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-rw-r--r--vendor/parakeet-rs/src/audio.rs179
1 files changed, 179 insertions, 0 deletions
diff --git a/vendor/parakeet-rs/src/audio.rs b/vendor/parakeet-rs/src/audio.rs
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+++ b/vendor/parakeet-rs/src/audio.rs
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+use crate::config::PreprocessorConfig;
+use crate::error::{Error, Result};
+use hound::{WavReader, WavSpec};
+use ndarray::Array2;
+use std::f32::consts::PI;
+use std::path::Path;
+
+pub fn load_audio<P: AsRef<Path>>(path: P) -> Result<(Vec<f32>, WavSpec)> {
+ let mut reader = WavReader::open(path)?;
+ let spec = reader.spec();
+
+ let samples: Vec<f32> = match spec.sample_format {
+ hound::SampleFormat::Float => reader
+ .samples::<f32>()
+ .collect::<std::result::Result<Vec<_>, _>>()
+ .map_err(|e| Error::Audio(format!("Failed to read float samples: {e}")))?,
+ hound::SampleFormat::Int => reader
+ .samples::<i16>()
+ .map(|s| s.map(|s| s as f32 / 32768.0))
+ .collect::<std::result::Result<Vec<_>, _>>()
+ .map_err(|e| Error::Audio(format!("Failed to read int samples: {e}")))?,
+ };
+
+ Ok((samples, spec))
+}
+
+pub fn apply_preemphasis(audio: &[f32], coef: f32) -> Vec<f32> {
+ let mut result = Vec::with_capacity(audio.len());
+ result.push(audio[0]);
+
+ for i in 1..audio.len() {
+ result.push(audio[i] - coef * audio[i - 1]);
+ }
+
+ result
+}
+
+fn hann_window(window_length: usize) -> Vec<f32> {
+ (0..window_length)
+ .map(|i| 0.5 - 0.5 * ((2.0 * PI * i as f32) / (window_length as f32 - 1.0)).cos())
+ .collect()
+}
+
+// We use proper FFT here instead of naive DFT because the model was trained
+// on correctly computed spectrograms. Naive DFT produces wrong frequency bins
+// and the model outputs all blank tokens. RustFFT gives us O(n log n) performance
+// and numerically correct results that match what the model expects.
+pub fn stft(audio: &[f32], n_fft: usize, hop_length: usize, win_length: usize) -> Array2<f32> {
+ use rustfft::{num_complex::Complex, FftPlanner};
+
+ let window = hann_window(win_length);
+ let num_frames = (audio.len() - win_length) / hop_length + 1;
+ let freq_bins = n_fft / 2 + 1;
+ let mut spectrogram = Array2::<f32>::zeros((freq_bins, num_frames));
+
+ let mut planner = FftPlanner::<f32>::new();
+ let fft = planner.plan_fft_forward(n_fft);
+
+ for frame_idx in 0..num_frames {
+ let start = frame_idx * hop_length;
+
+ let mut frame: Vec<Complex<f32>> = vec![Complex::new(0.0, 0.0); n_fft];
+ for i in 0..win_length.min(audio.len() - start) {
+ frame[i] = Complex::new(audio[start + i] * window[i], 0.0);
+ }
+
+ fft.process(&mut frame);
+
+ for k in 0..freq_bins {
+ let magnitude = frame[k].norm();
+ spectrogram[[k, frame_idx]] = magnitude * magnitude;
+ }
+ }
+
+ spectrogram
+}
+
+fn hz_to_mel(freq: f32) -> f32 {
+ 2595.0 * (1.0 + freq / 700.0).log10()
+}
+
+fn mel_to_hz(mel: f32) -> f32 {
+ 700.0 * (10.0_f32.powf(mel / 2595.0) - 1.0)
+}
+
+fn create_mel_filterbank(n_fft: usize, n_mels: usize, sample_rate: usize) -> Array2<f32> {
+ let freq_bins = n_fft / 2 + 1;
+ let mut filterbank = Array2::<f32>::zeros((n_mels, freq_bins));
+
+ let min_mel = hz_to_mel(0.0);
+ let max_mel = hz_to_mel(sample_rate as f32 / 2.0);
+
+ let mel_points: Vec<f32> = (0..=n_mels + 1)
+ .map(|i| mel_to_hz(min_mel + (max_mel - min_mel) * i as f32 / (n_mels + 1) as f32))
+ .collect();
+
+ let freq_bin_width = sample_rate as f32 / n_fft as f32;
+
+ for mel_idx in 0..n_mels {
+ let left = mel_points[mel_idx];
+ let center = mel_points[mel_idx + 1];
+ let right = mel_points[mel_idx + 2];
+
+ for freq_idx in 0..freq_bins {
+ let freq = freq_idx as f32 * freq_bin_width;
+
+ if freq >= left && freq <= center {
+ filterbank[[mel_idx, freq_idx]] = (freq - left) / (center - left);
+ } else if freq > center && freq <= right {
+ filterbank[[mel_idx, freq_idx]] = (right - freq) / (right - center);
+ }
+ }
+ }
+
+ filterbank
+}
+
+/// Extract mel spectrogram features from raw audio samples.
+///
+/// # Arguments
+///
+/// * `audio` - Audio samples as f32 values
+/// * `sample_rate` - Sample rate in Hz
+/// * `channels` - Number of audio channels
+/// * `config` - Preprocessor configuration
+///
+/// # Returns
+///
+/// 2D array of mel spectrogram features (time_steps x feature_size)
+pub fn extract_features_raw(
+ mut audio: Vec<f32>,
+ sample_rate: u32,
+ channels: u16,
+ config: &PreprocessorConfig,
+) -> Result<Array2<f32>> {
+ if sample_rate != config.sampling_rate as u32 {
+ return Err(Error::Audio(format!(
+ "Audio sample rate {} doesn't match expected {}. Please resample your audio first.",
+ sample_rate, config.sampling_rate
+ )));
+ }
+
+ if channels > 1 {
+ let mono: Vec<f32> = audio
+ .chunks(channels as usize)
+ .map(|chunk| chunk.iter().sum::<f32>() / channels as f32)
+ .collect();
+ audio = mono;
+ }
+
+ audio = apply_preemphasis(&audio, config.preemphasis);
+
+ let spectrogram = stft(&audio, config.n_fft, config.hop_length, config.win_length);
+
+ let mel_filterbank =
+ create_mel_filterbank(config.n_fft, config.feature_size, config.sampling_rate);
+ let mel_spectrogram = mel_filterbank.dot(&spectrogram);
+ let mel_spectrogram = mel_spectrogram.mapv(|x| (x.max(1e-10)).ln());
+
+ let mut mel_spectrogram = mel_spectrogram.t().to_owned();
+
+ // Normalize each feature dimension to mean=0, std=1
+ let num_frames = mel_spectrogram.shape()[0];
+ let num_features = mel_spectrogram.shape()[1];
+
+ for feat_idx in 0..num_features {
+ let mut column = mel_spectrogram.column_mut(feat_idx);
+ let mean: f32 = column.iter().sum::<f32>() / num_frames as f32;
+ let variance: f32 =
+ column.iter().map(|&x| (x - mean).powi(2)).sum::<f32>() / num_frames as f32;
+ let std = variance.sqrt().max(1e-10);
+
+ for val in column.iter_mut() {
+ *val = (*val - mean) / std;
+ }
+ }
+
+ Ok(mel_spectrogram)
+}