1422 lines
56 KiB
Rust
1422 lines
56 KiB
Rust
use std::path::{Path, PathBuf};
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use std::sync::Arc;
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use std::f32::consts::PI;
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use serde::{Deserialize, Serialize};
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use crate::time::Seconds;
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/// Per-output-channel mix coefficients to fold a multichannel source down to
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/// stereo, indexed `[out_channel(0=L,1=R)][src_channel]`.
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///
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/// Assumes the conventional interleave order for each channel count (FL, FR, FC,
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/// LFE, BL, BR, SL, SR …). Uses standard ITU/AC-3-style coefficients: full level
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/// for the matching front channel, `1/√2` (≈ −3 dB) for centre and each surround,
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/// LFE dropped. Each row is then normalized so its absolute-coefficient sum ≤ 1,
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/// which prevents clipping (matching FFmpeg's default `normalize` behaviour).
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///
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/// Returns `None` for layouts we don't special-case (caller falls back to taking
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/// the front L/R pair).
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fn stereo_downmix_matrix(src_channels: usize) -> Option<[Vec<f32>; 2]> {
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const C: f32 = std::f32::consts::FRAC_1_SQRT_2; // ≈ 0.7071
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// (L row, R row); each entry is the gain applied to that source channel.
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let (l, r): (Vec<f32>, Vec<f32>) = match src_channels {
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3 => (vec![1.0, 0.0, C], vec![0.0, 1.0, C]), // FL FR FC
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4 => (vec![1.0, 0.0, C, 0.0], vec![0.0, 1.0, 0.0, C]), // quad: FL FR BL BR
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5 => (vec![1.0, 0.0, C, C, 0.0], vec![0.0, 1.0, C, 0.0, C]), // FL FR FC BL BR
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// 5.1: FL FR FC LFE BL BR (LFE dropped)
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6 => (vec![1.0, 0.0, C, 0.0, C, 0.0], vec![0.0, 1.0, C, 0.0, 0.0, C]),
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// 6.1: FL FR FC LFE BC SL SR (BC → both)
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7 => (vec![1.0, 0.0, C, 0.0, C, C, 0.0], vec![0.0, 1.0, C, 0.0, C, 0.0, C]),
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// 7.1: FL FR FC LFE BL BR SL SR
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8 => (
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vec![1.0, 0.0, C, 0.0, C, 0.0, C, 0.0],
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vec![0.0, 1.0, C, 0.0, 0.0, C, 0.0, C],
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),
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_ => return None,
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};
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let normalize = |row: Vec<f32>| -> Vec<f32> {
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let sum: f32 = row.iter().map(|c| c.abs()).sum();
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if sum > 1.0 {
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row.into_iter().map(|c| c / sum).collect()
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} else {
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row
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}
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};
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Some([normalize(l), normalize(r)])
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}
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/// Windowed sinc interpolation for high-quality time stretching
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/// This is stateless and can handle arbitrary fractional positions
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#[inline]
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fn sinc(x: f32) -> f32 {
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if x.abs() < 1e-5 {
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1.0
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} else {
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let px = PI * x;
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px.sin() / px
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}
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}
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/// Blackman window function
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#[inline]
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fn blackman_window(x: f32, width: f32) -> f32 {
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if x.abs() > width {
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0.0
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} else {
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let a0 = 0.42;
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let a1 = 0.5;
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let a2 = 0.08;
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// Map x from [-width, width] to [0, 1] for proper Blackman window evaluation
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let n = (x / width + 1.0) / 2.0;
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a0 - a1 * (2.0 * PI * n).cos() + a2 * (4.0 * PI * n).cos()
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}
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}
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/// High-quality windowed sinc interpolation
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/// Uses a 32-tap windowed sinc kernel for smooth, artifact-free interpolation
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/// frac: fractional position to interpolate at (0.0 to 1.0)
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/// samples: array of samples centered around the target position
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#[inline]
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fn windowed_sinc_interpolate(samples: &[f32], frac: f32) -> f32 {
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let mut result = 0.0;
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let kernel_size = samples.len();
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let half_kernel = (kernel_size / 2) as f32;
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for i in 0..kernel_size {
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// Distance from interpolation point
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// samples[half_kernel] is at position 0, we want to interpolate at position frac
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let x = frac + half_kernel - (i as f32);
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let sinc_val = sinc(x);
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let window_val = blackman_window(x, half_kernel);
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result += samples[i] * sinc_val * window_val;
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}
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result
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}
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/// PCM sample format for memory-mapped audio files
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#[derive(Debug, Clone, Copy, PartialEq, Eq)]
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pub enum PcmSampleFormat {
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I16,
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I24,
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F32,
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}
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/// How audio data is stored for a pool entry
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#[derive(Debug, Clone)]
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pub enum AudioStorage {
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/// Fully decoded interleaved f32 samples in memory
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InMemory(Vec<f32>),
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/// Memory-mapped PCM file (WAV/AIFF) — instant load, OS-managed paging
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Mapped {
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mmap: Arc<memmap2::Mmap>,
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data_offset: usize,
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sample_format: PcmSampleFormat,
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bytes_per_sample: usize,
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total_frames: u64,
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},
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/// Compressed audio — playback handled by disk reader's stream decoder.
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/// `decoded_for_waveform` is progressively filled by a background thread.
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Compressed {
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decoded_for_waveform: Vec<f32>,
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decoded_frames: u64,
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total_frames: u64,
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},
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/// Audio track of a video container, decoded on demand via FFmpeg
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/// (`VideoAudioReader`). The source video file is `AudioFile::path`. Like
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/// `Compressed`, playback is streamed through the disk reader and
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/// `decoded_for_waveform` is filled progressively for the overview.
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VideoAudio {
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decoded_for_waveform: Vec<f32>,
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decoded_frames: u64,
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total_frames: u64,
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},
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}
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/// Audio file stored in the pool
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#[derive(Debug, Clone)]
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pub struct AudioFile {
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pub path: PathBuf,
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pub storage: AudioStorage,
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pub channels: u32,
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pub sample_rate: u32,
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pub frames: u64,
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/// Original file format (mp3, ogg, wav, flac, etc.)
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/// Used to determine if we should preserve lossy encoding during save
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pub original_format: Option<String>,
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/// Original compressed file bytes (preserved across save/load to avoid re-encoding)
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pub original_bytes: Option<Vec<u8>>,
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/// When `Some`, this entry's bytes are packed in the project container (not on
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/// disk at `path`); the disk reader opens them via the host's
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/// `AudioBlobSourceFactory` using this media id. `None` ⇒ stream from `path`.
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pub packed_media_id: Option<String>,
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}
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impl AudioFile {
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/// Create a new AudioFile with in-memory interleaved f32 data
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pub fn new(path: PathBuf, data: Vec<f32>, channels: u32, sample_rate: u32) -> Self {
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let frames = (data.len() / channels as usize) as u64;
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Self {
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path,
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storage: AudioStorage::InMemory(data),
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channels,
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sample_rate,
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frames,
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original_format: None,
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original_bytes: None,
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packed_media_id: None,
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}
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}
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/// Create a new AudioFile with original format information
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pub fn with_format(path: PathBuf, data: Vec<f32>, channels: u32, sample_rate: u32, original_format: Option<String>) -> Self {
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let frames = (data.len() / channels as usize) as u64;
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Self {
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path,
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storage: AudioStorage::InMemory(data),
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channels,
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sample_rate,
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frames,
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original_format,
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original_bytes: None,
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packed_media_id: None,
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}
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}
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/// Create an AudioFile backed by a memory-mapped WAV/AIFF file
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pub fn from_mmap(
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path: PathBuf,
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mmap: memmap2::Mmap,
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data_offset: usize,
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sample_format: PcmSampleFormat,
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channels: u32,
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sample_rate: u32,
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total_frames: u64,
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) -> Self {
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let bytes_per_sample = match sample_format {
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PcmSampleFormat::I16 => 2,
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PcmSampleFormat::I24 => 3,
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PcmSampleFormat::F32 => 4,
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};
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Self {
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path,
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storage: AudioStorage::Mapped {
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mmap: Arc::new(mmap),
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data_offset,
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sample_format,
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bytes_per_sample,
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total_frames,
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},
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channels,
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sample_rate,
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frames: total_frames,
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original_format: Some("wav".to_string()),
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original_bytes: None,
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packed_media_id: None,
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}
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}
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/// Create a placeholder AudioFile for a compressed format (playback via disk reader)
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pub fn from_compressed(
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path: PathBuf,
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channels: u32,
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sample_rate: u32,
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total_frames: u64,
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original_format: Option<String>,
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) -> Self {
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Self {
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path,
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storage: AudioStorage::Compressed {
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decoded_for_waveform: Vec::new(),
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decoded_frames: 0,
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total_frames,
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},
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channels,
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sample_rate,
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frames: total_frames,
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original_format,
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original_bytes: None,
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packed_media_id: None,
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}
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}
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/// Create a placeholder AudioFile for a video's audio track. `path` is the
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/// source video file; the audio is streamed on demand by the disk reader's
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/// FFmpeg-backed `VideoAudioReader`.
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pub fn from_video_audio(
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path: PathBuf,
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channels: u32,
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sample_rate: u32,
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total_frames: u64,
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) -> Self {
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Self {
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path,
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storage: AudioStorage::VideoAudio {
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decoded_for_waveform: Vec::new(),
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decoded_frames: 0,
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total_frames,
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},
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channels,
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sample_rate,
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frames: total_frames,
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original_format: None,
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original_bytes: None,
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packed_media_id: None,
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}
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}
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/// Get interleaved f32 sample data.
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///
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/// - **InMemory**: returns the full slice directly.
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/// - **Mapped F32**: reinterprets the mmap'd bytes as `&[f32]` (zero-copy).
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/// - **Mapped I16/I24 or Compressed**: returns an empty slice (use
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/// `read_samples()` or the disk reader's `ReadAheadBuffer` instead).
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pub fn data(&self) -> &[f32] {
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match &self.storage {
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AudioStorage::InMemory(data) => data,
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AudioStorage::Mapped {
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mmap,
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data_offset,
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sample_format,
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total_frames,
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..
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} if *sample_format == PcmSampleFormat::F32 => {
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let byte_slice = &mmap[*data_offset..];
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let ptr = byte_slice.as_ptr();
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// Check 4-byte alignment (required for f32)
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if ptr.align_offset(std::mem::align_of::<f32>()) == 0 {
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let len = (*total_frames as usize) * self.channels as usize;
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let available = byte_slice.len() / 4;
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let safe_len = len.min(available);
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// SAFETY: pointer is aligned, mmap is read-only and outlives
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// this borrow, and we clamp to the available byte range.
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unsafe { std::slice::from_raw_parts(ptr as *const f32, safe_len) }
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} else {
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&[]
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}
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}
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_ => &[],
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}
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}
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/// Read samples for a specific channel into the output buffer.
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/// Works for InMemory and Mapped storage. Returns the number of frames read.
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pub fn read_samples(
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&self,
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start_frame: usize,
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count: usize,
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channel: usize,
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out: &mut [f32],
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) -> usize {
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let channels = self.channels as usize;
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let total_frames = self.frames as usize;
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match &self.storage {
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AudioStorage::InMemory(data) => {
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let mut written = 0;
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for i in 0..count.min(out.len()) {
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let frame = start_frame + i;
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if frame >= total_frames { break; }
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let idx = frame * channels + channel;
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out[i] = data[idx];
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written += 1;
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}
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written
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}
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AudioStorage::Mapped { mmap, data_offset, sample_format, bytes_per_sample, .. } => {
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let mut written = 0;
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for i in 0..count.min(out.len()) {
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let frame = start_frame + i;
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if frame >= total_frames { break; }
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let sample_index = frame * channels + channel;
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let byte_offset = data_offset + sample_index * bytes_per_sample;
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let end = byte_offset + bytes_per_sample;
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if end > mmap.len() { break; }
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let bytes = &mmap[byte_offset..end];
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out[i] = match sample_format {
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PcmSampleFormat::I16 => {
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let val = i16::from_le_bytes([bytes[0], bytes[1]]);
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val as f32 / 32768.0
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}
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PcmSampleFormat::I24 => {
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// Sign-extend 24-bit to 32-bit
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let val = ((bytes[0] as i32)
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| ((bytes[1] as i32) << 8)
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| ((bytes[2] as i32) << 16))
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<< 8
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>> 8;
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val as f32 / 8388608.0
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}
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PcmSampleFormat::F32 => {
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f32::from_le_bytes([bytes[0], bytes[1], bytes[2], bytes[3]])
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}
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};
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written += 1;
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}
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written
|
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}
|
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AudioStorage::Compressed { .. } | AudioStorage::VideoAudio { .. } => {
|
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// Streamed through the disk reader, not via read_samples().
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0
|
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}
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}
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}
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/// Get duration in seconds
|
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pub fn duration_seconds(&self) -> f64 {
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self.frames as f64 / self.sample_rate as f64
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}
|
||
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/// Generate a waveform overview with the specified number of peaks
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/// This creates a downsampled representation suitable for timeline visualization
|
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pub fn generate_waveform_overview(&self, target_peaks: usize) -> Vec<crate::io::WaveformPeak> {
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self.generate_waveform_overview_range(0, self.frames as usize, target_peaks)
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}
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/// Generate a waveform overview for a specific range of frames
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///
|
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/// # Arguments
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/// * `start_frame` - Starting frame index (0-based)
|
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/// * `end_frame` - Ending frame index (exclusive)
|
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/// * `target_peaks` - Desired number of peaks to generate
|
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pub fn generate_waveform_overview_range(
|
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&self,
|
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start_frame: usize,
|
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end_frame: usize,
|
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target_peaks: usize,
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) -> Vec<crate::io::WaveformPeak> {
|
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if self.frames == 0 || target_peaks == 0 {
|
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return Vec::new();
|
||
}
|
||
|
||
let total_frames = self.frames as usize;
|
||
let start_frame = start_frame.min(total_frames);
|
||
let end_frame = end_frame.min(total_frames);
|
||
|
||
if start_frame >= end_frame {
|
||
return Vec::new();
|
||
}
|
||
|
||
let range_frames = end_frame - start_frame;
|
||
let frames_per_peak = (range_frames / target_peaks).max(1);
|
||
let actual_peaks = (range_frames + frames_per_peak - 1) / frames_per_peak;
|
||
|
||
let mut peaks = Vec::with_capacity(actual_peaks);
|
||
|
||
for peak_idx in 0..actual_peaks {
|
||
let peak_start = start_frame + peak_idx * frames_per_peak;
|
||
let peak_end = (start_frame + (peak_idx + 1) * frames_per_peak).min(end_frame);
|
||
|
||
let mut min = f32::MAX;
|
||
let mut max = f32::MIN;
|
||
|
||
// Scan all samples in this window
|
||
let data = self.data();
|
||
for frame_idx in peak_start..peak_end {
|
||
// For multi-channel audio, combine all channels
|
||
for ch in 0..self.channels as usize {
|
||
let sample_idx = frame_idx * self.channels as usize + ch;
|
||
if sample_idx < data.len() {
|
||
let sample = data[sample_idx];
|
||
min = min.min(sample);
|
||
max = max.max(sample);
|
||
}
|
||
}
|
||
}
|
||
|
||
// If no samples were found, clamp to safe defaults
|
||
if min == f32::MAX {
|
||
min = 0.0;
|
||
}
|
||
if max == f32::MIN {
|
||
max = 0.0;
|
||
}
|
||
|
||
peaks.push(crate::io::WaveformPeak { min, max });
|
||
}
|
||
|
||
peaks
|
||
}
|
||
}
|
||
|
||
/// Pool of shared audio files (audio clip content)
|
||
pub struct AudioClipPool {
|
||
files: Vec<AudioFile>,
|
||
/// Waveform chunk cache for multi-resolution waveform generation
|
||
waveform_cache: crate::audio::waveform_cache::WaveformCache,
|
||
}
|
||
|
||
/// Type alias for backwards compatibility
|
||
pub type AudioPool = AudioClipPool;
|
||
|
||
impl AudioClipPool {
|
||
/// Create a new empty audio clip pool
|
||
pub fn new() -> Self {
|
||
Self {
|
||
files: Vec::new(),
|
||
waveform_cache: crate::audio::waveform_cache::WaveformCache::new(100), // 100MB cache
|
||
}
|
||
}
|
||
|
||
/// Get the number of files in the pool
|
||
pub fn len(&self) -> usize {
|
||
self.files.len()
|
||
}
|
||
|
||
/// Check if the pool is empty
|
||
pub fn is_empty(&self) -> bool {
|
||
self.files.is_empty()
|
||
}
|
||
|
||
/// Get file info for waveform generation (duration, sample_rate, channels)
|
||
pub fn get_file_info(&self, pool_index: usize) -> Option<(f64, u32, u32)> {
|
||
self.files.get(pool_index).map(|file| {
|
||
(file.duration_seconds(), file.sample_rate, file.channels)
|
||
})
|
||
}
|
||
|
||
/// Generate waveform overview for a file in the pool
|
||
pub fn generate_waveform(&self, pool_index: usize, target_peaks: usize) -> Option<Vec<crate::io::WaveformPeak>> {
|
||
self.files.get(pool_index).map(|file| {
|
||
file.generate_waveform_overview(target_peaks)
|
||
})
|
||
}
|
||
|
||
/// Generate waveform overview for a specific range of a file in the pool
|
||
///
|
||
/// # Arguments
|
||
/// * `pool_index` - Index of the file in the pool
|
||
/// * `start_frame` - Starting frame index (0-based)
|
||
/// * `end_frame` - Ending frame index (exclusive)
|
||
/// * `target_peaks` - Desired number of peaks to generate
|
||
pub fn generate_waveform_range(
|
||
&self,
|
||
pool_index: usize,
|
||
start_frame: usize,
|
||
end_frame: usize,
|
||
target_peaks: usize,
|
||
) -> Option<Vec<crate::io::WaveformPeak>> {
|
||
self.files.get(pool_index).map(|file| {
|
||
file.generate_waveform_overview_range(start_frame, end_frame, target_peaks)
|
||
})
|
||
}
|
||
|
||
/// Add an audio file to the pool and return its index
|
||
pub fn add_file(&mut self, file: AudioFile) -> usize {
|
||
let index = self.files.len();
|
||
self.files.push(file);
|
||
index
|
||
}
|
||
|
||
/// Get an audio file by index
|
||
pub fn get_file(&self, index: usize) -> Option<&AudioFile> {
|
||
self.files.get(index)
|
||
}
|
||
|
||
/// Get a mutable reference to an audio file by index
|
||
pub fn get_file_mut(&mut self, index: usize) -> Option<&mut AudioFile> {
|
||
self.files.get_mut(index)
|
||
}
|
||
|
||
/// Get number of files in the pool
|
||
pub fn file_count(&self) -> usize {
|
||
self.files.len()
|
||
}
|
||
|
||
/// Render audio from a file in the pool with high-quality windowed sinc interpolation
|
||
/// start_time: position in the audio file to start reading from (in seconds)
|
||
/// clip_read_ahead: per-clip-instance read-ahead buffer for compressed audio streaming
|
||
/// Returns the number of samples actually rendered
|
||
pub fn render_from_file(
|
||
&self,
|
||
pool_index: usize,
|
||
output: &mut [f32],
|
||
start_time: Seconds,
|
||
gain: f32,
|
||
engine_sample_rate: u32,
|
||
engine_channels: u32,
|
||
clip_read_ahead: Option<&super::disk_reader::ReadAheadBuffer>,
|
||
) -> usize {
|
||
let start_time_seconds = start_time.0;
|
||
let Some(audio_file) = self.files.get(pool_index) else {
|
||
return 0;
|
||
};
|
||
|
||
let audio_data = audio_file.data();
|
||
let read_ahead = clip_read_ahead;
|
||
let use_read_ahead = audio_data.is_empty();
|
||
let src_channels = audio_file.channels as usize;
|
||
|
||
// Nothing to render: no data and no read-ahead buffer
|
||
if use_read_ahead && read_ahead.is_none() {
|
||
// Log once per pool_index to diagnose silent clips
|
||
static LOGGED: std::sync::atomic::AtomicU64 = std::sync::atomic::AtomicU64::new(u64::MAX);
|
||
let prev = LOGGED.swap(pool_index as u64, std::sync::atomic::Ordering::Relaxed);
|
||
if prev != pool_index as u64 {
|
||
eprintln!("[RENDER] pool={}: data empty, no read_ahead! storage={:?}, frames={}",
|
||
pool_index, std::mem::discriminant(&audio_file.storage), audio_file.frames);
|
||
}
|
||
return 0;
|
||
}
|
||
|
||
// In export mode, block-wait until the disk reader has filled the
|
||
// frames we need, so offline rendering never gets buffer misses.
|
||
if use_read_ahead {
|
||
let ra = read_ahead.unwrap();
|
||
if ra.is_export_mode() {
|
||
let src_start = (start_time_seconds * audio_file.sample_rate as f64) as u64;
|
||
// Tell the disk reader where we need data BEFORE waiting
|
||
ra.set_target_frame(src_start);
|
||
// Pad by 64 frames for sinc interpolation taps
|
||
let frames_needed = (output.len() / engine_channels as usize) as u64 + 64;
|
||
// Spin-wait with small sleeps until the disk reader fills the buffer
|
||
let mut wait_iters = 0u64;
|
||
while !ra.has_range(src_start, frames_needed) {
|
||
std::thread::sleep(std::time::Duration::from_micros(100));
|
||
wait_iters += 1;
|
||
if wait_iters > 100_000 {
|
||
// Safety valve: 10 seconds of waiting
|
||
eprintln!("[EXPORT] Timed out waiting for disk reader (need frames {}..{})",
|
||
src_start, src_start + frames_needed);
|
||
break;
|
||
}
|
||
}
|
||
}
|
||
}
|
||
|
||
// Snapshot the read-ahead buffer range once for the entire render call.
|
||
// This ensures all sinc interpolation taps within a single callback
|
||
// see a consistent range, preventing crackle from concurrent updates.
|
||
let (ra_start, ra_end) = if use_read_ahead {
|
||
read_ahead.unwrap().snapshot()
|
||
} else {
|
||
(0, 0)
|
||
};
|
||
|
||
// Buffer-miss counter: how many times we wanted a sample the ring
|
||
// buffer didn't have (frame in file range but outside buffer range).
|
||
let mut buffer_misses: u32 = 0;
|
||
|
||
// Read a single interleaved sample by (frame, channel).
|
||
// Uses direct slice access for InMemory/Mapped, or the disk reader's
|
||
// ReadAheadBuffer for compressed files.
|
||
macro_rules! get_sample {
|
||
($frame:expr, $ch:expr) => {{
|
||
if use_read_ahead {
|
||
let f = $frame as u64;
|
||
let s = read_ahead.unwrap().read_sample(f, $ch, ra_start, ra_end);
|
||
if s == 0.0 && (f < ra_start || f >= ra_end) {
|
||
buffer_misses += 1;
|
||
}
|
||
s
|
||
} else {
|
||
let idx = ($frame) * src_channels + ($ch);
|
||
if idx < audio_data.len() { audio_data[idx] } else { 0.0 }
|
||
}
|
||
}};
|
||
}
|
||
let dst_channels = engine_channels as usize;
|
||
let output_frames = output.len() / dst_channels;
|
||
|
||
// Fold a multichannel source (5.1, 7.1, …) down to stereo with proper
|
||
// coefficients (centre + surrounds mixed in, LFE dropped) instead of just
|
||
// taking the front L/R pair. `None` ⇒ no downmix needed / unknown layout.
|
||
let downmix = if dst_channels == 2 && src_channels > 2 {
|
||
stereo_downmix_matrix(src_channels)
|
||
} else {
|
||
None
|
||
};
|
||
|
||
let src_start_position = start_time_seconds * audio_file.sample_rate as f64;
|
||
|
||
// Tell the disk reader where we're reading so it buffers the right region.
|
||
if use_read_ahead {
|
||
read_ahead.unwrap().set_target_frame(src_start_position as u64);
|
||
}
|
||
|
||
let mut rendered_frames = 0;
|
||
|
||
if audio_file.sample_rate == engine_sample_rate {
|
||
// Fast path: matching sample rates — direct sample copy, no interpolation
|
||
let src_start_frame = src_start_position.floor() as i64;
|
||
|
||
// Continuity check: detect gaps/overlaps between consecutive callbacks (DAW_AUDIO_DEBUG=1)
|
||
if std::env::var("DAW_AUDIO_DEBUG").is_ok() {
|
||
use std::sync::atomic::{AtomicI64, Ordering as AO};
|
||
static EXPECTED_NEXT: AtomicI64 = AtomicI64::new(-1);
|
||
static DISCONTINUITIES: AtomicI64 = AtomicI64::new(0);
|
||
let expected = EXPECTED_NEXT.load(AO::Relaxed);
|
||
if expected >= 0 && src_start_frame != expected {
|
||
let count = DISCONTINUITIES.fetch_add(1, AO::Relaxed) + 1;
|
||
eprintln!("[RENDER CONTINUITY] DISCONTINUITY #{}: expected frame {}, got {} (delta={})",
|
||
count, expected, src_start_frame, src_start_frame - expected);
|
||
}
|
||
EXPECTED_NEXT.store(src_start_frame + output_frames as i64, AO::Relaxed);
|
||
}
|
||
|
||
for output_frame in 0..output_frames {
|
||
let src_frame = src_start_frame + output_frame as i64;
|
||
if src_frame < 0 || src_frame as u64 >= audio_file.frames {
|
||
break;
|
||
}
|
||
let sf = src_frame as usize;
|
||
|
||
for dst_ch in 0..dst_channels {
|
||
let sample = if src_channels == dst_channels {
|
||
get_sample!(sf, dst_ch)
|
||
} else if src_channels == 1 {
|
||
get_sample!(sf, 0)
|
||
} else if dst_channels == 1 {
|
||
let mut sum = 0.0f32;
|
||
for src_ch in 0..src_channels {
|
||
sum += get_sample!(sf, src_ch);
|
||
}
|
||
sum / src_channels as f32
|
||
} else if let Some(ref mat) = downmix {
|
||
// Surround → stereo with proper coefficients.
|
||
let mut s = 0.0f32;
|
||
for (src_ch, &c) in mat[dst_ch].iter().enumerate() {
|
||
if c != 0.0 {
|
||
s += c * get_sample!(sf, src_ch);
|
||
}
|
||
}
|
||
s
|
||
} else {
|
||
get_sample!(sf, dst_ch % src_channels)
|
||
};
|
||
|
||
output[output_frame * dst_channels + dst_ch] += sample * gain;
|
||
}
|
||
|
||
rendered_frames += 1;
|
||
}
|
||
} else {
|
||
// Sample rate conversion with windowed sinc interpolation
|
||
let rate_ratio = audio_file.sample_rate as f64 / engine_sample_rate as f64;
|
||
const KERNEL_SIZE: usize = 32;
|
||
const HALF_KERNEL: usize = KERNEL_SIZE / 2;
|
||
|
||
for output_frame in 0..output_frames {
|
||
let src_position = src_start_position + (output_frame as f64 * rate_ratio);
|
||
let src_frame = src_position.floor() as i32;
|
||
let frac = (src_position - src_frame as f64) as f32;
|
||
|
||
if src_frame < 0 || src_frame as usize >= audio_file.frames as usize {
|
||
break;
|
||
}
|
||
|
||
// Sinc-interpolate a single source channel at the current position.
|
||
macro_rules! sinc_ch {
|
||
($ch:expr) => {{
|
||
let mut channel_samples = [0.0f32; KERNEL_SIZE];
|
||
for (j, i) in (-(HALF_KERNEL as i32)..(HALF_KERNEL as i32)).enumerate() {
|
||
let idx = src_frame + i;
|
||
if idx >= 0 && (idx as usize) < audio_file.frames as usize {
|
||
channel_samples[j] = get_sample!(idx as usize, $ch);
|
||
}
|
||
}
|
||
windowed_sinc_interpolate(&channel_samples, frac)
|
||
}};
|
||
}
|
||
|
||
for dst_ch in 0..dst_channels {
|
||
let sample = if let Some(ref mat) = downmix {
|
||
// Surround → stereo: interpolate each contributing channel.
|
||
let mut s = 0.0f32;
|
||
for (ch, &c) in mat[dst_ch].iter().enumerate() {
|
||
if c != 0.0 {
|
||
s += c * sinc_ch!(ch);
|
||
}
|
||
}
|
||
s
|
||
} else if dst_channels == 1 {
|
||
let mut sum = 0.0;
|
||
for ch in 0..src_channels {
|
||
sum += sinc_ch!(ch);
|
||
}
|
||
sum / src_channels as f32
|
||
} else {
|
||
let src_ch = if src_channels == dst_channels {
|
||
dst_ch
|
||
} else if src_channels == 1 {
|
||
0
|
||
} else {
|
||
dst_ch % src_channels
|
||
};
|
||
sinc_ch!(src_ch)
|
||
};
|
||
|
||
output[output_frame * dst_channels + dst_ch] += sample * gain;
|
||
}
|
||
|
||
rendered_frames += 1;
|
||
}
|
||
}
|
||
|
||
if use_read_ahead && buffer_misses > 0 {
|
||
static MISS_COUNT: std::sync::atomic::AtomicU64 = std::sync::atomic::AtomicU64::new(0);
|
||
let total = MISS_COUNT.fetch_add(buffer_misses as u64, std::sync::atomic::Ordering::Relaxed) + buffer_misses as u64;
|
||
// Log every 100 misses to avoid flooding
|
||
if total % 100 < buffer_misses as u64 {
|
||
eprintln!("[RENDER] buffer misses this call: {}, total: {}, snap=[{}..{}], src_start_frame={}",
|
||
buffer_misses, total, ra_start, ra_end,
|
||
(start_time_seconds * audio_file.sample_rate as f64) as u64);
|
||
}
|
||
}
|
||
|
||
rendered_frames * dst_channels
|
||
}
|
||
|
||
/// Generate waveform chunks for a file in the pool
|
||
///
|
||
/// This generates chunks at a specific detail level and caches them.
|
||
/// Returns the generated chunks.
|
||
pub fn generate_waveform_chunks(
|
||
&mut self,
|
||
pool_index: usize,
|
||
detail_level: u8,
|
||
chunk_indices: &[u32],
|
||
) -> Vec<crate::io::WaveformChunk> {
|
||
let file = match self.files.get(pool_index) {
|
||
Some(f) => f,
|
||
None => return Vec::new(),
|
||
};
|
||
|
||
let chunks = crate::audio::waveform_cache::WaveformCache::generate_chunks(
|
||
file,
|
||
pool_index,
|
||
detail_level,
|
||
chunk_indices,
|
||
);
|
||
|
||
// Store chunks in cache
|
||
for chunk in &chunks {
|
||
let key = crate::io::WaveformChunkKey {
|
||
pool_index,
|
||
detail_level: chunk.detail_level,
|
||
chunk_index: chunk.chunk_index,
|
||
};
|
||
self.waveform_cache.store_chunk(key, chunk.peaks.clone());
|
||
}
|
||
|
||
chunks
|
||
}
|
||
|
||
/// Generate Level 0 (overview) chunks for a file
|
||
///
|
||
/// This should be called immediately when a file is imported.
|
||
/// Returns the generated chunks.
|
||
pub fn generate_overview_chunks(
|
||
&mut self,
|
||
pool_index: usize,
|
||
) -> Vec<crate::io::WaveformChunk> {
|
||
let file = match self.files.get(pool_index) {
|
||
Some(f) => f,
|
||
None => return Vec::new(),
|
||
};
|
||
|
||
self.waveform_cache.generate_overview_chunks(file, pool_index)
|
||
}
|
||
|
||
/// Get a cached waveform chunk
|
||
pub fn get_waveform_chunk(
|
||
&self,
|
||
pool_index: usize,
|
||
detail_level: u8,
|
||
chunk_index: u32,
|
||
) -> Option<&Vec<crate::io::WaveformPeak>> {
|
||
let key = crate::io::WaveformChunkKey {
|
||
pool_index,
|
||
detail_level,
|
||
chunk_index,
|
||
};
|
||
self.waveform_cache.get_chunk(&key)
|
||
}
|
||
|
||
/// Check if a waveform chunk is cached
|
||
pub fn has_waveform_chunk(
|
||
&self,
|
||
pool_index: usize,
|
||
detail_level: u8,
|
||
chunk_index: u32,
|
||
) -> bool {
|
||
let key = crate::io::WaveformChunkKey {
|
||
pool_index,
|
||
detail_level,
|
||
chunk_index,
|
||
};
|
||
self.waveform_cache.has_chunk(&key)
|
||
}
|
||
|
||
/// Get waveform cache memory usage in MB
|
||
pub fn waveform_cache_memory_mb(&self) -> f64 {
|
||
self.waveform_cache.memory_usage_mb()
|
||
}
|
||
|
||
/// Get number of cached waveform chunks
|
||
pub fn waveform_chunk_count(&self) -> usize {
|
||
self.waveform_cache.chunk_count()
|
||
}
|
||
}
|
||
|
||
impl Default for AudioClipPool {
|
||
fn default() -> Self {
|
||
Self::new()
|
||
}
|
||
}
|
||
|
||
/// Embedded audio data stored as base64 in the project file
|
||
#[derive(Debug, Clone, Serialize, Deserialize)]
|
||
pub struct EmbeddedAudioData {
|
||
/// Base64-encoded audio data
|
||
pub data_base64: String,
|
||
/// Original file format (wav, mp3, etc.)
|
||
pub format: String,
|
||
}
|
||
|
||
/// Serializable audio pool entry for project save/load
|
||
#[derive(Debug, Clone, Serialize, Deserialize)]
|
||
pub struct AudioPoolEntry {
|
||
/// Index in the audio pool
|
||
pub pool_index: usize,
|
||
/// Original filename
|
||
pub name: String,
|
||
/// Path relative to project file (None if embedded)
|
||
pub relative_path: Option<String>,
|
||
/// Duration in seconds
|
||
pub duration: f64,
|
||
/// Sample rate
|
||
pub sample_rate: u32,
|
||
/// Number of channels
|
||
pub channels: u32,
|
||
/// Embedded audio data (for files < 10MB)
|
||
pub embedded_data: Option<EmbeddedAudioData>,
|
||
/// Stable media id (UUID string) for the SQLite `.beam` container. When set,
|
||
/// the audio bytes live in the container's `media` table keyed by this id
|
||
/// (packed storage). `None` for referenced entries (use `relative_path`) or
|
||
/// legacy ZIP-loaded entries. Populated by the file_io save/load layer.
|
||
#[serde(default, skip_serializing_if = "Option::is_none")]
|
||
pub media_id: Option<String>,
|
||
/// Transient carrier for this entry's serialized waveform-pyramid blob (LBWF
|
||
/// bytes). Never serialized into project.json — the bytes live in the
|
||
/// container's `media` table (kind `Waveform`). Set by the file_io save layer
|
||
/// (in) and load layer (out); `None` everywhere else.
|
||
#[serde(skip)]
|
||
pub waveform_blob: Option<Vec<u8>>,
|
||
/// This entry is a video container's audio track (`relative_path` points at the
|
||
/// video file). It is always stored as a path reference (never packed/embedded
|
||
/// — the `VideoClip` already references the file) and reloaded by re-probing
|
||
/// the video via FFmpeg, so multichannel (5.1/7.1) audio survives the round-trip
|
||
/// (Symphonia reconstitution would otherwise collapse it).
|
||
#[serde(default, skip_serializing_if = "std::ops::Not::not")]
|
||
pub is_video_audio: bool,
|
||
}
|
||
|
||
impl AudioClipPool {
|
||
/// Serialize the audio clip pool for project saving
|
||
///
|
||
/// Files smaller than 10MB are embedded as base64.
|
||
/// Larger files are stored as relative paths to the project file.
|
||
pub fn serialize(&self, project_path: &Path) -> Result<Vec<AudioPoolEntry>, String> {
|
||
let project_dir = project_path.parent()
|
||
.ok_or_else(|| "Project path has no parent directory".to_string())?;
|
||
|
||
let mut entries = Vec::new();
|
||
|
||
for (index, file) in self.files.iter().enumerate() {
|
||
// Skip placeholder pool slots: `load_from_serialized` resizes the pool to
|
||
// `max_index + 1` filled with empty `AudioFile::new(PathBuf::new(), …)` to
|
||
// cover index gaps (and creates one even when there are no entries at all).
|
||
// Such a slot has an empty path and no packed media — there's nothing to
|
||
// persist, and emitting it yields an entry whose empty `relative_path`
|
||
// resolves to the project directory itself (unreadable on the next save).
|
||
if file.path.as_os_str().is_empty() && file.packed_media_id.is_none() {
|
||
continue;
|
||
}
|
||
|
||
// Video's audio track: reference the video file (it's also referenced
|
||
// by the VideoClip) and re-probe it via FFmpeg on load. Never pack or
|
||
// embed it as audio media — that both wastes space and loses the 5.1+
|
||
// layout when Symphonia later decodes it.
|
||
if matches!(file.storage, AudioStorage::VideoAudio { .. }) {
|
||
let relative_path = pathdiff::diff_paths(&file.path, project_dir)
|
||
.map(|r| r.to_string_lossy().to_string())
|
||
.or_else(|| Some(file.path.to_string_lossy().to_string()));
|
||
entries.push(AudioPoolEntry {
|
||
pool_index: index,
|
||
is_video_audio: true,
|
||
waveform_blob: None,
|
||
name: file
|
||
.path
|
||
.file_name()
|
||
.map(|n| n.to_string_lossy().to_string())
|
||
.unwrap_or_else(|| format!("file_{}", index)),
|
||
relative_path,
|
||
duration: file.duration_seconds(),
|
||
sample_rate: file.sample_rate,
|
||
channels: file.channels,
|
||
embedded_data: None,
|
||
media_id: None,
|
||
});
|
||
continue;
|
||
}
|
||
|
||
// Packed-in-container streaming entry: its bytes already live in the
|
||
// `.beam` media table (kept in place across re-saves). Emit just the
|
||
// media id — no path, no embedded bytes, nothing to decode.
|
||
if let Some(media_id) = &file.packed_media_id {
|
||
entries.push(AudioPoolEntry {
|
||
pool_index: index,
|
||
is_video_audio: false,
|
||
waveform_blob: None,
|
||
name: file
|
||
.path
|
||
.file_name()
|
||
.map(|n| n.to_string_lossy().to_string())
|
||
.unwrap_or_else(|| format!("file_{}", index)),
|
||
relative_path: None,
|
||
duration: file.duration_seconds(),
|
||
sample_rate: file.sample_rate,
|
||
channels: file.channels,
|
||
embedded_data: None,
|
||
media_id: Some(media_id.clone()),
|
||
});
|
||
continue;
|
||
}
|
||
|
||
let file_path = &file.path;
|
||
let file_path_str = file_path.to_string_lossy();
|
||
|
||
// Check if this is a temp file (from recording) or previously embedded audio
|
||
// Always embed these
|
||
let is_temp_file = file_path.starts_with(std::env::temp_dir());
|
||
let is_embedded = file_path_str.starts_with("<embedded:");
|
||
|
||
// Try to get relative path (unless it's a temp/embedded file)
|
||
let relative_path = if is_temp_file || is_embedded {
|
||
None // Don't store path for temp/embedded files, they'll be embedded
|
||
} else if let Some(rel) = pathdiff::diff_paths(file_path, project_dir) {
|
||
Some(rel.to_string_lossy().to_string())
|
||
} else {
|
||
// Fall back to absolute path if relative path fails
|
||
Some(file_path.to_string_lossy().to_string())
|
||
};
|
||
|
||
// Check if we should embed this file
|
||
// Always embed temp files (recordings) and previously embedded audio,
|
||
// otherwise use size threshold
|
||
let embedded_data = if is_temp_file || is_embedded || Self::should_embed(file_path) {
|
||
// Embed from memory - we already have the audio data loaded
|
||
Some(Self::embed_from_memory(file))
|
||
} else {
|
||
None
|
||
};
|
||
|
||
let entry = AudioPoolEntry {
|
||
pool_index: index,
|
||
is_video_audio: false,
|
||
waveform_blob: None,
|
||
name: file_path
|
||
.file_name()
|
||
.map(|n| n.to_string_lossy().to_string())
|
||
.unwrap_or_else(|| format!("file_{}", index)),
|
||
relative_path,
|
||
duration: file.duration_seconds(),
|
||
sample_rate: file.sample_rate,
|
||
channels: file.channels,
|
||
embedded_data,
|
||
media_id: None,
|
||
};
|
||
|
||
entries.push(entry);
|
||
}
|
||
|
||
Ok(entries)
|
||
}
|
||
|
||
/// Check if a file should be embedded (< 10MB)
|
||
fn should_embed(file_path: &Path) -> bool {
|
||
const TEN_MB: u64 = 10_000_000;
|
||
|
||
std::fs::metadata(file_path)
|
||
.map(|m| m.len() < TEN_MB)
|
||
.unwrap_or(false)
|
||
}
|
||
|
||
/// Embed audio from memory (already loaded in the pool)
|
||
fn embed_from_memory(audio_file: &AudioFile) -> EmbeddedAudioData {
|
||
use base64::{Engine as _, engine::general_purpose};
|
||
|
||
// Check if this is a lossy format that should be preserved
|
||
let is_lossy = audio_file.original_format.as_ref().map_or(false, |fmt| {
|
||
let fmt_lower = fmt.to_lowercase();
|
||
fmt_lower == "mp3" || fmt_lower == "ogg" || fmt_lower == "aac"
|
||
|| fmt_lower == "m4a" || fmt_lower == "opus"
|
||
});
|
||
|
||
// Check for preserved original bytes first (from previous load cycle)
|
||
if let Some(ref original_bytes) = audio_file.original_bytes {
|
||
let data_base64 = general_purpose::STANDARD.encode(original_bytes);
|
||
return EmbeddedAudioData {
|
||
data_base64,
|
||
format: audio_file.original_format.clone().unwrap_or_else(|| "wav".to_string()),
|
||
};
|
||
}
|
||
|
||
if is_lossy {
|
||
// For lossy formats, read the original file bytes (if it still exists)
|
||
if let Ok(original_bytes) = std::fs::read(&audio_file.path) {
|
||
let data_base64 = general_purpose::STANDARD.encode(&original_bytes);
|
||
return EmbeddedAudioData {
|
||
data_base64,
|
||
format: audio_file.original_format.clone().unwrap_or_else(|| "mp3".to_string()),
|
||
};
|
||
}
|
||
// If we can't read the original file, fall through to WAV conversion
|
||
}
|
||
|
||
// For lossless/PCM or if we couldn't read the original lossy file,
|
||
// convert the f32 interleaved samples to WAV format bytes
|
||
let wav_data = Self::encode_wav(
|
||
audio_file.data(),
|
||
audio_file.channels,
|
||
audio_file.sample_rate
|
||
);
|
||
|
||
let data_base64 = general_purpose::STANDARD.encode(&wav_data);
|
||
|
||
EmbeddedAudioData {
|
||
data_base64,
|
||
format: "wav".to_string(),
|
||
}
|
||
}
|
||
|
||
/// Encode f32 interleaved samples as WAV file bytes
|
||
fn encode_wav(samples: &[f32], channels: u32, sample_rate: u32) -> Vec<u8> {
|
||
let num_samples = samples.len();
|
||
let bytes_per_sample = 4; // 32-bit float
|
||
let data_size = num_samples * bytes_per_sample;
|
||
let file_size = 36 + data_size;
|
||
|
||
let mut wav_data = Vec::with_capacity(44 + data_size);
|
||
|
||
// RIFF header
|
||
wav_data.extend_from_slice(b"RIFF");
|
||
wav_data.extend_from_slice(&(file_size as u32).to_le_bytes());
|
||
wav_data.extend_from_slice(b"WAVE");
|
||
|
||
// fmt chunk
|
||
wav_data.extend_from_slice(b"fmt ");
|
||
wav_data.extend_from_slice(&16u32.to_le_bytes()); // chunk size
|
||
wav_data.extend_from_slice(&3u16.to_le_bytes()); // format code (3 = IEEE float)
|
||
wav_data.extend_from_slice(&(channels as u16).to_le_bytes());
|
||
wav_data.extend_from_slice(&sample_rate.to_le_bytes());
|
||
wav_data.extend_from_slice(&(sample_rate * channels * bytes_per_sample as u32).to_le_bytes()); // byte rate
|
||
wav_data.extend_from_slice(&((channels * bytes_per_sample as u32) as u16).to_le_bytes()); // block align
|
||
wav_data.extend_from_slice(&32u16.to_le_bytes()); // bits per sample
|
||
|
||
// data chunk
|
||
wav_data.extend_from_slice(b"data");
|
||
wav_data.extend_from_slice(&(data_size as u32).to_le_bytes());
|
||
|
||
// Write samples as little-endian f32
|
||
for &sample in samples {
|
||
wav_data.extend_from_slice(&sample.to_le_bytes());
|
||
}
|
||
|
||
wav_data
|
||
}
|
||
|
||
/// Load audio pool from serialized entries
|
||
///
|
||
/// Returns a list of pool indices that failed to load (missing files).
|
||
/// The caller should present these to the user for resolution.
|
||
pub fn load_from_serialized(
|
||
&mut self,
|
||
entries: Vec<AudioPoolEntry>,
|
||
project_path: &Path,
|
||
) -> Result<Vec<usize>, String> {
|
||
let fn_start = std::time::Instant::now();
|
||
eprintln!("📊 [LOAD_SERIALIZED] Starting load_from_serialized with {} entries...", entries.len());
|
||
|
||
let project_dir = project_path.parent()
|
||
.ok_or_else(|| "Project path has no parent directory".to_string())?;
|
||
|
||
let mut missing_indices = Vec::new();
|
||
|
||
// Clear existing pool
|
||
let clear_start = std::time::Instant::now();
|
||
self.files.clear();
|
||
eprintln!("📊 [LOAD_SERIALIZED] Clear pool took {:.2}ms", clear_start.elapsed().as_secs_f64() * 1000.0);
|
||
|
||
// Size the pool to hold the highest pool_index (slots are addressed by index,
|
||
// so gaps are filled with placeholders). No entries → length 0, NOT 1: the old
|
||
// `max().unwrap_or(0) + 1` produced a spurious placeholder for an empty pool.
|
||
let pool_size = entries.iter()
|
||
.map(|e| e.pool_index + 1)
|
||
.max()
|
||
.unwrap_or(0);
|
||
|
||
// Ensure we have space for all entries
|
||
let resize_start = std::time::Instant::now();
|
||
self.files.resize(pool_size, AudioFile::new(PathBuf::new(), Vec::new(), 2, 44100));
|
||
eprintln!("📊 [LOAD_SERIALIZED] Resize pool to {} took {:.2}ms", pool_size, resize_start.elapsed().as_secs_f64() * 1000.0);
|
||
|
||
for (i, entry) in entries.iter().enumerate() {
|
||
let entry_start = std::time::Instant::now();
|
||
eprintln!("📊 [LOAD_SERIALIZED] Processing entry {}/{}: '{}'", i + 1, entries.len(), entry.name);
|
||
|
||
let success = if entry.is_video_audio {
|
||
// Re-probe the video's audio track via FFmpeg → a streaming
|
||
// VideoAudio entry (keeps full 5.1/7.1; no decode-to-RAM).
|
||
match entry.relative_path.as_ref() {
|
||
Some(rel) => {
|
||
let full = if std::path::Path::new(rel).is_absolute() {
|
||
PathBuf::from(rel)
|
||
} else {
|
||
project_dir.join(rel)
|
||
};
|
||
if full.exists() {
|
||
match crate::audio::disk_reader::VideoAudioReader::open(&full) {
|
||
Ok(reader) => {
|
||
let file = AudioFile::from_video_audio(
|
||
full,
|
||
reader.channels(),
|
||
reader.sample_rate(),
|
||
reader.total_frames(),
|
||
);
|
||
if entry.pool_index < self.files.len() {
|
||
self.files[entry.pool_index] = file;
|
||
true
|
||
} else {
|
||
false
|
||
}
|
||
}
|
||
Err(e) => {
|
||
eprintln!("[AudioPool] Failed to reopen video audio {:?}: {}", full, e);
|
||
false
|
||
}
|
||
}
|
||
} else {
|
||
eprintln!("[AudioPool] Video file not found for audio: {:?}", full);
|
||
false
|
||
}
|
||
}
|
||
None => false,
|
||
}
|
||
} else if entry.media_id.is_some() && entry.embedded_data.is_none() {
|
||
// Packed-in-container streaming entry: build a Compressed placeholder
|
||
// backed by the host blob factory (opened at clip-activation time).
|
||
// No decode here — playback streams through the disk reader.
|
||
let media_id = entry.media_id.clone().unwrap();
|
||
let ext = std::path::Path::new(&entry.name)
|
||
.extension()
|
||
.and_then(|e| e.to_str())
|
||
.map(|s| s.to_lowercase());
|
||
let total_frames = (entry.duration * entry.sample_rate as f64).ceil() as u64;
|
||
let mut file = AudioFile::from_compressed(
|
||
PathBuf::from(&entry.name),
|
||
entry.channels,
|
||
entry.sample_rate,
|
||
total_frames,
|
||
ext,
|
||
);
|
||
file.packed_media_id = Some(media_id);
|
||
if entry.pool_index < self.files.len() {
|
||
self.files[entry.pool_index] = file;
|
||
true
|
||
} else {
|
||
false
|
||
}
|
||
} else if let Some(ref embedded) = entry.embedded_data {
|
||
// Load from embedded data
|
||
eprintln!("📊 [LOAD_SERIALIZED] Entry has embedded data (format: {})", embedded.format);
|
||
match Self::load_from_embedded_into_pool(self, entry.pool_index, embedded.clone(), &entry.name) {
|
||
Ok(_) => {
|
||
eprintln!("[AudioPool] Successfully loaded embedded audio: {}", entry.name);
|
||
true
|
||
}
|
||
Err(e) => {
|
||
eprintln!("[AudioPool] Failed to load embedded audio {}: {}", entry.name, e);
|
||
false
|
||
}
|
||
}
|
||
} else if let Some(ref rel_path) = entry.relative_path {
|
||
// Load from file path
|
||
eprintln!("📊 [LOAD_SERIALIZED] Entry has file path: {:?}", rel_path);
|
||
let full_path = project_dir.join(&rel_path);
|
||
|
||
if full_path.exists() {
|
||
Self::load_file_into_pool(self, entry.pool_index, &full_path).is_ok()
|
||
} else {
|
||
eprintln!("[AudioPool] File not found: {:?}", full_path);
|
||
false
|
||
}
|
||
} else {
|
||
eprintln!("[AudioPool] Entry has neither embedded data nor path: {}", entry.name);
|
||
false
|
||
};
|
||
|
||
if !success {
|
||
missing_indices.push(entry.pool_index);
|
||
}
|
||
|
||
eprintln!("📊 [LOAD_SERIALIZED] Entry {} took {:.2}ms (success: {})", i + 1, entry_start.elapsed().as_secs_f64() * 1000.0, success);
|
||
}
|
||
|
||
eprintln!("📊 [LOAD_SERIALIZED] ✅ Total load_from_serialized time: {:.2}ms", fn_start.elapsed().as_secs_f64() * 1000.0);
|
||
|
||
Ok(missing_indices)
|
||
}
|
||
|
||
/// Load audio from embedded base64 data
|
||
fn load_from_embedded_into_pool(
|
||
&mut self,
|
||
pool_index: usize,
|
||
embedded: EmbeddedAudioData,
|
||
name: &str,
|
||
) -> Result<(), String> {
|
||
use base64::{Engine as _, engine::general_purpose};
|
||
|
||
let fn_start = std::time::Instant::now();
|
||
eprintln!("📊 [POOL] Loading embedded audio '{}'...", name);
|
||
|
||
// Decode base64
|
||
let step1_start = std::time::Instant::now();
|
||
let data = general_purpose::STANDARD
|
||
.decode(&embedded.data_base64)
|
||
.map_err(|e| format!("Failed to decode base64: {}", e))?;
|
||
eprintln!("📊 [POOL] Step 1: Decode base64 ({} bytes) took {:.2}ms", data.len(), step1_start.elapsed().as_secs_f64() * 1000.0);
|
||
|
||
// Write to temporary file for symphonia to decode
|
||
let step2_start = std::time::Instant::now();
|
||
let temp_dir = std::env::temp_dir();
|
||
let temp_path = temp_dir.join(format!("lightningbeam_embedded_{}.{}", pool_index, embedded.format));
|
||
|
||
std::fs::write(&temp_path, &data)
|
||
.map_err(|e| format!("Failed to write temporary file: {}", e))?;
|
||
eprintln!("📊 [POOL] Step 2: Write temp file took {:.2}ms", step2_start.elapsed().as_secs_f64() * 1000.0);
|
||
|
||
// Load the temporary file using existing infrastructure
|
||
let step3_start = std::time::Instant::now();
|
||
let result = Self::load_file_into_pool(self, pool_index, &temp_path);
|
||
eprintln!("📊 [POOL] Step 3: Decode audio with Symphonia took {:.2}ms", step3_start.elapsed().as_secs_f64() * 1000.0);
|
||
|
||
// Clean up temporary file
|
||
let _ = std::fs::remove_file(&temp_path);
|
||
|
||
// Update the path to reflect it was embedded, and preserve original bytes
|
||
if result.is_ok() && pool_index < self.files.len() {
|
||
self.files[pool_index].path = PathBuf::from(format!("<embedded: {}>", name));
|
||
// Preserve the original compressed/encoded bytes so re-save doesn't need to re-encode
|
||
self.files[pool_index].original_bytes = Some(data);
|
||
self.files[pool_index].original_format = Some(embedded.format.clone());
|
||
}
|
||
|
||
eprintln!("📊 [POOL] ✅ Total load_from_embedded time: {:.2}ms", fn_start.elapsed().as_secs_f64() * 1000.0);
|
||
|
||
result
|
||
}
|
||
|
||
/// Load an audio file into a specific pool index
|
||
fn load_file_into_pool(&mut self, pool_index: usize, file_path: &Path) -> Result<(), String> {
|
||
use symphonia::core::audio::SampleBuffer;
|
||
use symphonia::core::codecs::{DecoderOptions, CODEC_TYPE_NULL};
|
||
use symphonia::core::formats::FormatOptions;
|
||
use symphonia::core::io::MediaSourceStream;
|
||
use symphonia::core::meta::MetadataOptions;
|
||
use symphonia::core::probe::Hint;
|
||
|
||
let file = std::fs::File::open(file_path)
|
||
.map_err(|e| format!("Failed to open audio file: {}", e))?;
|
||
|
||
let mss = MediaSourceStream::new(Box::new(file), Default::default());
|
||
|
||
let mut hint = Hint::new();
|
||
if let Some(ext) = file_path.extension() {
|
||
hint.with_extension(&ext.to_string_lossy());
|
||
}
|
||
|
||
let format_opts = FormatOptions::default();
|
||
let metadata_opts = MetadataOptions::default();
|
||
let decoder_opts = DecoderOptions::default();
|
||
|
||
let probed = symphonia::default::get_probe()
|
||
.format(&hint, mss, &format_opts, &metadata_opts)
|
||
.map_err(|e| format!("Failed to probe audio file: {}", e))?;
|
||
|
||
let mut format = probed.format;
|
||
let track = format
|
||
.tracks()
|
||
.iter()
|
||
.find(|t| t.codec_params.codec != CODEC_TYPE_NULL)
|
||
.ok_or_else(|| "No audio track found".to_string())?;
|
||
|
||
let mut decoder = symphonia::default::get_codecs()
|
||
.make(&track.codec_params, &decoder_opts)
|
||
.map_err(|e| format!("Failed to create decoder: {}", e))?;
|
||
|
||
let track_id = track.id;
|
||
let sample_rate = track.codec_params.sample_rate.unwrap_or(44100);
|
||
let channels = track.codec_params.channels.map(|c| c.count()).unwrap_or(2) as u32;
|
||
|
||
let mut samples = Vec::new();
|
||
let mut sample_buf = None;
|
||
|
||
loop {
|
||
let packet = match format.next_packet() {
|
||
Ok(packet) => packet,
|
||
Err(_) => break,
|
||
};
|
||
|
||
if packet.track_id() != track_id {
|
||
continue;
|
||
}
|
||
|
||
match decoder.decode(&packet) {
|
||
Ok(decoded) => {
|
||
if sample_buf.is_none() {
|
||
let spec = *decoded.spec();
|
||
let duration = decoded.capacity() as u64;
|
||
sample_buf = Some(SampleBuffer::<f32>::new(duration, spec));
|
||
}
|
||
|
||
if let Some(ref mut buf) = sample_buf {
|
||
buf.copy_interleaved_ref(decoded);
|
||
samples.extend_from_slice(buf.samples());
|
||
}
|
||
}
|
||
Err(_) => continue,
|
||
}
|
||
}
|
||
|
||
// Detect original format from file extension
|
||
let original_format = file_path.extension()
|
||
.and_then(|ext| ext.to_str())
|
||
.map(|s| s.to_lowercase());
|
||
|
||
let audio_file = AudioFile::with_format(
|
||
file_path.to_path_buf(),
|
||
samples,
|
||
channels,
|
||
sample_rate,
|
||
original_format,
|
||
);
|
||
|
||
if pool_index >= self.files.len() {
|
||
return Err(format!("Pool index {} out of bounds", pool_index));
|
||
}
|
||
|
||
self.files[pool_index] = audio_file;
|
||
Ok(())
|
||
}
|
||
|
||
/// Resolve a missing audio file by loading from a new path
|
||
/// This is called from the UI when the user manually locates a missing file
|
||
pub fn resolve_missing_file(&mut self, pool_index: usize, new_path: &Path) -> Result<(), String> {
|
||
Self::load_file_into_pool(self, pool_index, new_path)
|
||
}
|
||
}
|