Most such audio codecs are based to some degree on variants of fourier transforms, so this modification is done by dropping or reducing resolution of parts of the output.
CD audio is perfectly lossless - it encodes the signal that you put in by measuring a voltage 44100 times per second and recording that exactly. When you play it back you get exactly the same signal back out. The only problem is, this takes up a lot of space, roughly 10MB per minute for stereo audio.
MP3 audio is lossy in that rather than storing the exact values of a waveform, it stores a description of how short segments of the waveform change. The higher the bitrate, the better the description, and the more detailed the reproduction. A low bitrate MP3 is like trying to redraw the original waveform from a vague description with a paint roller, a high bitrate MP3 is like trying to draw it with a mapping pen from a really detailed description.
FLAC audio is lossless because it takes the precise values of the audio, and uses a technique similar to zip files to find similar-looking blocks of data. Think in terms of having a one-second silence recorded as "Zero, then 44099 more of 'em" rather than "zero zero zero zero zero..." and so on 44100 times.
Not at all! CD audio amplitude is quantized to 16 bits, and temporally sampled at 44100 Hz as you say. Certainly very high quality, but there's absolutely a loss (that nobody can really hear).
Otherwise image formats could not be lossless for the same reasons (at the very least very few of them store the polarity of the light)
I know that when I was about 10yo, I could hear a tiny suggestion of something audible when I tested my hearing at around 23-24kHz. I guess that if you had some really loud content around these frequencies and had equipment that could reproduce it perfectly, it would be not impossible for it to influence my listening experience those many years ago :)
Testing with https://www.audiocheck.net/blindtests_frequency.php, my hearing limit in the white noise case for example is at least 1 to 2 kHz lower than my hearing limit for a single frequency sinus tone.
It's got way more audio bandwidth than most of the analogue masters that everyone raves about.
No, there's nothing magically audible happening with phase shifts near the steep cutoff of the antialiasing filter that isn't happening with the gentle rolloff of tape, either. Not that you could hear anyway, and even though my hearing is better than most 48-year-old industrial music enthusiasts, not that I could hear either.
For cats, even your very best equipment with perfect reproduction sounds like AM radio because their hearing tops out at 80kHz.
https://en.wikipedia.org/wiki/Lossless_compression
https://en.wikipedia.org/wiki/Audio_Lossless_Coding
https://en.wikipedia.org/wiki/Category:Lossless_compression_...
This is very common terminology.
Thank you
Just like any general purpose compression: everything - flac could be used just like zip/zlib/gzip as a general purpose compressor it just wouldn’t compress as well on most data that isn’t audio.
> dropped when "LOSS" occurs?
Lossy compression generally employs some type of perceptual coding where data is reorganized such that the signal data is sorted or localized according to its perceptual importance. This does partly involve removing or reducing the density of signal in the higher frequencies but it also exploits things such as masking both in time - inability to perceive signals occurring close in time of similar frequency. And frequency masking - our inability to hear quieter signals that are close in frequency to a louder one.
You cannot do the same with MP3, AAC, OGG, Opus or any other lossy codec.
The exact frequency/amplitude relationships where masking effects come into play were studied by the telecoms early on (meaning in the 1950s-1960s era), and are still a key part of most lossy encoding models these days.
See https://en.wikipedia.org/wiki/Critical_band and https://en.wikipedia.org/wiki/Psychoacoustics as an introduction to the principles.
Cutting down the bandwidth (lowering the sample rate) wont sound as good as an actual lossy codec - otherwise we could just resample.