I’ve a suggestion that could be easily implemented:
Currently in the audio recording dialog box you can set the latency compensation in a value of ms. This is ideal for re-syncing recorded audio at the same delay that your soundcard’s buffer is set to. But, in recent times, precise timing right down to the sample has become very important to me (and will do so for anyone else getting deeply into analogue recording), thus the need to sort this problem has arose:
For example I usually set my buffer at 256 samples for a 44.1khz clock. This roughly equates to 5.8 ms. The current problem I have with latency compensation is that integers of ms, with or without decimals, is still inaccurate (atm you can only set it to 6ms). Given that latency selection for cards is in samples, it would be ideal to have the latency compensation parameter is be in samples. Please
A current work around is to not use the latency compensation feature at all: instead manually cutting off the number of samples (e.g. 256) from the start of each recording manually. Thank god (Taktik?) that 1.9’s sample ruler can display samples.
NB, later on I will request a new feature of the audio-recording interface or master preferences called ‘ADC fixed group delay compensation amount’ - but I’ll have to do a well presented InDepth article first to clearly outline the need.