Microphone record sample rate problem

Good day. Our company is developing a Flash-application for audio processing
purporses. We encountered the following problem: when we use flash.media.Microphone
object (ActionScript 3) with "rate" property set to 22kHz or 44kHz then
using a spectrum analysis tool it is clear that
actual sampling frequency of recording is 16kHz (instead of 22kHz or 44kHz) which is
too low for high quality voice recording.
Is there any way to solve this problem?
PS. We trying to compile our flash movie under version of player 10.3,11.1,11.2 in CS5 and in CS6 and we have the same result.
PPS. More tested getMicrophone and getEnhancedMicrophone. When microphone initialized with getMicrophone, quality of recorded data is same as rate property of microphone object. When initialized with getEnhancedMicrophone the sample rate of recorded data is 16kHz (view using a spectrum analysis tool. Adobe audition for example). looks like a BUG in flash player.

TheJackAttack wrote:
Some of this is a bit disinenuous. I don't know any "professional" audio engineer that is using on-chip audio. All pro engineers I know-on PC based machines-are using ASIO and professional level interfaces whether PCIe or 1394. The OS is a non starter in that regard. For all the crying about Win7, I just haven't had the issues or problems that seem to make it grossly inadequate for some of my colleagues on this forum.
Unfortunately (perhaps for them?) I know many many professional audio users who mix and match external asio, external wdm, internal on-board wdm drivers. They may not be engineers, although some will be, but they are professionals. An external audio interface running wdm drivers may well be affected by Microsoft settings.
The people I know use laptops, and include professional music writers and critics, broadcast and print journalists, composers, forensic experts and so on. They need to be able to play back occasionally over the laptop speakers and, if they have to listen or work with material from on-line sources, they will often have to use the wdm drivers with their professional audio interface.
Many of them have moved successfully from XP to Windows 7, but they do have to be aware of the possible problems. I find that it takes time to get them set up and running comfortably in Windows 7, but it can be done.
The trouble is that this software is used by so many people in so many different circumstances.

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Audigy 2 NX recording sample rate question

Hi, I have two questions about the sample rate used when recording in the Audigy 2 NX -
When recording an analog signal (Line In), is the actual?sampling rate at which the analog signal is being sampled different from the requested sample rate ? For example, when recording at XX Khz, does the actual sampling of the analog signal really?happen at XX Khz or does it get sampled at a different fixed rate(48khz?) and then gets re-sampled to XX Khz ?
When recording a digital signal (spdif in), if the recording sampling rate specified is the same as the sampling rate of the input signal then does the card record this digital input signal in a bit-perfect fashion ? For example, if the digital input signal is 6bit and 44.Khz, and the recording is also being done at 6bit and 44.Khz then does the card actually record the signal at exactly that rate without modifying the data in any way ? or does it record it at a different rate and then re-sample internally to arri've at the requested recording rate ?
Now I know this card works a little different in terms of re-sampling when the 'Digital Input only' checkbox is checked. So I am looking for the answers to these questions keeping in my mind this option as well.
Thanks!
Message Edited by whatcreative on 02-8-2009 08: PMMessage Edited by whatcreative on 02-8-2009 08: PM

thanks for your replies.
katman,
I checked that thread. Although I wonder if the same thing applies to recording at rates higher than 48khz as well ?as I am more interested in finding out if when recording 96khz does the signal really get sampled at true 96khz or something else ? But, Upsampling a recording (not playback)?to a higher frequency doesn't make sense?because you can't create what you lost by recording at a?lower sampling rate. A test for checking this would be to try and record a signal that has frequency content well above 24khz but less than 48khz. If that content gets captured that means it is indeed recording at 96khz, if it gets lost then its recording at 48khz.
flipflop,
I have opened up the soundcard and messed around with a lot. Here's a list of the chips alongwith their descriptions based on whatever I could find?about them -
Sigmatel STAC9460S - Codec
JRC 4556A - Headphone output opamp
ST MC33078 - Line level output/input dual opamp. (Total 4).
Philips UDA334 - Low Power DAC
Philips ISP58BD - USB device controller
Creative CA086 - DSP
Cirrus Logic EP7309 - High-Performance, Low-Power System-on-Chip Enhanced Digital Audio Interface
ST M29W400BB - 4Mbit Flash Memory
ST LD33 - 3.3v regulator (?)
ST LD8 - .8v regulator (?)
Although the datasheet of the codec suggests that it supports all sampling rates, I'd have to dig deeper to really find out if it is handicapped to only one standard rate for recording - http://www.alsa-project.org/~james/d...tac9460-ds.pdf
Message Edited by whatcreative on 02-2-2009 :35 AM

Sample Rate problem as well as CoreAudio overload and anyone using DFH?

I've only switched from Pro Tools to Logic Pro recently and am confused about a lot of things. I am running Toontrack DFH Superior with 7 audio tracks and one more software instrument track. I keep getting message about sample rate problem and something about conflict betwwen logic and external device. The only device I use is a M-Audio 49e MIDI controller and the M-Box, and I am running these on a Quad PowerMac G5 wih 4 GB RAM... I've tried to adjust the setting here and there, but none of them seem to work... Please help!

Often MIDI driver conflicts and processor overloads, or bottlenecks too. Freezing, removing old drivers, if you really are running Jaguar and if it is actually installed from those days then it is time to do a fresh install and maybe use Panther or Tiger. Panther can be found really cheap these days. As far as old drivers you don't use for any audio or MIDI interfaces you once used with your sytem, are there any? If so which ones? You may be able to direct yourself to your old driver uninstall by looking at the receipt of whichever driver you no longer use. Also, Jaguar was just really good at getting this message to come out, in my experience. Some plugins require alot of MIDI I/O or CPU power, some of NI's stuff really (momentarily) spikes on MIDI data input.

Sample rate problem

I have an odd sample rate problem: I have put in some music at 44.1 (while all other audio is at 48kHz.)
At most points this is, surprisingly, not an issue. But if I haved 4 channels of 48kHz sound then EITHER the 44.1 sound OR the 48kHz sound will not play.
Anyone know why this would be? I obviously need to redo the 44.1kHz sound - is there a quick and easy way to do this, or do I need to reimport everything seperately?

Export the 44.1khz audio as an AIF, 48khz, 16 bit file themn import the resulting file. Normally you'd want to do this with QT Pro before importing it into FCP.
-DH

Sample rate problems while re-opening .PCM file I recorded

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What is the sample rate and bit depth of the file?

Sample Rate Problems going from Logic to Garage Band

I've downloaded the Trial Version of Logic Express. When it launched it changes the i/o sample Rate that is required by Garage Band (I'm using a digi002 i/o). The result is that garage band plays back at a fraction of it normal rate (sounds like crap).
It wasn't a problem until the trial ran out on Logic Express. I could change the sample rate back inside Logic. Hummm ... what to do? I don't see any place to change the rate in the garage band preferences or in the core audio interface. Any suggestions would be appreciated.
G4 1.25 dual, mirror Mac OS X (10.4.6)
G4 1.25 dual, mirror Mac OS X (10.3.9)

updated

Sample Rate Problems going from Garage Band to Logic Express

I've downloaded the Trial Version of Logic Express. When it launched it changes the i/o sample Rate that is required by Garage Band (I'm using a digi002 i/o). The result is that garage band plays back at a fraction of it normal rate (sounds like crap).
It wasn't a problem until the trial ran out on Logic Express. I could change the sample rate back inside Logic. Hummm ... what to do? I don't see any place to change the rate in the garage band preferences or in the core audio interface. Any suggestions would be appreciated.
G4 1.25 dual, mirror Mac OS X (10.4.6)
G4 1.25 dual, mirror Mac OS X (10.3.9)
G4 1.25 dual, mirror Mac OS X (10.4.6)

I've never had any such problem with Logic Express. GB files just load up, and work fine. Did you change anything in Logic Express? There are setting for that there, but even songs I open after I changed LE to 24 bit, work fine.
Wait, are you saying LE is changing the sample rate, and then it stays that way when you open GB? Hmmm, What do you have LE set for? I think mine is set at 24/96, but I'd have to look to be sure. But I am not using the same interface as you are. It might be something you just have to live with. Have you tried posting this in the LE7 forum? Maybe one the gurus there knows what you can do.
Oh, and I am using GB 2, not GB 1, but I am not sure that is relevant.

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:
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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:
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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.

Audigy 2 platinum sample rate problem

People i need help with my audigy... Its connect to my Ht sony ddw880 via coax cable... In audio console, when selected spdif digital out (PCM) sample rate for 48 khz... everything goes fine..But when i select sample rate for 96khz, my receiver detects the signal as 96 pcm but theres no LFE signal... The sound plays normal but without subwoofer... Ive contacted sony and the tell me thats a problem with the signal that the pc is outputting in 96khz mode... cause in 48 khz its ok... I need help with this problem... please anyone.....!!!Thanks a lot.. PS: Sorry about my english (Brazilian)

People i need help with my audigy... Its connect to my Ht sony ddw880 via coax cable... In audio console, when selected spdif digital out (PCM) sample rate for 48 khz... everything goes fine..But when i select sample rate for 96khz, my receiver detects the signal as 96 pcm but theres no LFE signal... The sound plays normal but without subwoofer... Ive contacted sony and the tell me thats a problem with the signal that the pc is outputting in 96khz mode... cause in 48 khz its ok... I need help with this problem... please anyone.....!!!Thanks a lot.. PS: Sorry about my english (Brazilian)

Sample rate problems

I recorded my band at a studio at 48k in protools, then I took the ausio files and loaded them into logic pro, it worked great, I mixed a few songs, spend a huge amount of time mixing and always thought to myself I cant believe we played the songs so slow. at any rate I realized now that I brought all the 48k ausio files into Logic and logic was set at 44k, so it converted them. I tried changing the sample rate in the perferences to 48k and now everything does not line up like it should etc...
Is there anyway to now change my mixes to 48k without losing any infomation including automation?
Thanks

Try making a copy of the song, Erasing your bad tracks and removing them from the audio window. Then reimport at the correct rate. I think you will have to shift the automation.

Non-Uniformed Sampling Rate Problem

Hi All,
I have 2 analog inputs being sampled at 500 Hz. But after looking at the plot of my samples, it doesn't seemed like the intervals are of the same size? Can any one tell me what might cause that and how I can make them all the same?
Thanks much.
hellolv
Attachments:
FILE_4.vi ‏51 KB

hellolv,
I have taken a look at your program, and I have made a few modifications.
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First, you are performing your DAQmx read in two positions inside your
while loop. This means that for every loop iteration you read in two
samples (not one). This makes the logic you use to stop the loop
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I have modified the VI so that only 1 DAQmx read is used. I also changed
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Next, the interval between samples should be constant. The DAQmx device
will acquire data at the specified sampling rate and store the data in the PC
buffer. The graph will display data based upon waveform data type.
This data type only has a t0 (start time), dt (period between each sample), and
the data. I do not know how the graph would display data points that are
not spaced equally based upon only having this data. If you still see this in the code I posted,
can you please explain what you are seeing?
Regards,
Jesse O.
Applications Engineering
National Instruments
Jesse O. | National Instruments R&D
Attachments:
FILE_4_new.vi ‏49 KB

Tuner / sample-rate problem

I'm using MainStage 2.0.1 on OS X 10.6.2 with an Echo AudioFire12 interface.
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Also the audio output is sharp by the same amount when I use the built-in ouput of the mac with interface input.
Seems to be a sample rate issue but I don't know how to fix it.
Any suggestions?

Turned out this was a hardware issue with the Echo AudioFire12.
It was repaired under warranty with excellent customer service from Echo Audio.

1212M PCI sample rate problem

i! I have two audio PC's with 1212M's. One PCI-express, and one PCI, recently bought.
I have some hi-res audio files that I'd like to play at 96 khz, but the PCI version gives no sound if I make a new session at a higher sample rate.
Any ideas?
THanks, Daan

Try to fresh the driver or Profile session. How do you feel about the PCI-Express and Audio Response compared to PCI?

CANOpen - sample rate problem

Hi,
I am using the CANOpen toolkit to communicate with a sensor which can be set to a sample rate of 1kHz. Before I've used the standard Frame API and there I've received my PDO-objects every ms. Because I want to make the programming more easy I would like to use the CANOpen. But I can't set the sample rate to 1ms. When using 2ms the data isn't transmitted with the correct timestamp intervall.
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Maybe somebody can give me a hint.
Best Regards,
Joachim
Solved!
Go to Solution.
Attachments:
canopen_block-diagram.JPG ‏26 KB
canopen_error_1ms.JPG ‏16 KB

Hello DirkW,
thanks for your reply. I've attached 2 example VIs. One with the communication via Frame API (Interface NET) and one via CanOPEN (Frame API (Interface Object)). When running the example with Interface NET I can poll my data every ms. I only have to start the communication via ncWriteNET.vi. After that I am receiving my PDO objects. That means my hardware (slave) is able to deliver data every ms. I've seen that the CanOpen library uses the method 'coPdoCreate' from the ni_cano.* - library. How can I adapt this CanOpen object to communicate via the Interface NET method. I need for starting the PDO-transfer the PDO-object reference for the PDO Create Object.
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Attachments:
MAIN_MP55_RECEIVE_CANFrame.vi ‏52 KB
MAIN_MP55_RECEIVE_CANOpen.vi ‏49 KB

Recording Sample rates

I was thinking of upgrading to a 2nd gen touch to get recording functions, but does anyone know when or if there will be any way of recording @ 48khz on the in the near future? Also, what format(s) does the ipod record in and what apps are good to record with?

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

Microphone record sample rate problem

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