[linux-audio-dev] question about libalsaplayer's functionality (and its possible incorporation in my app)

Hi all, I've tried contacting the maintainer of the project (Andy), but he never bothered to reply, so now I am taking this question to all of you out there that might have had exposure to this interesting library. Does anyone know what is the current status of the whole project anyhow? Any help on this matter is greatly appreciated! When answering any of the given questions, please include the question in your reply. Thank you very much! So here's the excerpt from the letter: I've recently decided to incorporate libalsaplayer-like functionality the upcoming version of my app RTMix. However, with having a particular feature in mind I am wondering whether libalsaplayer provides that and if not, whether you'd be willing to add it (or let me add it, although that might get messy since I am not too familiar with the inner workings of the lib). So, here it goes: Apart from all the wonderful features libalsaplayer offers, I am looking for some additional ones: I would like to be able to indefinitely loop specific "ranges" of a particular soundfile (i.e. 2.2secs-4.12secs or whatever). 1. Is it possible to do this and have music continually loop even if one changes the direction of playing so that when the alsaplayer runs out of the looping material in each direction that it just jumps back onto the other end of the loop and contiues on (kind of like a ring-buffer)? 2. Is it also possible to do this kind of looping on the whole soundfile (the gui version of alsaplayer always stops if I let it play 'til the end or the beginning, depending in which direction I am going). 3. Is it possible to define loop points by addressing a particular sample number rather than giving time in seconds? 4. How stable is alsaplayer when looping really small chunks of sound (i.e. like 5 sample loop)? 5. Is alsaplayer capable of ramping such loop points by attenuating let's say ending 20 samples (it would be cool if this number could be user-selectable) and then ramping up the beginning 20 samples in order to alleviate the "pop" that happens when looping a sound where waveforms at the beginning and the end do not align? 6. If the feature in question 5 does not exist is there a way to control output level of a player on a per-sample basis via callback so that one can implement that outside of the lib? 7. If neither 5 or 6 are possible, would you be willing to implement such functionality (i.e. a toggle_ramp( bool ); and set_ramp_length( int ); callbacks or something similar. I would greatly appreciate your feedback on this matter as that will greatly assist me in determining how to go about implementing such functionality in my app. Thank you very much! Looking forward to hearing from you. Sincerely, Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico P.S. How do you implement reading of a sound faster and slower than it sounds in such a gradual fashion? Do you adjust the sampling rate of the DSP and if so does that affect other streams coming from the libalsaplayer, or is this something that is sound-specific (and if so, how)?