Help please? Logic playing back wrong sample rate

So I created a new session and recorded a new song that I have already done some work to this song and have had no problems whatsoever. Last night I went to open this song and work on it some more and when I pressed play the song started playing back very slowly. This session was recorded at 88.2k and I am getting a Logic error message that says sample rate 44.1k is recognized however my audio interface (a Metric Halo 2882 +DSP 2d unit) is set correctly to 88.2k. I look at my bin and all the files are correct, my preferences are all correctly set at 88.2k but Logic is playing the song at about half speed!!
What happened?? I have a couple projects going and I need to get this fixed.. Anyone??
Thank you in advance!!
DDD
Mac Book Pro Dual Core, OSX 10.6.8, Logic 9.1.5 (do not want to upgrade Logic to 9.1.6 I tried that and it was a BAD experience so Im sticking with 9.1.5), Metric Halo MIO Console 5.4

Sounds to me that your best bet is to use a utility like SoundHack (http://music.ucsd.edu/~tre/soft//SH893.hqx) to change the audio file headers. What you want is to flag the files as 48k, not do any sample rate conversion--they should play back natively at 48k. The only SRC you'll need to do is if you actually do want the files to end up as 44.1k at some point. (In your case, I don't think Logic is converting the sample rate in real time, by the way--what it's doing is playing back the files with a clock setting of 48k, which only sounds right because they've been mis-labeled.)
Either way, you should fix the file headers now, I think. And even if your final product needs to be 44.1k, since your files are quite clearly at 48k now, I'd do all the necessary editing and processing at 48k, then convert to 44.1k as the final step. And until the SRC in Logic is improved (Apple ?), maybe use some other app to convert to 44.1k in the end. Peak gets high marks--not sure what else out there sounds good and doesn't cost a mint.
Best luck!
James
[email protected]

Similar Messages

Audio tracks play back wrong audio

at some point in my mix sessions , in different sessions , one of my audio tracks starts playing back audio from both its channel and another channel simultaineously. there not always adjacent tracks , not assigned to the same voice and have seemingly nothing in common. if i change voices i can sometimes allieviate the problem or it will start playing a different track audio simultaineously. i have two records pending and need some immediate help please...

D,
Just to warn you (or throw some experience in). Logic 7 has for me, operated poorly with sessions created from older versions of Logic. It is often fine if you just want to duck in and grab something but the sessions do go weird if you just keep using them they go batty and can often get kind of self destructive.
Updating/merging/replacing the environment is one of the first steps to take, just creating new updated objects helps, even using presets and channel strips from older versions seem to give problems. Copying objects and configurations over can give you different results, but basically the more you replace and create as new the less problems you'll have.
So you know you have alot of reworking to do if you really want to do alot of work with these in the current version. Determine where a good transition point would be and try to follow with that. Often mixing or sooner than later but plan on taking some time reworking things. It's almost easiest to just grab everything and paste it over into a 7 built autoload-without the plugins. Audio Objects and plugin instances seem to give alot of problems. I usually end up redoing a bunch of stuff if I have a ways to go and export away. Things really get fixed along the way but it is a pain and better to do it sooner. Unless you're coming up on a mixing stage or something soon I'd just do it (and keep a backup of the older versions just in case I want to retrieve something.
I am working on an album that was created on a 6 autoload and 12 of 13 tracks had to be reformatted in 7 from forward INcompatabilities-some of them twice because they did not get a 100% clean autoload (as in the instruments and/or audio objects were opied over). Maybe I'm just weird, don't take only my word on this because it is alot of work for many projects to just redo things like this but you should be aware of what some people have encountered and be prepared to take some of those steps if there are more people saying their experience has been like this.
Best of Luck

Wrong sampling rate from Sound Input Read VI on Dell Latitude E6510

Hi,
When I
receive data from the sound card with the Sound Input Read VI, the actual
sampling rate is not equal to the sampling rate set in the Sound Input
Configure VI. I estimate the sampling ratio by dividing the number of samples received
by the difference between the first and the last t0 from Sound Input Read. For
example, if the sampling rate is set to 7000 S/s, the actual sampling rate is
around 6900 S/s. If the sampling rate is set to 7050 S/s, the actual sampling
rate is around 7200.
This problem appears when the
code is run on my Dell Latitude E6510 machine. When I run the
same code on my Lenovo ThinkPad, the sampling rate is correct.
The Labview application I
am developing is to be used to acquire data synchronously from several sources,
including the sound card input. It is therefore important that the sampling rate
is correct.
If anybody has any idea
about what may cause this problem, and a way to fix it, I would be grateful.
Ole

It has been a trend the last years that sound card only support some sampling rates, not a range. You should refer to your sound card manual for this kind of information. You should also be aware of that the NI sound interface is junk, and they do care about it either. Use the WaveIO interface instead See the link. But I am quite sure your problem is related to the sound card hardware not some NI driver issue
http://www.zeitnitz.de/Christian/waveio
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)

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 ~ $

Help - Audition only plays back static!

I'm trying to import a clip from a movie file to reduce the noise.
I tried transfering the file from premiere over to audition "edit in audition". It only imports static. There is a waveform that looks like it could be correct, but all you can hear is static.
Trying to pinpoint the problem I have tried all types of differnt files, ".mov,.mp4,.wav,.m4a,etc..." all are opening in the same fasion (not importing from premiere, direct opening).
I have edited these same type of files a month ago without an issue. They only thing I can think is that one of the adobe updates I have run changed something.
specs:
adobe cs6
audition 5.0.2 build 5
premiere 6.0.2
Mac pro - lion 10.7.3
Any ideas??

I'm trying to import a clip from a movie file to reduce the noise.
I tried transfering the file from premiere over to audition "edit in audition". It only imports static. There is a waveform that looks like it could be correct, but all you can hear is static.
Trying to pinpoint the problem I have tried all types of differnt files, ".mov,.mp4,.wav,.m4a,etc..." all are opening in the same fasion (not importing from premiere, direct opening).
I have edited these same type of files a month ago without an issue. They only thing I can think is that one of the adobe updates I have run changed something.
specs:
adobe cs6
audition 5.0.2 build 5
premiere 6.0.2
Mac pro - lion 10.7.3
Any ideas??

Changing sample rate corrupted space designer.... somehow.... HELP!

This is a new one to me, so it's very difficult to explain whats happened. I'll start from the beginning.
I created a new, blank logic file with a sample rate of 96.
I created a new software instrument and began to fill it out, adding some compression, some eq, and a little bit of reverb. The reverb came from space designer, using the small room preset "Thicken Vocals". So far so good. No problems, everything's working fine.
I open Reason to rewire in (and ultimately track and record) some drum samples out of reason. The sound is coming out of reason into an aux bus in Logic, and that aux bus is being used as the input for a new audio track. I lay down some midi notes, play back the drum audio out of Reason and hit record on the audio track, essentially making an audio file of the drum samples coming out of logic.
During this, something happens. Rewire is doing a LOT of work balancing the sync, the sample rate, etc.... and somehow, SOMETHING freaks out in the space designer i was using as some subtle reverb for my software instrument in logic (this software instrument is UNRELATED to the Reason audio).
Now, when i hit play, the space designer just produces a loud, annoying pop. If I CHANGE the space designer preset, everything is cool. If i change it back to "Thicken Vocals" under small rooms.... POP.
Worse, it seems to be a permanent problem.
SO --- Ive done a LOT of trying to figure out whats going on. I create a new document and recreate the software instrument. I browse to Thicken Vocals... POP! I tried to replace the IR that "thicken vocals" is referencing by overwriting the file from a system backup I have (an IR called "0.03s_Room Vocals")... still no dice.
Here's where it gets strange. It works FINE if the sample rate is 44.1... I can browse to "Thicken Vocals" and it loads up and init's fine, and everything is golden. As soon as I go to 96, or load it up when the document is set to 96, that's when it freaks out. NOW, if i adjust the falloff curves of the reverb just enough, THEN it suddenly works again, even under 96k.
So, in a nutshell, at 96k, when I load "thicken vocals"... it almost appears (even according to the waveform Space designer draws) to misread the IR waveform, and it plays back as a POP. If I adjust the Falloff curve, it corrects itself, and the waveform even redraws! At 44.1, there are no problems.
I did NOT have this problem before... does anyone have ANY clue what could be tweaking my system right now? I can post more details if I haven't posted enough... including pictures.
THANKS!!
Chris

You might have a corrupted song file. Possibly your autoload as well. Try trashing logic prefs and rebuilding you autoload. Also, repair permissions and maybe even try updating the prebinding in the terminal. Another possibility could be a conflict with the driver of the Audiowerk8, unlikely, but possible.

FCP 6 with M-audio fastrack pro and sample rates

hello, having a weird problem in FCP 6.02 under OS X 10.4.11 on a MacPro where if i record a voiceover through my m-audio fastrack at 48khz, which matches the sequence and project settings, it comes back at the wrong sample rate (plays back higher pitch), but if i record it at 44.1 it sounds right. any ideas on this?
thanks in advance
david

Perhaps the M-Audio is still recording at 44.1, and FCP is set to convert the audio to 48k, but is unable to resample as it would with an analog source. I would try to access the control panel for the M-Audio box and see if you can get it to force its sample rate to 48k outside of FCP. If so, you can probably get FCP to record correctly at 48k. If not, you might be better off just letting it record in 44.1, and maybe batch exporting later to upconvert.
Hope this helps -
Max Average

How is Core Audio sample rate set?

When I play a particular movie in QuickTime, the audio and video are out of sync. The movie plays fine on other computers.
This Mac Pro (OS 10.6.8) is used exclusively for Pro Tools and Final Cut Pro. The audio hardware is a Pro Tools HD Native card:
Native card --> Pro Tools Digital I/O boxes --> Lavry D/A converters --> monitors
The sample rate of the Digital I/O boxes is set by an external master clock, a Lavry Gold A/D converter.
I opened the DigiDesign Core Audio Manager and it says:
Connected @ 44.1K, 32 In/32 Out, Buffer Size 512
Yet the movie is at 48K (I know because I created it in Final Cut Pro), and the external clock is set at 48K. So, I don't know where the 41K in the Digidesign Core Audio Manager came from. This is why I am guessing the problem is sample rate.
Please help me understand what sets the sample rate.
Does the application using Core Audio set it?
If so, how is this made consistent with the sample rate set for the hardware by the external master clock?
Do I have to be sure I always change the external clock setting to be consistent with the movie being played?
BTW, I have not had a sync problem when I play video in Final Cut Pro..........it is always in sync. So, I have never worried about how Core Audio works. The sync problem is only with QuickTime.

You might ask on the GarageBand, Logic or Final Cut forums. There is not much audio traffic here.

How does Core Audio set sample rate?

When I play a particular movie in QuickTime, the audio and video are out of sync. The movie plays fine on other computers.
This Mac Pro (OS 10.6.8) is used exclusively for Pro Tools and Final Cut Pro. The audio hardware is a Pro Tools HD Native card:
Native card --> Pro Tools Digital I/O boxes --> Lavry D/A converters --> monitors
The sample rate of the Digital I/O boxes is set by an external master clock, a Lavry Gold A/D converter.
I opened the DigiDesign Core Audio Manager and it says:
Connected @ 44.1K, 32 In/32 Out, Buffer Size 512
Yet the movie is at 48K (I know because I created it in Final Cut Pro), and the external clock is set at 48K. So, I don't know where the 41K in the Digidesign Core Audio Manager came from. This is why I am guessing the problem is sample rate.
Please help me understand what sets the sample rate.
Does the application using Core Audio set it?
If so, how is this made consistent with the sample rate set for the hardware by the external master clock?
Do I have to be sure I always change the external clock setting to be consistent with the movie being played?
BTW, I have not had a sync problem when I play video in Final Cut Pro..........it is always in sync. So, I have never worried about how Core Audio works. The sync problem is only with QuickTime.

Well, here is more information:
- I reinstalled QuickTime 7 and did many other things (trashed preferences,etc)
- I set the audio hardware (Pro Tools/HD Native card) to 48k
- I checked the Digi CoreAudio Manager and it said "Connected at 44.1K", even though the hardware was set to 48K
- I played the problem videos, and all are now in sync
- I check the Digi CoreAudio Manager and it still says 44.1K
- I checked the hardware and it is now set to 44.1K
I know that the videos are 48K, because I created them in Final Cut Pro X.
Does this mean that QuickTime 7 always converts videos to 44.1K, regardless of their sample rate?

Optical digital input (S/PDIF) problem (sampling rates)

I have a huge problem with my 2009 iMac's digital input that I didn't run into on my older MacBook running the same OS (10.6.x), and it's driving me nuts!
Basically, I'm trying to record a digital feed from an ADC hooked up between my hand-built tube amps/turntable and my iMac, with the goal being the transfer of some of my LP records to 48/24 LPCM for archiving and editing. The ADC's output rate is fixed at 48000Hz and cannot be changed. My iMac, however, refuses to lock on to the device rate, instead forcing 44.1kHz in the audio/midi setup. In fact, I can't get it to do anything but 44.1kHz on anything I plug in (even my DAC's passthrough) despite the fact that I need to use 48kHz. The audio is correct for a few seconds, then forces back to the wrong rate again. No matter how many times I try, the computer insists on using the wrong sampling rate, which has rendered impossible any hope of using my iMac as an audio recorder.
I've tried setting up an aggregate device, but to no avail. Nothing I do seems to work, and since the problem is with Core Audio, any program used to catch the audio records gibberish. Is there a way to force Core Audio to use 48kHz in the terminal? It's not like the computer can't do it, as I get about 2 or 3 seconds of correct audio when changing the sampling rate manually. I just can't get it to stick!

Seeing the exact same thing here. External digital source is 48kHz, and this Mac Pro Desktop will only stay on that setting very temporarily (for about .10 or .15 seconds) then it reverts back to 44.1kHz. Even if I don't have a signal or even a Toslink cable plugged in to the input.
(I'm surprised you are getting 2 or 3 seconds of correct audio, when I'm getting about 1/10th that before it reverts back to the wrong sample rate. Maybe related to a buffer size option somewhere?)
I have tried every known option to reset or re-configure this, and even to the point of creating a new user, resetting PRAM, resetting SMC, etc. I was going to try upgrading to 10.6, thinking maybe it was a 10.5.x problem, but now I'm not sure that will help.
Had I the choice to set the setting to be 48kHz and lock it in, it should stay on that setting - (even if the chance exists that what I'm getting is not correct.)
More often than not, I know what I'm doing more than Mac OS does, and I want to force it to do what it needs to.
There really needs to be a "do it because I told you so" option for most of the Mac audio configuration system. This automatic "the hardware decided to do something for you" stuff is absolutely horrid for a system that is marketed as "Pro". A real "Pro" system would let me edit the configuration once, with the tool (text editor hopefully) of my choice, rather than attempting (and failing) to automate things.
+1 for another instance of this being "a huge problem" and "it's driving me nuts!"

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

OMF - Mixing Different Sample Rates

Hello -
So, I've got an OMF file from a FCP project that I opened in Logic and have been mixing without any issues. Interestingly, all of the audio files associated with this project are 44.1 kHz, but I am mixing in 48k. But everything is right (sounds right, looks right, syncs with video correctly).
But... if I open the same files in an external editor or quicktime, they play back incorrectly. And if I save a file in a different program, even if it is still at 44.1, and bring it back into logic, it plays back incorrectly. This is problematic if I need to edit an audio file somewhere other than within logic (say I want to do some noise reduction in soundtrack pro).
Anyone run into this issue or have any ideas about how this happened?

There are a few ways to look at this.
1) Regions in the arrange all play back at the session sample rate. Example: 44k session, 96k audio file in arrange=slower playback
2) Logic automatically converts output sample rate so you can record independent of CA devices. Example: You have a session recorded at 96k, your interface is not connected, Logic will load Built in Audio, Logic runs tyhe sessions at 96k and converts the SRate of the session to match the supported sample rate. So you can run sessions at unsupported sample rates, this rarely makes sense if you cannot capture your audio at session sample rate (if needed).
3) There are a few other options for handling this, such as EXS24, which automatically handles SRC.
4) When Importing Audio Files there is a song preference which you can en/disable to automatically convert SR upon import.
I think your friend may have referred to point #2 and it was interpreted as point #1...perhaps. Hope this clears things up. J

How to coerce the sampling rate??

I think I found my problem with sampling rate.
I'm using a PCI-5122 scope card, and in many of my aquisitions, I'm setting the sample rate to 40MS/s. Apparently, this is not a valid number and the scope reverts to 50 MS/s
Later when I try to calculate cycles per second based on cycles per sample, I need the actual sample rate, and 40 MS/s ain't it.
I'm trying to coerce sampling rate.
please,.

The digitizer coerces the sample rate because of how the sample clock is derrived from the Reference Clock. The following information is on page 13 of the specifications:
http://digital.ni.com/manuals.nsf/websearch/C6B059C1BDD70101862574C8005567F1
The sample clock is created by dividing down the Reference clock (internal reference clock is 100MS/s) by decimation, and it divides it by N, which is an integer between 2 and 65530.
Thus 50MS/s uses a decimation factor of 2, and 33.3MS/s is the next valid sample rate with a decimation factor of 3. So when you specify a sample rate that is not possible, the driver automatically coerces the requested sample rate up to the next valid rate. You can obtain the actual sample rate used in an acquisition using the NI-SCOPE property "Actual Sample Rate", or the LabVIEW VI "niScope Sample Rate.vi". Using this property, you can get the values you need for your calculations.
I hope this helps!
Nathan
Product Support Engineer
National Instruments

Actions won't play back transformations properly

The transformations 'skew', 'distort', and 'perspective' wont' play back properly in PS CC2014. You can record an action, but it will play back differently. Not just a few pixels different, but a lot. I also tested a number of existing actions that I used in PS CS3, but they play back wrong too, even though they worked fine previously.
How is this possible? Is it a bug? Anyone knows something about this issue?

When you record action that deal with sizes. You need the think about what your recording. The thing is there is a lot going on. You need to record the action to work the way you want them to. You need to take into consideration that you may not many things about the document the action is being used on. Unless you know somthing like the action will only be use on un-cropped landscape images from a particular. In that case you know every document is x pixel wide and y pixel high.
More often you have no idea of what size a document is you do not know the number of pixel wide, high or the resolution the document is. Size can be tricky to deal with. You can record things in absolute pixel sizes, relative to resolution and relative to document size. How Photoshop units is set determines how many steps are recorded. Steps like transform are prime examples. Is your transform recorded Absolute pixel sized a location, Relative to resolution or relative to Document size.
Crafting Actions Package UPDATED Aug 10, 2014 Added Conditional Action steps to Action Palette Tips.
Contains
Action Actions Palette Tips.txt
Action Creation Guidelines.txt
Action Dealing with Image Size.txt
Action Enhanced via Scripted Photoshop Functions.txt
CraftedActions.atn Sample Action set includes an example Watermarking action
Sample Actions.txt Photoshop CraftedActions set saved as a text file.
More then a dozen Scripts for use in actions
Example
Download

High sampling rate is not accurate

Dear alls,
Any one know why the time loop is not accurate at defined period at the RealTime Trace Viewer ? In figure of VI, I set the period for the time loop is 250 us, but when I analyze in Real time Trace Viewer, it shows the real period time about 676 us. That means my data acquisition will be sampled in wrong sampling rate.
Could you give me some explane, please? And how can I solve this problem? I tested this code with MyRIO board.
Best Regards,
Kien

crossrulz wrote:
Why do you have a wait in your timed loop? The point of the Timed Loop is to state the loop rate.
??? Bigger problems than that! Why do you have any timing other that the TASK timing? And really, The task timing is most likely the loop rate.
Can you post your actual VI? I suspect something else is happening here that we cannot see (configurations, channels used, etc).
And a snip of the Task needs posting

Help please? Logic playing back wrong sample rate

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