Audio sample rate converter

Purpose & User Intent: The tool provides precise calculation and planning for audio sample rate conversion, solving the problem of estimating output length, required processing, and quality considerations when changing the sampling rate. Target users include audio engineers, music producers, researchers, and QC technicians who need to plan resampling workflows, verify duration changes, and compare quality impacts without performing full file I/O. Core Logic & Features: Domain capabilities include computing resampling ratios (R = target_rate / source_rate), estimating output samples (new_samples = floor(original_samples * target_rate / source_rate)), and presenting options for quality modes (basic linear vs polyphase FIR with optional dithering). Required features include input validation, sample rate pair specification, channel handling (mono/stereo), and deterministic outcome. Optional/advanced features include dithering, anti-aliasing flags, and batch or chained conversions across multiple rate pairs. Outputs are designed for downstream file planning rather than on-device playback. Inputs & Outputs: Inputs include source_rate (Hz, integer), target_rate (Hz, integer), channels (mono|stereo), original_samples (integer, optional for duration estimation), dithering (boolean, optional), and quality (low|medium|high, optional). Outputs include target_rate (Hz), resampling_ratio, new_samples (integer), estimated_duration_seconds (float), and notes on potential quality impacts. Algorithms & Calculations: Core calculation uses R = target_rate / source_rate and new_samples = floor(original_samples * R). When duration is derived from samples, duration = new_samples / target_rate. If dithering is enabled, a simple noise shaping step is conceptually applied to preserve perceived quality; for high quality, a polyphase FIR interpolation approach is assumed in back-end processing. Error & Edge Cases: Invalid rates (≤0) or non-numeric inputs produce explicit errors in the outputs (no UI messaging). If original_samples is missing, duration cannot be computed but output samples can still be shown for a given assumed duration. Large rate gaps may imply higher computational load in actual conversion; this tool focuses on planning, not real-time processing. Industry/Region & Localization: Uses SI units (Hz, seconds). Assumes PCM-like audio data; compatible with common formats (WAV/PCM) and standard sampling conventions. Localization considerations include language-neutral numeric formatting for sample counts and rates. Localization notes: supports international decimal separators in numeric fields if configured. Assumptions & Exclusions: Assumes input samples are discrete PCM frames and that only sample rate conversion is performed (no bit-depth or channel count changes). File I/O, metadata handling, and actual audio resampling are outside scope; this tool provides planning and mathematical outputs for downstream processing.