I need an audio processing expert (NOT a general developer) to build a fully adaptive audio pipeline. Goal: No matter the input (YouTube, Facebook, speech, music, noisy recordings), the output should always have a consistent sound: - Warm - Clear vocals (strong mid presence) - Clean and controlled dynamics - Pleasant listening experience Technical requirements: - MP3 output (~35–40 kbps VBR, LAME) - Mono - 44.1 kHz - FFmpeg / Linux environment Target metrics: - Integrated loudness: ~ -16 LUFS (±1 LU) - Loudness range (LRA): ~ 3–5 LU - True peak: below -1.5 dBTP Processing expectations: - Adaptive pipeline (NOT static presets) - Audio analysis (loudness, dynamics, frequency balance) - Decision-based processing depending on input - EQ + compression + filtering chain Tonal shaping guidelines: - Control low-end (avoid muddy bass <120 Hz) - Add warmth (200–400 Hz) - Enhance vocal clarity (2–4 kHz) - Smooth highs (avoid harshness >10 kHz) Important: If you do not have real experience in audio processing, DSP, or sound engineering, please DO NOT apply. Bonus if you worked with: - FFmpeg audio filters - Loudness normalization (LUFS) - Psychoacoustics / low bitrate optimization I can provide reference audio samples (original + processed). Budget: $50–$150 (open to discussion based on experience and approach) Looking for someone experienced in sound, not just coding.