Understand what audio compression is
A container format is a data format that “encapsulates” other encoded data. It often contains “meta information” about the encoded data, or has a way of storing 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 an Ogg file and inside there are these little segments that contain encoded data. Each block contains a stream of Vorbis-encoded data and nothing else. For example, a cube might have the name of an artist and the title of a song stamped on it.
So, back to technology:
If you already have lossy music like mp3 or ogg / vorbis, converting it to lossless format will only take up (a lot) of disk space and will NOT, at all, NOT improve the audio quality at all. You can’t create loyalty when it’s already lost. Unless you’re writing a Visual Basic GUI on some popular TV show called CSI, but that’s fantasy, not reality.
If you have music in other lossless formats and want to convert it to FLAC, you can.
Be careful when using 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 Layer III (lossy) data in a WAV file.
It is possible to lose data when converting from one lossless format to another if you reduce the precision of the data. For example, if you convert an unsigned 16-bit PCM data stream at 48000 Hz to 8-bit PCM data at 44100 Hz, you lose precision in two ways: samples are merged from 48000 to just 44100 at a time. second (leading to data loss), and the data needs to be scrambled to fit the information into just 8 bits instead of 16 per sample, which will drastically degrade the quality.
Every digital audio stream, even encoded with a compressed encoder (lossy or lossless), has the following sample format properties, which are important elements that describe the properties of the stream:
An example of bit width and depth, i.e. 8 bit, 16 bit, etc. Bit widths and depths are slightly different and there is also big endian / endian byte order (which does not affect quality) and signed or unsigned sign (which does not matter either) affects quality but does affect encoder / decoder operation with data). The key point to remember is “the more bits the better”. So 32-bit is better than 16-bit, etc.
Frequency, also known as the sample rate. The more the better, because more “samples” of sound are played per second. Imagine sliding your finger across a deck of cards and seeing the cards blur; this is essentially how digital sound occurs. Each sample is a map, and if you have more maps flying per second, the sound is softer. For example, you would really notice if you were flipping only 5 cards per second, but everything would be blurry if you were flipping 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, up to Planck units, but this is debatable 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.
Therefore, if you change from 16-bit to 32-bit sample format, you will not lose data. But if you go from 32 bit to 16 bit, you will lose data.
So the answer to your question about whether it makes sense to use FLAC depends on the original data: if you have 64-bit WAV files that were originally recorded in this 192,000 Hz (or 192 kHz) sample format, and you convert them to “format Standard 16-bit 44.1kHz FLAC, you’ll lose a 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 you can even increase the file size depending on whether you gain lossless compression or a smaller sample format.
The sample format will affect the amount of space the file takes up, so “bigger” bits and a “faster” sample rate will take up more space.
When it comes to practical considerations and human hearing, you won’t notice if you convert very high-quality originals to 16-bit FLAC at 44.1 kHz. But you won’t notice any improvement if you convert MP3 to FLAC either. As such, you need to evaluate what format your raw data is in before deciding what to do.
