I am currently recording radio programs in a computer using a radio card. I actually need to capture various advertisements which are broadcasts within each program. The recorded programs are in WAV format.

Can you please recommend a software that I could use to locate selected advertisements in the recorded programs and save them in MP3 format. As you may have guessed I am not too IT-literate. Thanks in advance.

Excellent explanation. Thanks. --Reader in toronto

I am converting some of my records. Yes I said records to my computer to eventually save them to a CD. I know the 3 formats are wav. oog. and mp3. Which format should I use to burn to CD's?

In response to anon28062, it is not a true statement to say you cannot convert an mp3 to wav and it is not pointless if you are dealing with something that can only handle a wav format. As expected quality will be the same as the mp3, and re-sampling the mp3 with a high quality wav setting will not yield better results as indicated you cannot put back data that was removed, but you can change formats which I think is what anon27459 wanted to know.

To 27459, no you cannot convert .mp3 to .wav ... or rather it's pointless as it remains an mp3 file essentially, with a .wav extension.

Waves are lossless files... they contain more data than MP3s. So you can take a wave and convert it to mp3, and in the process it loses some data. But you can't take a .mp3 and convert it to .wav, cause it would have to put data BACK that is not there.

Kinda like, you can take a tree branch and whittle it down into a small stick, but you can't whittle the stick UP into a branch. ;)

Do you need a license to make new software that decodes (new codecs) for WAV files? Is it covered by a patent? I know you do for MP3s.. I'd appreciate any help! Thanks!

is there anyway to convert mp3 to wav?

For those who want to know more, I'll add a few lines about what .wav and .mp3 files really are. MP3 is a codec (codec = coder+decoder). It is named MP3 because it's part 3 of the MPEG1 suite of codecs. It is specifically designed for audio files.

Next, a codec's job is to code(encode) a raw file with data in order to compress it but retain quality and then to play it back the codec must decode the result.

Now, the 'encoder' part of the codec has (very broadly speaking) two tasks: first, 'Quantize' all the input signals. Second, 'Encode' this result. In other words, take the result of the Quantization phase and convert it to 1s and 0s to compress it.

Quantizer: Think of the Quantizer's job as smearing an image. The result is an image that's less sharp, more blurry but still has the same size. Mathematically, let's pretend the audio file inputs can take on values from 1-100 (RGB images take on values from 1-255 for example). Next, the Quantizer maps all inputs from 1-10 to the an output of 5. Then, it maps all inputs from 11-20 to 15. The result is much less detailed and smoothed out. This is where the error or quality loss occurs! Why do this then? It's all a trade off between quality loss and file size reduction. Quantization is lossy, but it's not compression!

Eg. inputs: 1 5 10 20 50 100

output: 5 5 5 15 45 95

We still have the same number of symbols, so we lost information but we did not compress the file!

Encoder: The 'Encoders' job is to perform lossless compression. It takes in input symbols and converts them to 1s and 0s. How does it do it? This is what Information Theory is really all about! In general, out of the input symbols, 5 5 5 15 45 95, 5 repeats the most. So, the encoder will assign it the symbol 0! In other words, assign the most frequently occurring symbols as few bits as possible! Putting it another way, the input symbols with the least information get assigned the fewest bits. There are many encoding methods out there (Huffman Encoding, Arithmetic Encoding). All work differently and produce different results but the bottom line is *encoding is a lossless compression* process.

Finally, you should be able to guess that .WAV files are much bigger and have higher quality because they are not quantized (or not significantly). Because of this, the quality does not suffer. Also, there are fewer repeating symbols at the input of the encoder and it is more difficult to encode/compress the file (highest compression is achieved when symbols repeat more often).

Using the MP3 codec however, the input files are quantized much more, and so a quality loss occurred. However, because the data is smeared, more input symbols repeat and the encoder is able to compress the data much more. Hope this helped!

Thanks for explaining WAV files. Now then, why can't I open WAV files (in Windows Media Player 11) from a certain source although I can open them in Gold Wave (audio editor)? After opening them in Gold Wave (or Twisted Wave on my iMac), I can do a simple save, without changing the format, and they play just fine. Any ideas?

Hey just wanted to say thank you for posting something so straight forward and in plain english. I'm 22 but even I have a hard time trying to understand a lot of this computer lingo and things like that. I'm a musician and I am trying to get more into the new equipment they have, but a lot of it i dont understand to its full capacity. Reading posts like this thou should help me greatly i think. Thanks again!