Specialist in real time low latency AUDIO processing (speech) C++ / ASIO: envelope follower, onset detection, volume equalisation, pitch shifting: URGENT

I want to hire someone specialising in low latency real time audio processing. The software I have is a custom made application written with C++ and using ASIO. It processes speech from a microphone real time and directs feedback to headphones (online monitoring or playback). Several things need to be to be improved or implemented: - quick onset detection (simple detection) function being able to trigger a playback file imperceptibly (under 10ms); the file is activated and being given as feedback when a person starts speaking through a microphone without being perceived as delayed (hence 10ms); the existing code may need some improvement to lower latency or a new code may be needed; - envelope follower needs to adjust a prerecorded playback to speaking (real time processing, low latency) in terms of duration and amplitude: the existing code may need some improvement or a new one may be needed; - possibly an existing real-time processing pitch shift code may require some improvement or an other low latency code needs to be installed (close to 10ms of latency) - volume levels for online monitoring and playback may need to be equalised, a 10dB gain or a level of 80dB -85dB in phones should be kept constant The offer is only for coders who know well how to do that and have experience with low latency real time audio processing. The hardware is RME Fireface UCX, microphone AKG, in-ear phones Etymotic, the system is Windows XP. I can show you how the software works using Teamviewer and Skype. Start: immediate. It is VERY URGENT. Please, contact me as quick as possible.