Filtering Audio Files In Obj-C

I am trying to write a cocoa app that will import a wav file, filter it through octaves then display the information in tabular form as well as graph it on a grid. How would i go about importing the wav file? Thanks.

Hi,
there are different possibilites, one is CoreAudio which might take some time to learn, a second is to use linux libraries like libsndfile (has to be installed first), which for example is used in the Musickit as well. A third way would be to read the Wavefile as a NSData object, next extracting everything per hand... and fourth Quicktime API.
How did you implement the filter? I was wondering since I am alsways looking for better audio filters (I am working on a bat call analysis tool and good filters are quite useful )
Volker

Similar Messages

Audio File (Wav) Filtering

Hi.
I am looking for a way to filter the white noise from a wav file.
There is one input for the wav file, then i added a white noise for it, and i have added a play waveform to the filtered wav file. I have changed the filter settings many times but it is still noisy. Please help me, im noob in labview. :/
Thanks.
P3tson
Here is the VI and the wav file:
Attachments:
pack.zip ‏539 KB

First, the concept you expressed in the original post appears to be flawed.
Since white noise is uniformly distributed across the spectrum, a filter will not remove it from a signal. A filter will only reduce the amplitude in the stop band of the filter.
Second, I looked at the spectrum of your audio signal without added noise. From about 200 Hz to 16 kHz the signal drops about 60-70 dB. In any small segment of the spectrum the local peaks are 20-30 dB above the background. There is a series of harmonically related peaks spaced 120 Hz and several other peaks without clealry obvious harmonic relationships. A complicated signal like this is much more difficult to denoise than a well defined signal like a sine or square wave.
The amplitude of your signal peaks at about 0.5 and the average is probably about 0.2-0.3 while your noise has an amplitude of about 3-4. Extracting a signal with a -30 dB signal to noise ratio is moderately difficult if the signal is sinusoidal. If the decoder does not have access to the original, complicated, signal, it will be nearly impossible to recover it.
Lynn

Audio File - Specific frequency range filtering

Hi! Im very noob in labview, i need help for my exam! The task is to do a audio filtering program in labview. Sorry about my bad english. :/
here it is:
First thing: We must do a wav file with some noise then we need to describe with some aspects:
-The noise's specific frequency range
- The wav file's specific frequency range
We need to with the program:
-Specific frequency range filtering ( with windowed -sinc)
- One input : the wav file
- One output: the filtered wav file
- Compare the filtering with the built in digital filters
( we can change the parameters in the front panel! )

Hello.
I suggest you to start with examples from Example Finder (in Labview go to Help>> Example Finder). There you can do a search for function you are interested in and figure out how those simple example are implemented. Combination of those examples will bring you result.
Functions which can be usefull for your task, you can find on Sound and Vibration subpalette (block diagram).
Links for examples, which explains how to use those functions, you can find in detailed help of those functions.
Do a serach how to write in wav file in LV, how create signals with specified features, how to add noice to signal, how to filter signals,...
I do not think that asking for a code, without trying firstly, is good aproach...Did you try to do your task by yourself?
Best regards,
S.P.

Export Media in Premiere Pro CS6 ONLY produces an Audio File in Media Encoder

First, all the system info:
Windows 7 Home Premium Service Pack 1
Intel HD Graphics 3000 Display Adapter
I’m using Premiere Pro CS6 Version 6.0.5 (001 (MC: 264587)) “Mighty Kilt”
Version 6.0.3.1 (64 Bit) of Adobe Media Encoder CS6
Other Adobe applications installed: AIR, Creative Cloud, Download Assistant, Encore CS6 Library, Flash Player 12 ActiveX, Flash Player 12 Plugin, Help Manager, Premiere Pro CS6, Premiere Pro CS6 Functional Content, Reader XI
I’ve run check for updates on both Premiere Pro and Media Encoder and everything is up to date
I'm producing a two disc, dual layer DVD set for 100 families of my son's marching band.
I have a half hour "masterwork" of interviews, performance segments, rehearsal sequences, credits, etc that I've edited together complete with sound, etc.
I also then have various other video segments from the season, some high quality (1920x1080) and some lower quality.
Rather than import everything into Encore and have Encore transcode them all at the same quality, I wanted to ensure that the half-hour piece and certain other segments were transcoded as high quality as possible and given the largest file sizes I could and then transcode other segments at lower quality levels as needed so everything would fit on the discs.
The company I'm using to duplicate/replicate the 100 DVDs from my Master DVD advised me to encode them FIRST in Premiere Pro as desired, and THEN import those files into Encore.
I've used Adobe Media Encoder a lot to export the individual segments into files I could upload to Youtube and have had no problem for the last several months.
But now, two things happen:
1) AME automatically designates the "Output File" with an .aac extension (see attached)
2) When I do an "Import As" of the AME encoded file in Encore it only loads an audio file (obviously)
Huh?
Here are the settings I use in my Export Media dialogue in Premiere Pro CS6:
Format: MPEG2-DVD
Preset: NTSC Widescreen High Quality
Output Name (and filetype): ______ name with Save as Type of "Video Files(*.mpg)
BOTH the "Export Video" and "Export Audio" boxes are checked (see attached screen print)
Filters Tab: left as is, don't do anything here
Multiplexer Tab:
choose the "DVD" button (vs the "None" button)
Bitrate Type: left as Variable which is what is selected as a default
Video Tab:
Codec: Reads MainConcept MPEG Video
Quality: increase it to the max of 5
TV Standard: NTSC
Frame Rate: 29.97
Field Order: None (Progressive)
Pixel Aspect Ration: Widescreen 16:9
Render at Maximum Depth box checked
Bitrate Encoding: I choose CBR for my high quality segments and VBR 2-pass for my lower quality segments
Bitrate: kick it up to the max for high qulaity segments, adjust as needed to get the filesize I want for my lower quality segments
Audio Tab:
Audio Format: Dolby Digital
Codec: Dolby Digital
Bitrate: At least 320, sometimes higher for the segments I want to be the highest quality
I check the "Use Maximum Render Quality" box
I chose "Queue" for everything and then let it run overnight
And yes, the original media has BOTH a video track and an audio track that works (see Screenprint)
So... what am I doing wrong, helpful experts????

Larry, Mark, Jim & Bill -
It is 1 am Tuesday morning and I'd do a primal scream if my family weren't all sleeping upstairs.
Somewhere along the way, I either read or was given the advice to just skip AME, load my videos as timelines in Encore, and just do the transcoding there.
I spent almost a day trying to figure out how to get "buttons" for my menus I was building (they weren't appearing in Encore's library like they should) and eventually found a simply workaround.
So I then spent the last 24 hours or so building all my menus, creating small clips for my "motion menu" buttons, getting all that straightened away.
Got really excited earlier tonight since it seemed like I was nearly finished and coming down the home stretch.
I hit "build" in Encore, went away for a couple hours, and came back to find a big error message awaiting me telling me that "Encoding Failed."
I have read a lot of posts, and the outlook is not good, or at least the possible solution not simple or short from a time perspective.
I have a 100 families awaiting their DVD they've paid for that they initially were told they'd have by mid-February. I've been telling everybody that they would have it - guaranteed - by this weekend when there's a band festival that would be ideal for distribution. The DVD duplication company says they have to have it by Wed morning at the absolute latest. Meaning I've got about 24 hours to figure this out and burn my dual-layer masters of the two DVDs or once again tell people "I know I guaranteed you'd have it by this weekend, but I don't have it ready yet."
Almost as importantly, I can't afford to keep spending the hours and hours and hours on this that I have. It's impacting my personal and professional life.
Thus the primal scream.
The posts I read about the encoding failure all pretty much advise that I use AME to do all the transcoding work. Which of course brings me squarely back to my original problem: I cannot get a video file out of AME.
I'm embedding a screenshot of what I see when I click on the pull-down menu in the AME dialogue window. The problem: there is no MPEG2-DVD option listed to be chosen.
Although I'm relatively new to forums, I've read enough posts by now that it seems like most people are in crisis mode when they post. So I realize my situation is not all that unique. But from a newbie rookie in the field to you experts, I REALLY could use some assistance right now, because between this and the long, cold winter we've had, I'm about to lose it. :-)
P.S. Mark, in answer to your question about frame rates, I have 18 different videos/films/projects that I'm including in my two DVD-set. There have probably been nearly 6-7 different cameras that have contributed footage to one or more of those films by borrowing camcorders of different brands and types and quality levels from people. Nearly every one of those 18 films have multi-camera sequences in them. And many films are collections of sequences which themselves are "nested sequences" sometimes three deep. In otherwords, there is no way I can tell you all the various frame rates that may be represented among these 18 separate projects and I certainly can't go back and hope to change them all to the same one.
Help. Please. :-)

Reconnect clips to source audio file

I have a FCP project with audio clips cut from a longer one hour clip.
When I send my first audio clip to Soundtrack, filter it, I see that the entire one hour audio clip has been filtered. I send the clip back to FCP and continue editing.
When I get to me next audio clip, which has an in and out point later in the one hour source clip, I am having problems simply connecting that audio clip to the previous one I worked on.
Is the only way to filter the next clip to add filters? It seems a waiste as STP has already applied filters to the entire source clip. All I should need to do is reconnect the media.
Has anyone done this?
Thanks.

If you CTRL - CLICK a clip in the timeline and choose to SEND to a Soundtrack Audio File Project, you are prompted to choose where you want to save the (sent) audio file. Remember this location.
Any changes made to the audio from this clip in STP will affect only the (sent) file (as long as you didn't choose to overwrite the "raw" original).
When you go back to FCP and click on the timeline, the audio of the clip you first selected should appear with the new (sent) filename and your changes will be included in it.
Other clips in your timeline which originate from the same original "raw" audio clip won't be affected because it's only the (sent) file which has changed. The other clips are referencing the "raw" files in your capture folder.
You could marry the (sent) audio to the full video sequence and then chop it into clips, or you could find the relevant portion of the (sent) audio for each other clip and put them together.
What we'd both like to know is whether there's a shortcut way of associating these clips with the (sent) files.
Andy
Quad 2.5 G5. 8 gigs. Maxtor 6L250M0 + 6H500F0. G-TECH G-RAID 1 TB Mac OS X (10.4.5) FCP 5.0.4 (FC Studio). Photoshop CS2. Sony HVR-Z1E and HVR-M10E

Having trouble with a audio file. Need help please!

Hello.
This may sound funny and not to related to any of this at first, but if anyone here knows about the game portal, then they know about the ARG that went on during last week and may even still be continuing. None of thats really important but if anyone cares to know, basically VALVe released a new update with 30 new sounds, 26 of them moarse code/sstv pictures. 3 of them called dinosaur_fizzle 1 2 and 3. Infact, each one was named dino_1 dino_2 and so on.
The main sound that no ones been able to crack is called Dinosaur_noise. Its obvious there are voices in it and its some sort of major clue, this file is way out of league with the other ones. No ones solved it 100% or cleaned it enough to make basically anything audible.
Now this started March 1st. Since then i have worked hours on the audio file. I have gotten close, but not close enough.
Thats why im turning to someone here, that can do a through analysis of the audio that adobe or reaper just cannot do.
In the attachments is the original file itself with no editing what so ever.
Heres some collected data.
Frequencies of interest
An analysis was undertaken to find the strongest frequencies acting within "dinosaur_noise". Linear and Logarithmic curves showed sets of dominant figures, with the most being:
Frequency (Hz)
Comments
9092
Largest spike before silence
7862
Very faint
7812
Very faint
7657
Very faint
7502
Largest spike in the 4000-9000Hz band
3502
various noise between 3502 and 3510
360
Another very visible wave. This was the most pronounced visible wave in the Logarithmic scale
35
The most dominant low frequency wave
There are a few standout segments, not of voice data underneath, but visible clicks or something happening about 5 times. These could be indications of splices or other manipulations going on. The largest noticed broad-spectrum spikes were:
Time(in samples)
Comments
55402
First broad spike
422931
Second broad spike
751643
Third broad spike
1081537
Fourth broad spike
1110841
Fifth broad spike
Silence in the frequency ranges is located in the band of 10.567 - 15.337 KHz. All frequencies above that are likely to be high frequency poinoning from the FFT process.
Heres what i know personally.
-Noise reduction makes it a lot worse and adds strange harmonics and artifacts.
-Most filters do the same.
-Clear voice around 15 that says THATS AMAZING.
If anyone cares to help I would appreciate it so much.
Here are some links that might help.
http://www.snotmonkey.com/work/school/405/methods.html
http://pastebin.com/r90fEtnd C++ code
Actual files in attachments
Attachments:
dinonoise.zip ‏2554 KB

LabVIEW has a sound and vibration toolkit. Do you have that? Do you have LabVIEW?
The forum is made up of volunteers who will help others with their LabVIEW problems. But you need to have a specific problem and ask clear questions.
What you are asking for is very unclear. Why do you even want to do a "thorough analysis of the audio"? No one is going to do your project for you.

Cant listen to audio files

For some reason whenever I try and listen to an audio file on the highly filtered network the file never seems to completely play. Perhaps this is a drawback to the high filters around here. The evangelical college that In attended really only filtered sex, ****, games, and other inapropiate websites. But at this school they filter that and much more. This includes Direct Connect access to CitiBank, TurboTax, Apple Software Update, FTP, Microsoft Technical Nessgroups, and even live audio streaming. Although the IT admins around here wont listen to anyone but their own voices, I realize that I cant convince them.
But anyways can any of you try and listen to these audio files and tell me if they play? I think there is a 1% chance the issue is with Yahoo, but its about 1%. Thanks..
http://www.cerm.info/audio_files/KJVO.mp3
http://www.cerm.info/audio_files/Trinity.mp3
http://www.cerm.info/audio_files/winter.wav
http://www.cerm.info/audiofiles/randyclark.mp3
Also can anyone tell me technically how they are able to filter audio files?
If they filter audio its strange why they do not also filter video.
http://www.cerm.info/movies/welcome.mov
John

You need to raise this issue with your school's tech support group. I tried playing the first mp3 link and although I didn't listen to the entire thing, I was able to fast forward to a few minutes prior to the end and it played fine via my cable modem. If your school's network administrators are placing a filter on your network port, only they would be able to assist you in opening it so you can listen to that content.

Insert an audio file (MP3) into a BLOB

I want to insert an audio file (MP3) into a db table.
i have declared the db field to be blob.
can anybody tell me what java code to write to insert an audio obj into db blob field.
thanxx in advance.
Raheel

I have never tried an MP3 file but have done images.
I assume the same type of methods would work.
Try looking at some of these forum answers.
http://forum.java.sun.com/thread.jsp?forum=48&thread=446603
http://forum.java.sun.com/thread.jsp?forum=48&thread=448964
http://forum.java.sun.com/thread.jsp?forum=31&thread=458059
http://search.java.sun.com/search/java/index.jsp?and=blob+store&phr=&qt=&not=&field=&since=&nh=10&col=javaforums&col=wireless&rf=0&Search.x=15&Search.y=8
rykk

Filtering audio in LabView

Hi,
I'm still working on my LabView project to obtain the numerical value of the peak frequency in a spectrum.
I'm using an example VI, and I've done some slight modification to it to learn how to implement filtering.
I've added a filter, but cannot seem to get it to make any difference in the output. I'd like to cut out the lower part of the spectrum.
If anyone has any advice as to why it is behaving as if I don't have a filter in at all, I would very much appreciate it.
Thanks,
Wes
Attachments:
Sound Card Spectrum Modified.vi ‏114 KB

nvaloor wrote:
Hi
I need to do the following
I have a audio file which has a person speaking and typing simultaneously.
What I need is to filter out the voice of the person and just get the keystroke sounds.
By analysing the sprectrum you can probably detect the key stroke, and use the spectrum before and after the stroke to estinate a filter to filter out the speaker during the stroke.
Look at the spectrogram.
Mono or stereo input? Using the phase information might help too.
Powerful algorythms have been created to do such tasks (ask the NSA ?? ) but a slightly similar task is the party-problem for hearing aids->multiple sound sources, where our brain (together with both ears) is able to filter out the sound (usually person) of interest, but people who wear hearing aids lost that feature. Search for puplications in the acustics fields.
Kalman-filters might do the job (Have fun )
Greetings from Germany
Henrik
LV since v3.1
“ground” is a convenient fantasy
'˙˙˙˙uıɐƃɐ lɐıp puɐ °06 ǝuoɥd ɹnoʎ uɹnʇ ǝsɐǝld 'ʎɹɐuıƃɐɯı sı pǝlɐıp ǝʌɐɥ noʎ ɹǝqɯnu ǝɥʇ'

How to process overly compressed high volume audio files

I used a process in scientific filters in audition 3 to process overly compressed high volume audio files normalized to 0 db. It was like magic.
How would you process this sort of file in Audition 6.

Below was the process used in Adobe Audition 3. The sound was much improved. Try the steps on a file you have to hear the difference.
Effects- Amplitude - Normalize - uncheck the decibles format box - normalize to and normalize L/R boxes checked, enter 48 into percentage box.
After normalization was complete
Effects - Filters - Scientific filters - select Chebychev 1 tab - band pass button and select the following parameters.
Cutoff =4, high cutoff = 22050, order = 18 all other settings at defaults values (0). Delay radio button sellected, check the extended range box, press OK.
Increase to appropriate volume after processing if necessary.

Cant play audio files

I am connecting to the Internet via a highly filtered ISP at a fundamentalist school.
Below are some files that DO NOT play via FireFox (latest version) on the network here on my ibook.
http://www.cerm.info/audio_files/scientology.mp3
http://www.cerm.info/audiofiles/randyclark.mp3
etc...
While I cannot play the above audio files at school (the files only play for a few seconds and then cut out) I CAN play them on the same ibook OFF CAMPUS at the local library.
Below are some files that DO PLAY on the NETWORK AT THE SCHOOL in FIREFOX.
http://download1-cm.edgesuite.net/crownradio/public/mm/mmlo6-18.mp3?sauth=118224 6151_69a429f064b7cbb195c71151c22dbf87&ext=.mp3
http://www.coastsidebaptist.org/home/900/900/audio/6-17-07%20Main.wav
Why is it that some audio files play just fine on my Ibook in FireFox and others do not?
And why can I play all the audio files in Safari?
John

...but it doesn't work for me.
downloaded perian onto my macbook pro, but still only audio.
next best bet?
also, any way to play mpeg-3?
thanks--
mm

Only seeing Audio File in AME on Import or Export to Queue from Premiere

Hello All...
I don't have alot of experience with AME... I"ve exported a sequence from Premiere Pro, Selected setting and added to the Queue in AME... all I see show up once the AME window opens is an Audio File.
When I open a Premiere project directly from AME (using the dropdown menu) I get the same result... I've tried this with a few Premiere Projects. Am I missing a step here???
Also... I'm confused as to why I select all the transcode settings in Premiere Pro when exporting media... and then there are presets again in AME... don't the settings carry over from Premiere's export window?
Thyanks for any help...

I'm wrestling with this same issue.
I'm on Windows 7using Adobe Premiere Pro CS6.
I'm producing a two disc, dual layer DVD set for 100 families of my son's marching band.
I have a half hour "masterwork" of interviews, performance segments, rehearsal sequences, credits, etc that I've edited together complete with sound, etc.
I also then have various other video segments from the season, some high quality (1920x1080) and some lower quality.
Rather than import everything into Encore and have Encore transcode them all at the same quality, I wanted to ensure that the half-hour piece and certain other segments were transcoded as high quality as possible and given the largest file sizes I could and then transcode other segments as needed so everything would fit on the discs.
The company I'm using to duplicate/replicate the 100 DVDs from my Master DVD advised me to encode them FIRST in Premiere Pro as desired, and THEN import those files into Encore.
I've used Adobe Media Encoder a lot to export the individual segments into files I could upload to Youtube and have had no problem.
But now, two things happen:
1) AME automatically designates the "Output File" with an .aac extension
2) When I do an "Import As" of the AME encoded file in Encore it only loads an audio file (obviously)
Whassup?
Here are the settings I use in my Export Media dialogue in Premiere Pro CS6:
Format: MPEG2-DVD
Preset: NTSC Widescreen High Quality
Output Name (and filetype): ______ name with Save as Type of "Video Files(*.mpg)
BOTH the "Export Video" and "Export Audio" boxes are checked
Filters Tab: left as is, don't do anything here
Multiplexer Tab:
choose the "DVD" button (vs the "None" button)
Bitrate Type: left as Variable which is what is selected as a default
Video Tab:
Codec: Reads MainConcept MPEG Video
Quality: increase it to the max of 5
TV Standard: NTSC
Frame Rate: 29.97
Field Order: None (Progressive)
Pixel Aspect Ration: Widescreen 16:9
Render at Maximum Depth box checked
Bitrate Encoding: I choose CBR for my high quality segments and VBR 2-pass for my lower quality segments
Bitrate: kick it up to the max for high qulaity segments, adjust as needed to get the filesize I want for my lower quality segments
Audio Tab:
Audio Format: Dolby Digital
Codec: Dolby Digital
Bitrate: At least 320, sometimes higher for the segments I want to be the highest quality
I check the "Use Maximum Render Quality" box
I chose "Queue" for everything and then let it run overnight
So... what am I doing wrong, helpful experts????

Synching a separate audio file with video

Hello, I was hoping someone might be able to help.
I have a video file but I am trying to sync a audio file to the video.. Not the video's original audio recording but a different file. Basically I have a recorded live band performance and I also have a direct out audio recording of same song (which is better quality than the video recorders internal mic) I want to sync the direct audio to the video.. The biggest issue is the tempos of the video and the direct audio recording are different. Can you set the tempo of the video file to the audio or vice versa? Can this be done in FCE or is this something for Logic Express.?
Thanks for the help

This is a BEAR!
First need to know much more and even then, oh well...
(a) You say the sound from the board has a different "tempo" than the sound from your on board Mic. or camera. Were both recorded at the same time and place? (b) When you say "tempo" do you mean that when you import the sound from the board (that was recorded to a CD?) into FCE and place on the timeline that the synch "drifts"? (c) Are you first finding a common "synch point" to begin with? (d) Your camera was set to 16 bit 48 kHz, right? What are "the properties" of the sound from the CD created on the board?
If you are indeed finding a common synch point - that is, if you are synching by eye and the synch then "drifts" because the 2 have different "properties", then you will have to either use a cutaway every time you re-synch by eye to a common "cue" or use a D.A.W. or sound editing program that allows you to import a "movie" and has speed and pitch controls that you will have to constantly tweak. This is a formidable undertaking even for a pro but can be done. Say you do that, tho. This "tweaking" will change the quality of the sound at best and sound "wavy" at worse, so...
bottom line - I'd try to enhance the sound you recorded in camera with filters and an equalizer in any DAW that you can import a "movie" into like Garage Band or just do it in FCExpress itself and learn that next time you are shooting a live musical event, plug your camera into the board to begin with... via balanced cable. Careful to tape your XLR cable run down well.
Maybe a sound engineer might be able to help more. Ask on a Pro Tools or Logic board?
Good Luck -
B.S.

Drawable Pitch enveloppe on audio file as in cubase sx3 & ableton live ????

Hello
In cubase sx and live , you can find pitch enveloppe ( or transpose ) .
Live makes it real time with possibility to draw a pitch curve , SX makes it off line on an audio file.
I am looking for the same kind of tool in Logic . Does it exist ????
So far , the only way i could reproduce this effect ( i like to play long delay with endless feedback and automate the pitch of the feedback , or make a 1 osc riser sound with flat osc sound that i pitch afterwards ) was to insert apple's plugins : Pitchshifter II or apple AUpitch and automate them .
The problem is realtime automation of these plugins cause clips and glitch , and the pitch is going up step by step and not in a linear way . ( you can hear every semitones variated , no transition between tones ) .
So is there a drawable pitch enveloppe option in Logic 8 ?
If yes , where would that be ??
Otherwise is there a better alternative to pitchshifter II & AUpitch ???
I ve been serching in the manual already + i am aware that the Delay Designer fx can do the same work , but it s only working with delayed sounds.
I hope it was clear enough.
Thx in advance.

Once you load the audio into the EXS sampler and assign it to a MIDI note, use either or both envelopes as well as up to three LFO's to modulate pitch of your audio. You can also modulate the various envelopes and LFO's with each other. While you can't draw the envelope, you can get very complex shapes with the various routing possibilities. This is also a way (among many) to do filtering, stutter cuts, looping, anything else you can ordinarily do with a decent sampler.
Of course this is a work around; you'll need to trigger your audio with MIDI, then print it if you would prefer to have it sitting on an audio track. But at least you can do it within L8. Possibly the most direct way to do what I think you're trying to do would involve buying Melodyne.

How can I see previously stored tunes no longer on my computer? Where are iCloud audio files?

Maybe my question has an answer too "obvious" for me to see it. But lately I've become aware of tunes that, for one reason or another, are not showing in my iMac's iTunes program. I went to System Preferences and clicked on "iCloud." Immediately, I saw how to check iCloud for the following files and documents: Mail, Contacts, Calendar, Notes, Safari (not sure what this category could be for--seeing what was on the web at some prior point in history?), Photo Stream, Documents & Data, Back to My Mac, Find My Mac.
In other words, going to iCloud appears to reveal categories for all of your iCloud content EXCEPT your music files. I know about iTunes Match, and I think I like it, now that I've gotten used to a few of its quirks. But certainly there are times when the owner of downloaded, commercial audio files (i.e. Tunes) will not have them on his computer. And certainly there must be a way for me to see what I've previously stored in the Cloud--if only to remind myself of the music I've purchased and have available to me.
I may be missing something about the whole iCloud concept. Is there a way for me to see tunes I've previously stored in the iCloud but no longer have on my computer? (I'm always careful not to delete tunes from my iTunes program unless I see 2 or more identical files appearing on my computer).
Thanks for any directions to the iCloud musical holdings / archives--or clarification of any misunderstandings.
Cap
(I see that I'm currently running OS 10.8.2. I have only 500 gigs of memory, and my music collection is so larg, I'm almost dependent on being able to access songs in the iCloud. Next time, I'll certainly pony up for the largest memory Apple offers on an iMac.)

They are not showing up in Lightroom at all. I already deleted them.
When I try to import them again into Lightroom it wants to add another 2012 folder to wherever I am trying to insert them and then it does the month/day folder when it already has the month folder it should automatically go into and not be adding a seperate 2012 folder first.
I can't even get it to do the above right now. Now all it does is it wants to make this seperate 2012 folder with the month/day folders but has little boxes beside each folder showing that there are no images to import. The images do show up on the import screen but they are all darkened and unchecked. When I click the proper folder to import them they come on bright and have little boxes checked, but that lasts for less than a second before it decides they are not new. They are new images, I deleted them all out of Lightroom, and probably missed deleting some through Lightroom but instead deleted them straight from the folder outside of Lightroom. This is what is probably causing me such grief and I have found nothing in my searches to tell me how to correct this. *sigh*

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