LTC is straightforward, so If nothing else, you could just scan the audio stream for LTC data, as documented on wikipedia. Each 80 bit frame ends with 0011 1111 1111 1101, just scan for that byte sequence to synchronize, then cast the buffer data starting after that sync sequence to be an array of 80 bit struct timecode_t elements. If your buffer is sized as a multiple of 80 your calculations will be easier (but you do need to test for sync lossage, because soundcards lose bits with overruns).
The hard part is that if I am not mistaken, the time code "bits" are not the same of the bits of the sampled audio stream, so you would have to implement logic to detect the bit sequence. This can just be a for loop checking for the proper signal changes and appending bits to the buffer as appropriate (and then calling the function to interpret the buffer when it is full).