Audio quality improvement?

I have tested the latency of a 1-to-1 video chat between US and China. The latency is acceptable, less than 1s. But the audio quality is poor. A lot of noise. Changing the audio quality setting helped a bit but still not good enough.
Is there a way to improve audio quality?

BUMP
It is vista x64 btw

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"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]

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.

How do I improve audio quality in iMovie?

I have heard that you can strip out the audio track in a video clip, improve it in GarageBand and then put it back into iMovie. Is this true? I'd like to improve the audio quality on a series of video talks.

Publish your movie first by going to Share and then select Media Browser.
After it is "published" open up Garageband.
Click "New Project"
Select "Movie"
Choose a name for your project.
Select your movie from the browser on the right.
Adjust your audio.
When done, you can export your movie with new audio from Garageband

How Do I Improve Audio Quality on Face Time

Can anyone advise how I can improve the audio quality on Face Time?
Video is acceptable, but audio sounds like a long distance phone call back in the bad old days, lots of background noise and voices barely audible.
I don't think its my internet connection, currently ADSL2+, Line speed 4.9Mbps, download speed 617KB/s

Little Eddy wrote:
Can anyone advise how I can improve the audio quality on Face Time?
Video is acceptable, but audio sounds like a long distance phone call back in the bad old days, lots of background noise and voices barely audible.
I don't think its my internet connection, currently ADSL2+, Line speed 4.9Mbps, download speed 617KB/s
Welcome to Discussions, Little Eddy
Depending on whether you mean sound you hear or how you sound to others, here are some ideas for your consideration:
(1) Call from an area with less background noise.
(2) Use an iPhone. Its sound quality is superior to iPod touch.
(3) Try http://store.apple.com/us/product/MB770G/B It sounds better in my tests than internal mic.
(4) For FaceTime on your iMac, sit closer to the Mac's mic. Also test speaking louder.
(5) For FaceTime on your Mac, try a USB mic like http://store.apple.com/us/product/T6831LL/A
(6) Both you and your connection should test with other users to determine who has the problem.
(7) If sound you hear from other end, ask your contact to double check that his mic is not obstructed by a case, fingers, etc.
Mac Pro Quad Core (Early 2009) 2.93Ghz Mac OS X (10.6.5); MacBook Pro (13 inch, Mid 2009) 2.26GHz (10.6.5)
LED Cinema Display; G4 PowerBook 1.67GHz (10.4.11); iBookSE 366MHz (10.3.9); External iSight; iPod4touch4.2.1

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.

Improving audio quality of existing recording to better understand speaker

I recorded about a 1 hour recording using imovie's "capture" mode with the built-in internal microphone. On the recording there are 4 speakers talking and background noises (music). The music is making it such that it's hard to understand the 4 speakers at times. Inside of movie it seems the only thing I'm able to do is increase the volume of the overall track.
Using this or other software, is there a way for me to improve the audio quality such that the speakers are more clear?
Thanks.

I think voices tend to be in the midrange frequencies, something like 1 kHz. Maybe if you filtered out high and low and boost the midrange, you might get an improvement. Of course a lot of music is in the midrange too.
I know you can filter audio in Final Cut products. I've never done it in iMovie.

1.1.4 firmware update - does it improve audio quality?

I'm really confused here as I've read different opinions on various Apple ipod touch forum sites that some people swear after updating the firmware the audio quality has improved greatly (e.g. better bass/mid-range responses, louder volume). I called Apple retail store and was told no, that the only thing the firmware update fixes are bugs. Well, I still have the 1.1.3 version (mine came with the January updates) and so am debating whether to update or not. Anyone who has on this forum, have you noticed any improvement made to the audio quality?

To all: so do you guys recommend I still upgrade to the 1.1.4? FWIW, I haven't experienced any bug issue with my touch (I have yet to test the internet access of the touch, and from what I've heard some people have reported that their safari crashed with the 1.1.3 version; when they upgraded to 1.1.4 all was said to be fine).
Message was edited by: iBLUE

Audio quality in adobe connectnow vs skype

can i improve the audio quality in afcs than skype?
if i am using adobe connectnow, the audio quality is lower than skype.
is there any way to improve?

The answer is simple... with Logic you have to know what you're doing as opposed to Garageband which does some of the work for you. Logic is a program (generally) designed for professionals, it assumes you know a little bit about EQ, Compressors, Limiters...etc. If you don't, you need to learn if you wish to use Logic effectively. It also means you will need to study the ART of mixing, there's audio pros who say mixing can only be learned by doing it over and over and over....etc and listening to your mixes on as many different playback systems as possible.
Which version of Logic are you using? (important)
And what do you mean your CPU won't let you see the manual, the manual is online, if you have Safari you can see the manual.
Set Logic's master fader so your song never drives it over 0.0, but remember, louder is not necessarily better, first get a good mix/blend then worry about getting it louder.
Something else to consider, commercial recordings are mostly done using processing hardware that costs thousands of dollars. To compete with that you really have to know what you're doing.

Will an optical digital audio cable improve the sound when I am viewing content thru Apple TV

I have a 2nd generation Apple TV. When I view content via my Apple TV the sound gets "fuzzy" as I increase the volume on my TV. If i connect Apple TV to my Samsung HDTV with an optical digital audio cable, with the sound quality improve??

people say that if the atv2 dont have a connection to the tv optical is muted
I have a atv1 and been using the optical out with my sony stereo system for ages mute the tv so only 1 audio source and on atv1 it play out the optical even with the tv off
of cause one have to choose the optical input as input source on the receiver

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.

Audio Quality

Hey folks,
We've got a client who is looking for better audio quality than we've previously been able to get out of LCCS. Looking into it a bit, it looks like the Flash microphone class has two main quality settings: encodeQuality and rate. The MicrophoneManager allows you to set the encodeQuality (although it seems to default to the max: 10 (~42kbps)) but not the sampling rate, which defaults to something like 8 or 11 kHz. Is there a reason for this? Would it be a particularly bad idea to up the sampling rate on the mic manually if we'd like to improve audio quality?
Also, in the Flash Microphone class documentation (http://help.adobe.com/en_US/FlashPlatform/reference/actionscript/3/flash/media/Microphone. html) it looks like the rate can be set to 5, 8, 11, 22, 44 kHz. Another article I found on audio capture (http://help.adobe.com/en_US/as3/dev/WS5b3ccc516d4fbf351e63e3d118a9b90204-7d1d.html) seems to suggest that those settings are available with the Nellymoser codec only, and that the SPEEX codec, which is recommended for LCCS, can only do 16 kHz sampling. Was wondering if that article might not be a bit outdated and if we might not be able to set the rate to something higher (the wikipedia entry on SPEEX seems to suggest that it can handle 32 kHz).
Our client was hoping for relatively high grade audio quality - I believe he wanted, in an ideal world, something like 44kHz sampling frequency and 96kbps bitrate. I don't know if these values are possible, but I'd hope that we can improve the audio quality to some degree.
Do you have any suggestions on how to produce higher quality audio from LCCS? Are there any other factors in the audio quality that I'm neglecting? Any help would be greatly appreciated.
Thanks,
Davis

like this....
                    private var audioCodec:String="Nellymoser";
                    private function onTimerComplete(p_evt:TimerEvent):void
                                        activityTimer.start();
                                        if ( audioPub.isPublishing && !audioPub.isPaused ) {
                                                  if (audioPub.microphone.codec.toLocaleLowerCase()=="speex" && myCodec=="Nellymoser") {audioPub.microphone.codec=myCodec;audioPub.microphone.setSilenceLevel(0,666666);}
                    blablabla....
Even if I apply the codec this way: <rtc:AudioPublisher id="audioPub" codec="Nellymoser"
when a new user enters or leaves the room the audioPub.codec switch to Speex. To correct this, I can reapply the codec and also setSilencelevel but the audio stream will be stopped for 1 or 2 sec.

Audio Quality Poor in Windows, OK in MAC

I made a Quicktime movie from an iPhoto slideshow. It plays fine on the MAC, but in Windows the audio quality is poor. The high frequencies seem to be gone. (Perhaps the audio data are being undersampled?) Any ideas as to how I can improve the Windows performance?

Use the QuickTime Control Panel / Audio tab to check the settings used by the machine (sound out). You want to use 44100 Hz and 16 bit sample size.
Clicking on the "Devices" you can also edit other audio settings and properly configure them.
Updating audio drivers may also help and last resort is to use "Safe mode".

Captivate 8, NeoSpeech Spanish poor audio quality

Hello everyone,
     I was demoing some of NeoSpeech's non-English voices when I had Captivate 7 and they were operating fine. Recently I licensed a Spanish language voice AND upgraded to Captivate 8. Now the audio quality for the Spanish voice is very bad. There's a lot of noise and it sounds like the treble is being over-driven. This does NOT occur with the standard 5 NeoSpeech voices already included in the language pack.
     I reverted my licensed version back to a demo version and the quality is still poor. Audio using the same voice through Microsoft's SAPI sounds clear and fine. It is only when running through Captivate 8 that the audio degrades.
     Does anyone have an idea why this worked fine in Cp7 and fails in Cp8, yet other NeoSpeech voices are fine?

I had exactly the same problem on my new, beefed-up laptop and couldn't figure out how an "upgrade" could ruin my music (which I take pretty seriously). I am embarrassed to say what actually fixed the problem. I started by turning off the EQ and it improved somewhat. I kept searching the web without luck until I noticed something by accident. My system volume was low and the iTunes volume was peaked at max. I turned the iTunes volume to 40% and turned up the computer volume to compensate and voila! The bass distortion and indistinct vocal range was immediately fixed. I turned the EQ back on and was able to use it however I liked. I guess it is like turning up the volume on a radio to max. You get sound, but it gets distorted in the upper range. Hope that helps!

Poor audio quality with iTunes 9

Hi.
This is my first posting and I am hoping members of the iTunes development team look in here.
I recently upgraded to Snow Leopard and I noticed a marked improvement in sound quality using iTunes 8 . Please note I am playing AIFF files through the digi out of my Macbook into a high quality Dac, amp and speaker separates system, in fact the sound was as good as any cd transport I have ever used. I then "upgraded" to iTunes 9 and noticed a deterioration in audio quality, the bass has become woolly and indistinct and small details in songs are harder to hear resulting in a un-involving sound.
I know I am not imagining this as I have done repeated tests using the same file played back via iTunes9, Songbird and Cog and iTunes is clearly the inferior of the three and the poor quality of 9 is being discussed on many hifi forums.
So come on iTunes team either sort out the audio output of 9 or give us audiophiles an easy way to get rid of 9 and go back to 8 without losing our libraries.
ps. love all the bells and whistles but sound quality is paramount.
Cheers
Andy.

I had exactly the same problem on my new, beefed-up laptop and couldn't figure out how an "upgrade" could ruin my music (which I take pretty seriously). I am embarrassed to say what actually fixed the problem. I started by turning off the EQ and it improved somewhat. I kept searching the web without luck until I noticed something by accident. My system volume was low and the iTunes volume was peaked at max. I turned the iTunes volume to 40% and turned up the computer volume to compensate and voila! The bass distortion and indistinct vocal range was immediately fixed. I turned the EQ back on and was able to use it however I liked. I guess it is like turning up the volume on a radio to max. You get sound, but it gets distorted in the upper range. Hope that helps!

Audio quality improvement?

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