Files
Psychotoxical-psysonic/src-tauri/crates/psysonic-audio/src/engine.rs
T
cucadmuh 184e87a469 fix(audio): release idle output stream after 60s (#1071) (#1073)
* fix(audio): release idle output stream after 60s (#1071)

Lazy-open CPAL on first playback and close the device handle after one
minute without active audio so Windows can sleep; emit output-released
for cold resume and skip post-wake reopen when idle.

* docs: CHANGELOG and credits for idle audio stream fix (PR #1073)

* fix(audio): satisfy clippy if-same-then-else in idle watcher

* fix(audio): silence rodio DeviceSink drop unless logging is debug

Gate log_on_drop(false) on runtime should_log_debug() so normal/off
logging modes avoid stderr noise from intentional idle stream release.

* feat(audio): cold-start paused restore and silent engine prepare

After getPlayQueue on startup, apply saved seek position to the UI,
prefetch the current track to hot cache, and load the engine paused via
new audio_play startPaused so playback does not audibly start before
pause. Shared engineLoadTrackAtPosition with queue-undo restore.

* fix(audio): satisfy clippy too_many_arguments on stream arm helper

Bundle spawn_legacy_stream_start_when_armed parameters into
LegacyStreamStartWhenArmed so workspace clippy passes.

* fix(audio): release output stream immediately on stop (#1071)

Stop and natural queue end call audio_stop; close the CPAL device right
away instead of waiting for the 60s idle timer. Pause keeps the grace
period for warm resume.

* fix(audio): keep waveform mounted after stop (#1071)

Stop preserves currentTrack, so its cached analysis waveform stays valid.
Stop no longer nulls waveformBins for the still-shown track and re-hydrates
them from the analysis DB, instead of dropping to flat placeholder bars.

* test(audio): cover output_stream_is_needed branches; harden audio_play arg (#1071)

- Add unit tests for the idle-keepalive decision: empty/playing/paused main
  sink, preview and fading-out sinks, and radio playing/paused. Players are
  built device-less via rodio's Player::new + a Zero source, so empty()/state
  are exercised without an audio device.
- Make audio_play's `start_paused` an Option<bool> defaulting to false, so the
  new field is strictly additive (omitting startPaused no longer fails serde).
- Drop the unused `_engine` parameter from start_stream_idle_watcher; it
  resolves the engine from the AppHandle each poll.

* refactor(audio): extract sink-swap lifecycle into sink_swap module (#1071)

Move SinkSwapInputs/swap_in_new_sink and the legacy stream-arm helper
(LegacyStreamStartWhenArmed/spawn_legacy_stream_start_when_armed) out of
play_input.rs into a focused sink_swap.rs, so source selection and source
building stay separate from sink lifecycle. play_input.rs drops from 953 to
799 lines. No behavior change.
2026-06-12 17:13:51 +03:00

562 lines
25 KiB
Rust

//! `AudioEngine` / `AudioCurrent`, stream thread, and HTTP client refresh.
use std::sync::atomic::{AtomicBool, AtomicU32, AtomicU64};
use std::sync::{Arc, Mutex, RwLock};
use std::time::{Duration, Instant};
use rodio::Player;
use super::state::{ChainedInfo, PreloadedTrack, StreamCompletedSpill};
/// Reply channel handed back to the audio-stream thread once an open finishes.
pub type StreamOpenReply =
std::sync::mpsc::SyncSender<(Arc<rodio::MixerDeviceSink>, u32)>;
/// Requests handled on the dedicated audio-stream thread (open / idle release).
pub enum StreamThreadMsg {
Open {
desired_rate: u32,
is_hi_res: bool,
device_name: Option<String>,
reply: StreamOpenReply,
},
Release {
reply: std::sync::mpsc::SyncSender<()>,
},
}
pub struct AudioEngine {
pub stream_handle: Arc<std::sync::Mutex<Option<Arc<rodio::MixerDeviceSink>>>>,
/// Sample rate the output stream was last opened at (updated on every re-open).
pub stream_sample_rate: Arc<AtomicU32>,
/// The rate the device was opened at on cold start — used to restore the
/// stream when Hi-Res is toggled off while a hi-res rate is active.
pub device_default_rate: u32,
/// Open or release the CPAL output stream on the audio-stream thread.
pub stream_thread_tx: std::sync::mpsc::SyncSender<StreamThreadMsg>,
/// User-selected output device name (None = follow system default).
pub selected_device: Arc<Mutex<Option<String>>>,
pub current: Arc<Mutex<AudioCurrent>>,
/// Monotonically incremented on each audio_play (non-chain) / audio_stop call.
pub generation: Arc<AtomicU64>,
pub http_client: Arc<RwLock<reqwest::Client>>,
pub eq_gains: Arc<[AtomicU32; 10]>,
pub eq_enabled: Arc<AtomicBool>,
pub eq_pre_gain: Arc<AtomicU32>,
pub playback_rate: crate::playback_rate::PlaybackRateAtomics,
pub(crate) preloaded: Arc<Mutex<Option<PreloadedTrack>>>,
/// Last fully downloaded manual-stream track bytes (same playback identity),
/// used to recover seek/replay without waiting for network again.
pub(crate) stream_completed_cache: Arc<Mutex<Option<PreloadedTrack>>>,
/// On-disk spill for completed ranged streams above `TRACK_STREAM_PROMOTE_MAX_BYTES`.
pub(crate) stream_completed_spill: Arc<Mutex<Option<StreamCompletedSpill>>>,
/// True when the currently playing source supports seeking (in-memory bytes
/// or `RangedHttpSource`); false for the legacy non-seekable streaming
/// fallback (`AudioStreamReader`). `audio_seek` rejects with a "not
/// seekable" error when false so the frontend restart-fallback can engage.
pub(crate) current_is_seekable: Arc<AtomicBool>,
/// HTTP stream paths (`RangedHttpSource`, legacy `AudioStreamReader`): false
/// until `TRACK_STREAM_PLAY_START_BYTES` are buffered (or download ends).
/// Bytes / local file / radio keep true.
pub(crate) stream_playback_armed: Arc<AtomicBool>,
pub crossfade_enabled: Arc<AtomicBool>,
pub crossfade_secs: Arc<AtomicU32>,
pub fading_out_sink: Arc<Mutex<Option<Arc<Player>>>>,
/// When true, audio_play chains sources to the existing Sink instead of
/// creating a new one, achieving sample-accurate gapless transitions.
pub gapless_enabled: Arc<AtomicBool>,
/// 0=off, 1=replaygain, 2=loudness (future runtime loudness engine).
pub normalization_engine: Arc<AtomicU32>,
/// Target loudness in LUFS for loudness engine (future use).
pub normalization_target_lufs: Arc<AtomicU32>,
/// Extra attenuation (dB) when no loudness DB row exists at decode bind; also seeds streaming heuristics (Settings).
pub loudness_pre_analysis_attenuation_db: Arc<AtomicU32>,
/// Info about the next-up chained track (gapless mode).
/// The progress task reads this when `current_source_done` fires.
pub(crate) chained_info: Arc<Mutex<Option<ChainedInfo>>>,
/// Atomic sample counter — incremented by CountingSource in the audio thread.
/// Progress task reads this for drift-free position tracking.
pub samples_played: Arc<AtomicU64>,
/// Sample rate of the currently playing source (for samples → seconds).
pub current_sample_rate: Arc<AtomicU32>,
/// Channel count of the currently playing source.
pub current_channels: Arc<AtomicU32>,
/// Instant (as nanos since UNIX epoch via Instant hack) of the last gapless
/// auto-advance. Commands arriving within 500 ms are rejected as ghost commands.
pub gapless_switch_at: Arc<AtomicU64>,
/// Active radio session state. None for regular (non-radio) tracks.
/// Dropping the value aborts the HTTP download task via RadioLiveState::Drop.
pub(crate) radio_state: Mutex<Option<crate::stream::RadioLiveState>>,
/// URL last committed to `AudioCurrent` — used so `audio_update_replay_gain` can
/// resolve LUFS / startup trim when the frontend passes `loudnessGainDb: null`
/// (otherwise `compute_gain` would treat that as unity gain and playback "jumps").
pub(crate) current_playback_url: Arc<Mutex<Option<String>>>,
/// Subsonic song id last passed from JS with `audio_play` (trimmed). Used
/// for loudness/waveform cache when the URL is `psysonic-local://…`.
pub(crate) current_analysis_track_id: Arc<Mutex<Option<String>>>,
/// App server id (`playbackServerId ?? activeServerId`) of the current
/// playback, pinned by `audio_play`. Scopes analysis-cache reads (loudness
/// gain, replay-gain updates, device resume) to the right server so a switch
/// can't surface another server's blob for the same bare `track_id`.
pub(crate) current_playback_server_id: Arc<Mutex<Option<String>>>,
/// While a `RangedHttpSource` download task is filling the buffer for this
/// `(track_id, play_generation)`, skip `analysis_enqueue_seed_from_url` for the
/// same id — otherwise a parallel full GET + Symphonia competes with playback
/// decode (ALSA underruns). The ranged task clears this on exit; `gen` avoids a
/// late drop clearing a newer play of the same track.
pub(crate) ranged_loudness_seed_hold: Arc<Mutex<Option<(String, u64)>>>,
/// Secondary sink dedicated to track previews. Runs on the same `OutputStream`
/// as the main sink (rodio mixes both internally) so we don't open a second
/// device handle — important on ALSA-exclusive hardware.
pub(crate) preview_sink: Arc<Mutex<Option<Arc<Player>>>>,
/// Cancel token for the active preview. Bumped on every `audio_preview_play`
/// and `audio_preview_stop` so that orphan timer/progress tasks bail out.
pub(crate) preview_gen: Arc<AtomicU64>,
/// True when `audio_preview_play` paused the main sink and should resume it
/// on preview end. False if the main sink was already paused (or empty).
pub(crate) preview_main_resume: Arc<AtomicBool>,
/// Subsonic song id of the currently playing preview. Echoed back in
/// `audio:preview-end` so the frontend can clear UI state for that row.
pub(crate) preview_song_id: Arc<Mutex<Option<String>>>,
}
pub struct AudioCurrent {
pub sink: Option<Arc<Player>>,
pub duration_secs: f64,
pub seek_offset: f64,
pub play_started: Option<Instant>,
pub paused_at: Option<f64>,
pub replay_gain_linear: f32,
pub base_volume: f32,
/// Crossfade: trigger for sample-level fade-out of the current source.
pub fadeout_trigger: Option<Arc<AtomicBool>>,
/// Crossfade: total fade samples (set before triggering).
pub fadeout_samples: Option<Arc<AtomicU64>>,
}
impl AudioCurrent {
pub fn position(&self) -> f64 {
if let Some(p) = self.paused_at {
return p;
}
if let Some(t) = self.play_started {
let elapsed = t.elapsed().as_secs_f64();
(self.seek_offset + elapsed).min(self.duration_secs.max(0.001))
} else {
self.seek_offset
}
}
}
/// Open an output device at `desired_rate` Hz (0 = device default).
///
/// `device_name`: exact name from `audio_list_devices`. `None` → system default.
/// Falls back to the system default if the named device is not found.
///
/// Resolution order:
/// 1. Exact rate match in the device's supported config ranges.
/// 2. Highest available rate (for hardware that doesn't support the source rate).
/// 3. Device default.
/// 4. System default (last resort).
///
/// Rodio prints a stderr line on every intentional stream drop. Keep that only
/// when runtime logging is in **debug** mode; normal/off silence the noise.
fn finalize_mixer_device_sink(mut handle: rodio::MixerDeviceSink) -> Arc<rodio::MixerDeviceSink> {
if !crate::logging::should_log_debug() {
handle.log_on_drop(false);
}
Arc::new(handle)
}
/// Returns `(stream_handle, actual_sample_rate)`.
fn open_stream_for_device_and_rate(device_name: Option<&str>, desired_rate: u32) -> (Arc<rodio::MixerDeviceSink>, u32) {
use rodio::cpal::traits::{DeviceTrait, HostTrait};
// Suppress ALSA stderr noise while enumerating devices on Unix.
#[cfg(unix)]
let _guard = unsafe {
struct StderrGuard(i32);
impl Drop for StderrGuard {
fn drop(&mut self) { unsafe { libc::dup2(self.0, 2); libc::close(self.0); } }
}
let saved = libc::dup(2);
let devnull = libc::open(c"/dev/null".as_ptr(), libc::O_WRONLY);
libc::dup2(devnull, 2);
libc::close(devnull);
StderrGuard(saved)
};
let host = rodio::cpal::default_host();
// Resolve the target device: explicit name first, then (on Linux) prefer
// a "pipewire" or "pulse" ALSA alias before falling back to cpal's system
// default. On PipeWire-based distros the raw ALSA `default` alias can
// route to a null sink at app-start (issue #234 on Debian 13): the stream
// opens cleanly, progress ticks run, no audio reaches the user. The
// named-alias path goes through pipewire-alsa's real sink and just works.
// On systems where neither alias exists (pure ALSA, macOS, Windows),
// `find_by_name` returns None and we drop through to `default_output_device`
// unchanged — no regression.
let find_by_name = |name: &str| -> Option<_> {
host.output_devices().ok()?.find(|d| {
d.description()
.ok()
.map(|desc| desc.name().to_string())
.as_deref()
== Some(name)
})
};
let device = device_name
.and_then(find_by_name)
.or_else(|| {
#[cfg(target_os = "linux")]
{ find_by_name("pipewire").or_else(|| find_by_name("pulse")) }
#[cfg(not(target_os = "linux"))]
{ None }
})
.or_else(|| host.default_output_device());
if let Some(device) = device {
if desired_rate > 0 {
if let Ok(supported) = device.supported_output_configs() {
let configs: Vec<_> = supported.collect();
// 1. Exact rate match — prefer more channels (stereo > mono).
let exact = configs.iter()
.filter(|c| {
c.min_sample_rate() <= desired_rate
&& desired_rate <= c.max_sample_rate()
})
.max_by_key(|c| c.channels());
if exact.is_some() {
if let Ok(handle) = rodio::DeviceSinkBuilder::from_device(device.clone())
.and_then(|b| b.with_sample_rate(std::num::NonZeroU32::new(desired_rate).unwrap_or(std::num::NonZeroU32::MIN)).open_stream())
{
crate::app_eprintln!("[psysonic] audio stream opened at {} Hz (exact)", desired_rate);
return (finalize_mixer_device_sink(handle), desired_rate);
}
}
// 2. No exact match — use the highest supported rate.
let best = configs.iter()
.max_by_key(|c| c.max_sample_rate());
if let Some(cfg) = best {
let rate = cfg.max_sample_rate();
if let Ok(handle) = rodio::DeviceSinkBuilder::from_device(device.clone())
.and_then(|b| b.with_sample_rate(std::num::NonZeroU32::new(rate).unwrap_or(std::num::NonZeroU32::MIN)).open_stream())
{
crate::app_eprintln!(
"[psysonic] audio stream opened at {} Hz (highest, wanted {})",
rate, desired_rate
);
return (finalize_mixer_device_sink(handle), rate);
}
}
}
}
// 3. Device default.
if let Ok(handle) = rodio::DeviceSinkBuilder::from_device(device.clone()).and_then(|b| b.open_stream()) {
let rate = device
.default_output_config()
.map(|c| c.sample_rate())
.unwrap_or(44100);
crate::app_eprintln!("[psysonic] audio stream opened at {} Hz (device default)", rate);
return (finalize_mixer_device_sink(handle), rate);
}
}
// 4. Last resort: system default.
crate::app_eprintln!("[psysonic] audio stream falling back to system default");
let handle = rodio::DeviceSinkBuilder::open_default_sink()
.expect("cannot open any audio output device");
let rate = rodio::cpal::default_host()
.default_output_device()
.and_then(|d| d.default_output_config().ok())
.map(|c| c.sample_rate())
.unwrap_or(44100);
(finalize_mixer_device_sink(handle), rate)
}
fn probe_device_default_rate() -> u32 {
use rodio::cpal::traits::{DeviceTrait, HostTrait};
rodio::cpal::default_host()
.default_output_device()
.and_then(|d| d.default_output_config().ok())
.map(|c| c.sample_rate())
.unwrap_or(44_100)
}
/// Open the output stream (blocking). Updates `stream_handle` and `stream_sample_rate`.
pub(crate) fn open_output_stream_blocking(
engine: &AudioEngine,
desired_rate: u32,
is_hi_res: bool,
device_name: Option<String>,
) -> Result<Arc<rodio::MixerDeviceSink>, String> {
let rate = if desired_rate > 0 {
desired_rate
} else {
engine.device_default_rate
};
let (reply_tx, reply_rx) = std::sync::mpsc::sync_channel(0);
engine
.stream_thread_tx
.send(StreamThreadMsg::Open {
desired_rate: rate,
is_hi_res,
device_name,
reply: reply_tx,
})
.map_err(|e| e.to_string())?;
let (handle, actual_rate) = reply_rx
.recv_timeout(Duration::from_secs(5))
.map_err(|_| "audio stream open timed out".to_string())?;
engine
.stream_sample_rate
.store(actual_rate, std::sync::atomic::Ordering::Relaxed);
*engine.stream_handle.lock().unwrap() = Some(handle.clone());
Ok(handle)
}
/// Ensure a live output stream exists; lazy-opens on first playback.
pub(crate) fn ensure_output_stream_open(
engine: &AudioEngine,
) -> Result<Arc<rodio::MixerDeviceSink>, String> {
if let Some(handle) = engine.stream_handle.lock().unwrap().clone() {
return Ok(handle);
}
let rate = engine.stream_sample_rate.load(std::sync::atomic::Ordering::Relaxed);
let open_rate = if rate > 0 {
rate
} else {
engine.device_default_rate
};
let device = engine.selected_device.lock().unwrap().clone();
open_output_stream_blocking(engine, open_rate, false, device)
}
pub(crate) fn request_stream_release(engine: &AudioEngine) -> Result<(), String> {
let (reply_tx, reply_rx) = std::sync::mpsc::sync_channel(0);
engine
.stream_thread_tx
.send(StreamThreadMsg::Release { reply: reply_tx })
.map_err(|e| e.to_string())?;
reply_rx
.recv_timeout(Duration::from_secs(5))
.map_err(|_| "audio stream release timed out".to_string())?;
Ok(())
}
pub fn create_engine() -> (AudioEngine, std::thread::JoinHandle<()>) {
// macOS: request a smaller CoreAudio buffer to reduce output latency.
#[cfg(target_os = "macos")]
{
if std::env::var("COREAUDIO_BUFFER_SIZE").is_err() {
std::env::set_var("COREAUDIO_BUFFER_SIZE", "512");
}
}
// Channel: main thread ←→ audio-stream thread (lazy open + idle release).
let (ready_tx, ready_rx) = std::sync::mpsc::sync_channel::<()>(0);
let (stream_thread_tx, stream_thread_rx) =
std::sync::mpsc::sync_channel::<StreamThreadMsg>(4);
let device_default_rate = probe_device_default_rate();
let thread = std::thread::Builder::new()
.name("psysonic-audio-stream".into())
.spawn(move || {
// Set PipeWire / PulseAudio latency hints before the first open.
#[cfg(target_os = "linux")]
{
// Match cpal ALSA ~200 ms headroom: larger quantum reduces underruns when
// the decoder thread catches up after seek or competes with other work.
if std::env::var("PIPEWIRE_LATENCY").is_err() {
std::env::set_var("PIPEWIRE_LATENCY", "8192/48000");
}
if std::env::var("PULSE_LATENCY_MSEC").is_err() {
std::env::set_var("PULSE_LATENCY_MSEC", "170");
}
}
// Thread priority is kept at default during standard-mode playback.
// It is escalated to Max only when a Hi-Res stream reopen is requested,
// to prevent PipeWire underruns at high quantum sizes (8192 frames).
let mut _stream: Option<Arc<rodio::MixerDeviceSink>> = None;
ready_tx.send(()).ok();
while let Ok(msg) = stream_thread_rx.recv() {
match msg {
StreamThreadMsg::Release { reply } => {
_stream = None;
let _ = reply.send(());
}
StreamThreadMsg::Open {
desired_rate,
is_hi_res,
device_name,
reply,
} => {
// Escalate to Max for Hi-Res reopens (large PipeWire quanta need
// real-time scheduling to avoid underruns). No escalation for
// standard mode — the thread blocks on recv() between reopens so
// elevated priority would only waste scheduler budget.
if is_hi_res {
thread_priority::set_current_thread_priority(
thread_priority::ThreadPriority::Max,
)
.ok();
}
_stream = None;
// Scale the PipeWire quantum with the sample rate so wall-clock
// latency stays roughly constant (≈93 ms) at all rates.
#[cfg(target_os = "linux")]
if desired_rate > 0 {
let frames: u32 = if desired_rate > 48_000 { 8192 } else { 4096 };
std::env::set_var("PIPEWIRE_LATENCY", format!("{frames}/{desired_rate}"));
let latency_ms =
(frames as f64 / desired_rate as f64 * 1000.0).round() as u64;
std::env::set_var("PULSE_LATENCY_MSEC", latency_ms.to_string());
}
let (new_stream, actual_rate) =
open_stream_for_device_and_rate(device_name.as_deref(), desired_rate);
let new_handle = new_stream.clone();
_stream = Some(new_stream);
let _ = reply.send((new_handle, actual_rate));
}
}
}
})
.expect("spawn audio stream thread");
ready_rx.recv().expect("audio stream thread ready");
let engine = AudioEngine {
stream_handle: Arc::new(std::sync::Mutex::new(None)),
stream_sample_rate: Arc::new(AtomicU32::new(0)),
device_default_rate,
stream_thread_tx,
selected_device: Arc::new(Mutex::new(None)),
current: Arc::new(Mutex::new(AudioCurrent {
sink: None,
duration_secs: 0.0,
seek_offset: 0.0,
play_started: None,
paused_at: None,
replay_gain_linear: 1.0,
base_volume: 0.8,
fadeout_trigger: None,
fadeout_samples: None,
})),
generation: Arc::new(AtomicU64::new(0)),
http_client: Arc::new(RwLock::new(
reqwest::Client::builder()
.timeout(Duration::from_secs(30))
.use_rustls_tls()
.user_agent(psysonic_core::user_agent::subsonic_wire_user_agent())
.build()
.unwrap_or_default(),
)),
eq_gains: Arc::new(std::array::from_fn(|_| AtomicU32::new(0f32.to_bits()))),
eq_enabled: Arc::new(AtomicBool::new(false)),
eq_pre_gain: Arc::new(AtomicU32::new(0f32.to_bits())),
playback_rate: crate::playback_rate::PlaybackRateAtomics::new(),
preloaded: Arc::new(Mutex::new(None)),
stream_completed_cache: Arc::new(Mutex::new(None)),
stream_completed_spill: Arc::new(Mutex::new(None)),
current_is_seekable: Arc::new(AtomicBool::new(true)),
stream_playback_armed: Arc::new(AtomicBool::new(true)),
crossfade_enabled: Arc::new(AtomicBool::new(false)),
crossfade_secs: Arc::new(AtomicU32::new(3.0f32.to_bits())),
fading_out_sink: Arc::new(Mutex::new(None)),
gapless_enabled: Arc::new(AtomicBool::new(false)),
normalization_engine: Arc::new(AtomicU32::new(0)),
normalization_target_lufs: Arc::new(AtomicU32::new((-16.0f32).to_bits())),
loudness_pre_analysis_attenuation_db: Arc::new(AtomicU32::new((-4.5f32).to_bits())),
chained_info: Arc::new(Mutex::new(None)),
samples_played: Arc::new(AtomicU64::new(0)),
current_sample_rate: Arc::new(AtomicU32::new(0)),
current_channels: Arc::new(AtomicU32::new(2)),
gapless_switch_at: Arc::new(AtomicU64::new(0)),
radio_state: Mutex::new(None),
current_playback_url: Arc::new(Mutex::new(None)),
current_analysis_track_id: Arc::new(Mutex::new(None)),
current_playback_server_id: Arc::new(Mutex::new(None)),
ranged_loudness_seed_hold: Arc::new(Mutex::new(None)),
preview_sink: Arc::new(Mutex::new(None)),
preview_gen: Arc::new(AtomicU64::new(0)),
preview_main_resume: Arc::new(AtomicBool::new(false)),
preview_song_id: Arc::new(Mutex::new(None)),
};
(engine, thread)
}
/// `analysis_enqueue_seed_from_url` should bail while this track's ranged HTTP buffer
/// is still filling — playback will seed on completion with the same bytes.
pub fn ranged_loudness_backfill_should_defer(engine: &AudioEngine, track_id: &str) -> bool {
let tid = track_id.trim();
if tid.is_empty() {
return false;
}
let Ok(g) = engine.ranged_loudness_seed_hold.lock() else {
return false;
};
matches!(&*g, Some((t, _)) if t.as_str() == tid)
}
/// Stops the Rust audio engine cleanly (mirrors the logic in `audio_stop`).
/// Called before process exit on macOS to ensure audio stops immediately.
pub fn stop_audio_engine(app: &tauri::AppHandle) {
use std::sync::atomic::Ordering;
use tauri::Manager;
let engine = app.state::<AudioEngine>();
engine.generation.fetch_add(1, Ordering::SeqCst);
*engine.chained_info.lock().unwrap() = None;
drop(engine.radio_state.lock().unwrap().take());
let mut cur = engine.current.lock().unwrap();
if let Some(sink) = cur.sink.take() { sink.stop(); }
}
/// Subsonic id pinned for the playing source (`audio_play`). Used to prioritize
/// HTTP loudness backfill for the track the user is listening to.
pub fn analysis_track_id_is_current_playback(engine: &AudioEngine, track_id: &str) -> bool {
let needle = track_id.trim();
if needle.is_empty() {
return false;
}
let Ok(guard) = engine.current_analysis_track_id.lock() else {
return false;
};
let Some(cur) = guard.as_deref().map(str::trim).filter(|s| !s.is_empty()) else {
return false;
};
cur == needle
}
pub(crate) fn audio_http_client(state: &AudioEngine) -> reqwest::Client {
state
.http_client
.read()
.map(|c| c.clone())
.unwrap_or_default()
}
pub fn refresh_http_user_agent(state: &AudioEngine, ua: &str) {
let client = reqwest::Client::builder()
.timeout(Duration::from_secs(30))
.use_rustls_tls()
.user_agent(ua)
.build()
.unwrap_or_default();
if let Ok(mut slot) = state.http_client.write() {
*slot = client;
}
}