIT: Nevermind... Now it is working out of the blue...

Similar Messages

Audio quality changes when I drag to timeline

I selected one part of a video, placed it on my timeline and suddenly the audio changed to a terrible quality. The problem is that when I play the same part on the preview it sounds completely normal. I was using the same video(but not the same part) earlier and everything was fine, now every part I try to put on my timeline(I even tested the ones i've already used) gets the audio messed up. I am working with a video of a lesser quality along with an hd one, but I don't know if it has any influence in that, I was using both earlier and i've done it with other project and it was okay but I don't know... Can somebody help?

Is the FCE Sequence matching your imported Clips?
Right click the Sequence icon in the Browser and select Item Properties.
Al

Importing changes the audio quality of original video

When I import a video into a new project the audio sampling rate of the video goes down and the audio channel changes to mono though the video is stereo.
I have tried creating new projects changing the sequence presets to ones that include the higher sample rates and stereo, but this didn't work.
More info on the original raw video:
AVI
Frame size: 1280x720
Frame rate: 29/frames per second
Data rate: 5000kbps
Total bitrate 5192kbps
Audio:
Bit rate: 192 kbps
Channels: 2 (stereo)
Audio sample rate 44kHz
When I drag and drop this video into the project section of premiere, and I click the "Info" tab, I notice that the audio is "11025 Hz - 16-bit - Mono" which is different from the original video.
Does anyone know why this is and how I can fix it?
Thanks!

It might help if you were to post a screenshot with the
full details (codecs, etc.) of the .avi file in question.
Here are a couple of free utilities for media file inspection:
GSpot
http://www.headbands.com/gspot/
MediaInfo
http://mediainfo.sourceforge.net/en
Also, you neglected to mention your version of Premiere.

HP Slate 7 Voice Tab - Audio Quality Issue

I see drastic change in audio quality in HP Slate 7 Voice Tab after an upgrade to recent Kit Kat update. Songs and other media is not all audible , Volume through external speaker is very very less at maximum volume also.Tried with many song, HD videos but issue remains same and feels its related to new android version update on HP Slate 7 voice tab.
So would like to know is there any new update or solution for this issue.

Hey @S_Thomas ,
Welcome to the HP forums.
I understand you're experiencing issues with audio quality on your Slate 7 Voice Tab after updating to KitKat.
How to check for system updates:
You must be connected to the Internet to check for updates. Use the following steps to check for system updates on your tablet.
1. On the Home screen, touch the All apps icon, and then touch the Settings icon.
2. In the System section, touch About tablet.
Figure 1: About tablet.
3. Touch System updates.
Figure 2: System updates.
4. Touch Check now.
Figure 3: Check now.
5. Install any available system updates.
If no updates are available, or updates do not resolve the issue try the following.
Clear all cache on the tablet: Settings > Storage > Cached Data. You should have the option to erase all the cached data here.
Hard reset: While the tablet is on, press and hold the power button for 8+ seconds to force shut down. Turn the tablet back on and see if that helps.
Factory reset: Make sure you backup any important files or information first. Settings > Backup & reset > Factory data reset.
If this fails to resolve the issue please call our technical support at 800-474-6836. If you live outside the US/Canada Region, please click the following link to get the support number for your region: Phone Assist Worldwide.
Thanks.
Please click the "Kudos, Thumbs Up" at the bottom of this post if you want to say "Thanks" for helping!
Please click "Accept as Solution" if you feel my post solved your issue, it will help others find the solution.
The Great Deku Tree
I work on behalf of HP.

Poor audio quality in iChat

When I use iSight in Skype it delivers a way better audio quality in comparison with iChat Audio/Video chat. It's not that crisp, more like a poor telephone call. I wonder if I can change it or is it just an iChat nature?
My connection speed is 1 MB.
Straming in Quicktime changed to 1.5 MB
My Output/Input settings in Sound tab seems OK.
I don't use firewall.

Hi Sleeze,
I do not have Skype so I can not judge. Most of the posts I have read seem to suggest iChat Audio is better.
However Skype could have updated their app.
As long as iChat Bandwidth and Quicktime streaming setting in System Preferences are not capping your connection I can not see any other changes you can make. (unless you are "too far" from the camera). If the last is the case it may pay to adjust the volume in System Preferences > Sound >Input tab as this is the master for iChat. I am not sure if it is the same for Skype.
Ralph

Audio Quality in different mo

Is there a significant Audio Quality difference specifically in Entertainment and Gaming mode ? as i noticed better dynamics and high tones on entertainment mode but those can be easily replicated with the proper EQ changes on gaming mode, Its really a hassle to change modes everything i game or play movies/mp3. also im aware that theres a smart mode changers but dont wanna more apps running on my system as there are already tons of it

The mode changer is less than half a megabyte and only uses about 980kb of ram. Also, for music, Audio Creation mode with bit-matched playback enabled is the best thing to use. Preferably using winamp with an ASIO or kernel streaming output plugin.Message Edited by 003 on 05--2006 06:3 PM

A few questions - audio quality, and remote capability

1) What is the audio quality of songs streamed from airplay?
2) Is there any way to change the volume of a song using the iphone/ipad remote?
Thanks!

Hello..
Not sure about the actual specs of Airplay SQ.
Using the ATV2 to access your iTunes Library you must control the volume with the volume control of the device through which you have the audio routed.. ie TV or Stereo. There is no volume level on the ATV2 remote or in the interface of the ATV OS that I can see.
Using the ATV2 as a speaker for iTunes from your Mac or PC you can control the volume through iTunes on the Mac or PC.
Using the Remote App on iPhone 4 or other device you can control iTunes on your Mac or PC. In addition you can select the ATV2 as a speaker for iTunes directly from the Remote App and in turn have control of the volume from the iPhone 4.
I like the last way best. Easier to navigate library, make selections and control volume all from anywhere within range of your wifi connection.
Message was edited by: rz22g

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]

Sound Quality changes between itunes 8.2.1 and itunes 9.1

I am using itunes and a modified Airport Express as my audio source in my Hifi Stereo system.
It is working so well that I have discarded my high end audiophile CD player.
However I found an issue with itunes consistency from version to version in audio quality.
Here is the experience I conducted:
Using:
- a mac mini with Snow Leopard and itunes 8.2.1 and a macbook pro with Snow Leopard and itunes 9.1.
- Identical Wav (full res 16bit 44.1Khz) files local to each mac.
to be completely accurate the music used was:
Ben Webster Meets Oscar Peterson: T01 The Touch Of Your Lips
Diana Krall Live in Paris: T11 A Case of You
- in both cases itunes streams to the Airport Express (same one)
-> The result is that the there is a difference in sound between the two itunes, and I would like to understand what changed or why this difference?
If I were to describe the differences I would say that itunes 8.2.1 delivers a softer sound with more medium than itunes 9.1. Itunes 9.1 would have a brighter sound with less medium than itunes 8.2.1.
(On a side note, not relevant for the problem above, but could interest some people: When I access with my mac with itunes 9.1 the itunes 8.2.1 library on the other mac, the sound quality streamed to the AE is poor compared to either itunes version used with local files. I wonder how apple streams the music from one itunes to the next but that music it altered may be compressed in this case.)
Message was edited by: cdfr

Chris thanks for this answer.
I did check the settings on both Macs/itunes and they are identical. Everything is disabled. The remote volume is also disabled on both.
So the sound quality difference cannot be explained by that.

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;

Terrible audio quality in 5.1 mode

Hi !
This issue has been here for a while, but it's only now that I come to try to get a fix for it ...
So, it's been a few version that I've noted that the sound was really bad when playing videos on youtube and came to realize that this was fixed when I switched my audio configuration from 5.1 to just stereo in the windows configuration.
I've searched a bit and it seems that flash proudly converts stereo audio sources into 6 channels using the Dolby ProLogic II algorithm. This would probably be fine if it was customizable and de-activable, which it is not right now.
From what I hear, my subwoofer is overloaded with sounds and seems to be used to play low-medium sounds rather than just the deep basses it's meant to.
I have also noticed that, in the sound configuration of windows, I can specify whether my satellites have a broad spectrum or not, but this doesn't seem to change the issue.
Sure, I can get my audio quality back by going to the windows sound configuration and switch to stereo, but I don't want to do that everytime I use flash and then change it again everytime I play a movie or a video game.
The simplest way in my opinion would be to have an option to disable that audio conversion in the flash client.
In case the algorithm for upgrading stereo to 5.1 is customizable, could we perhaps also get access to these parameters (for this kind of algorithm, I would really expect a way to specify the ranges for the individual speakers)
Thanks,
WishGate.

Standard track height:
and no waveform. Now extend it vertically,
and voila, the whole waveform. Just drag down the bottom of the audio track to make it wider (vertically) and there you are.

Unexplained audio level changes

I keep running into a very strange problem. After I've put a sequence together, and done an audio mix (keyframing/rubberbanding audio tracks in FCP), certain pieces of music will be too loud during playback, even though I have them turned down low, or even all the way down. If I stop the playback and start it again from the same spot, or even back it up a few seconds, it plays back correctly.
It seems to do this only with music tracks. I've tried trashing render files, replacing the original files, and I repair my permissions and trash prefs on a regular basis.
Any suggestions?
Dual 5oo MHz G4 Mac OS X (10.3.9)

Try this...
In your user prefs, reduce the quality of the audio playback to "medium" or "low" and see if it helps. Using a "low" setting drastically reduces the processor load on your Mac during playback.
The audio quality won't be affected to any degree that will bother you for any listening purpose except serious review for post-production done through a "reference" monitor system of speakers. In my opinion, you should be doing that work outside of FCP in a dedicated audio editor anyway. So practically speaking, changing the playback quality setting shouldn't matter much to you in most cases.
So try setting the audio playback quality to "low" and see what happens.
Another question: are all your audio tracks 16-bit 48Khz sample rate, or are some of them different, which means they need to be rendered before playback?
That could affect real-time playback, too.

How to improve audio quality

I got the 8GB Nano as a gift. I already had good headphones -- Shure eC4's. The audio quality from the Nano is not as good as what I get off my laptop. Fooling around with the EQ settings gives a small improvement but overall the quality off the laptop still beats the Nano. Most of my tunes were ripped at 192 using Musicmatch several years ago. Did iTunes somehow transfer these to my Nano at 128? What is the default rip rate under iTunes? Can I change it? Or is the audio quality on the Nano just mediocre to begin with?
Thanks for any advice you can give me.

You must re-import everything at the new bitrate of 192kbps. You can clear your iTunes first before doing this to ensure there are no duplicate files.
The info contained in the other post is wrong. You cannot just sync your iPod at the new bit rate you selected, this will change nothing.
When you imported your CDs at 128kbps, a lot of info was lost from the original CD. This is how compression works. Put simply, when importing the songs iTunes discards any info that you cannot hear in order to make the file size smaller. That info can consist of anything such as frequencies that the human ear cannot hear (but which may affect harmonics). The theory being that if the ear cannot hear it, it's of no use.
Once imported into iTunes at 128kbps, increasing the bitrate to 192kbps without importing the CD all over again will not replace the info that was lost originally because it was discarded and cannot be recovered, and it will not improve the sound quality of your already imported files.
If sound quality is important to you, import the CDs again at the new bitrate.

Audio Quality Changes

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