1. No, that's not how processing works! You will NOT get either higher precision or less quality loss. Precision is dictated by the plugin's internal bit depth which is either 32 or 64 bit float. So whether you feed a plugin with 16 bits of data or 16 bits plus 8 additional zeros, the precision and result are identical. Increasing the sample rate is likewise pointless, there's no data above the Nyquist point to process and you don't magically get from upsampling. The only time a higher sampling rate could make a difference is in the case of some non-linear processes, in which case any decent plugin will internally upsample anyway. Actually, upsampling could (theoretically) make matters worse, some plugins operate at a single internal sample rate. Say for example a plugin operates at 96kHz, you upsample to say 176kHz, the plugin down samples to 96, performs it's algorithm and when complete upsamples the result back to 176kHz again, to match the incoming sample rate. What have you gained by upsampling in the first place? You'd have been better if you'd left it at 44.1kHz! You've fallen into the old audiophile trap of confusing the bit depth of the audio file with the bit depth of the processing environment! The bit depth (and precision) has nothing to do with the bit depth of the file but the bit depth of the internal processing of the plugins and the bit depth of the data connections between those plugins.
2. Dither should be the last step here, not the first! If you are changing the volume in your processing environment, which is presumably a 32 or 64bit environment, then the result of that volume change is obviously a 32 or 64bit word length, which you are then truncating to 16 or 24bit because you applied dither before the volume change. However, if you're outputting 24bit to your DAC there is no point in dithering, as truncation artefacts would be way below audibility, even at 16bit truncation artefacts would probably be inaudible.
G
