This guide was written with the intent of use on a Linux distro, if you have questions, contact me.
Step #1:
Install or compile the required dependencies and packages.
sudo apt-get -y install ffmpeg
sudo apt-get -y install lame
sudo apt-get update
Step #2:
Upload your audio file to your Linux or Windows server.
For example, my audio file is going to be track.mp3
Here is some metadata and stream information for my specified file:
Bitrate:
96 kbps
Sample Rate:44.1kHz
Channels:1 (Mono)
Audio Codec:MPEG-1 Audio Layer III
Compression Mode:Lossy
Step #3:
Now that you have your audio file and dependencies/packages installed, we can begin the encoding process.
Just for first-timer purposes, check if you have the required packages installed by running these two commands.
ffmpeg
lame
After you have ensured that these two dependencies are installed, the next step is to encode your audio file into a lossless format.
Run this command to begin the encoding process of your audio file into a 24-Bit/48kHz FLAC file. (Replace track.mp3 with the name and format of your audio file. keep in mind that certain formats have limited transcoding properties).
ffmpeg -i track.mp3 -sample_fmt s32 -codec:a flac -ar 48k -b:a 1411k -ac 2 track.flac
The output should look like something around this subject.
[flac @ 0x5616606c4880] encoding as 24 bits-per-sample
Output #0, flac, to 'track.flac':
Metadata:
encoder : Lavf58.45.100
Stream #0:0: Audio: flac, 48000 Hz, stereo, s32 (24 bit), 1411 kb/s
Metadata:
encoder : Lavc58.91.100 flac
size= 34551kB time=00:03:10.98 bitrate=1482.0kbits/s speed= 158x
video:0kB audio:34543kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.023431%
If FFmpeg does not print a similar output, you most likely forgot or did a step incorrectly, or the format of your file could not be transcoded into a high quality, lossless, FLAC format. If done correctly, just go right ahead and download your new, transcoded audio file and enjoy the crisp and immersive experience with CD quality.
Unfortunately, lossless audio files can be quite large so if you would like to combat that, we will use the LAME Encoder to transcode your audio file into a high qualtiy MP3 format instead.
To begin the encoding process, paste this command into your console. (Remember to change track.mp3 to the name of your audio file.)
lame -h --abr 320k -b 320k -B 320k -q 0 --resample 48 track.mp3 highqualitytrack.mp3
The output should look something like this.
LAME 3.100 64bits (http://lame.sf.net)
Using polyphase lowpass filter, transition band: 20323 Hz - 20903 Hz
Encoding track.mp3 to highqualitytrack.mp3
Encoding as 48 kHz j-stereo MPEG-1 Layer III (4.8x) average 320 kbps qval=0
Useful Information to Know
It would probably be wise of you to know what each flag/switch in these encoders do or mean, so here, I'm going to list out the most important ones.
-ar <number>k - Sample Rate (48kHz, 96kHz, 192kHz)
-b:a <number>k - Audio Bitrate (320kbps, 1411kbps, 9126kbps)
-i - Name of Input File (Comes before ALL encoding options)