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Work on daw backend

This commit is contained in:
Skyler Lehmkuhl 2025-10-18 18:09:07 -04:00
parent 87d2036f07
commit 9414bdcd74
758 changed files with 5749 additions and 0 deletions
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[package]
name = "daw-backend"
version = "0.1.0"
edition = "2021"
[dependencies]
cpal = "0.15"
symphonia = { version = "0.5", features = ["all"] }
rtrb = "0.3"
[dev-dependencies]
[profile.release]
opt-level = 3
lto = true
[profile.dev]
opt-level = 1 # Faster compile times while still reasonable performance

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/// Clip ID type
pub type ClipId = u32;
/// Audio clip that references data in the AudioPool
#[derive(Debug, Clone)]
pub struct Clip {
pub id: ClipId,
pub audio_pool_index: usize,
pub start_time: f64, // Position on timeline in seconds
pub duration: f64, // Clip duration in seconds
pub offset: f64, // Offset into audio file in seconds
pub gain: f32, // Clip-level gain
}
impl Clip {
/// Create a new clip
pub fn new(
id: ClipId,
audio_pool_index: usize,
start_time: f64,
duration: f64,
offset: f64,
) -> Self {
Self {
id,
audio_pool_index,
start_time,
duration,
offset,
gain: 1.0,
}
}
/// Check if this clip is active at a given timeline position
pub fn is_active_at(&self, time_seconds: f64) -> bool {
let clip_end = self.start_time + self.duration;
time_seconds >= self.start_time && time_seconds < clip_end
}
/// Get the end time of this clip on the timeline
pub fn end_time(&self) -> f64 {
self.start_time + self.duration
}
/// Set clip gain
pub fn set_gain(&mut self, gain: f32) {
self.gain = gain.max(0.0);
}
}

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use crate::audio::clip::ClipId;
use crate::audio::pool::AudioPool;
use crate::audio::track::{Track, TrackId};
use crate::command::{AudioEvent, Command};
use std::sync::atomic::{AtomicU64, Ordering};
use std::sync::Arc;
/// Audio engine for Phase 4: timeline with clips and audio pool
pub struct Engine {
tracks: Vec<Track>,
audio_pool: AudioPool,
playhead: u64, // Playhead position in samples
sample_rate: u32,
playing: bool,
channels: u32,
// Lock-free communication
command_rx: rtrb::Consumer<Command>,
event_tx: rtrb::Producer<AudioEvent>,
// Shared playhead for UI reads
playhead_atomic: Arc<AtomicU64>,
// Event counter for periodic position updates
frames_since_last_event: usize,
event_interval_frames: usize,
// Mix buffer for combining tracks
mix_buffer: Vec<f32>,
}
impl Engine {
/// Create a new Engine with communication channels
pub fn new(
sample_rate: u32,
channels: u32,
command_rx: rtrb::Consumer<Command>,
event_tx: rtrb::Producer<AudioEvent>,
) -> Self {
let event_interval_frames = (sample_rate as usize * channels as usize) / 10; // Update 10 times per second
Self {
tracks: Vec::new(),
audio_pool: AudioPool::new(),
playhead: 0,
sample_rate,
playing: false,
channels,
command_rx,
event_tx,
playhead_atomic: Arc::new(AtomicU64::new(0)),
frames_since_last_event: 0,
event_interval_frames,
mix_buffer: Vec::new(),
}
}
/// Add a track to the engine
pub fn add_track(&mut self, track: Track) -> TrackId {
let id = track.id;
self.tracks.push(track);
id
}
/// Get mutable reference to audio pool
pub fn audio_pool_mut(&mut self) -> &mut AudioPool {
&mut self.audio_pool
}
/// Get reference to audio pool
pub fn audio_pool(&self) -> &AudioPool {
&self.audio_pool
}
/// Get a handle for controlling playback from the UI thread
pub fn get_controller(&self, command_tx: rtrb::Producer<Command>) -> EngineController {
EngineController {
command_tx,
playhead: Arc::clone(&self.playhead_atomic),
sample_rate: self.sample_rate,
channels: self.channels,
}
}
/// Process audio callback - called from the audio thread
pub fn process(&mut self, output: &mut [f32]) {
// Process all pending commands
while let Ok(cmd) = self.command_rx.pop() {
self.handle_command(cmd);
}
if self.playing {
// Ensure mix buffer is sized correctly
if self.mix_buffer.len() != output.len() {
self.mix_buffer.resize(output.len(), 0.0);
}
// Clear mix buffer
self.mix_buffer.fill(0.0);
// Convert playhead from samples to seconds for timeline-based rendering
let playhead_seconds = self.playhead as f64 / (self.sample_rate as f64 * self.channels as f64);
// Check if any track is soloed
let any_solo = self.tracks.iter().any(|t| t.solo);
// Mix all active tracks using timeline-based rendering
for track in &self.tracks {
if track.is_active(any_solo) {
track.render(
&mut self.mix_buffer,
&self.audio_pool,
playhead_seconds,
self.sample_rate,
self.channels,
);
}
}
// Copy mix to output
output.copy_from_slice(&self.mix_buffer);
// Update playhead
self.playhead += output.len() as u64;
// Update atomic playhead for UI reads
self.playhead_atomic
.store(self.playhead, Ordering::Relaxed);
// Send periodic position updates
self.frames_since_last_event += output.len() / self.channels as usize;
if self.frames_since_last_event >= self.event_interval_frames / self.channels as usize
{
let position_seconds =
self.playhead as f64 / (self.sample_rate as f64 * self.channels as f64);
let _ = self
.event_tx
.push(AudioEvent::PlaybackPosition(position_seconds));
self.frames_since_last_event = 0;
}
} else {
// Not playing, output silence
output.fill(0.0);
}
}
/// Handle a command from the UI thread
fn handle_command(&mut self, cmd: Command) {
match cmd {
Command::Play => {
self.playing = true;
}
Command::Stop => {
self.playing = false;
self.playhead = 0;
self.playhead_atomic.store(0, Ordering::Relaxed);
}
Command::Pause => {
self.playing = false;
}
Command::Seek(seconds) => {
let samples = (seconds * self.sample_rate as f64 * self.channels as f64) as u64;
self.playhead = samples;
self.playhead_atomic
.store(self.playhead, Ordering::Relaxed);
}
Command::SetTrackVolume(track_id, volume) => {
if let Some(track) = self.tracks.iter_mut().find(|t| t.id == track_id) {
track.set_volume(volume);
}
}
Command::SetTrackMute(track_id, muted) => {
if let Some(track) = self.tracks.iter_mut().find(|t| t.id == track_id) {
track.set_muted(muted);
}
}
Command::SetTrackSolo(track_id, solo) => {
if let Some(track) = self.tracks.iter_mut().find(|t| t.id == track_id) {
track.set_solo(solo);
}
}
Command::MoveClip(track_id, clip_id, new_start_time) => {
if let Some(track) = self.tracks.iter_mut().find(|t| t.id == track_id) {
if let Some(clip) = track.clips.iter_mut().find(|c| c.id == clip_id) {
clip.start_time = new_start_time;
}
}
}
}
}
/// Get current sample rate
pub fn sample_rate(&self) -> u32 {
self.sample_rate
}
/// Get number of channels
pub fn channels(&self) -> u32 {
self.channels
}
/// Get number of tracks
pub fn track_count(&self) -> usize {
self.tracks.len()
}
}
/// Controller for the engine that can be used from the UI thread
pub struct EngineController {
command_tx: rtrb::Producer<Command>,
playhead: Arc<AtomicU64>,
sample_rate: u32,
channels: u32,
}
impl EngineController {
/// Start or resume playback
pub fn play(&mut self) {
let _ = self.command_tx.push(Command::Play);
}
/// Pause playback
pub fn pause(&mut self) {
let _ = self.command_tx.push(Command::Pause);
}
/// Stop playback and reset to beginning
pub fn stop(&mut self) {
let _ = self.command_tx.push(Command::Stop);
}
/// Seek to a specific position in seconds
pub fn seek(&mut self, seconds: f64) {
let _ = self.command_tx.push(Command::Seek(seconds));
}
/// Set track volume (0.0 = silence, 1.0 = unity gain)
pub fn set_track_volume(&mut self, track_id: TrackId, volume: f32) {
let _ = self
.command_tx
.push(Command::SetTrackVolume(track_id, volume));
}
/// Set track mute state
pub fn set_track_mute(&mut self, track_id: TrackId, muted: bool) {
let _ = self.command_tx.push(Command::SetTrackMute(track_id, muted));
}
/// Set track solo state
pub fn set_track_solo(&mut self, track_id: TrackId, solo: bool) {
let _ = self.command_tx.push(Command::SetTrackSolo(track_id, solo));
}
/// Move a clip to a new timeline position
pub fn move_clip(&mut self, track_id: TrackId, clip_id: ClipId, new_start_time: f64) {
let _ = self.command_tx.push(Command::MoveClip(track_id, clip_id, new_start_time));
}
/// Get current playhead position in samples
pub fn get_playhead_samples(&self) -> u64 {
self.playhead.load(Ordering::Relaxed)
}
/// Get current playhead position in seconds
pub fn get_playhead_seconds(&self) -> f64 {
let samples = self.playhead.load(Ordering::Relaxed);
samples as f64 / (self.sample_rate as f64 * self.channels as f64)
}
}

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pub mod clip;
pub mod engine;
pub mod pool;
pub mod track;
pub use clip::{Clip, ClipId};
pub use engine::{Engine, EngineController};
pub use pool::{AudioFile as PoolAudioFile, AudioPool};
pub use track::{Track, TrackId};

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use std::path::PathBuf;
/// Audio file stored in the pool
#[derive(Debug, Clone)]
pub struct AudioFile {
pub path: PathBuf,
pub data: Vec<f32>, // Interleaved samples
pub channels: u32,
pub sample_rate: u32,
pub frames: u64,
}
impl AudioFile {
/// Create a new AudioFile
pub fn new(path: PathBuf, data: Vec<f32>, channels: u32, sample_rate: u32) -> Self {
let frames = (data.len() / channels as usize) as u64;
Self {
path,
data,
channels,
sample_rate,
frames,
}
}
/// Get duration in seconds
pub fn duration_seconds(&self) -> f64 {
self.frames as f64 / self.sample_rate as f64
}
}
/// Pool of shared audio files
pub struct AudioPool {
files: Vec<AudioFile>,
}
impl AudioPool {
/// Create a new empty audio pool
pub fn new() -> Self {
Self {
files: Vec::new(),
}
}
/// 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 number of files in the pool
pub fn file_count(&self) -> usize {
self.files.len()
}
/// Render audio from a file in the pool with sample rate and channel conversion
/// start_time_seconds: position in the audio file to start reading from (in seconds)
/// Returns the number of samples actually rendered
pub fn render_from_file(
&self,
pool_index: usize,
output: &mut [f32],
start_time_seconds: f64,
gain: f32,
engine_sample_rate: u32,
engine_channels: u32,
) -> usize {
if let Some(audio_file) = self.files.get(pool_index) {
// Calculate starting frame position in the source file (frame = one sample per channel)
let src_start_frame = start_time_seconds * audio_file.sample_rate as f64;
// Calculate sample rate conversion ratio (frames)
let rate_ratio = audio_file.sample_rate as f64 / engine_sample_rate as f64;
let src_channels = audio_file.channels;
let dst_channels = engine_channels;
// Render frame by frame
let output_frames = output.len() / dst_channels as usize;
let mut rendered_frames = 0;
for frame_idx in 0..output_frames {
// Calculate the corresponding frame in the source file
let src_frame_pos = src_start_frame + (frame_idx as f64 * rate_ratio);
let src_frame_idx = src_frame_pos as usize;
// Check bounds
if src_frame_idx >= audio_file.frames as usize {
break;
}
// Calculate source sample index (interleaved)
let src_sample_idx = src_frame_idx * src_channels as usize;
// Check bounds for interpolation
if src_sample_idx + src_channels as usize > audio_file.data.len() {
break;
}
// Linear interpolation for better quality
let frac = src_frame_pos - src_frame_idx as f64;
let next_frame_idx = src_frame_idx + 1;
let next_sample_idx = next_frame_idx * src_channels as usize;
let can_interpolate = next_sample_idx + src_channels as usize <= audio_file.data.len() && frac > 0.0;
// Read and convert channels
for dst_ch in 0..dst_channels {
let sample = if src_channels == dst_channels {
// Same number of channels - direct mapping
let ch = dst_ch as usize;
let s0 = audio_file.data[src_sample_idx + ch];
if can_interpolate {
let s1 = audio_file.data[next_sample_idx + ch];
s0 + (s1 - s0) * frac as f32
} else {
s0
}
} else if src_channels == 1 && dst_channels > 1 {
// Mono to multi-channel - duplicate to all channels
let s0 = audio_file.data[src_sample_idx];
if can_interpolate {
let s1 = audio_file.data[next_sample_idx];
s0 + (s1 - s0) * frac as f32
} else {
s0
}
} else if src_channels > 1 && dst_channels == 1 {
// Multi-channel to mono - average all source channels
let mut sum = 0.0f32;
for src_ch in 0..src_channels {
let s0 = audio_file.data[src_sample_idx + src_ch as usize];
let s = if can_interpolate {
let s1 = audio_file.data[next_sample_idx + src_ch as usize];
s0 + (s1 - s0) * frac as f32
} else {
s0
};
sum += s;
}
sum / src_channels as f32
} else {
// Mismatched channels - use modulo for simple mapping
let src_ch = (dst_ch % src_channels) as usize;
let s0 = audio_file.data[src_sample_idx + src_ch];
if can_interpolate {
let s1 = audio_file.data[next_sample_idx + src_ch];
s0 + (s1 - s0) * frac as f32
} else {
s0
}
};
// Mix into output with gain
let output_idx = frame_idx * dst_channels as usize + dst_ch as usize;
output[output_idx] += sample * gain;
}
rendered_frames += 1;
}
rendered_frames * dst_channels as usize
} else {
0
}
}
}
impl Default for AudioPool {
fn default() -> Self {
Self::new()
}
}

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use super::clip::Clip;
use super::pool::AudioPool;
/// Track ID type
pub type TrackId = u32;
/// Audio track for Phase 4 with clips
pub struct Track {
pub id: TrackId,
pub name: String,
pub clips: Vec<Clip>,
pub volume: f32,
pub muted: bool,
pub solo: bool,
}
impl Track {
/// Create a new track with default settings
pub fn new(id: TrackId, name: String) -> Self {
Self {
id,
name,
clips: Vec::new(),
volume: 1.0,
muted: false,
solo: false,
}
}
/// Add a clip to this track
pub fn add_clip(&mut self, clip: Clip) {
self.clips.push(clip);
}
/// Set track volume (0.0 = silence, 1.0 = unity gain, >1.0 = amplification)
pub fn set_volume(&mut self, volume: f32) {
self.volume = volume.max(0.0);
}
/// Set mute state
pub fn set_muted(&mut self, muted: bool) {
self.muted = muted;
}
/// Set solo state
pub fn set_solo(&mut self, solo: bool) {
self.solo = solo;
}
/// Check if this track should be audible given the solo state of all tracks
pub fn is_active(&self, any_solo: bool) -> bool {
!self.muted && (!any_solo || self.solo)
}
/// Render this track into the output buffer at a given timeline position
/// Returns the number of samples actually rendered
pub fn render(
&self,
output: &mut [f32],
pool: &AudioPool,
playhead_seconds: f64,
sample_rate: u32,
channels: u32,
) -> usize {
let buffer_duration_seconds = output.len() as f64 / (sample_rate as f64 * channels as f64);
let buffer_end_seconds = playhead_seconds + buffer_duration_seconds;
let mut rendered = 0;
// Render all active clips
for clip in &self.clips {
// Check if clip overlaps with current buffer time range
if clip.start_time < buffer_end_seconds && clip.end_time() > playhead_seconds {
rendered += self.render_clip(
clip,
output,
pool,
playhead_seconds,
sample_rate,
channels,
);
}
}
rendered
}
/// Render a single clip into the output buffer
fn render_clip(
&self,
clip: &Clip,
output: &mut [f32],
pool: &AudioPool,
playhead_seconds: f64,
sample_rate: u32,
channels: u32,
) -> usize {
let buffer_duration_seconds = output.len() as f64 / (sample_rate as f64 * channels as f64);
let buffer_end_seconds = playhead_seconds + buffer_duration_seconds;
// Determine the time range we need to render (intersection of buffer and clip)
let render_start_seconds = playhead_seconds.max(clip.start_time);
let render_end_seconds = buffer_end_seconds.min(clip.end_time());
// If no overlap, return early
if render_start_seconds >= render_end_seconds {
return 0;
}
// Calculate offset into the output buffer (in interleaved samples)
let output_offset_seconds = render_start_seconds - playhead_seconds;
let output_offset_samples = (output_offset_seconds * sample_rate as f64 * channels as f64) as usize;
// Calculate position within the clip's audio file (in seconds)
let clip_position_seconds = render_start_seconds - clip.start_time + clip.offset;
// Calculate how many samples to render in the output
let render_duration_seconds = render_end_seconds - render_start_seconds;
let samples_to_render = (render_duration_seconds * sample_rate as f64 * channels as f64) as usize;
let samples_to_render = samples_to_render.min(output.len() - output_offset_samples);
// Get the slice of output buffer to write to
if output_offset_samples + samples_to_render > output.len() {
return 0;
}
let output_slice = &mut output[output_offset_samples..output_offset_samples + samples_to_render];
// Calculate combined gain
let combined_gain = clip.gain * self.volume;
// Render from pool with sample rate conversion
// Pass the time position in seconds, let the pool handle sample rate conversion
pool.render_from_file(
clip.audio_pool_index,
output_slice,
clip_position_seconds,
combined_gain,
sample_rate,
channels,
)
}
}

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pub mod types;
pub use types::{AudioEvent, Command};

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use crate::audio::{ClipId, TrackId};
/// Commands sent from UI/control thread to audio thread
#[derive(Debug, Clone)]
pub enum Command {
// Transport commands
/// Start playback
Play,
/// Stop playback and reset to beginning
Stop,
/// Pause playback (maintains position)
Pause,
/// Seek to a specific position in seconds
Seek(f64),
// Track management commands
/// Set track volume (0.0 = silence, 1.0 = unity gain)
SetTrackVolume(TrackId, f32),
/// Set track mute state
SetTrackMute(TrackId, bool),
/// Set track solo state
SetTrackSolo(TrackId, bool),
// Clip management commands
/// Move a clip to a new timeline position
MoveClip(TrackId, ClipId, f64),
}
/// Events sent from audio thread back to UI/control thread
#[derive(Debug, Clone)]
pub enum AudioEvent {
/// Current playback position in seconds
PlaybackPosition(f64),
/// Playback has stopped (reached end of audio)
PlaybackStopped,
/// Audio buffer underrun detected
BufferUnderrun,
}

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use std::path::Path;
use symphonia::core::audio::SampleBuffer;
use symphonia::core::codecs::DecoderOptions;
use symphonia::core::errors::Error;
use symphonia::core::formats::FormatOptions;
use symphonia::core::io::MediaSourceStream;
use symphonia::core::meta::MetadataOptions;
use symphonia::core::probe::Hint;
pub struct AudioFile {
pub data: Vec<f32>,
pub channels: u32,
pub sample_rate: u32,
pub frames: u64,
}
impl AudioFile {
/// Load an audio file from disk and decode it to interleaved f32 samples
pub fn load<P: AsRef<Path>>(path: P) -> Result<Self, String> {
let path = path.as_ref();
// Open the media source
let file = std::fs::File::open(path)
.map_err(|e| format!("Failed to open file: {}", e))?;
let mss = MediaSourceStream::new(Box::new(file), Default::default());
// Create a probe hint using the file extension
let mut hint = Hint::new();
if let Some(extension) = path.extension() {
if let Some(ext_str) = extension.to_str() {
hint.with_extension(ext_str);
}
}
// Probe the media source
let probed = symphonia::default::get_probe()
.format(&hint, mss, &FormatOptions::default(), &MetadataOptions::default())
.map_err(|e| format!("Failed to probe file: {}", e))?;
let mut format = probed.format;
// Find the default audio track
let track = format
.tracks()
.iter()
.find(|t| t.codec_params.codec != symphonia::core::codecs::CODEC_TYPE_NULL)
.ok_or_else(|| "No audio tracks found".to_string())?;
let track_id = track.id;
// Get audio parameters
let codec_params = &track.codec_params;
let channels = codec_params.channels
.ok_or_else(|| "Channel count not specified".to_string())?
.count() as u32;
let sample_rate = codec_params.sample_rate
.ok_or_else(|| "Sample rate not specified".to_string())?;
// Create decoder
let mut decoder = symphonia::default::get_codecs()
.make(&codec_params, &DecoderOptions::default())
.map_err(|e| format!("Failed to create decoder: {}", e))?;
// Decode all packets
let mut audio_data = Vec::new();
let mut sample_buf = None;
loop {
let packet = match format.next_packet() {
Ok(packet) => packet,
Err(Error::ResetRequired) => {
return Err("Decoder reset required (not implemented)".to_string());
}
Err(Error::IoError(e)) if e.kind() == std::io::ErrorKind::UnexpectedEof => {
// End of file
break;
}
Err(e) => {
return Err(format!("Failed to read packet: {}", e));
}
};
// Skip packets for other tracks
if packet.track_id() != track_id {
continue;
}
// Decode the packet
match decoder.decode(&packet) {
Ok(decoded) => {
// Initialize sample buffer on first packet
if sample_buf.is_none() {
let spec = *decoded.spec();
let duration = decoded.capacity() as u64;
sample_buf = Some(SampleBuffer::<f32>::new(duration, spec));
}
// Copy decoded audio to sample buffer
if let Some(ref mut buf) = sample_buf {
buf.copy_interleaved_ref(decoded);
audio_data.extend_from_slice(buf.samples());
}
}
Err(Error::DecodeError(e)) => {
eprintln!("Decode error: {}", e);
continue;
}
Err(e) => {
return Err(format!("Decode failed: {}", e));
}
}
}
let frames = (audio_data.len() / channels as usize) as u64;
Ok(AudioFile {
data: audio_data,
channels,
sample_rate,
frames,
})
}
}

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daw-backend/src/io/mod.rs Normal file
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pub mod audio_file;
pub use audio_file::AudioFile;

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// DAW Backend - Phase 4: Clips & Timeline
//
// A DAW backend with timeline-based playback, clips, and audio pool.
// Supports multiple tracks, mixing, per-track volume/mute/solo, and shared audio data.
// Uses lock-free command queues, cpal for audio I/O, and symphonia for audio file decoding.
pub mod audio;
pub mod command;
pub mod io;
// Re-export commonly used types
pub use audio::{AudioPool, Clip, ClipId, Engine, EngineController, PoolAudioFile, Track, TrackId};
pub use command::{AudioEvent, Command};
pub use io::AudioFile;

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use cpal::traits::{DeviceTrait, HostTrait, StreamTrait};
use daw_backend::{AudioEvent, AudioFile, Clip, Engine, PoolAudioFile, Track};
use std::env;
use std::io::{self, Write};
use std::path::{Path, PathBuf};
use std::sync::Arc;
use std::sync::Mutex;
use std::thread;
use std::time::Duration;
fn main() -> Result<(), Box<dyn std::error::Error>> {
// Get audio file paths from command line arguments
let args: Vec<String> = env::args().collect();
if args.len() < 2 {
eprintln!("Usage: {} <audio_file1> [audio_file2] [audio_file3] ...", args[0]);
eprintln!("Example: {} track1.wav track2.wav", args[0]);
return Ok(());
}
println!("DAW Backend - Phase 4: Clips & Timeline\n");
// Load all audio files
let mut audio_files = Vec::new();
let mut max_sample_rate = 0;
let mut max_channels = 0;
for (i, path) in args.iter().skip(1).enumerate() {
println!("Loading file {}: {}", i + 1, path);
match AudioFile::load(path) {
Ok(audio_file) => {
let duration = audio_file.frames as f64 / audio_file.sample_rate as f64;
println!(
" {} Hz, {} channels, {} frames ({:.2}s)",
audio_file.sample_rate, audio_file.channels, audio_file.frames, duration
);
max_sample_rate = max_sample_rate.max(audio_file.sample_rate);
max_channels = max_channels.max(audio_file.channels);
audio_files.push((
Path::new(path)
.file_name()
.unwrap()
.to_string_lossy()
.to_string(),
PathBuf::from(path),
audio_file,
));
}
Err(e) => {
eprintln!(" Error loading {}: {}", path, e);
eprintln!(" Skipping this file...");
}
}
}
if audio_files.is_empty() {
eprintln!("No audio files loaded. Exiting.");
return Ok(());
}
println!("\nProject settings:");
println!(" Sample rate: {} Hz", max_sample_rate);
println!(" Channels: {}", max_channels);
println!(" Files: {}", audio_files.len());
// Initialize cpal
let host = cpal::default_host();
let device = host
.default_output_device()
.ok_or("No output device available")?;
println!("\nUsing audio device: {}", device.name()?);
// Get the default output config to determine sample format
let default_config = device.default_output_config()?;
let sample_format = default_config.sample_format();
// Create a custom config matching the project settings
let config = cpal::StreamConfig {
channels: max_channels as u16,
sample_rate: cpal::SampleRate(max_sample_rate),
buffer_size: cpal::BufferSize::Default,
};
println!("Output config: {:?} with format {:?}", config, sample_format);
// Create lock-free command and event queues
let (command_tx, command_rx) = rtrb::RingBuffer::<daw_backend::Command>::new(256);
let (event_tx, event_rx) = rtrb::RingBuffer::<AudioEvent>::new(256);
// Create the audio engine
let mut engine = Engine::new(max_sample_rate, max_channels, command_rx, event_tx);
// Add all files to the audio pool and create tracks with clips
let mut track_ids = Vec::new();
let mut clip_info = Vec::new(); // Store (track_id, clip_id, name, duration)
let mut max_duration = 0.0f64;
let mut clip_id_counter = 0u32;
println!("\nCreating tracks and clips:");
for (i, (name, path, audio_file)) in audio_files.into_iter().enumerate() {
let duration = audio_file.frames as f64 / audio_file.sample_rate as f64;
max_duration = max_duration.max(duration);
// Add audio file to pool
let pool_file = PoolAudioFile::new(
path,
audio_file.data,
audio_file.channels,
audio_file.sample_rate,
);
let pool_index = engine.audio_pool_mut().add_file(pool_file);
// Create track
let track_id = i as u32;
let mut track = Track::new(track_id, name.clone());
// Create clip that plays the entire file starting at time 0
let clip_id = clip_id_counter;
let clip = Clip::new(
clip_id,
pool_index,
0.0, // start at beginning of timeline
duration, // full duration
0.0, // no offset into file
);
clip_id_counter += 1;
track.add_clip(clip);
engine.add_track(track);
track_ids.push(track_id);
clip_info.push((track_id, clip_id, name.clone(), duration));
println!(" Track {}: {} (clip {} at 0.0s, duration {:.2}s)", i, name, clip_id, duration);
}
println!("\nTimeline duration: {:.2}s", max_duration);
let mut controller = engine.get_controller(command_tx);
// Wrap engine in Arc<Mutex> for thread-safe access
let engine = Arc::new(Mutex::new(engine));
// Build the output stream
let stream = match sample_format {
cpal::SampleFormat::F32 => build_stream::<f32>(&device, &config, engine)?,
cpal::SampleFormat::I16 => build_stream::<i16>(&device, &config, engine)?,
cpal::SampleFormat::U16 => build_stream::<u16>(&device, &config, engine)?,
_ => return Err("Unsupported sample format".into()),
};
// Start the audio stream
stream.play()?;
println!("\nAudio stream started!");
print_help();
print_status(0.0, max_duration, &track_ids);
// Spawn event listener thread
let event_rx = Arc::new(Mutex::new(event_rx));
let event_rx_clone = Arc::clone(&event_rx);
let _event_thread = thread::spawn(move || {
loop {
thread::sleep(Duration::from_millis(50));
let mut rx = event_rx_clone.lock().unwrap();
while let Ok(event) = rx.pop() {
match event {
AudioEvent::PlaybackPosition(pos) => {
// Clear the line and show position
print!("\r\x1b[K");
print!("Position: {:.2}s / {:.2}s", pos, max_duration);
print!(" [");
let bar_width = 30;
let filled = ((pos / max_duration) * bar_width as f64) as usize;
for i in 0..bar_width {
if i < filled {
print!("=");
} else if i == filled {
print!(">");
} else {
print!(" ");
}
}
print!("]");
io::stdout().flush().ok();
}
AudioEvent::PlaybackStopped => {
print!("\r\x1b[K");
println!("Playback stopped (end of timeline)");
print!("> ");
io::stdout().flush().ok();
}
AudioEvent::BufferUnderrun => {
eprintln!("\nWarning: Buffer underrun detected");
}
}
}
}
});
// Simple command loop
loop {
let mut input = String::new();
print!("\r\x1b[K> ");
io::stdout().flush()?;
io::stdin().read_line(&mut input)?;
let input = input.trim();
// Parse input
if input.is_empty() {
controller.play();
println!("Playing...");
} else if input == "q" || input == "quit" {
println!("Quitting...");
break;
} else if input == "s" || input == "stop" {
controller.stop();
println!("Stopped (reset to beginning)");
} else if input == "p" || input == "play" {
controller.play();
println!("Playing...");
} else if input == "pause" {
controller.pause();
println!("Paused");
} else if input.starts_with("seek ") {
// Parse seek time
if let Ok(seconds) = input[5..].trim().parse::<f64>() {
if seconds >= 0.0 {
controller.seek(seconds);
println!("Seeking to {:.2}s", seconds);
} else {
println!("Invalid seek time (must be >= 0.0)");
}
} else {
println!("Invalid seek format. Usage: seek <seconds>");
}
} else if input.starts_with("volume ") {
// Parse: volume <track_id> <volume>
let parts: Vec<&str> = input.split_whitespace().collect();
if parts.len() == 3 {
if let (Ok(track_id), Ok(volume)) = (parts[1].parse::<u32>(), parts[2].parse::<f32>()) {
if track_ids.contains(&track_id) {
controller.set_track_volume(track_id, volume);
println!("Set track {} volume to {:.2}", track_id, volume);
} else {
println!("Invalid track ID. Available tracks: {:?}", track_ids);
}
} else {
println!("Invalid format. Usage: volume <track_id> <volume>");
}
} else {
println!("Usage: volume <track_id> <volume>");
}
} else if input.starts_with("mute ") {
// Parse: mute <track_id>
if let Ok(track_id) = input[5..].trim().parse::<u32>() {
if track_ids.contains(&track_id) {
controller.set_track_mute(track_id, true);
println!("Muted track {}", track_id);
} else {
println!("Invalid track ID. Available tracks: {:?}", track_ids);
}
} else {
println!("Usage: mute <track_id>");
}
} else if input.starts_with("unmute ") {
// Parse: unmute <track_id>
if let Ok(track_id) = input[7..].trim().parse::<u32>() {
if track_ids.contains(&track_id) {
controller.set_track_mute(track_id, false);
println!("Unmuted track {}", track_id);
} else {
println!("Invalid track ID. Available tracks: {:?}", track_ids);
}
} else {
println!("Usage: unmute <track_id>");
}
} else if input.starts_with("solo ") {
// Parse: solo <track_id>
if let Ok(track_id) = input[5..].trim().parse::<u32>() {
if track_ids.contains(&track_id) {
controller.set_track_solo(track_id, true);
println!("Soloed track {}", track_id);
} else {
println!("Invalid track ID. Available tracks: {:?}", track_ids);
}
} else {
println!("Usage: solo <track_id>");
}
} else if input.starts_with("unsolo ") {
// Parse: unsolo <track_id>
if let Ok(track_id) = input[7..].trim().parse::<u32>() {
if track_ids.contains(&track_id) {
controller.set_track_solo(track_id, false);
println!("Unsoloed track {}", track_id);
} else {
println!("Invalid track ID. Available tracks: {:?}", track_ids);
}
} else {
println!("Usage: unsolo <track_id>");
}
} else if input.starts_with("move ") {
// Parse: move <track_id> <clip_id> <new_start_time>
let parts: Vec<&str> = input.split_whitespace().collect();
if parts.len() == 4 {
if let (Ok(track_id), Ok(clip_id), Ok(time)) =
(parts[1].parse::<u32>(), parts[2].parse::<u32>(), parts[3].parse::<f64>()) {
// Validate track and clip exist
if let Some((_tid, _cid, name, _)) = clip_info.iter().find(|(t, c, _, _)| *t == track_id && *c == clip_id) {
controller.move_clip(track_id, clip_id, time);
println!("Moved clip {} ('{}') on track {} to {:.2}s", clip_id, name, track_id, time);
} else {
println!("Invalid track ID or clip ID");
println!("Available clips:");
for (tid, cid, name, dur) in &clip_info {
println!(" Track {}, Clip {} ('{}', duration {:.2}s)", tid, cid, name, dur);
}
}
} else {
println!("Invalid format. Usage: move <track_id> <clip_id> <time>");
}
} else {
println!("Usage: move <track_id> <clip_id> <time>");
}
} else if input == "tracks" {
println!("Available tracks: {:?}", track_ids);
} else if input == "clips" {
println!("Available clips:");
for (tid, cid, name, dur) in &clip_info {
println!(" Track {}, Clip {} ('{}', duration {:.2}s)", tid, cid, name, dur);
}
} else if input == "help" || input == "h" {
print_help();
} else {
println!("Unknown command: {}. Type 'help' for commands.", input);
}
}
// Drop the stream to stop playback
drop(stream);
println!("Goodbye!");
Ok(())
}
fn print_help() {
println!("\nTransport Commands:");
println!(" ENTER - Play");
println!(" p, play - Play");
println!(" pause - Pause");
println!(" s, stop - Stop and reset to beginning");
println!(" seek <time> - Seek to position in seconds (e.g. 'seek 10.5')");
println!("\nTrack Commands:");
println!(" tracks - List all track IDs");
println!(" volume <id> <v> - Set track volume (e.g. 'volume 0 0.5' for 50%)");
println!(" mute <id> - Mute a track");
println!(" unmute <id> - Unmute a track");
println!(" solo <id> - Solo a track (only soloed tracks play)");
println!(" unsolo <id> - Unsolo a track");
println!("\nClip Commands:");
println!(" clips - List all clips");
println!(" move <t> <c> <s> - Move clip to new timeline position");
println!(" (e.g. 'move 0 0 5.0' moves clip 0 on track 0 to 5.0s)");
println!("\nOther:");
println!(" h, help - Show this help");
println!(" q, quit - Quit");
println!();
}
fn print_status(position: f64, duration: f64, track_ids: &[u32]) {
println!("Position: {:.2}s / {:.2}s", position, duration);
println!("Tracks: {:?}", track_ids);
}
fn build_stream<T>(
device: &cpal::Device,
config: &cpal::StreamConfig,
engine: Arc<Mutex<Engine>>,
) -> Result<cpal::Stream, Box<dyn std::error::Error>>
where
T: cpal::Sample + cpal::SizedSample + cpal::FromSample<f32>,
{
let err_fn = |err| eprintln!("Audio stream error: {}", err);
// Preallocate a large buffer for format conversion to avoid allocations in audio callback
// Size it generously to handle typical buffer sizes (up to 8192 samples = 2048 frames * stereo * 2x safety)
let conversion_buffer = Arc::new(Mutex::new(vec![0.0f32; 16384]));
let stream = device.build_output_stream(
config,
move |data: &mut [T], _: &cpal::OutputCallbackInfo| {
// Lock the engine
let mut engine = engine.lock().unwrap();
// Lock and reuse the conversion buffer (no allocation - buffer is pre-sized)
let mut f32_buffer = conversion_buffer.lock().unwrap();
// Safety check - if buffer is too small, we have a problem
if f32_buffer.len() < data.len() {
eprintln!("ERROR: Audio buffer size {} exceeds preallocated buffer size {}",
data.len(), f32_buffer.len());
return;
}
// Get a slice of the preallocated buffer
let buffer_slice = &mut f32_buffer[..data.len()];
buffer_slice.fill(0.0);
engine.process(buffer_slice);
// Convert f32 samples to output format
for (i, sample) in data.iter_mut().enumerate() {
*sample = cpal::Sample::from_sample(buffer_slice[i]);
}
},
err_fn,
None,
)?;
Ok(stream)
}

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