A CD-Digital Audio uses 44.1 kHz 16 Bit Stereo, making 44100*16*2 = 1411,2 kBit/s.
Fairly the algorithm of an AudioCD compresses the overall data rate to about 1223 kBit/s. I don't know how.
If you don't wanna use .wav, you can use lossless codecs such as AppleLossless (ALAC), WMA Lossless or FLAC to reduce this data rate to approximately 750 to 1150 kBit/s (variable data rate). The same or very similar procedure is used with .rar oder .7z archiving.
This does a little problem to the audio world: While common lossy audio compressors like MP3, AAC or OPUS do acceptable quality at 64-256 kBit/, the artefact-free lossless world never realy gets close to that.
UNLESS you use a little trick. Since lossy codecs really fuck up the audio quality, a little reduction in bit depth is acceptable. FLAC compresses way better when using less than 16 Bit audio depth. However, you can not simply store a 16 Bt .wav file with 12 or 10 bit - only the noisy 8 Bits are available. But you can turn down the volume. Every 6 dB volume halfing means 1 Bit less in data dynamics. Turn down the volume of your audio by 36 dB and you effectifely achive a 10 Bit storing. FLAC will eat this!
Also you can turn down the sample rate from 44100 Hz to 36000 Hz, if possible. Or go lower to 32000 Hz.
This way flac stores your quiet PCMs with around 350 kBit/s - achieving better quality than the famous ADPCM code and having clearer sound then MP3 and AAC!
Only problem is you have to raise volume by those 36 dB before playing. Runtime? I don't know how this would be possible. Does anybody know how to raise the volume of audio files above 100% while running an app? You could simply calculate the RAW data ( you must multiply the PCM data with the value 2^6 = 64) but this requires an RAW player or a .wav header lib.
What do you think? This is the bridge between CD audio and lossy compression. Data reduced FLAC. Great idea, i think.
