3gs has very low audio sample rate

Every time I listen to Pandora on my 3gs the sample rate is so low that i can barely tell what song is playing. The same thing happens on sirius and even on locally stored apps played on the ipod app. In Pandora there's a setting to increase sample rate, but that didn't work. Anyone else have this problem?

Try another camera, it could be a quirk of your Sony.
happy cutting.
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Similar Messages

Highest audio sampling rate in CS4?

Hello,
I apologize if this has already been asked, but I have been searching everywhere and I simply cannot find the answer to this.
What is the highest audio sampling rate that can be utilized in Premiere Pro CS4? Can it import and export 192kHz 24-bit audio?
Thanks in advance

Hey Hacienda,
I might not have the experience in audio work you have since I've only been doing this for the past 6 years or so. But I've been a musician for far longer than that, and I've learned A LOT mostly from really smart people in the industry. So, I'm not gonna lie to you and say that I've done extensive testing in this area because I simply do not have the equipment, nor the money to buy it (WAY too expensive). But we do share the neophyte status when it comes to video editing :-P
Anyways, the Nyquist Theorem is not a theory, which is what people are led to believe. It is a theorem, meaning it's already mathematically proven. It is proven that, as long as you follow the premise of capturing twice the highest frequency of the sound source, you'll get a perfect reproduction of it. To capture more than that is a waste of bandwith specially because most people won't even hear above 18KHz, nor do they have the equipment to reproduce such frequencies. Most consumer systems and audio gear, including those found in professional studios, go up to about 22KHz. You need to spend BIG dolars for anything that goes beyond that. So, who are we really making music for here? The super rich? Dolphins?
Now, I know you're not just talking about higher frequencies, but the amount of samples needed to recronstruct a perfect copy of the original waveform. OK, well, this is the kind of snake oil marketing BS I was talking about. The biggest one being that 1bit DSD crap that Sony/Phillips is pushing. Adding more samples to the recording will not make any difference on how faithfully you can reproduce a sound. It'll just make the files bigger for no reason. Again, the Nyquist Theorem already proves this. This is FACT! Here's a link I found interesting regarding these audio industry lies, maybe you will too: http://theaudiocritic.com/back_issues/The_Audio_Critic_26_r.pdf It starts on page 5, but the one pertaining this discussion is lie #3 on page 6. :-D
Don't forget that modern converters already sample at much higher frequencies than the target sampling rate. I believe my RME Fireface 400 samples at 5.6MHz, which is twice the amount of samples compared to DSD technology, before going back down to the target rate. But, like I said, it does so for other reasons and NOT because it needs that many samples in order to faithfully reproduce a waveform. Of more importance are the quality of the FIR (Finite Impulse Response) filter and the clock inside the converters. These components are what make a converter high grade, among others. The converter chips themselves are very inexpensive (in the tens of dolars) which why you hear some companies advertizing having the same converter chip as a ProTools HD rig (not the best example I know).
By the way, I didn't say humans can only hear up to 20KHz. I'm sure there are people who can hear above that. My point was that the 20Hz - 20KHz range is what's generally accepted as an average for humans (which implies that there are people who can hear avobe/below that). Also, the reason why modern-day pop records causes headaches and sound horrible is because of a totally different issue known as "The Loudness War" (I'm sure you know about it so I won't go into details). However, I do agree with you as far as compressed audio goes. Unfortunately, there's a reason for that and there's nothing we can do about it until the day Internet bandwith becomes more accessible and cheaper. Eventually it'll get to the point where uncompressed audio can be streamed reliably through the net. But, until then, we're stuck with MP3, AAC, DTS and other audio compression formats. As far as digital media distribution goes, it's the future and companies are seeing that. More and more people download music rather than buying CDs, so I do believe those numbers are accurate. Just look at sales from iTunes and even games like Guitar Hero and Rock Band. It's just a matter of time.
Take care!

How can I find out the audio sampling rate of BetacamSP tape?

Hi guys
I'm trying to digitize BetacamSP tape. But I'm afraid if I might choose wrong setting...
This tape is from very long time ago so we don't know which audio sampling rate we recorded with..
How can I find out the audio sampling rate of this BetacamSP tape?
Thanks:)

The sampling rate is set by the Sony DVMCDA2 you are using, when the conversion is made from the analog input to the digital (DV) output. You should be outputting standard DV which is 16bit 48khz audio.
Assuming you are in the US, your Easy Setup for FCP should be DV-NTSC, and then open the Log and Capture Pane and set the Capture Settings Device Control to Non-Controlable Device and you should be good to go.
You will have to roll the deck manually and start and end your capture manually.
You can download a user manual for the DVMCDA2 by clicking here.
MtD

Need help. Mac OS 10.5.8 crashes when changing audio sample rate.

Anyone have an answer to this problem?
Profire 2626 interface, when switching audio sample rate, crashes Mac Os

Usually when I disconnect the audio interface from my computer and then open up a song that has a different sample rate, the crash doesn't happen. But when the interface is hooked up (this problem started just a few weeks ago...), I get this error, pls bear with me, its long:
Interval Since Last Panic Report: 2159 sec
Panics Since Last Report:          1
Anonymous UUID:                    B03C4E76-CB50-4839-8FB4-A6A511E50492
Fri Sep 23 20:10:24 2011
panic(cpu 0 caller 0x001AB0FE): Kernel trap at 0x00d67395, type 14=page fault, registers:
CR0: 0x80010033, CR2: 0xb795e750, CR3: 0x011a4000, CR4: 0x00000660
EAX: 0x00000000, EBX: 0x00000004, ECX: 0x00000040, EDX: 0x00000000
CR2: 0xb795e750, EBP: 0x2e76fbd8, ESI: 0xe00007c8, EDI: 0x3795c830
EFL: 0x00010202, EIP: 0x00d67395, CS: 0x00000008, DS: 0x00000010
Error code: 0x00000002
Backtrace (CPU 0), Frame : Return Address (4 potential args on stack)
0x2e76f9e8 : 0x12b4c6 (0x45f91c 0x2e76fa1c 0x13355c 0x0)
0x2e76fa38 : 0x1ab0fe (0x469a98 0xd67395 0xe 0x469248)
0x2e76fb18 : 0x1a1713 (0x2e76fb30 0x2e76fc4c 0x2e76fbd8 0xd67395)
0x2e76fb28 : 0xd67395 (0xe 0x48 0x10 0x10)
0x2e76fbd8 : 0xd65b1b (0x37a70000 0x3795c830 0x4 0x10)
0x2e76fc58 : 0xd519be (0x4678000 0x37b00bc0 0x37a70000 0x22f)
0x2e76fc88 : 0xd51bcb (0x4678000 0x37b00bc0 0x37a70000 0x22f)
0x2e76fcd8 : 0xb528a0 (0x4678000 0x22f 0x100 0x41cfb68)
0x2e76fd28 : 0xb52687 (0x41cfb00 0x22f 0x100 0x0)
0x2e76fd68 : 0xb52463 (0x41cfb00 0x3782c010 0x37af8000 0x22f)
0x2e76fde8 : 0xb4db00 (0x41cfb00 0x47c5908 0x22f 0x1)
0x2e76fe68 : 0xb4d18a (0x4371200 0x22f 0x1 0x455d580)
0x2e76fea8 : 0x4437c8 (0x4371200 0x22f 0x1 0x0)
0x2e76ff08 : 0x198fa3 (0x2e76ff44 0x0 0x0 0x0)
0x2e76ffc8 : 0x1a1cfa (0x533c6c0 0x1 0x10 0x0)
No mapping exists for frame pointer
Backtrace terminated-invalid frame pointer 0xb035fb58
      Kernel loadable modules in backtrace (with dependencies):
         com.m-audio.driver.firewire.dice(2.3.2)@0xd4d000->0xd77fff
            dependency: com.apple.iokit.IOFireWireFamily(3.4.9)@0x795000
            dependency: com.apple.iokit.IOAudioFamily(1.6.9fc5)@0xb43000
         com.apple.iokit.IOAudioFamily(1.6.9fc5)@0xb43000->0xb59fff
            dependency: com.apple.kext.OSvKernDSPLib(1.1)@0xb40000
BSD process name corresponding to current thread: Logic Pro
Mac OS version:
9L31a
Kernel version:
Darwin Kernel Version 9.8.0: Wed Jul 15 16:55:01 PDT 2009; root:xnu-1228.15.4~1/RELEASE_I386
System model name: MacPro3,1 (Mac-F42C88C8)
System uptime in nanoseconds: 1278160883646
unloaded kexts:
com.apple.driver.InternalModemSupport          2.4.0 - last unloaded 190309720723
loaded kexts:
com.m-audio.driver.firewire.dice          2.3.2
com.apple.driver.InternalModemSupport          2.4.0 - last loaded 9798699115
com.apple.iokit.IOBluetoothSerialManager          2.1.9f10
com.apple.filesystems.autofs          2.0.2
com.apple.driver.AppleHDAPlatformDriver          1.7.1a2
com.apple.driver.AppleHDAHardwareConfigDriver          1.7.1a2
com.apple.driver.AppleUpstreamUserClient          2.7.5
com.apple.driver.AppleHDA          1.7.1a2
com.apple.kext.ATY_Lamna          5.4.8
com.apple.driver.AppleHWSensor          1.9d0
com.apple.Dont_Steal_Mac_OS_X          6.0.3
com.apple.driver.AppleTyMCEDriver          1.0.0d28
com.apple.driver.AppleHDAController          1.7.1a2
com.apple.iokit.IOFireWireIP          1.7.7
com.apple.driver.AppleUSBDisplays          2.0.2
com.apple.driver.AudioIPCDriver          1.0.6
com.apple.ATIRadeonX2000          5.4.8
com.apple.driver.AppleMCEDriver          1.1.7
com.apple.driver.ACPI_SMC_PlatformPlugin          3.4.0a17
com.apple.driver.AppleLPC          1.3.1
com.apple.driver.CSRUSBBluetoothHCIController          2.1.9f10
com.apple.driver.AppleHIDKeyboard          1.0.9b4
com.apple.driver.AppleUSBMergeNub          3.5.2
com.apple.driver.CSRHIDTransitionDriver          2.1.9f10
com.apple.driver.PioneerSuperDrive          2.0.9
com.apple.iokit.SCSITaskUserClient          2.1.1
com.apple.iokit.IOATAPIProtocolTransport          1.5.3
com.apple.driver.XsanFilter          2.7.91
com.apple.iokit.IOAHCIBlockStorage          1.2.2
com.apple.driver.AppleFileSystemDriver          1.1.0
com.apple.driver.AppleUSBHub          3.4.9
com.apple.driver.AppleAHCIPort          1.7.0
com.apple.driver.AppleIntelPIIXATA          2.0.1
com.apple.driver.AirPortBrcm43xx          367.91.22
com.apple.iokit.IOUSBUserClient          3.5.2
com.apple.driver.AppleIntel8254XEthernet          2.1.2b1
com.apple.driver.AppleFWOHCI          3.9.7
com.apple.driver.AppleUSBEHCI          3.4.6
com.apple.driver.AppleUSBUHCI          3.5.2
com.apple.driver.AppleEFINVRAM          1.2.0
com.apple.driver.AppleACPIButtons          1.2.5
com.apple.driver.AppleRTC          1.2.3
com.apple.driver.AppleHPET          1.4
com.apple.driver.AppleACPIPCI          1.2.5
com.apple.driver.AppleSMBIOS          1.4
com.apple.driver.AppleACPIEC          1.2.5
com.apple.driver.AppleAPIC          1.4
com.apple.security.seatbelt          107.12
com.apple.nke.applicationfirewall          1.8.77
com.apple.security.TMSafetyNet          3
com.apple.driver.AppleIntelCPUPowerManagement          76.2.0
com.apple.driver.DiskImages          199
com.apple.BootCache          30.4
com.apple.iokit.IOSerialFamily          9.4
com.apple.driver.DspFuncLib          1.7.1a2
com.apple.iokit.IOHDAFamily          1.7.1a2
com.apple.iokit.IOAudioFamily          1.6.9fc5
com.apple.kext.OSvKernDSPLib          1.1
com.apple.iokit.IONDRVSupport          1.7.3
com.apple.iokit.IOGraphicsFamily          1.7.3
com.apple.driver.IOPlatformPluginFamily          3.4.0a17
com.apple.driver.AppleSMC          2.3.1d1
com.apple.driver.AppleUSBBluetoothHCIController          2.1.9f10
com.apple.iokit.IOBluetoothFamily          2.1.9f10
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com.apple.iokit.IOACPIFamily          1.2.0

What is the best way to deal with different audio sample rates on the same timeline ?

what is the best way to deal with different audio sample rates on the same timeline ?

You don't have to do anything special. If possible, start your project with a clip that has the desired target frame rate and audio sample rate, and your project parameters will be set automatically. Other sample rates will be converted under the covers.
For example, if your video is shot at 48khz, you can add music files at 44.1khz with no problem.
If you are recording audio that you want to synch with video (multicam), you will get best results if everything is 48khz, but you can use 44.1 if that is all you have. Once I forgot to reset my Zoom to 48,000 and it still worked.

Problem with audio sample rate when exporting

I am having an issue with my audio sample rate. For some reason it is set at 8 khz by defualt when I am exporting. This was not an issue until the latest update (2 weeks ago). I'll try and give all the info I can up front. Also I tried customer support...that was a nightmare.
Ok the issue I am addressing is that the audio sample rate when exporting is at 8khz by default. Which to my knowledge has NO relevant use what so ever. Even when I set the file format to h.264 it insists on resetting to 8khz. Even if I manually set everything look at the render queue and click on output module again it changes the sample rate back to 8 khz automatically which I must adjust again. Why can it not just be set at 48000 like every other adobe product is and like it used to be two weeks ago. Does anybody have an answer?

This problem is addressed by the After Effects CC (12.2.1) bug-fix update, which is now available:
http://adobe.ly/AE_CC_1221
Note the part at the end of that page about a crucial update for the Creative Cloud desktop application, which addresses some severe problems with AME, Premiere Pro, and After Effects.

IDT High Definition Audio CODEC v. 6.10.6395.0 Upon installation I have very low audio output

IDT High Definition Audio CODEC v. 6.10.6395.0. 5/21/ 2012
Upon installation I have very low audio output! Screen Shot Screen Shot 2
I have tried:
un-installing and reinstalling the drivers.
Installed the original drivers via HP's website sp53222 for my particular model. HP Pavilion HPE h8-1070t
Any suggestions? would be greatly appreciated. It would appear IDT who I also emailed and they never responded is responsible for a codec pack that is not fully functional with this motherboard.
was wir lieben meistens nicht kriegen

Did you access the Catalyst Control Center in programs and look at the latest updates for your graphics card and make sure you didn't download the audio drivers they self install and are checked automatically each time you attempt an update for the card.
Did you right click on the speaker in the system tray and go into recodring devices make sure everything is setup propper there sometimes they too can change.
I'm familiar with cakewalk but never used it on Win7 its been many years since I played around with that. If you are using the onboard audio card and it has worked before and not now if I were in your situtation I would attempt to reisntall the dirvers via device manager they do sometimes become corrupt especially if you picked up any malware that was removed by your antivirus software due to the drivers and malware both reside inwindows\system32.
After the reisntall of the drivers if the issue is not resolved I would upgrade the audio card with a PCIe x1 component such as:
Creative Labs SB1040 PCI Express Sound Blaster X-Fi Xtreme Audio Sound Card
http://www.amazon.com/Creative-Labs-SB1040-Express-Blaster/dp/B000W7PNZI/ref=sr_1_1?ie=UTF8&qid=1357...
After I paid for a real soundcard with Recording: 24-bit/96kHz and Sound Output Mode - 7.1 and bypassed my onboard audiocard I never had any issues again. My onbaord worked too in the begining for certain things than stopped mysteryously and the reinstallation became ineffective after a while so I decided the money for the tower was invested mainly in the CPU i7 the RAM and HDD.
*** Note if you did install the audio drivers for the graphics card you will have to change the onboard audio in the BIOS from automattic to enabled. Otherwise you will have no sound at all. So never take the audio drivers for the graphics card unless you need them for DVR recording or something. The PCIe audio cards are never effected by this however only the onboard audio. Another reason onboard is a cheap way to go and HP bundles it as cheap as it can my friend.
Good Luck again!
was wir lieben meistens nicht kriegen

DV 16:9 but Only Exports 4:3 WHY? Also Audio Sample Rate Problem

I'm quite new to Final Cut and have FC 6.01. I use PAL 25fps and a 16:9 SD DV Camcorder with FireWire SONY DV VTR Deck. I have two problems:
1. When i capture my 16:9 DV footage the Logging Window shows only 4:3 with a distorted image in it. Though I have chosen the easy set up and told Final Cut that im editing Anamorphic 16:9 I also found that FC will only export 4:3. However, during the editing process i see 16:9 in the Browser Preview and Canvas Window.
2.(Not sure if its related) I get the following message everytime after I capture individual clips or if i press the Escape key during a capture:
"The audio sample rate of one or more of your captured media files does not match the sample rate on your source tape. This may cause the video and audio of these media files to be out of sync. Make sure the audio sample rate of your captured preset matches the sample rate of your tape"
Does any one else have this problem?
Apple, as yet have not given me any answers.

Danny Boy.. Thanks for your reply and I'll be happy if it is my fault and not FCP's. Actually i made an error in my post. It does indeed display correctly in DVDSP, it's iDVD it does not even when told to display it in 16:9. However, when i used the PAL DVD Anamorphic file preset, iDVD still couldn't display it. To get it working I had to tell Compressor specifically (in the additional settings) to encode 16:9 despite what the presets stated! No matter how one looks at this, this is confusing to say the least! If a preset says 16:9 then one should expect 16:9! Remember, Im using the display window of iDVD to show me the output.
To detail my steps as hanumang has said, im doing the following:
1. in FCP I encode to QuickTime Move
2. Open iDVD & Create a Project
3. 'Drag' The QT file into the menu
4. Using iDVD Preview function, Preview the QT file
5. ITS STILL 4:3
now, the above was done with QT Conversion which as also set to encode 16:9 and still had the same result.
Thnaks to you both.

Capturing-audio sample rate

I have fce hd v 3.5. My camera is a sony dcr-trv11. I am trying to "capture now" but it looks like I am dropping frames. When i hit "esc" it stops capture of course, but I get a message that reads "the audio sample rate of one or more of your captured media files does not match the sample rate on your source tape. This may cause the video ans audio of these media files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape." Okay, so how do I do that? When I originally captured my footage with imovie I had no problems. I had trouble importing them to FCE so I scrapped them all and decided to start from scratch with FCE. I have 199.5 GB of free space.

Most miniDV cameras can be set to record audio as 12-bit or 16-bit; and most of them come set to 12-bit as the default.
Your preset (easy setup) in FCE has to match both the video & audio setup of your camera. Normally, you would select the DV-NTSC easy setup in FCE, which would give you a sequence that expects DV-NTSC video and 16-bit (aka 48KHz) audio. If your video was recorded as 12-bit (aka 32KHz) audio, but you captured into a 16-bit (48KHz) sequence in FCE, that would give you the mismatch.
Check your camera - if it was set to 12 bit audio for the tape you are trying to capture, then in FCE you should select the DV-NTSC 32KHz easy setup for your sequence before you capture your tape.
There are many different reasons you might be getting dropped frames - can you tell us more about your exact setup, esp. if you have an external HD connected to your system. Oh, and by the way, 512MB is the bare minimum to run FCE, you will find things much better overall if you upgrade to at least 1GB.

Geting audio sample rate error, help

Hey all, been doing a massive project where I ma bring in tons of old 8 mm tapes, hi8 and digital 8. This one tape I brought on though however is giving me grief. I keep getting this error:
The audio sample rate of one or more of your captured media files does not match the sample rate on your source tape. This may cause the video and audio of these files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape.
If find this odd though cause the tape should be standard, NTSC dv 48 K. Any suggestions. Also how do I reset my final cut so that when I plug in a camera it always reads it as what it recognizes. I ask this because I had to used the setting uncontrollable device because on the original old 8mm tapes there is no time code so I had to capture that way.
Anyhow, any suggestions on this would be great for if I recapture. Cause I could line it up by eye but want to find the problem so I know for the future. Thank you.
Nathan

This is a recent problem that seems related to a recent upgrade of QuickTime. Here's why.
In the last month, a rash of these posts have begun to appear:
"DV Capture Audio problem"
http://discussions.apple.com/thread.jspa?messageID=6708693&#6708693
"audio/video"
http://discussions.apple.com/thread.jspa?messageID=6591262&#6591262
Plus this thread, plus my own.
In my case, nothing changed in my operating system or Final Cut Pro version. I upgraded to QuickTime 7.1.6, and the problem began. I have upgraded all the way to 7.4 to no avail. When I attempt to import a DV clip using the same Sony DVCAM deck that imported the same clip in December, I get the error every time. Nothing has changed in the tape, the deck, the Project or Final Cut. I am simply unable to import DV video. I can import other kinds (Panasonic P2, for example), but DV is a no go. I cannot get rid of this error.

Problems with audio sample rate when capturing DV50

Hello from Spain, I have a problem when capturing DV50 source footage: I setup my sequence settings and capturing settings like always, DV50 codec and 48k sample rate for audio, then when the capturing process finish the result is a file that contains audio to a non-48k audio sample rate, then video and audio don´t match. The sample rate is 47945 and other similars, not always the same, sometimes 47214 sometimes 48000, and there is no rules...My capturing device works ok in other computers and the footage has been capture ok in other Mac...I don´t know what happens.
Sebas, kind regards
G5 Quad Mac OS X (10.4.6)

Hello!, first of all thanks for your reply. Yes, the camera is the same, the problem is on my computer, I have discovered a solution: first I capture the audio, only the audio, and then the video, only the video. After this I soncronize the two separate files on Final Cut timeline. It´s a slow process, but I am really desesperated. I don´t know what happens, I few minutes ago I have captured a 30 mins. video and the sample rate of the audio was 42.145k (???). It´s a crazy thing...

MiniDV audio sample rate issue

I cannot get the audio sample rate to match. I have captured miniDV footage (32mhz/12bit). In Sequence settings the audio options for bit are 8, 16, 24. My audio is out of sync!!!! What can I do? I have 2 hours of raw footage and can't proceed! Any help is VERY appreciated. thanks in advance!

Hi again!
Hum ... that´s not as simple as it could be!
That way i´ll have to capture twice?!
There must be another way of doing it , i don´t think Avid can do it and FCP don´t .
As i told you i use both systems and i know they´re limitions (or i think i know!), but it´s strange because Avid Xpress DV 3.5 (from the age of stone ) do it in a blink ... ok ... found an FCP limitation!
Thank you!
If there´s another way ... please feel free to post it!
(and is was supposed to "reply" a "miniDV audio sample rate issue.
HBars

Error message stops .wmv creation: "Inconsistent Audio Sample Rate..."

Just recently when attempting to export a video into the .wmv format, QuickTime Pro 7.2 gives me this alert message, "Inconsistent Audio Sample Rate. The media you are exporting contains audio with multiple sample rates." QuickTime will not compress (export) to .wmv. It will export to .mov. and .m4v.
This problem is new. For more than a year my weekly video exported to the .wmv format without fail.
I am working on a Mac G4 Dual Processor and iMovieHD.
Any help you can offer would be greatly appreciated.
Mac G4 Dual Processor Mac OS X (10.4.10)

Just in from flip4mac support:
The recent release of QuickTime 7.2 created an audio compression error for Flip4Mac WMV Studio, Studio Pro and Studio Pro HD customers. Specifically, when encoding WMV files with Flip4Mac WMV v.2.1.2 using uncompressed audio sources on QuickTime 7.2 systems, customers were receiving "Inconsistent Audio Sample Rate" error and the encode will fail. This problem has been resolved with the beta release of Flip4Mac WMV v2.1.33. If you are using QuickTime 7.2, please upgrade to this release at http://www.flip4mac.com/wmv_beta.htm

Error Msg.: "Inconsistent Audio Sample Rate" with Flip4Mac

If you're using Flip4Mac to produce WMV files, the latest QT updates will create a conflict with Flip4Mac, which produces the error msg. referenced in the subj. heading above.
If you're using Flip4Mac, you may want to wait until Telestream resolves the issue before upgrading QT.
More info is here: http://www.flip4mac.com/knowledge/kb_0070.htm
and here: http://forum.flip4mac.com/forum/messageview.aspx?catid=10&threadid=270&enterthre ad=y
MacPro, MBP 17in (2 GB RAM), PB G4 15in, PB G4 12in, iMac G3 Mac OS X (10.4.9) Loyal to Apple since the Apple ][+

Just in from flip4mac support:
The recent release of QuickTime 7.2 created an audio compression error for Flip4Mac WMV Studio, Studio Pro and Studio Pro HD customers. Specifically, when encoding WMV files with Flip4Mac WMV v.2.1.2 using uncompressed audio sources on QuickTime 7.2 systems, customers were receiving "Inconsistent Audio Sample Rate" error and the encode will fail. This problem has been resolved with the beta release of Flip4Mac WMV v2.1.33. If you are using QuickTime 7.2, please upgrade to this release at http://www.flip4mac.com/wmv_beta.htm

Why is FME audio sample rate restrictive?

I am unable to select higher than 22kHz audio sample rate on
the FME Control Panel - this results in very poor audio quality
delivered! The input device delivers up to 48kHz and is succesfully
addressed by WME - the only reason why I'm still using WME.
In case you are interested, I have listed the input device
spec below:
DVBETA (LairdTelemedia)
Output Ports available (I use FireWire)
SDI Digital Video: SMPTE 259, PAL or NTSC, on BNC a
connector
AES/EBU Digital Stereo Audio: 16 Bit @ 48 kHz, Embedded on
SDI connector
* IEEE-1394 (FireWire) x2: maximum 400 Mbps, PAL or NTSC, on
a 4-Pin and 6-Pin connectorsMachine Control, RS-422, on a 9-Pin
connector
Time Code: DVITC or VITC, on a BNC connector

I have the same issue when using my video capture card in FME
on Vista.
I am having a lot of issues actually.
1. CPU while encoding video & audio is only reaching
around 15% but I am dropping frames when I try and use high quality
encoding settings.
2. Audio and Video get out of sync very easily.
3. Audio sounds VERY odd, especially during music. It seems
like the speed of the audio is constantly shifting. Makes things
sound like oooOOOoooOOoo etc..
I hope Adobe is addressing these issues.
Thanks and take care.

3gs has very low audio sample rate

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