hanzo-audio 0.6.1

//! Audio utilities for `hanzo`.
//!
//! This crate mirrors the `hanzo-vision` crate and focuses on audio specific
//! functionality such as reading audio data, resampling and computing
//! mel spectrogram features.

use anyhow::Result;
use symphonia::core::{
    audio::SampleBuffer, codecs::DecoderOptions, formats::FormatOptions, io::MediaSourceStream,
    meta::MetadataOptions, probe::Hint,
};

/// Raw audio input consisting of PCM samples and a sample rate.
#[derive(Clone, Debug, PartialEq)]
pub struct AudioInput {
    pub samples: Vec<f32>,
    pub sample_rate: u32,
    pub channels: u16,
}

impl AudioInput {
    /// Read a wav file from disk.
    pub fn read_wav(wav_path: &str) -> Result<Self> {
        let mut reader = hound::WavReader::open(wav_path)?;
        let spec = reader.spec();
        let samples: Vec<f32> = match spec.sample_format {
            hound::SampleFormat::Float => reader
                .samples::<f32>()
                .collect::<std::result::Result<_, _>>()?,
            hound::SampleFormat::Int => reader
                .samples::<i16>()
                // Match libsndfile/soundfile normalization for PCM16 by
                // dividing by the full signed range, not by `i16::MAX`.
                .map(|s| s.map(|v| v as f32 / 32768.0))
                .collect::<std::result::Result<_, _>>()?,
        };
        Ok(Self {
            samples,
            sample_rate: spec.sample_rate,
            channels: spec.channels,
        })
    }

    /// Decode audio bytes using `symphonia`.
    pub fn from_bytes(bytes: &[u8]) -> Result<Self> {
        let cursor = std::io::Cursor::new(bytes.to_vec());
        let mss = MediaSourceStream::new(Box::new(cursor), Default::default());
        let hint = Hint::new();
        let probed = symphonia::default::get_probe().format(
            &hint,
            mss,
            &FormatOptions::default(),
            &MetadataOptions::default(),
        )?;
        let mut format = probed.format;
        let track = format
            .default_track()
            .ok_or_else(|| anyhow::anyhow!("no supported audio tracks"))?;
        let codec_params = &track.codec_params;
        let sample_rate = codec_params
            .sample_rate
            .ok_or_else(|| anyhow::anyhow!("unknown sample rate"))?;
        #[allow(clippy::cast_possible_truncation)]
        let channels = codec_params.channels.map(|c| c.count() as u16).unwrap_or(1);
        let mut decoder =
            symphonia::default::get_codecs().make(codec_params, &DecoderOptions::default())?;
        let mut samples = Vec::new();
        loop {
            match format.next_packet() {
                Ok(packet) => {
                    let decoded = decoder.decode(&packet)?;
                    let mut buf =
                        SampleBuffer::<f32>::new(decoded.capacity() as u64, *decoded.spec());
                    buf.copy_interleaved_ref(decoded);
                    samples.extend_from_slice(buf.samples());
                }
                Err(symphonia::core::errors::Error::IoError(e))
                    if e.kind() == std::io::ErrorKind::UnexpectedEof =>
                {
                    break;
                }
                Err(e) => return Err(e.into()),
            }
        }
        Ok(Self {
            samples,
            sample_rate,
            channels,
        })
    }

    /// Convert multi channel audio to mono by averaging channels.
    pub fn to_mono(&self) -> Vec<f32> {
        if self.channels <= 1 {
            return self.samples.clone();
        }
        let mut mono = vec![0.0; self.samples.len() / self.channels as usize];
        for (i, sample) in self.samples.iter().enumerate() {
            mono[i / self.channels as usize] += *sample;
        }
        for s in &mut mono {
            *s /= self.channels as f32;
        }
        mono
    }

    /// Normalize audio to prevent clipping
    pub fn normalize(&mut self) -> &mut Self {
        let max_amplitude = self.samples.iter().map(|s| s.abs()).fold(0.0f32, f32::max);
        if max_amplitude > 0.0 && max_amplitude != 1.0 {
            let scale = 1.0 / max_amplitude;
            for sample in &mut self.samples {
                *sample *= scale;
            }
        }
        self
    }

    /// Apply fade in/out to reduce audio artifacts
    pub fn apply_fade(&mut self, fade_in_samples: usize, fade_out_samples: usize) -> &mut Self {
        let len = self.samples.len();
        // Fade in
        for i in 0..fade_in_samples.min(len) {
            let factor = i as f32 / fade_in_samples as f32;
            self.samples[i] *= factor;
        }
        // Fade out
        for i in 0..fade_out_samples.min(len) {
            let factor = (fade_out_samples - i) as f32 / fade_out_samples as f32;
            self.samples[len - 1 - i] *= factor;
        }
        self
    }

    /// Remove DC offset (audio centered around 0)
    pub fn remove_dc_offset(&mut self) -> &mut Self {
        if self.samples.is_empty() {
            return self;
        }
        let mean = self.samples.iter().sum::<f32>() / self.samples.len() as f32;
        for sample in &mut self.samples {
            *sample -= mean;
        }
        self
    }
}

#[cfg(test)]
mod tests {
    use super::AudioInput;
    use hound::{SampleFormat, WavSpec, WavWriter};
    use std::io::Cursor;

    #[test]
    fn read_wav_roundtrip() {
        let spec = WavSpec {
            channels: 1,
            sample_rate: 16000,
            bits_per_sample: 16,
            sample_format: SampleFormat::Int,
        };
        let mut writer = WavWriter::create("/tmp/test.wav", spec).unwrap();
        for _ in 0..160 {
            writer.write_sample::<i16>(0).unwrap();
        }
        writer.finalize().unwrap();
        let input = AudioInput::read_wav("/tmp/test.wav").unwrap();
        assert_eq!(input.samples.len(), 160);
        assert_eq!(input.sample_rate, 16000);
        std::fs::remove_file("/tmp/test.wav").unwrap();
    }

    #[test]
    fn read_wav_matches_pcm16_full_scale_normalization() {
        let spec = WavSpec {
            channels: 1,
            sample_rate: 16000,
            bits_per_sample: 16,
            sample_format: SampleFormat::Int,
        };
        let mut writer = WavWriter::create("/tmp/test_full_scale.wav", spec).unwrap();
        writer.write_sample::<i16>(i16::MIN).unwrap();
        writer.write_sample::<i16>(i16::MAX).unwrap();
        writer.finalize().unwrap();

        let input = AudioInput::read_wav("/tmp/test_full_scale.wav").unwrap();
        assert_eq!(input.samples, vec![-1.0, 32767.0 / 32768.0]);

        std::fs::remove_file("/tmp/test_full_scale.wav").unwrap();
    }

    #[test]
    fn from_bytes() {
        let spec = WavSpec {
            channels: 1,
            sample_rate: 8000,
            bits_per_sample: 16,
            sample_format: SampleFormat::Int,
        };
        let mut buffer: Vec<u8> = Vec::new();
        {
            let mut writer = WavWriter::new(Cursor::new(&mut buffer), spec).unwrap();
            for _ in 0..80 {
                writer.write_sample::<i16>(0).unwrap();
            }
            writer.finalize().unwrap();
        }
        let input = AudioInput::from_bytes(&buffer).unwrap();
        assert_eq!(input.samples.len(), 80);
        assert_eq!(input.sample_rate, 8000);
    }

    #[test]
    fn test_normalize() {
        let mut input = AudioInput {
            samples: vec![0.2, -0.5, 0.8, -1.0],
            sample_rate: 16000,
            channels: 1,
        };
        input.normalize();
        let max = input.samples.iter().map(|s| s.abs()).fold(0.0f32, f32::max);
        assert!((max - 1.0).abs() < 1e-6);
    }

    #[test]
    fn test_remove_dc_offset() {
        let mut input = AudioInput {
            samples: vec![1.0, 1.0, 1.0, 1.0],
            sample_rate: 16000,
            channels: 1,
        };
        input.remove_dc_offset();
        for s in input.samples {
            assert!((s - 0.0).abs() < 1e-6);
        }
    }
}