Files
psysonic/src-tauri/src/audio/commands.rs
T
Psychotoxical ff4271181c diag(audio): trace ranged-stream supersedion + abort paths
Skipping tracks while a RangedHttpSource is still in initial probe leaves
no diagnostic trail today: RangedHttpSource::read returns Ok(0) on
gen-mismatch (silent), ranged_download_task drops out the same way, and
the `dl done` summary only fires under app_deprintln so release users
never see partial/aborted downloads either.

Add focused logs at the bail points without changing behaviour:

- RangedHttpSource::read — log on each Ok(0) return that isn't the normal
  pos>=total_size EOF (superseded before/during wait, download done with
  no bytes ahead of cursor). Symphonia stops reading after Ok(0), so at
  most one log per source, no spam.
- ranged_download_task — log on the gen-mismatch bail with track id +
  gen transition + downloaded/total bytes so we can tell "user skipped
  while downloading" apart from "stream stalled".
- track_download_task — same gen-mismatch log for the legacy
  non-seekable path (consistency with ranged).
- dl-done summary — split into release-visible `[stream] ranged dl
  ABORTED: …` (downloaded < total_size) vs the existing dev-only
  `[stream] dl done` for full completions.
- audio_play — log on both supersedion-bail points around
  select_play_input so a silently-ending audio_play call leaves a trace.

No semantics changed: every existing return / store / branch is intact.
Pure additive logging to make the next reproduction of cucadmuh's
ranged-stream toast diagnosable.
2026-05-09 01:47:02 +02:00

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//! Tauri commands: audio_play / chain_preload / preload + the shared
//! spawn_progress_task helper. Transport (pause/resume/stop/seek), device,
//! radio, mix-mode and AutoEQ commands live in sibling modules.
use std::sync::atomic::{AtomicBool, AtomicU64, Ordering};
use std::sync::Arc;
use std::time::Duration;
use futures_util::StreamExt;
use rodio::Player;
use rodio::Source;
use tauri::{AppHandle, Emitter, State};
use super::decode::build_source;
use super::engine::{audio_http_client, AudioEngine};
use super::helpers::*;
use super::ipc::{maybe_emit_normalization_state, NormalizationStatePayload};
use super::play_input::{
build_source_from_play_input, select_play_input, swap_in_new_sink, url_format_hint,
PlayInputContext, SinkSwapInputs,
};
use super::preview::preview_clear_for_new_main_playback;
use super::progress_task::spawn_progress_task;
use super::state::{ChainedInfo, PreloadedTrack};
// ─── Commands ─────────────────────────────────────────────────────────────────
/// `analysis_track_id`: Subsonic `song.id` from the UI — ties waveform/loudness
/// cache to the track when playing `psysonic-local://` (hot/offline). Optional
/// for HTTP streams (`playback_identity` is used as fallback).
///
/// `stream_format_suffix`: Subsonic `song.suffix` (e.g. m4a); `stream.view` URLs have no
/// file extension, so this helps pick a Symphonia `format_hint` for ranged HTTP.
#[tauri::command]
pub async fn audio_play(
url: String,
volume: f32,
duration_hint: f64,
replay_gain_db: Option<f32>,
replay_gain_peak: Option<f32>,
loudness_gain_db: Option<f32>,
pre_gain_db: f32,
fallback_db: f32,
manual: bool, // true = user-initiated skip → bypass crossfade, start immediately
hi_res_enabled: bool, // false = safe 44.1 kHz mode; true = native rate (alpha)
analysis_track_id: Option<String>,
stream_format_suffix: Option<String>,
app: AppHandle,
state: State<'_, AudioEngine>,
) -> Result<(), String> {
let gapless = state.gapless_enabled.load(Ordering::Relaxed);
// ── Ghost-command guard ───────────────────────────────────────────────────
// After a gapless auto-advance, the frontend may fire a stale playTrack()
// call via IPC. If we're within 500 ms of the last gapless switch AND the
// requested URL matches the already-playing chained track, reject it.
{
let switch_ms = state.gapless_switch_at.load(Ordering::SeqCst);
if switch_ms > 0 {
let now_ms = std::time::SystemTime::now()
.duration_since(std::time::UNIX_EPOCH)
.unwrap_or_default()
.as_millis() as u64;
if now_ms.saturating_sub(switch_ms) < 500 {
// Within the guard window — suppress this ghost command.
return Ok(());
}
}
}
// Cancel any active preview before starting fresh main playback so the
// two sinks don't end up mixed.
preview_clear_for_new_main_playback(&state, &app);
// ── Gapless pre-chain hit ─────────────────────────────────────────────────
// audio_chain_preload already appended this URL to the Sink 30 s in
// advance. The source is live in the queue — just return and let the
// progress task handle the state transition when the previous source ends.
//
// Never for manual skips: the UI already jumped to this track in JS, but
// the current source is still playing until the chain drains. User-initiated
// play must clear the chain and start this URL immediately (standard path).
if gapless && !manual {
let already_chained = state.chained_info.lock().unwrap()
.as_ref()
.map(|c| same_playback_target(&c.url, &url))
.unwrap_or(false);
if already_chained {
return Ok(());
}
}
// ── Standard (new-sink) path ─────────────────────────────────────────────
// Used for: manual skip, gapless OFF, first play, or gapless when the
// proactive chain was not set up in time.
// Bump generation first so the old progress task stops before we peel
// chained_info (avoids a race where it sees current_done + empty chain).
let gen = state.generation.fetch_add(1, Ordering::SeqCst) + 1;
// Manual skip onto the gapless-pre-chained track: reuse raw bytes (no HTTP;
// preload cache was already consumed when the chain was built). Otherwise
// clear any stale chain metadata.
let reuse_chained_bytes: Option<Vec<u8>> = if gapless && manual {
let mut ci = state.chained_info.lock().unwrap();
if ci.as_ref().is_some_and(|c| same_playback_target(&c.url, &url)) {
ci.take().map(|info| {
Arc::try_unwrap(info.raw_bytes).unwrap_or_else(|a| (*a).clone())
})
} else {
*ci = None;
None
}
} else {
*state.chained_info.lock().unwrap() = None;
None
};
// Stop fading-out sink from previous crossfade.
if let Some(old) = state.fading_out_sink.lock().unwrap().take() {
old.stop();
}
// Pin the logical playback URL immediately so `audio_update_replay_gain` (e.g. from
// a fast `refreshLoudness` after `playTrack`) resolves LUFS for **this** track, not
// the previous URL still stored until the sink swap completes.
*state.current_playback_url.lock().unwrap() = Some(url.clone());
let logical_trim = analysis_track_id
.as_ref()
.map(|s| s.trim().to_string())
.filter(|s| !s.is_empty());
*state.current_analysis_track_id.lock().unwrap() = logical_trim.clone();
let cache_id_for_tasks = analysis_cache_track_id(logical_trim.as_deref(), &url);
let format_hint = url_format_hint(&url);
let play_input = match select_play_input(
PlayInputContext {
url: &url,
gen,
duration_hint,
stream_format_suffix: stream_format_suffix.as_deref(),
format_hint: format_hint.as_deref(),
cache_id_for_tasks: cache_id_for_tasks.as_deref(),
reuse_chained_bytes,
},
&state,
&app,
).await? {
Some(input) => input,
None => {
crate::app_deprintln!(
"[audio] audio_play superseded inside select_play_input: gen={} cur={} track_id={:?}",
gen, state.generation.load(Ordering::SeqCst), cache_id_for_tasks
);
return Ok(());
}
};
if state.generation.load(Ordering::SeqCst) != gen {
crate::app_deprintln!(
"[audio] audio_play superseded after select_play_input: gen={} cur={} track_id={:?}",
gen, state.generation.load(Ordering::SeqCst), cache_id_for_tasks
);
return Ok(());
}
let gain_inputs = resolve_track_gain_inputs(&state, &app, &url, logical_trim.as_deref(), loudness_gain_db);
let (gain_linear, effective_volume) = compute_gain(
gain_inputs.norm_mode,
replay_gain_db,
replay_gain_peak,
gain_inputs.effective_loudness_db,
pre_gain_db,
fallback_db,
volume,
);
let current_gain_db = loudness_ui_current_gain_db(gain_linear);
crate::app_deprintln!(
"[normalization] audio_play track_id={:?} engine={} replay_gain_db={:?} replay_gain_peak={:?} loudness_gain_db={:?} gain_linear={:.4} current_gain_db={:?} target_lufs={:.2} volume={:.3} effective_volume={:.3}",
playback_identity(&url),
normalization_engine_name(gain_inputs.norm_mode),
replay_gain_db,
replay_gain_peak,
gain_inputs.cache_loudness_db,
gain_linear,
current_gain_db,
gain_inputs.target_lufs,
volume,
effective_volume
);
maybe_emit_normalization_state(
&app,
NormalizationStatePayload {
engine: normalization_engine_name(gain_inputs.norm_mode).to_string(),
current_gain_db,
target_lufs: gain_inputs.target_lufs,
},
);
// Manual skips (user-initiated) bypass crossfade — the track should start immediately.
let crossfade_enabled = state.crossfade_enabled.load(Ordering::Relaxed) && !manual;
let crossfade_secs_val = f32::from_bits(state.crossfade_secs.load(Ordering::Relaxed)).clamp(0.5, 12.0);
// Measure how much audio Track A actually has left right now.
// By the time audio_play is called, near_end_ticks (2×500ms) + IPC latency
// have consumed ~500800ms from Track A's tail — so its true remaining time
// is always less than crossfade_secs_val. Using the measured remaining time
// for BOTH fade-out (Track A) and fade-in (Track B) keeps them in sync and
// guarantees Track A reaches 0 exactly when its source exhausts.
let actual_fade_secs: f32 = if crossfade_enabled {
let cur = state.current.lock().unwrap();
let remaining = (cur.duration_secs - cur.position()) as f32;
remaining.clamp(0.1, crossfade_secs_val)
} else {
0.0
};
// Fade-in duration for Track B:
// crossfade → equal-power sin(t·π/2) over actual remaining time of Track A
// hard cut → 5 ms micro-fade to suppress DC-offset click
let fade_in_dur = if crossfade_enabled {
Duration::from_secs_f32(actual_fade_secs)
} else {
Duration::from_millis(5)
};
// Build source: decode → trim → resample → EQ → fade-in → fade-out → notify → count.
let done_flag = Arc::new(AtomicBool::new(false));
// Reset sample counter for the new track.
state.samples_played.store(0, Ordering::Relaxed);
let playback_source = build_source_from_play_input(
play_input,
&state,
format_hint.as_deref(),
done_flag.clone(),
fade_in_dur,
hi_res_enabled,
duration_hint,
)
.await
.map_err(|e| {
app.emit("audio:error", &e).ok();
e
})?;
state.current_is_seekable.store(playback_source.is_seekable, Ordering::SeqCst);
let built = playback_source.built;
let source = built.source;
let duration_secs = built.duration_secs;
let output_rate = built.output_rate;
let output_channels = built.output_channels;
// Store the actual output rate/channels for position calculation.
state.current_sample_rate.store(output_rate, Ordering::Relaxed);
state.current_channels.store(output_channels as u32, Ordering::Relaxed);
if state.generation.load(Ordering::SeqCst) != gen {
return Ok(());
}
// ── Stream rate management ────────────────────────────────────────────────
// Hi-Res ON: open device at file's native rate (bit-perfect, no resampler).
// Hi-Res OFF: if the stream was previously opened at a hi-res rate (e.g. the
// toggle was just turned off mid-session), restore the device
// default rate so playback is no longer at 88.2/96 kHz etc.
// If already at the device default — skip entirely (no IPC, no
// PipeWire reconfigure, no scheduler cost).
{
let current_stream_rate = state.stream_sample_rate.load(Ordering::Relaxed);
let target_rate = if hi_res_enabled {
output_rate // native file rate
} else {
state.device_default_rate // restore device default
};
let needs_switch = target_rate > 0 && target_rate != current_stream_rate;
if needs_switch {
let (reply_tx, reply_rx) = std::sync::mpsc::sync_channel::<Arc<rodio::MixerDeviceSink>>(0);
let dev = state.selected_device.lock().unwrap().clone();
if state.stream_reopen_tx.send((target_rate, hi_res_enabled, dev, reply_tx)).is_ok() {
match reply_rx.recv_timeout(std::time::Duration::from_secs(5)) {
Ok(new_handle) => {
*state.stream_handle.lock().unwrap() = new_handle;
state.stream_sample_rate.store(target_rate, Ordering::Relaxed);
// Give PipeWire time to reconfigure at the new rate before
// we open a Sink — only needed for large hi-res quanta.
if hi_res_enabled && target_rate > 48_000 {
tokio::time::sleep(Duration::from_millis(150)).await;
}
}
Err(_) => {
crate::app_eprintln!("[psysonic] stream rate switch timed out, keeping {current_stream_rate} Hz");
}
}
}
}
// Re-check gen: a rapid skip during the settle sleep would have bumped it.
if state.generation.load(Ordering::SeqCst) != gen {
return Ok(());
}
}
let sink = Arc::new(Player::connect_new(state.stream_handle.lock().unwrap().mixer()));
sink.set_volume(effective_volume);
// ── Sink pre-fill for hi-res tracks ──────────────────────────────────────
// At sample rates > 48 kHz the hardware quantum is larger and the first
// period demands more decoded frames than at 44.1/48 kHz.
// Strategy: pause the sink before appending so rodio's internal mixer
// decodes into its ring buffer ahead of the hardware. After a short delay
// we resume — the buffer is already full and the hardware gets its frames
// without an underrun on the very first period.
// Standard mode: no pre-fill needed — default 44.1/48 kHz quantum is small.
let needs_prefill = hi_res_enabled && output_rate > 48_000;
if needs_prefill {
sink.pause();
}
// Gapless OFF: prepend a short silence so tracks are clearly separated.
// Only when this is an auto-advance (near end), not on manual skip.
//
// Use a frame-aligned `SamplesBuffer` rather than `Zero + take_duration` —
// the latter computes its sample count via integer-nanosecond division
// (1_000_000_000 / (sr * ch)), which at common rates leaks an odd number
// of samples (e.g. 44103 at 44.1 kHz / 2 ch / 500 ms = 22051.5 frames).
// The half-frame leak shifts the next source's L/R parity in the device
// stream and can manifest as a dead channel for the rest of the track
// (reported by users for natural-end-without-gapless transitions only).
if !gapless {
let cur_pos = {
let cur = state.current.lock().unwrap();
cur.position()
};
let cur_dur = {
let cur = state.current.lock().unwrap();
cur.duration_secs
};
let is_auto_advance = cur_dur > 3.0 && cur_pos >= cur_dur - 3.0;
if is_auto_advance {
let ch = source.channels();
let sr = source.sample_rate();
// 500 ms in whole frames, then expand to interleaved samples.
let frames = (sr.get() / 2) as usize;
let total_samples = frames.saturating_mul(ch.get() as usize);
let silence = rodio::buffer::SamplesBuffer::new(ch, sr, vec![0f32; total_samples]);
sink.append(silence);
}
}
sink.append(source);
if needs_prefill {
// 500 ms lets rodio decode several seconds of hi-res audio into its
// internal buffer while the sink is paused. The hardware sees no gap
// because the output is held — it only starts draining after sink.play().
// 500 ms gives ~5 quanta of headroom at 8192-frame/88200 Hz quantum size,
// absorbing scheduler jitter and PipeWire graph wake-up latency.
tokio::time::sleep(Duration::from_millis(500)).await;
if state.generation.load(Ordering::SeqCst) != gen {
return Ok(()); // skipped during pre-fill — abort silently
}
sink.play();
}
swap_in_new_sink(&state, SinkSwapInputs {
sink,
duration_secs,
volume,
gain_linear,
fadeout_trigger: built.fadeout_trigger,
fadeout_samples: built.fadeout_samples,
crossfade_enabled,
actual_fade_secs,
});
app.emit("audio:playing", duration_secs).ok();
// ── Progress + ended detection ────────────────────────────────────────────
spawn_progress_task(
gen,
state.generation.clone(),
state.current.clone(),
state.chained_info.clone(),
state.crossfade_enabled.clone(),
state.crossfade_secs.clone(),
done_flag,
app,
state.samples_played.clone(),
state.current_sample_rate.clone(),
state.current_channels.clone(),
state.gapless_switch_at.clone(),
state.current_playback_url.clone(),
);
Ok(())
}
/// Proactively appends the next track to the current Sink ~30 s before the
/// current track ends. Called from JS at the same trigger point as preload.
///
/// Because this runs well before the track boundary, the IPC round-trip is
/// irrelevant — by the time the current track actually ends, the next source
/// is already live in the Sink queue and rodio transitions at sample accuracy.
///
/// audio_play() checks chained_info.url on arrival: if it matches, it returns
/// immediately without touching the Sink (pure no-op on the audio path).
#[tauri::command]
pub async fn audio_chain_preload(
url: String,
volume: f32,
duration_hint: f64,
replay_gain_db: Option<f32>,
replay_gain_peak: Option<f32>,
loudness_gain_db: Option<f32>,
pre_gain_db: f32,
fallback_db: f32,
hi_res_enabled: bool,
analysis_track_id: Option<String>,
app: AppHandle,
state: State<'_, AudioEngine>,
) -> Result<(), String> {
// Idempotent: already chained this track → nothing to do.
{
let chained = state.chained_info.lock().unwrap();
if chained.as_ref().is_some_and(|c| same_playback_target(&c.url, &url)) {
return Ok(());
}
}
// Gapless must be enabled and a sink must exist.
if !state.gapless_enabled.load(Ordering::Relaxed) {
return Ok(());
}
let snapshot_gen = state.generation.load(Ordering::SeqCst);
// Fetch bytes — use preload cache if available, otherwise HTTP.
let data: Vec<u8> = {
let cached = {
let mut preloaded = state.preloaded.lock().unwrap();
if preloaded.as_ref().is_some_and(|p| same_playback_target(&p.url, &url)) {
preloaded.take().map(|p| p.data)
} else {
None
}
};
if let Some(d) = cached {
d
} else {
if let Some(path) = url.strip_prefix("psysonic-local://") {
tokio::fs::read(path).await.map_err(|e| e.to_string())?
} else {
let resp = audio_http_client(&state).get(&url).send().await
.map_err(|e| e.to_string())?;
if !resp.status().is_success() {
return Ok(()); // silently fail — audio_play will retry
}
let hint = resp.content_length().unwrap_or(0) as usize;
let mut stream = resp.bytes_stream();
let mut buf = Vec::with_capacity(hint);
while let Some(chunk) = stream.next().await {
if state.generation.load(Ordering::SeqCst) != snapshot_gen {
return Ok(()); // superseded by manual skip — abort download
}
buf.extend_from_slice(&chunk.map_err(|e| e.to_string())?);
}
buf
}
}
};
// Bail if the user skipped to a different track while we were downloading.
if state.generation.load(Ordering::SeqCst) != snapshot_gen {
return Ok(());
}
let raw_bytes = Arc::new(data);
let logical_trim = analysis_track_id
.as_ref()
.map(|s| s.trim().to_string())
.filter(|s| !s.is_empty());
// Only `gain_linear` is needed — `effective_volume` is intentionally NOT
// applied to the Sink here. `audio_chain_preload` runs ~30 s before the
// current track ends, and `Sink::set_volume` affects the WHOLE Sink (incl.
// the still-playing current source). Volume for the chained track is
// applied at the gapless transition in `spawn_progress_task`, not here.
let gain_inputs = resolve_track_gain_inputs(&state, &app, &url, logical_trim.as_deref(), loudness_gain_db);
let (gain_linear, _effective_volume) = compute_gain(
gain_inputs.norm_mode,
replay_gain_db,
replay_gain_peak,
gain_inputs.effective_loudness_db,
pre_gain_db,
fallback_db,
volume,
);
let done_next = Arc::new(AtomicBool::new(false));
// Use a dedicated counter for the chained source — it will be swapped into
// samples_played when the chained track becomes active.
let chain_counter = Arc::new(AtomicU64::new(0));
// Always 0 — no application-level resampling (same as audio_play).
let target_rate: u32 = 0;
let format_hint = url.rsplit('.').next()
.and_then(|ext| ext.split('?').next())
.map(|s| s.to_lowercase());
let built = build_source(
(*raw_bytes).clone(),
duration_hint,
state.eq_gains.clone(),
state.eq_enabled.clone(),
state.eq_pre_gain.clone(),
done_next.clone(),
Duration::ZERO, // gapless: no fade-in — sample-accurate boundary, no click
chain_counter.clone(),
target_rate,
format_hint.as_deref(),
hi_res_enabled,
).map_err(|e| e.to_string())?;
let source = built.source;
let duration_secs = built.duration_secs;
// Final gen check — reject if a manual skip happened during decode.
if state.generation.load(Ordering::SeqCst) != snapshot_gen {
return Ok(());
}
// In hi-res mode: if the next track's native rate differs from the current
// output stream, we cannot chain gaplessly — audio_play will do a hard cut
// with a stream re-open. Store raw bytes to avoid re-downloading.
// In safe mode (44.1 kHz locked): the stream rate is always 44100, so the
// chain proceeds and rodio resamples internally — no bail needed.
let next_rate = if hi_res_enabled { built.output_rate } else { 44_100 };
let stream_rate = state.stream_sample_rate.load(Ordering::Relaxed);
if hi_res_enabled && stream_rate > 0 && next_rate != stream_rate {
crate::app_eprintln!(
"[psysonic] gapless chain skipped: next track rate {} Hz ≠ stream {} Hz",
next_rate, stream_rate
);
*state.preloaded.lock().unwrap() = Some(PreloadedTrack {
url,
data: Arc::try_unwrap(raw_bytes).unwrap_or_else(|a| (*a).clone()),
});
return Ok(());
}
// Append to the existing Sink. The audio hardware stream never stalls.
// Note: `set_volume` is deliberately NOT called here (see comment above).
{
let cur = state.current.lock().unwrap();
match &cur.sink {
Some(sink) => {
sink.append(source);
}
None => return Ok(()), // playback stopped — bail
}
}
*state.chained_info.lock().unwrap() = Some(ChainedInfo {
url,
raw_bytes,
duration_secs,
replay_gain_linear: gain_linear,
base_volume: volume.clamp(0.0, 1.0),
source_done: done_next,
sample_counter: chain_counter,
});
Ok(())
}