I am on Windows 10.
I am using Audacity version 2.1.2, .exe installer version.
I was tinkering with a specific .wav file from a game of mine, and noticed that it has a really small size for it’s quality. The bitrate is only 80kbs, yet it sounds very high quality. I’ll upload it since it’s less than 2MB.
Can someone tell me exactly how this bitrate was obtained and how it sounds so good?
So someone forced a name change from music_omega_danger.mp3 to music_omega_danger.wav?
Or the real name is music_omega_danger.wav.mp3 and Windows is hiding the real extension?
Please note 60 is the lowest quality stereo MP3 will go without audible distortion. Mono is 30. Even with 80, you will not be able to do production, effects or filtering and make a new MP3. The new file will have serious distortion. That’s why it’s a poor idea to do production in MP3. It’s a time bomb.
Nevermind, I figured out how this was so small in size. All good. However, I have a slightly new problem.
I am attempting to convert soundtracks for a game from 2004, and this 2004 game… uses some slightly obsolete formats that Audacity doesn’t seem to know about.
MP3 files with a RIFF WAVE compression and Fraunhofer codec. I’ve been tinkering with this for the WHOLE day, but all I can find is stuff saying “use Windows Sound Recorder to convert it”, but obviously windows 10 doesn’t have that. Nothing seems to work, but I imagine audacity can somehow do it.
Where did you find that information? It’s not quite right. “RIFF WAVE” is not a compression format, it’s a container format.
The Fraunhofer codec is MP3.
If the format is MP3 with a RIFF header, then I’d have expected Audacity to be able to open it.
Neither. The file is RIFF (WAVE) format, which is a container format, and the data is encoded as MP3. The file has a RIFF header.
For music, that is generally the case, but whether the distortion is noticeable depends on the material. If you listen to the file that mrliberty90 posted, it doesn’t sound too bad, even though it’s only 80 kbps stereo. Part of the reason for the (subjectively good) sound quality is that the sample rate is only 22050 Hz, so it is band limited to 11025 Hz. The lack of audio frequencies above 11 kHz makes it ‘easier’ to encode. Also, due to the fast tempo and deliberately “synthetic” (electronic) timbre, compression artefacts are less noticeable (they blend in with the timbre of the music).
However, 80 kbps is about as far as you can go for this music without it sounding bad, which may be a problem if mrliberty90 wants the final product of his editing in a compressed format. Because it’s already very compressed, compressing it again, even with fairly high settings, is quite likely to cause noticeable deterioration in the sound quality - it’s like making photocopies of photocopies; the quality always goes down.