haven’t a clue, i only mix couldn’t produce if my life depended on it lol

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Sample rate and bitrate are two completely different things.

The sample rate is the number of samples per second in a digital audio stream (we’re talking uncompressed PCM audio, like WAV). E.g if you record a guitar into your computer at 48kHz, the analogue-to-digital converter in your soundcard takes a “snapshot” of the analogue audio signal from the microphone 48000 times per second. The sample rate determines the highest possible audio frequency that can be recorded in the system – it is always half the sample rate (in the case of 48kHz, the highest audio frequency that can be represented is 24kHz). This is the “Nyquist” frequency you mention. In practice, everything above 24kHz would need to be filtered off prior to sampling in order to eliminate some very weird sounding alias components.

The bit depth (again different from bitrate), is the accuracy to which each sample represents the analogue signal at that moment in time (this is really easy to show on a diagram, but hard to put into words). With a bit depth of 8 bits, the value of each sample is 8 binary bits long, e.g. 10010110 or 00010101 etc (if you know about binary this means that the sample could have 2^8, or 256 possible values – not particularly accurate. With a bit depth of 16 bits (e.g. 1001101000110111) the sample could have 2^16 or 65536 possible values – allowing for much greater detail. In practice, recording at high bit depth gives you more dynamic range (or headroom, depending on how you look at it).

The bitrate of a digital audio stream is the number of bits per second. This is easily derived from the sample rate, bit depth and number of channels. E.g. for a 16-bit, 48kHz, 2-channel stereo audio file, the bit-rate = (48000 samples per second) of (16 bits each) for (two channels) = 48000 x 16 x 2 = 1536000, or 1536kbit/s. Note this is for uncompressed digital audio, and is considerably more than the 128kbit/s typical of MP3s.