Best Bitrate for Spoken Word Audio Files?

I'd like to import a bunch of lectures from CD's into iTunes to download onto my Mini. Some of these lectures are in the MP3 format and some are in regular audio CD format.
I would like to import them as MP3's into iTunes with a bitrate that minimizes the file size while still retaining decent sound. I certainly won't need something like 128, but do you have a recommendation? I'd like to cram as many files onto my Mini as possible, and still be able to hear the speakers clearly.
Thanks!

You could probably go down to 32K with no problems at all.
Give it a try at even lower (ho wlow can you go?) and see what happens.

Similar Messages

Spoken Word Audio CDs

How do I NOT include 'spoken word' audio cd tracks in shuffle songs. My language cd's are on Audio cd's rather than AudioBook format so they fall into my music library... Much thanks to anyone who can help!

Select all the tracks, use File -> Get Info (or right click menu: Get Info), and in the options tab choose Media Kind: Audiobook; all the tracks will move to the Audiobook category and, if they have track numbers, cover flow will show just one entry for the group, you still see the separate tracks in the list.

Good levels for spoken word?

Is there a general guideline for where levels should be for spoken word?

Jim Simon wrote:
The short answer is no, there is no one "general" guideline on this one.
Good answer...
To shooternz.... You can make a whisper as loud as a scream LEVEL WISE.... So they might both PEAK at -3db but the scream will still sound (RMS) MUCH louder than the whisper.
To OP... The most importing thing is to get the relationships between the audio events to blend in with each other. What I mean by that is, if you have a car dialog and are working with Foley talent
you might want to make sure that the spoken word mixes (blends) in with the environment they are supposed to be in... CLose your eyes and listen to background noise of the engine and the tooting car horns outside,
then up the gain on the spoken words until you find that they blend in nicely witht the whole atmosphere. After you have blended the various levels of one scene you can create a nested clips and make sure that is normalized.
Basically what that means is that you find the loudest passage of the clip and up the gain until that part hits the roof (I set mine to negative 0.5 dB) you can do -3db too, to be real safe.

Best setup for a non-Mac file server?

I have a dual xeon server, with a SATA RAID5 I want to use as a file server in a cross-platform environment.
*What I've tried and the issues...*
At first I considered using Windows 2003 (Win2k3) but Services for Macintosh (SFM) is an older version of AFP and thus only supports 31 character filenames. With all our Macs supporting SMB/Samba/CIFS and Apple touting that "Macs and PCs can co-exist harmoniously on the same network" I figured I would give that a try.
SMB doesn't work.
Sure I can create a connection, but transferring files is a completely different story. I'm trying to backup application and system data, but companies such as Adobe and Apple have named some of their files with special characters that can't be transferred over SMB. I know NTFS doesn't support these characters, but I though a Linux box using SMB would work fine. It doesn't. It's the protocol which keeps me from transferring the data. I end up with the lovely error message of "You cannot copy some of these items to the destination because their names are too long or contain invalid characters for the destination..." (what's sad is, if you google for " because their names are too long or contain invalid characters for the destination" you only get 6 results.)
So I thought I would give NFS a try. Apple says "Viewed from Mac OS X, [connecting via NFS] is just like connecting to an Apple or Windows server." No. It's not. NFS shares don't even show up in the Finder's Network listing. There are also a pile of other hurdles which are only tackled by savvy, command-line using users.
So that leaves me with AFP. Win2k3 doesn't support filenames longer than 31 characters, and Win2k8 is dropping SFM altogether. Off to choose a *nix flavor, but that requires Netatalk. It hasn't been updated in years, it has many bad performance reviews... and most distros have removed it. I can download and install it. Oh, but that requires I get the kernel source files. Then I have to create an RPM an that's not working... now I'm several levels deep in trying to figure out how to get Netatalk working and I'm not even sure it will work.
*What's the best setup for a non-Mac file server?*
FreeNAS seems promising, but it's in alpha/beta and they have all sorts of warnings regarding potential data loss. Sure there's ExtremeZ-IP, but I really don't want to spend $675 do something Apple claims OS X can already do. I can put just about any non-Mac OS on this thing... what's the best way to set it up so it works?
Thanks much.

Rick may be right because although i didnt think of it before i tend to have notoriuosly long classnames for my php classes and i have used samba on occasion (when rsync is out of the question for one reason or another) and never had a problem. I use kubuntu (feisty at the moment )with an ext3 filesystem. if i have a chance this evening ill give it a try and see what happens.
You could also possibly use FUSE to use an ssh filesystem for the shares... i don tknow how that would figure in your back up though.
Also if worse comes to worse you could tar or dmg the the necessary files... just some thoughts.
Ill be interested to know what you end up implementing....
OH one last thought... Compile Darwin from source and use that as your server

Audio bug with WAV files: use mp3, not wav for syncing long audio files

Just a tip:
I've noticed another audio bug in Presenter (including 7.05) but there is a workaround. I often export the audio as a single, long clip from pre-recorded lecture videos, and sync it with the powerpoint slides in Presenter. I've found it easiest to import the long audio file into the first slide, then sync using the Edit Audio window. I find that much easier to use than the Sync audio window.
However, If you use WAV format for the imported audio file, sometimes Presenter will simply refuse to play back anything beyond about 5 minutes into the timeline in the Edit Audio window. Nothing fixes it - restarting, re-importing, etc. However the problem goes away if you save the audio as mp3.
Ellen
http://thedesignspace.net

Why did you post this in the LiveType forum?
I would recommending posting your question in the Final Cut Pro forum.
Have you tried trashing your preferences?
Anyways... here is a thread with a similar problem:
http://discussions.apple.com/thread.jspa?threadID=656515

Fixing duplicate database entries for the same audio file

My iTunes Media folder is stored on a network accessible storage server. Since upgrading to 11.1.x, my computer seems to be duplicating database entries for the same audio files - for some albums, I have 4 entries for every song, each one pointing to the exact same file. The only difference between entries is rating info and play counts, which are specific to a particular entry (I think). Has anyone else seen this behavior? Is there an obvious way to stop this from happening? I can delete and rebuild the database from media files, but that starts getting old if I have to do it every week or so. Is there a script to delete duplicate database entries? (I found a VB script on the iTunes for Windows community, but the offending machine is a Mac).
Thank you for any assistance!
K

Select the photo stack under the People tab and click the rename icon (bottom left)

Best way to break up audio files for Captivate?

Hi guys, I have a dilemma here. I am getting ready to record the audio clilps for Captivate and I'm wondering how I should break up the audio. For example, the first slide has
First textbox with title "Introduction"
Second textbox with title "Learning Objectives"
Third textbox with "In this lesson you will learn: a) how to do something b) how to define something"
Fourth textbox with "It will take you 5 minutes to complete this lesson"
So, how does Captivate play the audio? Do I record only one audio file for all the text on screen and add the audio to the slide? Or, do I record smaller chunks (for each textbox) and add the audio file to each of the textbox? In which order will Captivate play the audio? I'm assuming if all the textboxes have separate pieces of audio, Captivate would play those at the same time.
Thanks

Hi. My recommendation would be to create one audio file for the slide, then use the timeline if necessary to time the text captions (and other objects) to appear/disappear to match the slide audio. If you record audio for each text caption, the audio will begin to play as soon as the text caption becomes visable on the slide. If you have more than one text caption appearing at the same time, both will play their audio at the same time.
Hope this helps.
Mister C.

PPro2 best bitrates for transcoding in AME to DVD

Hello. I have created a 165-minute movie (including 8 minutes of motion menus) to be authored with menus/chapters in TMPG Authoring Works 4. I'm exporting from PPro2 via Adobe Media Encoder for transcoding to mp2, and have space left over due to my current bitrate settings: 1.5 minimum, 6 target, 8 maximum. The resulting master will be factory-replicated with glass master. Audio will be transcoded to stereo AC3 only; no subtitles.
TMPG is showing that my current movie size is 7.323GB. I would like to increase my bitrate settings in AME for the best possible visual quality, but I've read various limit recommendations of 7, 8, or at most 9mbps for the max setting. I will burn to Verbatim DVD+R DL which shows a usable space of 8.025GB. Can I assume that factory replicating will avoid DVD player compatability problems due to high bitrates?
My Bit Budget Calculator v.2 gives a maximum video-only bitrate of 9.352mbps, average bitrate of 6.594, and a video-only target bitrate of 6.146. The target number is easily doable, but I wonder if I dare boost the max to 9mbps (which Bit Budget Calculator says should not be exceeded), while keeping the minimum at 1.5
Any advice about bitrates will be appreciated. Thank you.

Hi Jim et al.
I wanted to update this thread since I figured out where my problems originated and how to solve them. I received advice from Adobe forum, videohelp.com, and doom9.org, and I am grateful to all those who commented.
First, Ann, my last WinXP re-installation was in 11/08. Lacking an XP CD, it required taking my computer to a shop for hours of billable work, so I am not eager to repeat that. My experiences since Sept.2010 suggest that my computer was not causing the various problems.
I had been getting various artifacts in my AVI lossless encoding using Lagarith and UT. There is a "gallery" of screenshots of these artifacts at:
http://forum.videohelp.com/threads/326242-artifacts-created-by-Lagarith-UT-unable-to-impor t-into-HC?p=2021174#post2021174
I attempted my same workflows in PremiereCS3, but again got artifacts. Poisondeathray wrote: "I think this is a PP2 problem . The lagarith artifact looks very familiar - I remember reading about incompatibilities with PP2. I don't recall seeing any of these after CS4 or 5." A thread at doom9 commented on PPro2's shortcomings: http://forum.doom9.org/showthread.php?t=107558 This thread includes the suggestion: "I choose Microsoft AVI. Then I go to the video menu, and choose None for the codec." But for me none of the "uncompressed" settings ever worked--the output was always gibberish.
Through trial and error, I found that Huffyuv (version 211 only) produced no (or the fewest) artifacts. Huffyuv 211 worked for me, so I've been using it for processing crucial movies and for archiving.
I had been getting repeated error messages when trying to use an AviSynth script in HC-025. MT mode was not needed in the script, after all. Poisondeathray wrote: "If you used lagarith or UT directly from PPro2, it will be RGB and you need to convert to YV12 , if it's interlaced add interlaced=true
Avisource("F:\Exports\CopyofMontage1.post.avi")
ConvertToYV12()
or
Avisource("F:\Exports\CopyofMontage1.post.avi")
ConvertToYV12(interlaced=true)
The second script worked for me. For future readers, I stress the importance of filling-in HC's output and log names+locations before encoding, and of selecting "Make DVD compliant." Here are the settings for my letterboxed 4:3 movie:
Having learned how to use HC-025, I ended up not using it in my workflow. Since I definitely wanted to use TMPG Authoring Works 4 (TAW4) for authoring, I brought three samples into it for comparison:
1) plain AVI exported directly from PPro2 with no lossless encoding. 2) AVI with Huffyuv-211 lossless encoding. 3) AVI with Huffyuv-211 processed through HC-025. #3 produced the least clean footage--HC added barely visible "borders" around the feet of letters in text, an artifact that was not present in #1 and #2. Since #1 and #2 looked the same, I ended up using #2: AVI with Huffyuv-211 imported into TAW4.
TAW4 with its onboard Intercom MPEG-Video Transcoder and its video settings probably made a great contribution to my DVD's visual quality. TAW4's video settings are not obvious; they're at the upper left when in "Source" view. For each Track, click Settings>Video, and make your selections. 8000 bitrate is the highest recommended by Craig Hanna at DiscMakers, with 1550 bitrate as the minimum (which answers one of my earliest questions in 9/10), and 6050 for average bitrate for my particular file size. Then click Video Encoder Advanced Settings and choose Highest (with error correction). Here are my settings for my 172-minute DVD-9 (grab lower-right corner and stretch image to read better):
Thank you. --Best regards, SamB.

Workflow for syncing uncut audio files to a complete cut?

Hello,
I've been working on a narrative project (about 30 mins). For various reasons, I was unable to sync audio before editing, and have now done all the cutting with camera audio. We recorded separate audio on a stand alone recorder, and now I want to sync it to the project in CC. There is one audio file for each night of shooting, so the audio is in these giant files not separated by take. Can someone advise me as to the best workflow for this? Clearly it would be possible to do it manually by cutting up the audio files, identifying the proper audio for each clip and lining up waveforms, but this would be incredibly time consuming. It seems that there should be some way to use either Plural Eyes or the new audio sync feature (which I don't have experience with) to sync audio automatically, but I'm not sure how to do this.
Thanks for the help.

If it was my task, and I didn't have the recorder files in advance of having to edit, I would probably jot down on a notepad the in/out timecode from the camera for every cut.
I would then take the recorder audio and sync it up with the original camera audio on a different sequence. This might prove a lot easier than trying to do it in a bunch of steps by waveform.
Once I had it all lined up, I would make the razor cuts at exactly the same place as the timecode I jotted down on the notepad. Then I would delete all of the audio portions that I didn't want to keep. I would then copy over the audio track to the original sequence and line of the audio with the original sequences cuts, once again using the notepad to keep track of the places where everything fit.
Painful? Yes, perhaps. But easier than trying to sync up lots and lots of clips.

Separate drives for samples and audio files..?

Hello all
I am moving to Macbook Pro (previous gen) with Logic Studio after years of PC (feel like I’m standing on the edge of a precipice staring anxiously down!). Will be looking to use it live in our 3-piece band to run existing MIDI Files and live triggered BFD drum samples
Have decided on getting an Echo AudioFire 8 to handle the necessary Audio outs and MIDI connections and will be replacing our rather dated Korg 05R/W General MIDI module with virtual instruments / samples libraries
I am thinking that once the Bass (probably Orange Tree CoreBass samples through Kontakt 3) and Orchestration (still wide open to suggestions on that!) are done, they’ll be exported as Audio files or perhaps ‘frozen’ in Logic to ease the load on real-time processing
Here’s the query then:
*Assuming that the internal drive will be for OS and programs only, will having the sample libraries and exported audio song files on separate physical drives from each other pay dividends – particularly when considering that the audio tracks (probably no more than two or three stereo files) will be streaming whilst BFD is triggering live samples via MIDI input..? I’m wondering if the samples and audio separate drives situation will be considerable beneficial, or given the intended application, would this approach be merely over-speccing things?*
If that is thought to be the best approach, what do you believe to be the best way of hooking these up? I plan on using the FW400 port for the Echo AudioFire 8 and have ordered a Sonnet Express eSATA Card to hook up an external drive. Should any additional drive look to use the FW800 port or should I simply tag into the spare eSATA port on the Sonnet Express Card..? My concern with the latter is that a bottleneck would occur in the Express port and defeat the whole object
I would really appreciate any thoughts on this
Many thanks
Clive

My opinions:
If you're really only talking about 2-3 audio tracks per song, having them on a separate drive is not going to make that much difference. I'd recommend just using one drive for both audio and samples for now, you can always add another hard drive later if you're getting disk errors, or if you start using more audio tracks.
If you're looking for a good orchestral library, I recommend Vienna Symphonic Library Special Edition (about $500). VSL sounds great, and this particular set is much cheaper and still gives you a good amount of instruments/articulations.
If you do go with a third drive, use both eSATA ports. Even with a little bottlenecking, you should see substantially faster performance than with FW800. The data transfer rate of eSATA is over 3 times that of FW800.
Message was edited by: jdredge

Best program for mlti-track audio

Does anyone know what Adobe program would work to create a multi-track audio interface?
The idea is to allow a user on a website to select what tracks are played.
For example:
A four track audio file of a song including: track 1: lead guitar Track 2: Rythm guitar Track 3: Bass guitar Track 4: Drums
The user could select all 4 tracks...or, if they wanted to play along with a lead guitar...select tracks 2,3 and 4.
Can I do that in Flash? Or what would be the best program for that?
Thanks!!!
Dignon

I haven't used the Isotope plug-in, but I use Brian Davies' Denoise which is a stand-alone cross-platform program: I found it much superior to SoundSoap. It's available in a free trial.
Declaration of interest: I have no financial interest in this program but was involved in the beta testing.
Together with the same author's ClickRepair it's possible to get remarkable results: this sample of a noisy 78 was treated with both programs.
http://homepage.mac.com/rfwilmut/notes/noise/companions.mp3

Best technique for "stripping" an XML file

I've been coding Java for a while now, but this is my first foray into Java-based XML processing. Here's the problem:
I want to take an XML file, extract from it a simpler data set, and output that as a new XML file. (Parse, extract, serialize.) There are a ton of very powerful tools in the API, but I don't want an overcomplicated system. What are the best tools for simple processing?
(Why am I not using XSLT? I need to take in parameters from the command line, and the data set is large, so I want smart caching. Also, the server will need to read the XML file repeatedly to check for permission data.)

A little background info:
I use KimDaBa, a photo organizer for Linux. It stores metadata for the photos in a single large XML file. I've succeeded in having the 2000-photo database display directly in a browser (using XSLT). What I want to do is display a subset of the images, the ones tagged as being publishable under a Creative Commons license.
A lot of the metadata is sensitive. I don't want to reveal location information for everything, and not all the photos are for public release. As a result, serving the whole file to the browser and using XSLT to transform it is not an option. (I'm also not sure I want to get into server-side XSLT.)
One issue here is that the images directory is by default not servable to the web. I'll have a PHP script check each image request to see if the image is mentioned in the stripped database. That's what I meant by permission data.
Once I have a stripped version of the database, I can easily slap together some XSLT, CSS, and JS for convenient browser display.
So, I think pure XSLT won't cut it.

What's the best way for reading this binary file?

I've written a program that acquires data from a DAQmx card and writes it on a binary file (attached file and picture). The data that I'm acquiring comes from 8 channels, at 2.5MS/s for, at least, 5 seconds. What's the best way of reading this binary file, knowing that:
-I'll need it also on graphics (only after acquiring)
-I also need to see these values and use them later in Matlab.
I've tried the "Array to Spreadsheet String", but LabView goes out of memory (even if I don't use all of the 8 channels, but only 1).
LabView 8.6
Solved!
Go to Solution.
Attachments:
AcquireWrite02.vi ‏15 KB
myvi.jpg ‏55 KB

But my real problem, at least now, is how can I divide the information to get not only one graphic but eight?
I can read the file, but I get this (with only two channels):
So what I tried was, using a for loop, saving 250 elements in different arrays and then writing it to the .txt file. But it doesn't come right... I used 250 because that's what I got from the graphic: at each 250 points it plots the other channel.
Am I missing something here? How should I treat the information coming from the binary file, if not the way I'm doing?
(attached are the .vi files I'm using to save in the .txt format)
(EDITED. I just saw that I was dividing my graph's data in 4 just before plotting it... so It isn't 250 but 1000 elements for each channel... Still, the problem has not been solved)
Message Edited by Danigno on 11-17-2008 08:47 AM
Attachments:
mygraph.jpg ‏280 KB
Read Binary File and Save as txt - 2 channels - with SetFilePosition.vi ‏14 KB
Read Binary File and Save as txt - with SetFilePosition_b_save2files_with_array.vi ‏14 KB

Work flow for syncing separate Audio files with Video files in FCP

It may be too late for me to use this workflow that I want with my current project, but I need to figure out what it is anyway for future reference.
Right now I'm editing a movie which is near completion. When we recorded the movie, the camera's deck recorded reference audio from the sound guy at 48k @ 16bit. This audio also went through an additional set of preamps which added a small amount of noise. So we don't want to use the reference audio because of it's low resolution and less clean audio. The sound guy recorded straight to his deck at 48k @ 24bit, besides the better resolution, his audio is noticeably better.
The camera deck and 'clapper' (I think that's what you call it) were not connected or synced. So the video files and the soundsguy's audio files are slightly different lengths, but at least have a 'clap sound' to manually sync to. And I don't mean that the audio and video files won't play in sync with each other, I mean that the soundguy always started recording before the camera, so his audio files have slightly different start and stop times.
Could I have synced the audio files with the video files before I started editing this project? If so, could it have been in the way that the soundguy's audio would be connected to my video files while editing? What would that procedure have been?
My current workflow right now is I have my finished sequence, I change the bit rate to 24 in Sequence Settings, and start manually dragging the soundguy's files to the sequence for every little edited clip. This is going to take forever, there are 99 scenes in this movie. Anyway, at the end of the day these sequences are exported as .omf files and imported into Pro Tools for audio post production.
I was not thinking to figure this out before I started editing the movie, this is unfortunate, a newb move in a decent production.
Anyway, I don't know if there is a standard for this situation or if I just opened up a can of worms for many possible workflows. Please comment if you have experience in this area or please refer me to any reading material.
Thanks for your time!
-Monty

The normal workflow (besides manually syncing up before editing!)
is to have ' some ' way of auto syncing the clips.
So you must have common timecode.
With external audio, you can slave the cam's TC to the TC output of the BWF recorder.
Then you can use BWF / QT merge (one of my apps) to sync them, or use Sync 'n Link.
(another third party application)
If it is not possible cause you're shooting with a cam without a TC input, you can record the TC output on one of the audio tracks of your cam, and use FCPauxTC reader to decode the beeps and translate that to QT TC, and again use the former tricks to sync them.
It is difficult when you have never done it, but very easy once you get the hang of it, and it saves a huge amount of time to do it right from the start.
Have a look at the FCP aux TC reader here:
http://www.videotoolshed.com/?page=products&pID=26
Sync 'n Link can be found here:
http://www.assistedediting.com/Sync-N-Link/
/// disclaimer, i AM the owner of Videotoolshed, and i WILL (at least try) to make money on it. ///
/// i do NOT have any relation with AssistedEditing.com, besides friendship with the owner. ///

Best place for LR catalog (.lrcat) file?

I'm putting together a new PC running Vista and am trying to figure out the best place for the LR catalog (lrcat) file. This PC has multiple hard drives as follows:
C: OS and apps
D: non-photo files (docs, mp3s, etc)
E: photo files
F: video files
All the drives are comparable in read/write speed. The Lightroom app will be installed on the C: drive, and the LR catalog will consist of the images on the E: drive. While working in LR, I sometimes listen to mp3s stored on the D: drive. I'm assuming that the best place for the lrcat file is the F: drive (since I seldom access that drive while working in LR). Does this make sense? Or should the lrcat file go somewhere else?
On a side note, on my last PC, I had the lrcat file in what was originally a dedicated scratch drive (for PS), but I'm doing less and less in Photoshop these days, and my new system will have 12GB of ram, so I decided to do without a dedicated scratch drive for the new system.
Any advise appreciated.

Asking yourself the following questions may help determine which drive is best suited for storing the Lightroom Catalog:
1) Are you the only person using the "Lightroom computer"?
2) Do you have multiple log on accounts?
3) Do multiple log on accounts need access to the Lightroom Catalog?
If you answer yes to #1, its default location might be fine. If you answer yes to #2 or #3, you should concider a central location for the .lrcat file.
By default, the .lrcat file is located in a users document folder. This means that at least one (1) .lrcat file will be created per user account. Seeing that you usually keep photo files on the E Drive, creating a folder on the root of the E Drive might not be a bad idea. Using the root of E schema will then allow mutiple users to access the .lrcat file when they log on. Using the E Drive keeps "photo related" files together, thought Drive D or Drive F would also work. It appears from your description that you keep O/S and Applications only on the C Drive, which I do myself.
Hope that helps!

Best Bitrate for Spoken Word Audio Files?

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