Sample rate 40938/playback -synchronization issue

Hi, my first attempts with my new Macbook Pro and LE9.1.1 are painfull ;-( After being forced to bounce in place every recorded region (see previous topic) , I am now after a few hours recording confronted with the fact that the monitored sound stops/ hampers and following error appears:
"Error while trying to synchronize audio and midi ; Sample rate 40938 recognized; check conflict between Logic Express and external device".This device is a Tascam US-122 USB interface , with latest drivers, which I use as input and monitoring tool. Restarting LE or even entire Macbook doesn't solve the issue. Tried changing sample rates etc.. in vein. No CPU overload or whatsoever.
NO clue however what/how to check. Any ideas or experience?

Yes I have the latest drivers and everything. Even on the latest release of Premiere Pro CC 2014, I have the same issue. It is so strange being that the high end sound card should be able to do this much better and faster than my system processor. I would like to use the benefits of ASIO for Everyhting, not only mixing 5.1. Anyone else try a similar setup?

Similar Messages

Force mpd to play at 41000 sample rate (high CPU usage issue)

Playing a mp3 file with mpd causes high cpu usage but mplayer plays it without significant cpu usage.
Googling has brought be to the inference that its the issue of mpd playing with different settings (ie at higher sample_rate).
I added .asoundrc but then also mpd is not respecting it.
Basically, I want to play mpd with the same settings as mplayer plays.
My .asoundrc file:-
defaults.pcm.rate_converter "44100"
When I play a file with mplayer.
shadyabhi@archlinux-N210 ~ $ cat /proc/asound/card0/pcm0p/sub0/hw_params
access: RW_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 1024
buffer_size: 16384
OSS format: S16_LE
OSS channels: 2
OSS rate: 44100
OSS period bytes: 4096
OSS periods: 16
OSS period frames: 1024
shadyabhi@archlinux-N210 ~ $
And when I have mpd running,
shadyabhi@archlinux-N210 ~ $ cat /proc/asound/card0/pcm0p/sub0/hw_params
access: MMAP_INTERLEAVED
format: S32_LE
subformat: STD
channels: 2
rate: 48000 (48000/1)
period_size: 1024
buffer_size: 8192
shadyabhi@archlinux-N210 ~ $

shadyabhi@archlinux-N210 ~ $ zgrep SND /proc/config.gz | grep "="
CONFIG_SND=m
CONFIG_SND_TIMER=m
CONFIG_SND_PCM=m
CONFIG_SND_HWDEP=m
CONFIG_SND_RAWMIDI=m
CONFIG_SND_JACK=y
CONFIG_SND_SEQUENCER=m
CONFIG_SND_SEQ_DUMMY=m
CONFIG_SND_OSSEMUL=y
CONFIG_SND_MIXER_OSS=m
CONFIG_SND_PCM_OSS=m
CONFIG_SND_PCM_OSS_PLUGINS=y
CONFIG_SND_SEQUENCER_OSS=y
CONFIG_SND_HRTIMER=m
CONFIG_SND_SEQ_HRTIMER_DEFAULT=y
CONFIG_SND_DYNAMIC_MINORS=y
CONFIG_SND_SUPPORT_OLD_API=y
CONFIG_SND_VERBOSE_PROCFS=y
CONFIG_SND_VMASTER=y
CONFIG_SND_DMA_SGBUF=y
CONFIG_SND_RAWMIDI_SEQ=m
CONFIG_SND_OPL3_LIB_SEQ=m
CONFIG_SND_EMU10K1_SEQ=m
CONFIG_SND_MPU401_UART=m
CONFIG_SND_OPL3_LIB=m
CONFIG_SND_VX_LIB=m
CONFIG_SND_AC97_CODEC=m
CONFIG_SND_DRIVERS=y
CONFIG_SND_DUMMY=m
CONFIG_SND_ALOOP=m
CONFIG_SND_VIRMIDI=m
CONFIG_SND_MTPAV=m
CONFIG_SND_MTS64=m
CONFIG_SND_SERIAL_U16550=m
CONFIG_SND_MPU401=m
CONFIG_SND_PORTMAN2X4=m
CONFIG_SND_AC97_POWER_SAVE=y
CONFIG_SND_AC97_POWER_SAVE_DEFAULT=0
CONFIG_SND_SB_COMMON=m
CONFIG_SND_SB16_DSP=m
CONFIG_SND_PCI=y
CONFIG_SND_AD1889=m
CONFIG_SND_ALS300=m
CONFIG_SND_ALS4000=m
CONFIG_SND_ALI5451=m
CONFIG_SND_ASIHPI=m
CONFIG_SND_ATIIXP=m
CONFIG_SND_ATIIXP_MODEM=m
CONFIG_SND_AU8810=m
CONFIG_SND_AU8820=m
CONFIG_SND_AU8830=m
CONFIG_SND_AW2=m
CONFIG_SND_AZT3328=m
CONFIG_SND_BT87X=m
CONFIG_SND_CA0106=m
CONFIG_SND_CMIPCI=m
CONFIG_SND_OXYGEN_LIB=m
CONFIG_SND_OXYGEN=m
CONFIG_SND_CS4281=m
CONFIG_SND_CS46XX=m
CONFIG_SND_CS46XX_NEW_DSP=y
CONFIG_SND_CS5530=m
CONFIG_SND_CS5535AUDIO=m
CONFIG_SND_CTXFI=m
CONFIG_SND_DARLA20=m
CONFIG_SND_GINA20=m
CONFIG_SND_LAYLA20=m
CONFIG_SND_DARLA24=m
CONFIG_SND_GINA24=m
CONFIG_SND_LAYLA24=m
CONFIG_SND_MONA=m
CONFIG_SND_MIA=m
CONFIG_SND_ECHO3G=m
CONFIG_SND_INDIGO=m
CONFIG_SND_INDIGOIO=m
CONFIG_SND_INDIGODJ=m
CONFIG_SND_INDIGOIOX=m
CONFIG_SND_INDIGODJX=m
CONFIG_SND_EMU10K1=m
CONFIG_SND_EMU10K1X=m
CONFIG_SND_ENS1370=m
CONFIG_SND_ENS1371=m
CONFIG_SND_ES1938=m
CONFIG_SND_ES1968=m
CONFIG_SND_ES1968_INPUT=y
CONFIG_SND_FM801=m
CONFIG_SND_FM801_TEA575X_BOOL=y
CONFIG_SND_FM801_TEA575X=m
CONFIG_SND_HDA_INTEL=m
CONFIG_SND_HDA_HWDEP=y
CONFIG_SND_HDA_RECONFIG=y
CONFIG_SND_HDA_INPUT_BEEP=y
CONFIG_SND_HDA_INPUT_BEEP_MODE=1
CONFIG_SND_HDA_INPUT_JACK=y
CONFIG_SND_HDA_PATCH_LOADER=y
CONFIG_SND_HDA_CODEC_REALTEK=y
CONFIG_SND_HDA_CODEC_ANALOG=y
CONFIG_SND_HDA_CODEC_SIGMATEL=y
CONFIG_SND_HDA_CODEC_VIA=y
CONFIG_SND_HDA_CODEC_HDMI=y
CONFIG_SND_HDA_CODEC_CIRRUS=y
CONFIG_SND_HDA_CODEC_CONEXANT=y
CONFIG_SND_HDA_CODEC_CA0110=y
CONFIG_SND_HDA_CODEC_CMEDIA=y
CONFIG_SND_HDA_CODEC_SI3054=y
CONFIG_SND_HDA_GENERIC=y
CONFIG_SND_HDA_POWER_SAVE=y
CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0
CONFIG_SND_HDSP=m
CONFIG_SND_HDSPM=m
CONFIG_SND_HIFIER=m
CONFIG_SND_ICE1712=m
CONFIG_SND_ICE1724=m
CONFIG_SND_INTEL8X0=m
CONFIG_SND_INTEL8X0M=m
CONFIG_SND_KORG1212=m
CONFIG_SND_LX6464ES=m
CONFIG_SND_MAESTRO3=m
CONFIG_SND_MAESTRO3_INPUT=y
CONFIG_SND_MIXART=m
CONFIG_SND_NM256=m
CONFIG_SND_PCXHR=m
CONFIG_SND_RIPTIDE=m
CONFIG_SND_RME32=m
CONFIG_SND_RME96=m
CONFIG_SND_RME9652=m
CONFIG_SND_SONICVIBES=m
CONFIG_SND_TRIDENT=m
CONFIG_SND_VIA82XX=m
CONFIG_SND_VIA82XX_MODEM=m
CONFIG_SND_VIRTUOSO=m
CONFIG_SND_VX222=m
CONFIG_SND_YMFPCI=m
CONFIG_SND_SPI=y
CONFIG_SND_USB=y
CONFIG_SND_USB_AUDIO=m
CONFIG_SND_USB_UA101=m
CONFIG_SND_USB_USX2Y=m
CONFIG_SND_USB_CAIAQ=m
CONFIG_SND_USB_CAIAQ_INPUT=y
CONFIG_SND_USB_US122L=m
CONFIG_SND_PCMCIA=y
CONFIG_SND_VXPOCKET=m
CONFIG_SND_PDAUDIOCF=m
CONFIG_SND_SOC=m
CONFIG_SND_SOC_I2C_AND_SPI=m
shadyabhi@archlinux-N210 ~ $ lsmod | grep snd
snd_seq_dummy 1479 0
snd_seq_oss 29240 0
snd_seq_midi_event 5516 1 snd_seq_oss
snd_seq 50562 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device 5281 3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_pcm_oss 39509 1
snd_mixer_oss 17730 1 snd_pcm_oss
snd_hda_codec_realtek 295589 1
snd_hda_intel 21837 1
snd_hda_codec 74609 2 snd_hda_codec_realtek,snd_hda_intel
snd_hwdep 6222 1 snd_hda_codec
snd_pcm 72321 3 snd_pcm_oss,snd_hda_intel,snd_hda_codec
snd_timer 19537 2 snd_seq,snd_pcm
snd 58906 11 snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer
soundcore 6161 2 snd
snd_page_alloc 7361 2 snd_hda_intel,snd_pcm
shadyabhi@archlinux-N210 ~ $

How do I determine that my audio sample rate matches my tape?

How do I determine that my audio sample rate of my capture preset matches the sample rate of my tape?

You've really got two issues.
DV records audio and video differently and if the camera is substandard, you will have drift over long captures. Solutions: new, better camera or capture in small (max 10 minutes) chunks.
The other issue is the 32k sample rate, which will exacerbate issue 1. You can duplicate the 48k preset and modify the copy for 32k and work in that until you are ready to deliver. Then copy it into a 48k sequence, render and output.
It's funny, sometimes I get that pop up but everything is fine.

Error while trying to synchronize audio and MIDI. Sample rate 42804 recognized. Check conflict between Garageband and external device.

Sometimes, while playing back my software instrument songs, I get an intermittent pop-up error message, saying "Error while trying to synchronize audio and MIDI. Sample rate 42804 recognized. Check conflict between Garageband and external device."
(Sometimes the five digit number is different, but remains a five-digit number beginning with "4".)
Simultaneously, the song stops until I press the "Okay" button in the pop-up window.
When I continue to play the song, the sound is jerky and clipped, and the playhead doesn't keep up with the song, and then suddenly jumps to the part of the song currently being played.
There's also a sound of static.
The issue seems to occur whether or not I have my MIDI controller turned on and plugged into my desktop Imac.
Tony

Hello,
open your Audio MIDI Setup utility and set the input to 44100
https://discussions.apple.com/message/12710638#12710638

Slow Playback Adobe Audition CS5.5 - not sample rate setting error!

I'm currently using Windows 7 and Adobe Audition CS5.5 build 1815. Soundcard is RME Digi96 Pad. Ok its an old card and the drivers date from 2005. In Adobe 3.1 and CS5.5 I get slow, around half speed, playback at 44k and 96k sample rates. YES the project sample rates and sound card sample rates are set correctly, clock is set to master on the soundcard.
However, in using Cockos Reaper 3 or 4 on exactly the same computer, same soundcard, same OS, I can play 44k and 96k wav files perfectly.
This suggests to me there is an error in the way Adobe Audition 3 / CS5.5 is handling the ASIO drivers in Windows 7 with some soundcards.
If it was a driver problem, or indeed a soundcard hardware issue, then Reaper would not perform properly.
So what's going on with the ASIO drivers in Adobe Audition with Windows 7?

I have two XLR wireless mics, which are great for my video work, but I need to do audio recordings for podcasts and wanted a mic that would be able to do just audio podcasts, as well as video podcasts, live via my laptop. So while I do agree that the audio recording from my XLR mics is great, I am not happy with the audio quality of standard computer mics I've tried and the only way I can get my XLR mics into the computer directly is through a USB connector (which I also have, I'm getting the Yeti set-up for my business partner to use).
Maybe Audition just isn't the best software option for this, but since we have CS6 Premium I had hoped I could utilize software I already had. (And for the live video casts I'm looking at starting out with Google hangouts...)
With that being said, I appreciate your reply and thought I had already set-up the hardware correctly in Windows & Audition. Here is a screenshot that shows the settings I have - my question at this point is whether the master clock needs to be changed (the audio channel mapping tab shows the left and right speaker of the default Realtek speaker):
I just double checked the properties of both the speakers and mic and they are both set to 48,000/16/stereo, as is the audio file in Audition.
I'm a journalist, not an audio pro, and I fully accept that the sound device must not be properly connected. The problem is that I cannot figure out why that is the case.
Since I need to use the USB mic and I want to be able to use the power of Audition to put together podcasts, using other software for the livecasts, I would really like to sort through this. Any and all help is greatly appreciated!

Tape synchronization at high sample rates + recording at 192 Khz 24 bit

I was expecting at least that Apple would fix Emagix synchronization problem that occurs when Logic is slave to a tape recorder at sample rate other than 44.1 or 48Khz.
Further more I do get click , pops and digital noise when I try to master @ 192Khz 24 bit most of the time (not synchronized to any code).
For archiving tapes or going back and forth to tape I use Digi 002 , Rosentahl WIF (SMPTE to MTC) and Pro tools , as Logic is incapable for such a task , and for mastering and mix-down I use Lynx AES 16 with dCS904 AD and dCS954 DA with LogicPro.

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.

Capturing issues-audio sample rate & locating timecode break

I am a first time FCP user.
I am working on an educational video with hours of DV source tapes.
The source tapes are from a 2R50MC Cannon mini DV camcorder also used as the deck I am capturing from via firewire. My scratch disk is Maxtor III 500GB external drive hooked up via a firewire (I'm realizing the firewire is years old could that be a problem?)
In my first capture test I captured 1 minute of tape by setting in and out points and after completeing the capture I received an error message—"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 capture preset matches the sample rate of your tape."
For the past day I have been scouring the user manual, this discussion group and training tapes from Lynda.com and can not resolve this issue. I have determined that the source tape is 12 bit or 32 kHz and yet can not find a way to set up capture preferences to 32 kHz. Is this the problem/resolution? Page 320 of the user manual shows the QuickTime Audio Settings dialog box but I can't find it in FCP or Quicktime. Is this where I make the change to 32kHz?
After doing the first test, I tried again and ran into a new problem—it stopped capturing with the message: Locating timecode break [press esc to abort]. It never seemed to resume capturing and I pressed escape to abort. If I'm reading FCP time code window correctly I received this time code break message within a section I had already captured previously (with the audio sync warning).
I have done several capture tests at different points in the tape and on different tapes all with the same results. I've used the capture clip and capture now buttons. I've tested with drop frame turned on and off. Confirmed my setting of: At timecode break "Make New Clip." Confirmed my Easy Setup as DV-NTSC. My capture preset is DV NTSC 48 kHz. I've turned off and on FCP. Restarted my computer. Restarted my computer with the shift key down and ran permissions.
Any ideas are GREATLY appreciated, T.

I am concerned about available memory when capturing to my hard drive and the babysitting and extra steps involved considering the amount of tape I want to capture...but it does work that's GREAT!
This is always a concern, but in your case I think having the camera and external on the same bus was causing your problem. You may have to capture a little and then transfer, rinse and repeat. Just don't try to do too much at once and let your System Drive get too full, you'll run into other problems there. Slow and steady is the pace!
I shouldn't have stacked my questions since you answered one and Chris answered the other. Don't know how to apply the answered question and who gets the points.
No problem, just mark it answered and divide up the solved and helpful points as you wish. All in all we really don't care too much about the points, but they do make us feel good! Thanks for your desire to use the forums properly.
K

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

The whole audio sync/weird sample rate/drift issue (FCP6)

I can tell this question is doomed to fail on account of viewing the other unanswered questions on the subject in the forums, but I'll try nonetheless!
So yeah, importing longer DV clips....audio rate winds up being 'weird' (as in 48003.9kHz instead of 48). Any way of straightening that out that doesn't require a buncha hoops? I've already tried re-exporting it from QT at 48kHz and a few other things. Do I just have to capture in separate chunks instead of one big go?
I found a doc that says, "In previous versions of Final Cut Pro, the "Sync Adjust Movies Over [n] Minutes" option could be enabled in the preferences, triggering an adjustment in the audio sample rate based on the amount of audio recorded to disk during capture. In Final Cut Pro 4 [and up, I'm guessing] this preference has been replaced by automatic audio rate analysis, specific to DV media..."
Apparently this rate analysis ain't working as well as it should...or something...
Anyone?
Thanks,
E

If your footage was shot with the Panasonic dvx100 camera, there is a known audio sync issue. You can install a patch from disc 1 of the FCP install discs inside of the "extras" folder. I've had this problem from time to time and I capture DVCPRO 50, so it's not just a mini DV issue. I usually get an audio rate slightly below 48, like 47.89999 or so. Are you using capture now? I used to get this problem a lot more when using capture now as opposed to log and capture. My solution has always been to always use log and capture and keep clips 15 minutes or shorter. I've tried trashing prefs and all that jazz, swapping out hard drives, but log and capture and shorter clip lengths always seem to clear it up.

LogicX - Maverick - Sample Rate issue with Audio Interface

As a test prior to updating my Mac Pro and iMac, I loaded ML so I could test out LogicX. I've been running Logic 9 either using the built in audio or a Line6 guitar Port. The MBP is a late 2009, non retina, 13 in with 4 GB Ram. As compared to my iMac, the laptop can barely play back my tracks which are mostly software instruments (I don't record audio until the midi arrangements are finalized).
So, I updated to ML to try out X and of course, had to deal with the 32 bit plugin issue. Luckily, 32 Lives and Vienna helped out with 90% of my plugins.
Now, I updated to MAV and had to go through an arduous process of re validating the AU plugins for both Logic 9 and X.
After doing so, I loaded up a song that did have a single bass audio track. I updated the Guitar Port driver from Line 6 to the latest driver. The song will start, but then I get a stuttering and an error message about running the audio at a weird sample rate, eg 28654. I tried using the built in audio and same issue.
I opened the song in Logic 9 and it works.
Any suggestions? Do you think one of the resurrected plugins is causing this problem? Everything worked OK in ML though.
Thanks and appreciate any help.
Todd
trademarkauthority

Let me start over ... the typical audio setting that FCP works with is 48khz, 16 bit stereo. If you check the sequence settings, it will probably show that to be the case unless you've created a sequence with other settings. The audio of any clip you bring into FCP really needs to match the sequence settings. To check the sequence settings go to Sequence->Settings.
So far, you've confirmed the clip is 48khz sampling rate but you didn't say what it's bit depth was. In FCP, right click on the clip and select "Item Properties." That will tell you the information I've been asking about.
-DH

External sound card sample rate issue

After installing Yosemite I am having problem with Sample Rate settings on my external audio interface. It was working perfectly before the upgrade.
The issue is that the sample rate on my system is set to 44.1 Khz. When a new application attempts to access the audio card it plays but breaks up. This is indicative of a sample rate issue so I tried to set the rate to 48 Khz and then back to 44.1 Khz. Once I do this the system works properly until another application tried to use the card and I have to repeat the same process over again. While it DOES work it is not correct functionality and there is obviously something wrong.
At first I thought is was just my system until my friend also commented on it happening to him too.
I am using an RME Fireface UFX with the latest drivers issued on October 8th of this year.
My friend is using an M-Audio unit and he is finding the same issue.
I have already checked everything in Audio / Midi Set-up control and reinstalled the drivers for my audio interface. I also looked for any new settings to Yosemite to see if there was something new that might affect this. No luck and no leads on where to look for the next option.
Anyone have any ideas?

Hi,
The sound card do not support all sample rates. Which sample rates your sound card support you will find in the the manual.goto help in the toolbar select find examples and search for sound Open the Sound Input to File.vi file.This will give you a template for recording sound.You have to set the sample rate then open a sound file for write.
Please post a sample VI that you have created.
also,please go through this:-
http://www.zeitnitz.de/Christian/waveio
Thanks as kudos only

Sample rate and audio-MIDI sync issues

Disclaimer: I did read other posts similar to this but couldn't find an answer to my specific situation. So here it is:
Logic was perfectly fine when everything was running at 44.1kHz sample rate. Then I got vocals at 48 kHz so I had to convert the sample rate in Logic Pro to match it.
Suddenly I get a slew of "Error trying to sync MIDI and audio" messages. After crying and changing the sample rate back to 44.1, then to 48, and over and over again, it finally works again.
So Logic is fine at 48kHz. But when I go to a track that's at 44.1kHz, I get the sync messages again and have to play "toggle the sample rate" for about 5-15 minutes before Logic decides whose master again.
Why is it doing this? Do I need to change Logic Pro to some kind of default settings every time I go from one song to another with a different sample rate? Or will this not be an issue if I upgrade to 7.2? (I have 7.1.0)
PowerPC G5 Mac OS X (10.4.6)

This is not a bug, but a nuisance.
You should upgrade to LP 7.1.1, which is way more stable than 7.1.0. No need for you to go to LP 7.2.
"Then I got vocals at 48 kHz so I had to convert the sample rate in Logic Pro to match it."
Are you certain, that in your song, in your regions, you used the newly converted 44.1kHz files, chosen from the Audio Window, and not (still) the old 48kHz files?
"So Logic is fine at 48kHz. But when I go to a track that's at 44.1kHz, I get the sync messages again"
LP doesn't do this well. And for a reason, but we'll not get into this now.
See: "Audio > Sample Rate > ..." and select one.
Perform proper conversion and make sure ALL of your audio files running in your song are congruent. Check your Audio Window and the files associated to the regions.
Set up your autoload to contain these settings. From then on, whenever you know that you will be importing other sample rates, change the settings in "Audio > Sample Rate > ..." before loading the sounds/files, if indeed you are starting from scratch. This will save you significant time.
Been there, done that.
sonther

AA 3 vs CS6 playback sample rate

Is there a way to ask CS6 to playback at the native sample rate, automatically, instead of resample? AA 3 would always playback at the native sample rate, if supported by the hardware, while CS6 resample even if the sound device supports the sample rate.
CS6 switching between projects I have to manually look at the sample rate, then change the hardware settings.
Thanks!
David

I think it's an operating system problem you've got here - Windows is notorious for resampling anything it thinks it can get away with. There's a setting on the audio hardware page that attempts to link Audition and the sound device hardware rather more firmly, but your best bet absolutely is to use an external sound device that uses ASIO - this bypasses most of the OS completely, and generally doesn't have these problems.

Issues writing varying sample rates to a continuous TDMS

Hi All,
I have a VI that uses DAQ.mx to aquire data where I can dynamically change the sampling rate without stopping the program and restarting it. The aquired data goes to to a queue and then gets written to a TDMS file in a separate VI. I know the aquisition part of the routine is properly changing the sampling rate, but when I view the TDMS files all of the data has been changed to one sampling rate for the entire file. Does anyone know what may be causing this and how I might fix it?
Thanks in advance!

"How are you changing the sampling rate without stopping the program and restarting it?"
I have tried changing the sampling rate two ways to see how they differed with system resource load. Both still leave me with the same TDMS file issue.
1) there is a case structure in the aquisition loop and a "low speed"/"high speed" switch. There is also an "equal to" comparison coming off and the switch's value from the last loop via shift register. If I flip the switch true case stops the device, changes the sampling rate through a datamx task property node to high speed or low speed (depending on the boolean, a second case structure inside the first), then starts the tast again at the new rate. The case then is false until the switch is flipped again.
2) the loop always samples at high speed and downsamples the waveforms when "low speed" is clicked.
What sampling rate are you coming from and going to?
High Speed = 50kS/second
Low Speed = 10kS/second

Inconsistent Audio Sample Rate - Export issue

I am attempting to export my project into a Windows Media file and I'm getting an error that says "inconsistent audio sample rate" Does anyone know how to fix this problem?

That error was introduced with the release of Quicktime 7.2. Flip4Mac released a beta with the fix for that particular audio error thing HERE. I don't think the final is ready yet, just the beta, but it's working for me. Don't be "put off" by it, but, before you can download the beta, they ask you to fill out some registration info so you can report back any problems you might encounter.

Sample rate 40938/playback -synchronization issue

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