Understand the difference between encoding and container format.
A container format is a data format that “encapsulates” other encoded data. It often contains “meta information” about the encoded data, or a way to store several separate streams of encoded data, or something like that.
The encoding produced by the codec is the real essence of the data stream.
The most common example I can think of is the Ogg / Vorbis format. Ogg is the container format and Vorbis is the encoding. So you have a file in Ogg format and inside it are these little buckets that contain encoded data. Inside each repository is a stream of Vorbis encoded data and nothing else. For example, in the cube, you can include the name of the artist and the title of the song.
So, back to technology:
If you already have lossy music like mp3 or ogg / vorbis, converting it to lossless format will only consume (a lot) of disk space, and W NOT NOT NOT NOT will improve the overall sound quality. You cannot create loyalty once it is lost. Unless you are writing a Visual Basic GUI in some successful program called CSI, but this is fantasy, not reality.
If you have music in other lossless formats and want to convert them to FLAC, you can.
Be careful when dismissing the term “WAV”. Wav doesn’t have to be lossless; In fact, WAV is just a container for the various possible formats. In this sense, it is similar to AVI. You can have lossless WAV if it is just raw PCM data, but you can also embed MPEG-1 (lossy) layer 3 data into a WAV file.
When converting from one lossless format to another, you may lose data if you reduce the accuracy of the data. For example, if you convert an unsigned 16-bit 48000 Hz data stream to an 8-bit 44100 Hz PCM data stream, you lose precision in two ways: samples are combined from 48000 to 44100 for every second (whichever results in data loss), and the data needs to be scrambled to match only 8 bits of information rather than 16 per sample, which will drastically degrade the quality.
Every digital audio stream, even those encoded with a lossy or lossless compression encoder, has the following sample format properties, which are important elements that describe the properties of the stream:
Sample width and bit depth, i.e. 8-bit, 16-bit, etc. Bit width and bit depth are slightly different from each other, and there are also little-endian / big-endian (which does not affect quality) and signed or unsigned (which also does not affect quality, but does affect the way the encoder / decoder handles the data). The main thing to remember is that “more bits the better”. So 32-bit is better than 16-bit, etc.
Frequency, also known as the sample rate. More is better because you have more “samples” of sound played per second. Imagine sliding your finger across a deck of cards and seeing the cards blur – this is how digital audio works. Each sample is one card, and if you have more cards flying per second, the sound is smoother. For example, you would really notice if you were only flipping 5 cards per second, but everything would go blurry if you flip thousands of cards per second. So it’s even better, because it’s more natural and closer to reality, which is analogous and infinitely divisible (well, down to Planck units, but this is a moot point and off-topic).
Lossless simply means that if you use the same or better sample format in the output that you used in the input, you won’t lose any data.
So if you go from 16-bit to 32-bit format, you don’t lose data. But if you go from 32 bit to 16 bit, you lose data.
So the answer to your question about whether FLAC makes sense depends on the original data: if you have 64-bit WAV files that were originally recorded in this format, with 192,000 Hz (aka 192 kHz) and you convert them to “FLAC format 16-bit 44.1 kHz standard, you will lose TON of data. But if your WAV file is 8-bit with 22100 samples per second and you convert it to 16-bit FLAC with 44100 samples in second, you won’t lose data and even You can increase the file size depending on whether you gain lossless compression or a smaller sample format.
The sample format affects the amount of space the file takes up, so “bigger” bits and a “faster” sample rate take up more space.
