There are many tools out there that can convert CD tracks, WAV and other audio file formats to
MP3 files. When encoding your MP3 files, keep the following in mind:
We recommend Audacity (Mac, Windows)
Quicktime 7 Pro (Mac, Windows)
The older version of quicktime is a great untility.
Convert to MP3
Mac and Windows
Apple's cool music jukebox. Robust and free.
By default iTunes converts audio to AAC, so you will have
to edit the preferences to get iTunes to convert the audio
To create mp3s using iTunes:
- - From the iTunes main menu > Edit > Preferences
- - Click the "Advanced" tab, then the "Importing" sub-tab.
Good, High, or Higher are fine. If you use the custom setting, you may experience issues with playback speed.
- - Click "OK"
- - Click the "Library" item in the left-hand menu.
- - Drag an audio file from the desktop to the open library window.
- - Right click on the listing for the new file and choose:
"Convert Selection to MP3"
- - When the file is done, right-click on the newly created file and choose
"Show Song File"
Wimpy Audio Encoder
simple program to convert WAV and AIF files to properly formatted MP3 files for Flash. Click here to download. Wimpy and free.
Can do some advanced encoding... plus you get a greate player!
A small, right-click type
utility that makes encoding an mp3 a snap. Good for one-off conversions.
We recommend DBpowerAMP:
When using DBpowerAMP we recommend these settings:
One of the most stable and comprehensive multi format audio file converters available and is very easy to use.
A nice CD to MP3 ripper / MP3 file encoder / batch MP3 encoder for Mac OS X. Click here to see the project directory. You'll want to download the file named LameBrain.dmg.gz.
Files don't stream (they download fully before playing)
Files take too long to start playing
Try re-creating the MP3 file using differnt software. Some MP3 files may be encoded imporperly, resulting in the file not playing back until the entire file has loaded.
Songs play "too fast" or "too slow"
The issue with the audio sounding distorted (aka the "chipmunk" effect)
is due to the way that the mp3 was originally encoded. Adobe
Flash can handle most standard mp3 encodings with any bitrate.
However, some mp3 encoders use "non-standard" encoding
techniques that Adobe Flash can not handle... Namely the MP3-PRO encoding scheme.
When a file is encoded with the MP3-PRO setting the file seems to
play too fast and makes the audio sound as if it were made by
chipmunks. The solutions is to re-encode your mp3's with a standard MP3
codecs. Wimpy will support CBR and VBR encoding -- at any bitrate.
I recommend using dbPowerAmp to encode MP3 files. It's a small, "right-mouse-click" type utility
that makes encoding an MP3 a snap. Using the standard settings in
dbPowerAmp works great with Wimpy.
James Roy has discovered
encoding MP3s at 96kbps using iTunes. An MP3 encoded at 128kbps
seemed ok, but anything else (even encoding the files first
at one bit rate, and then another) gave me either a faster
or slower playing speed. I was able to solve my problem by going into iTunes
prefs, choosing 'custom' for the MP3 encoding, and then choosing
44.1kHz for the sample rate instead of 'auto'. Apparently
when iTunes uses an auto bit rate, the Flash player is unable
to adjust its playing speeds to accommodate the optimized MP3
James Koenig discovered the following:
Flash goes all chipmunk on a LAME encode at 40kbps
mono, but works at 32kbps.
at Jukebox Alive notes:
For low bitrates (less than 32) I have the option of resampling
Of those, flash seems to only play nice with 11.025 or 22.05,
it was defaulting to 24
John Henry Mostyn notes:
...A slightly more robust answer to the resampling issue
users of Lame mp3 encoders, an additional call to -- resample
will force the sample rate flash seems to need for compact
"Choppy" audio playback
In general, dialup connections have trouble with streaming audio and streaming video. People who have modem connections usually understand the limitations on what they can actually do in this area :)
To correct the "stuttering" effect, try increasing the "buffer audio" option to a higher value. Something like 10-20 may work better. Or, you may want to consider reducing the overall size of your files by increasing the amount of compression you are suing to generate your MP3 files.
Initial MP3 playback for modem users may be "choppy." Once the entire file is downloaded, however, the file will play normally. In the current version there is no option to force the entire file to download before starting playback. This is something that will be incorporated into future versions. (Upgrading to newer versions of Wimpy is free.)
As for streaming videos, it all depends on how large the videos are in file size. For example, if you put a really large, high-quality photograph on the web, then modem users will have to wait longer to see it because the actual file size is large. Of course, people with faster internet connections will be able to download the file faster. The same thing applies with video, the larger the file, the longer it will take for modem users to download the file. The best thing to do is to test different compression sizes vs. quality.
There are two things you can do to minimize "choppiness" :
1) Use the Customizer Tool and increase the "Buffer Audio" to a higher value. Something like 10 or 30 seconds will help. For dial-up connections, you may even want to try a higher number such as 45 or 60.
2) Install Wimpy into two separate folders on your web site. One for "Broadband" users, which contains higher quality (and therefore larger files) and the other for "Dial-up" users, which contains lower quality, and therefore smaller files). Then present the user with a choice prior to launching Wimpy.
The following chart can be userd as a guide when creating MP3 files so that you can encode / compress your MP3 files for a "target" internet connection speed.
|Type of connection:
|Dial up modem
Audio artifacts will be present. Good for talk radio.
56k modems shouldn't have too much trouble with these files. 28.8k modems will probably choke.
|DSL / ISDN
Wider range of tone, little to no audio artifacts.
56k modems may experience some choppiness, but not much.
Decent sound, not quite as good as a CD, but close enough.
56k modems choke. DSL / ISDN may experience some skipping from time to time, but not much.
128 to 160
Bit Rates can range anywhere from 8 to 320. Higher numbers will result in higher quality, but larger file size, which means it will take longer to download and may stutter (audio stops and starts intermittently). Lower number cause the audio to sound it's coming through a tin can.
Don't use MP3pro, as these files may not be recognized by the myriad of players out there. One way to think about the word "codec" is "Code and Decode", which bascially means the method used to encode and decode the audio data. The same codec is used to create the file, as well as play it back.
Varible Bit Rate. Constant Bit Rate (CBR) is OK, but you can get higher quality and lower files sizes with VBR. With VBR, the bitrate setting varies depending on the kind of audio data that is being encoded. Most music has high points and low points, loud spots and soft spots. So during the low and soft points, the bitrate can drop down to a low number without affecting quality, while during the intense points, the highest bitrate is used to increae the quality. VBR allows for an overall higher quality track at smaller files sizes.
Usually there are two settings for this: Lossy and Lossless. Lossless means that the audio will remain identical to the source, which means that the file size will be enourmous. Lossy simply means that the data will be compressed, which means something's gotta give (quality) in order to make the file smaller.
2 channels means "stereo", 1 channel means "mono". There are three "stereo modes":
3. Dual Channel
Joint Stereo sounds better than regular Stereo. Dual Channel means that two files will be created, one for the left channel and the other for the right. Dual Channel files are only used for high-end audio mixing people.
You can use "mono" to really reduce the file size. (Using mono instead of stereo will cut your file sizes in half). Most computer speakers are cheap, and crammed into weird corners, so mono may work just fine.
22000 khz (kilo-hertz) may also sound just fine. The higher the Sample Rate, the better the quality, but will also cause the file sizes to be higher. Sample Rate refers to the number of slices each sample is broken down to. So if you take one second worth of music, it will consist of 44000 individual samples. Think of it like a paint brush, more hairs makes the stroke more smooth.
Ensure that the frequency rate is a multiple of 11,025. Flash is known to have "issues" (may sound distorted, like chipmonks or like a monster) when the MP3 uses a frequency that is NOT a multiple of 11,025 kbps.
- 11,025 kbps
- 22,050 kbps
- 44,100 kbps
- 8 kbps
- 16 kbps
- 22 kbps
- 48 kbps
Do not use 48000 or higher. These super high quality Sample Rates are very processor intensive and may not work properly in all players.
Tags are text data embedded into the file and contain information about the artist, title, album, etc. There are two versions of ID3 tags. Version 1 (ID3v1) and Version 2 (ID3v2). The difference being that in ID3v1 the data is located at the beginning of the file. In ID3v2 the data is located at the end of the file. ID3v2 also provides for the ability to include much more information. In ID3v1, space is limited.
Higher bit floats, such as 24, 32 and 48 are very processor intensive and may not work properly in all players.