Automating wav import, sample rate change, mp3 export

I do radio spots, lots of them, and every day I have to take my spot that is a 48k wav, convert it to 44.1k, then convert it to a mp3, them distribute via email.
Can I automate this task?

Rather than trying to automate iTunes, it might be easier to do the conversion in one go with a utility such as Sound Converter or Switch.
Hope this helps.

Similar Messages

Audio sample rate change

Hi; am importing audio into FCPX - AAC, 44k1 - along with the .mov vids, but on (trying to) export finished work am finding it nearly impossible to alter the 48kHz (default?) output sample rate .. how so ?
Any help gratefully received .. cheers
Peter

Why don't you want the 48kHz?
It's the standard for video.
This article may help (audio is at the bottom).
http://support.apple.com/kb/PH12526?viewlocale=en_US
Message was edited by: Ian R. Brown

Sample rate change from 44.100 to 48.000

Hi Guys,
I am preparing a sound design file for a film project. the audio files are originally in 44.100. Now, I am preparing the surround mix for the DVD release. For DVD I would need a sample rate of 48.000khz. However, when I change the sample rate from 44.100 to 48.000 in the Project settings, all my audio files seem to be shortened in the arrange and everything sounds as if it was sped up. How can I change the rate to 48000 without the audio files changing?
thanks

Man you have to convert the files to 48khz otherwise they will of course sound mickey moused and distorted when played in a 48khz project. You can easily do that in the Bin, batch process all your files and convert them to 48 using the "copy and convert" funtion from the menue. Compressor does the job also...

Gain changes on sample rate change

I wrote an application with 2 channel analog input with different gains. If I change the sample frequency the gain setting seems to mix up. But the sample frequencies work if used as first. So, only after changing the sampling frequency the gains are mixed and could not be 'reset' by calling Scan_Setup() from NIDAQ library again.
Any ideas?

Hello,
So it seems that changing the sampling frequency is affecting the gain of a channel?
I am not that sure what is causing this behavior, if you could answer the following questions, we could begin to figure this one out.
1.) How do we know that the gains are mixed up? Is one signal significantly larger then before? Is the code width decreased?
2.) Are you trying to change the sampling rate during acquisition? Are you running the acquisition to the end, then calling Scan_Setup again?
3.) Do you get this error if you call DAQ_Clear before reconfiguring the sampling rate?
Also, if you could post some pseudo code that explains which DAQ calls you are making, and their order, we should be able to get this problem figured out.
Best regards,
Jus
tin T.
National Instruments

Changing Wave File Sampling rates

Hello,
I was wondering if there was a java class or package that can do this. I want to up sample a 8000 Hz wave file into a 44100 Hz one, and vice versa.
Any info would be appreciated!
Vicki

Use the javax.sound.sampled package.
Try this:
// Suppose you got a file, or an url
File file = new File("Blablablabla.wav");
AudioInputStream oldstream = AudioSystem.getAudioInputStream(file);
AudioInputStream newstream = AudioSystem.getAudioInputStream(
new AudioFormat(44100.0f,16,2,true,false),
oldstream
// Use the following code if you want to write to file:
AudioSystem.write(
newstream,
AudioFileFormat.Type.WAVE,
new File("blabla.wav")
);The variable oldstrea can also be obtained with an URL.
Check the part where the AudioFormat is constructed, you may want to change something, but if im right, this is cd quality.
The last method is for if you want to write the new stream in a file.
Note that if you change the file extension where it says new File("blabla.wav") and change the AudioFileFormat.Type you can also convert between wav and aiff and au etc.
R. Hollenstein

Sample Rate Changes from 44k to 48k?

So who can tell me why my project in which all the peices are saved 41k are magically transfered to 48k when saved as an MP3?
Cheers.

Because your MP3 encoder isn't set up properly? Have you checked all the options available when you save as .mp3?

SCXI Sample rate changes when VI run as SubVI

I have a VI which acquires 7 channels of data from SCXI-1100 module and places the data into a global variable. The VI runs fine on it's own but once it is used as a SubVI the performance drops.

Hello Andy,
Using global variables might be the culprit here. You have to consider that for every read of a local or global variable in a VI or its subVIs, a copy of the data is made in memory. Each instance of a local or global variable will therefore make a copy of the data in memory.
So if you are reading from this global in a loop, this could be slowing down your CPU and program. I would suggest to just create an output of the SubVI that could pass the data from the SubVI to the main VI. This should help the efficiency of your program.
Does Reading a Local and Global Variable Create a Copy of the Data in Memory?
http://digital.ni.com/public.nsf/websearch/37002ACC84B961CA86256D9C00760EE2?OpenDocument
I hope this helps. Let me know if you have any add
itional questions.
Regards,
Todd D.
NI Applications Engineer

IPhone: AudioQueue - is it possible to change the sample rate?

I've been playing around with the AudioQueue stuff for a few days and it's all working fine.
I was trying to build a low-latency playback system by making the streaming buffers the same size as the audio file and pre-loading the buffers (which works fine) but I've hit a snag.
I've been trying to get the streaming to work at different sample rates so that I can play back the same sample at different pitches. I managed to do it by modifying the sample rate in the AudioStreamBasicDescription structure but in order to actually make the stream playback at the new rate it seems you have to create a new output, reload the audio file into the buffers and re-enqueue the output queue before starting playback again, otherwise the sample rate change has no effect.
There is a method to set queue properties; AudoQueueSetProperty() but unfortunately the sample rate Property (kAudioQueueDeviceProperty_SampleRate) is read-only
Can anyone suggest a way to achieve this with AudioQueue or do I need to move over to OpenAL?
Thanks,
Neil

Dan,
there is one point in your understanding, which i am not sure what you think about when talking about it: I understand E-series devices do not support this property change while the VI is running.
infact, you cannot change the sample clock rate during acquisition. but
this does not mean that you cannot change it while the VI is running.
you have only to interrupt the acquisition. since you want to acquire
continuous, this would have the same effect as stopping the vi, i asume.
so the best way to accomplish this task is to use an external clock.
this is e.g. often used for acquistion on rotating shafts. the
acquistionrate is always e.g. 24 points per revolution regardless of
the rotational speed of the shaft, except for a maximum frequency of
course.
Norbert B.
NI - Germany
Message Edited by Norbert B on 09-14-2005 04:16 AM
CEO: What exactly is stopping us from doing this?
Expert: Geometry
Marketing Manager: Just ignore it.

If your Lightroom 6 is crashing in the Slideshow module, check the SAMPLE RATE of your music track

After spending over 2 hours with an Adobe technician who had control over my Windows 7 machine, I've learned something that I don't think Adobe knows yet... I upgraded from LR 5.7 to LR 6 (not CC) and began having issues when I was working in the Slideshow module. When trying to add a music track, Lightroom would crash ("unexpectedly quit"). I couldn't for the life of me figure out why. I turned off the GPU since a fair amount of chatter on the forum related to that, but it didn't change anything.
So I finally waited patiently for the Adobe Chat line to finally get a tech to respond...and he was exceptionally detailed and methodical in his approach to diagnosing my issues. I could follow on my screen as he did a variety of things that indicated he understood what he was doing. Unfortunately he got to a point where he thought (as did I) that he had found the problem/solution...which he thought was related to the bit rate. He concluded that music with a Variable Bit Rate (VBR) as opposed to a Constant Bit Rate (CBR) was causing the problem. Also unfortunate was that this "discovery" was near the 2-1/2 hour mark and we did not apply the scientific method to property determine if we had narrowed down to cause/effect.
Based on my further analysis, he was close, but not right on. It isn't the BIT rate mode that is causing the crash, but rather the SAMPLE rate. Turns out that the vast majority of songs in my iTunes library have a sample rate of 44.100 kHz--and those work just fine. However, I have about 15 songs, mostly downloaded as free background music from the YouTube creation library, that have a sample rate of 48.000 kHz--every single one of those cause my LR6 to crash.
If you're having a problem with Lightroom crashing in the Slideshow module, check the SAMPLE RATE of your music track.

If you're on Windows, check out a free program called Audacity--it's fantastic. With Audacity the process of converting a 48.000 kHz to 44.100 kHz is as simply as opening, changing the sample rate, and then exporting again, as an MP3.

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

Project sample rate vs the audio interface clock

I used Audition for digitizing archival audio and I mostly work in 96 kHz because it's the current archival standard for analog audio, however I occasionally need to transfer DAT or ADAT tapes at 44.1 or 48 kHz. One thing I noticed when I went to change the project format was that my audio interface clock did not change and match the multitrack settings. I was able to change the clock manually but I had to go to the Audio Hardware preferences.
I find this behavior for a DAW very strange. If I have a multitrack open of one bit rate and my clock set to another, how does that work? How does Audition 6 play and record audio at differnt sample rates than the interface clock? When I open a multitrack that's 96 kHz and my clock is set to 48, what sample rate am I getting? The audio file being saved says 96 kHz but is it?
Audition 6
RME Fireface UCX
Mac Pro mid-2010

loneraver1 wrote:
If true, this is indeed strange. I have used just about every DAW under the sun for the last 10 years based on what my employer purchased and I have never seen a DAW allow you to record sample rates where the project didn't match the incoming clock.
There should be an option that locks the audio interface internal clock to the opened project. It makes it annoying any time I have to switch between recording in 96 kHz to 48 kHz. That's annoying on it's own right, but what's really annoying is that it's an extra step that I have to train our archival technicians who don't have a techical background in audio.
I can definitely say it's true for my set-up!
Surely, the "extra step" you refer to is not necessary if AA is resampling the audio to match the desired sample rate?
As for the "option" to "lock the audio interface clock to the opened project", for those cards, like mine, which cannot have their sample rate changed by the audio software, how is that going to work? I suspect this was how AA CS5.5 and earlier worked: unless session/project and interface sample rate matched, no audio. I much prefer the current "resampling" method; no necessity to have to think about soundcard/interface settings every time a different sample rate is being used.
FWIW, if I want to record something at 44.1 I always ensure the card and AA settings are 44.1; but for "playback" (or video editing which I do quite a lot of and which requires 48 - my video editor software will not "play" unless the audio sampling rate of the card is 48 - ) 48 is my "default" setting.
JMO!

About Logic's Sample Rate

When you choose a sample rate, I know that the sample rate will be applied to the incoming audio of your interface BUT what if you have a project of music that contains Logic software instruments? Does the sample rate change the audio coming from the software instruments also?

Hi,
I don't know, but I think this only applies to software instruments when they are a) frozen b) bounced. Because: The things you hear coming out of you monitors cannot have a sample rate - they are analog 20-20,000 Hz. Period. The question is which way Logic takes to create this analog signal. It's a question how the software instruments themselfes create the signal. So, when sample rate is 44,1kHz, do the software instruments "run" their output at a lower rate, too? Or do they always run at full pace (192)? Or always at 44,1kHz, since this is way enough for the demands of a synth?
For EXS, the sample rate is determined by the EXS samples. See how much they have: That's what Logic will feed into the D/A's.
Don't know how the synths are running, though.
Fox

Converting sample rate when exporting

CS5.5
I'm using Audition to do some final mastering to 16-bit stereo 44100 wav files, in 32 float multi-track sessions.
All I'm doing is increasing the clip gain, then exporting to 16-bit stereo 44100 wave and 320 mp3.
I'm not changing the sample rate, channels or bit depth, but I am applying triangular dither when exporting.
The thing I've noticed is even though I'm not changing any of the properties of the files, when exporting Audition still tells me its "converting sample rate", despite the sample rate being the same as the source (44100). The quality advanced tab on the sample rate is greyed out.
Also, when I check the history box for the newly exported files it tells me "convert sample type on save as"
What is Audition doing at this stage?
How can it be converting the sample rate, when the sample rate is the exact same as the source? And the quality tab for sample rate conversion is greyed out?
All I'm doing is upping the clip gain slightly and then applying dither to these files. No sample rates are changed.
Any help greatly appreciated.

brazil101 wrote:
I'm using Audition to do some final mastering to 16-bit stereo 44100 wav files, in 32 float multi-track sessions.
I'm not changing the sample rate, channels or bit depth,
Except that you are...
If you open a 16-bit file as a 32-bit float, then you've changed the bit depth. It's the right thing to do if you are doing amplitude changes, as it makes any change effectively lossless, but it will still have to be converted back to 16-bit if you elect to store it like that.

How do i change the audio sample rate in motion 5?

Hello,
here's a simple question. I import 48 Khz audio into the project, but when I go to export the summery says the audios will be exported at 44 Khz. Why is that and how do I export in 48 Khz?
Thanks!
Mike

I'll see if I can muddle through this.
According to the documentation, Motion converts everything internally to 44.1kHz 16-bit. There's no way to change this.
To export in 48kHz, you need to quit Motion. From the Finder > Go > Utilities. Open the Audio MIDI Setup application. For (all) outputs, set the sample rate(s) to 48.000 kHz (and 2 - 24 bit if available [32bit float might work as well]). Quit Audio MIDI Setup and restart Motion.
When you go to export from Motion, be aware that the export Summary will still show 44.1kHz (which is thoroughly annoying!), but the exported video will be the audio rate you set in the Setup app:
The Info from an exported project:
HTH

Having trouble with wav files and sample rates

Hi ,I am having trouble with wav files and sample rates .I have been sent multiple projects on wav as the main instrumental ; I wish to record in 48.000kHz .Now comes the problem.When I try to change the project to 48k It seems to pitch up the track.I can't have them send the logic/project file as most have outboard synths,different plug ins etc.This particular case the producer has recorded the synth task in 41.000 kHz .My successful outcome would be to be able t create a project file in 48 kHz .And NOT pitch up whne I add the instrumenta wav file .Any help would be gratefully recieved,this is my first post so any mistakes I may have made go easy

You'll have to convert the actual synth audio file file that the producer gave you to 48kHz. You can do this in the audio Bin in Logic.

Automating wav import, sample rate change, mp3 export

Similar Messages

Maybe you are looking for