I bought an mp3 player that just isn;t loud enough. Is there a way to boost the level in the file itself without rerecording it?
I believe you can do that with Goldwave. It’s called normalizing the volume. It’s been a long time since I needed it done, but I’m pretty sure Goldwave is what I used. Search for it on the net.
There is a plugin for winamp that normalizes on the fly, but I guess that’s not what you’re looking for.
What you’ll want to do is normalize the mp3, that is make raise the sound level of the mp3 to the max without clipping the wave.
Found a free program that should do it for you.
Joe Cool-
I wouldn’t mind having that program. In fact, I’ve been looking for something like that for awhile. Nothing I’ve been able to find so far fits the bill.
Got a link?
Sure thing:
Try MP3Trim:
http://www.logiccell.com/~mp3trim/
Thanks for the links everyone. I see that the first two seem to normalize a group of mp3’s to the same volume. Can i use the same app’s to make the volume higher on all of them?
Im gonna hijack my own thread. I see these guys tout their products as causing no quality loss. Why would there be if digital music is just a bunch of ones and zeroes anyway?
Cnote- go with the mp3gain that sig recommended it does everything you need and nothing more. You can also add whole folder’s to normalize and add gain too. I tried all three and this is the best.
The digital waveform is made up of 16-bit samples, so each sample can have a value from -32,768 to +32,767.
If you were to amplify the volume by simply multiplying each sample by 2, a sample with an original value of 20,000 would have a new value of 40,000. But that won’t fit in 16 bits, so it would have to be clipped to 32,767 or wrap around to -25,535. Either way, it would distort the waveform.
Normalizing without losing quality (that is, without distorting the waveform) means you find the ideal factor that you can multiply each sample by, without making any of the samples go outside the allowed range.
Mr2001,
learn something new everyday. Thanks
MP3 is a lossy compression format, meaning that every time you encode a file in MP3 format, some data is destroyed to save space. This is fine if you only encode a file once, but if you try to encode it a second or third time, sound quality will suffer.
A file should only be saved as MP3 once. After that, use a program like MP3Trim that allows you to directly modify the MP3 data so you don’t have to re-encode.
“…I see these guys tout their products as causing no quality loss. Why would there be if digital music is just a bunch of ones and zeroes anyway?..”
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- They’re lying: there is quality loss, but most of it occurred before their program:
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- With analog equipment, sound is a sinusoidal wave, and you can stop at any moment and measure the level of the sound, and increase or decrease it by a factor, and basically get a higher or lower-level exact copy of the exact same sound. This is why most commercial CD’s are still mastered on analog tape.
- For what you want to do, most of the quality was lost when the sound was converted to a digital signal, but even more distortion is introduced when you attempt to amplify very low-level/inaudible (low bit-count) sounds up to where they are audible. The processing simply multiplies the digital recording by some factor (for the entire recording) but for smaller-variations in the sound wave, they get converted from sinusoidal waves to square waves, and there’s no way for the PC to re-convert them back to their original shapes from the limited data. In the original recording level you may not have been able to hear these sine waves, but in the normalized soundfile, you may hear the amplified square waves.
- To help avoid this problem, enthusiasts prefer to record in 24-bit digital, reducing down to 16-bits only in the last processing step before burning to CD. - DougC
Joe Cool-
Thank ya sir, I appreciate it.