FLAC

FLAC

Is it storing a literal analog signal in a digital way?

Or...

Is FLAC just an approximation of analog signal by having an extremely high sampling rate that simulates a sine wave instead of digital flat waves?

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the latter
you can't store analog signal in a digital way dummy.

>Free Lossless Audio Codec

fucking idiot
>>>/google/

Of course you can, just store the formula instead of the sets of values, then it can be evaluated to get the analog value with some approximation error margin. Just finding it would be a fuckton of computing.

And yea flac is #2 since before signal gets to en/decoder it already got sampled.

>Is FLAC just an approximation of analog signal
FLAC isn't an approximation of any signal, it's a compression algorithm for a PCM bitstream.

Is it possible to do #1 before #2?

Doing #1 is practically impossible for audio.

chiru.no:8081/stream.flac

FLAC is digital flat waves. You need something like AAC's quantization to smooth the flatness and give you proper sounding audio.

>lossy compression

>what is ALAC
Leave.

not AAC?

Yes you can! You can have a digital file that describes where all the atoms are in a vinyl CD.

You can also have a digital file that perfectly describes an analog waveform.

Do people realize analog in terms of music is technically incorrect? You aren't actually listening to a "wave", your ears are being bombarded by discrete air molecules. You turn up your sampling rate to the point where you aren't missing information and boom you are done.

well, I think they are talking about the waveform of the signal before it hits the speaker.

But he literally said nothing about sound. The volume could be on mute for the purpose of this question.

flac has nothing to do with analog audio, it takes digital PCM audio, and makes it smaller in a lossless manner, by exploiting redundancies in the data

PCM audio is the most common type of raw, digital audio, it's used basically everywhere digital audio is used, for example, CD's store PCM audio directly, and compressed formats such as flac or mp3 are decoded into PCM, which is what goes into a DAC to be converted into an analog signal for speakers

an analog signal is converted into PCM by means of sampling, that is, you record a voltage level with a certain precision (bitdepth) and at a certain frequency (number of samples recorded per second). that's all PCM data is, a set of samples

also, as for the "certain" precision and frequency, these are chosen based on the requirements and limitations of the situation
the bitdepth/precision affects only the noise floor of the recorded signal, and is typically chosen in a multiple of 8bits, for convenient storage and manipulation purposes, 16bit is often picked as it has a hardly noticeable noise floor, higher is often used for masters as it allows for more manipulation headroom, 8bit was used when computers and storage devices were more limited
as for the frequency, you want double the samples of the highest analog frequency you want to capture, that is, a 40Khz sample rate can perfectly capture any frequency up to 20KHz (see en.wikipedia.org/wiki/Nyquist_rate)

Make signal spin forever in a coiled cable ok

>you can't store analog signal in a digital way dummy.
that... doesn't even make sense
you should read up what digital means