I recently noticed that some of my songs for my website are very low, so I decided to amplify them a bit to make them louder. I choose to use Audacity cause Goldwave is a pain to use. I use to use ACE HIGH MP3 WMA OGG WAV converter for my conversions to 24 bits; 44100 Hz; Mono. The quality isn't bad at all, and I think it's very good just for a converter. This is what the quality usually is:
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Recently something happened to my ACE HIGH converter and is no longer working, so I want to use Audacity for all my conversion/editing needs now. When I tried to convert to the same settings as above, the quality was HORRIBLE. I didn't like it one bit. This is how it sounds:
Link is GONE
I split the original file from stero into 2 separate audios. Then made both audios into mono. Changed the preferences to 24 bits for mp3. If you need any more into I can provide it. I just want the music to sound GOOD like the one I made from ACE HIGH. What should I do?
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To be honest, I thought the quality of the original was pretty poor, but that's mp3 for you.
Tye second file just sounds like it has had the volume increased too much, and been recompressed.
If the original is your normal volume, I don't see a need to make it loader.Read my blog here.
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I'm by no means an audio expert so I don't know all the proper terminology and I can't tell by listening what's wrong, but I believe, by default, that Audicity does clipping for a straight amplification. Meaning if you amplify it to above 0db, it will clip the wave to keep it at 0db. I think this can be over-ridden, but I'm not sure how.
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Guessing, 24 bit 44.1 kHz is a bit unusual for a recording format. Were or are the originals something else, & if so, did your former method apply dithering? This can make a pretty big difference, so might want to try turning it on in Audacity.
Another guess would be the encoder itself -- believe Audacity uses whatever you have on your system rather then something dedicated -- just checked and they have a section on mp3s in help files. If you're preferred convertor stopped working, could be that the encoder you were using got overwritten or Windows default changed etc... Might want to check that end of things out.
For amping up the audio, use normalize & you won't (or shouldn't) go over the limit. Any levels that exceed max (0 db) are just flat-lined there otherwise. -
OK, a little clearification. Sorry that I didn't mention any of this. The first file is already compressed to 24bit; 44100 Hz; Mono using ACE HIGH (doesn't work on my computer anymore). The second file is also the same as the above, but using Audacity which made it sound horrible. I also didn't amplify the settings for the second one, but if I had the quality of the first one, I'll like to ampify that since it's already in good quality for websites (xanga, blogspot, etc)
I just put the file I just did with Audacity, the file came out to be 24bit; 16000 Hz; Mono. How can I change it to be 44100?
@Mikiem: The original is 320bit; 44100 Hz; Stereo. While Using ACE HIGH, it really didn't have "options" per say to do any of that. It was a VERY simple converter, and just did what you wanted. It wasn't really an "editor". Well the dither that they have at default for Audacitu is:
Real-Time Dither: NONE
High Quality Dither: Triangle
Yea, I've been reading a bit on the help files, and I can't come up with anything. I have LAME MP3 ENCODER. Does Spectrogram have to do with any of this? My Spectrogram is:
FFT Size: 256
Maximum Frequency: 22050 (told me to do that in the help file, and I HAVE changed it to 44100, but still got 16000 Hz) -
Your specs are really confusing. You are mixing wave bitrate with killobit rate of MP3's.
You say you encoded at 24 bitrate. There is no need for this, 16bits is what CD's are so raising this isn't going to do anything for you. Then in another sentence you say the bitrate is 320. Impossible, there isn't hardware or software avialable to run a datastream of 320 bits. You are talking about 320kbps of the MP3 file.
You also shouldn't be using a wave editor such as Audacity. Wave editor have to decode the MP3 data for editing then re-ecode it to save. This causes further degradation of the quality.
You should really start over with the original wave files or MP3 files and use a dedicated MP3 editor that doesn't need to decode to amplify the signal.
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