Hydration:
- qbz-plex: additive plex_hydrate_album_quality(base, token, rating_keys) — per
key plex_get_track_metadata (batch 5, 5s/call timeout, skip failures) then the
COALESCE write-back plex_cache_update_track_quality; awaited before re-read.
- qbz-player/qbz-core: patch_plex_quality / patch_plex_queue_quality patch the
in-memory queue in place by rating_key (for the now-playing stamp).
- qbz-slint: album-open trigger (gated on DB-NULL, not a value heuristic) then
refresh of all badge surfaces — album-card grid (re-read aggregate + patch
every model + the raw filter source), album-detail header + rows, now-playing
stamp. Re-sync stays routed through plex_cache_save_tracks (carry-over).
Quality badge consistency (the four surfaces must agree — and pre-hydration
show the filetype, not nothing):
- One shared classifier crate::quality::badge(format, bit_depth, sample_rate) ->
(tier, detail, tooltip); the album card, album-detail header and track rows
all route through it (kills the per-surface duplication that produced
"96 kHz" vs "96000 kHz"). sample_rate normalized to kHz via a >=1000 guard so
Qobuz's already-kHz values are never re-divided (no "0.041 kHz").
- New "lossless" tier: an un-hydrated lossless file (FLAC/WAV/ALAC… with unknown
bit depth) shows a generic filetype badge ("FLAC") instead of an empty badge,
rendered in QualityBadge (compact chip) + QualityBadgeFull; upgrades to
"24-bit / 96 kHz" after hydration. Local/Qobuz/mp3 unchanged.
cc
- badge: gate is-remote on transport_connected so it clears on every session-end
path (transport drop / watchdog / reconnect-exhausted), not only toggle-off
- multi-track: route a whole album/playlist/favorites enqueue to the peer in one
QueueAddTracks/QueueInsertTracks (all-or-nothing admission), not single-track only
- shuffle inbound: SetShuffleMode renderer arm is flag-only (set_shuffle_flag) so the
renderer never invents a local order; the cloud's echoed order is authoritative
- shuffle outbound: send shuffle_seed + shuffle_pivot_queue_item_id (the cloud needs
the seed to build the order); hoist resolve_qconnect_shuffle_pivot to qconnect-app
- start index: resolve_remote_start_index validates the track_id before trusting a
qid, so a freshly-pushed album no longer inherits a stale cursor (wrong NowPlayingBar track)
- history: play_index records history only when actually moving, so the current track
is no longer duplicated in the History tab
- pause-to-peer: resolve current_queue_item from the core-aligned cursor instead of the
stale renderer snapshot, so pausing an iOS renderer (reports null qid) is accepted
cc
The poll loop's seamless-transition branch guessed whether a track-id change
while still playing was a gapless hand-off via a peek_upcoming(1) == track_id
heuristic. When that heuristic missed (queue pointer lagging the engine), the
meta was never refreshed: the seek bar kept advancing while the title/art/album
stayed on the previous track (the reported now-playing populate bug).
Replace the heuristic with reconciliation to what is ACTUALLY playing. New
frontend-agnostic primitive QueueManager::sync_current_to_id(id) moves the queue
pointer to the track the engine advanced to (recording history + keeping the
shuffle cursor aligned) and reports whether it moved; QbzCore wraps it and emits
a queue update only on a real move. The poll loop now syncs to the live track_id
and always refreshes the card, doing the record/sidebar/prefetch bookkeeping only
on a genuine advance (a manual play already moved the pointer, so it is idempotent).
cc
After bit-perfect ALSA Direct grabs a DAC exclusively (hw: + D-Bus
reservation), the device stays sequestered and is invisible to PipeWire,
so switching back to the PipeWire backend can't see it without restarting
the app. Tauri had a refresh affordance for hot-plugged DACs; this is the
refresh plus an explicit release, in one button.
Backend: Player::release_device() / AudioCommand::ReleaseDevice tears the
stream down immediately (cancelling the deferred drop), freeing the hw:
grab + its reservation, then un-suspends the parked sink and resets any
forced graph clock (both self-gating). It does NOT reopen, unlike
reinit_device which would re-grab the device.
UI: a refresh-cw icon button beside the Output device select, propagated
through the settings callback chain to handle_release_device, which
releases, waits for PipeWire to reclaim, then re-enumerates so a freed or
hot-plugged DAC reappears without an app restart.
cc
Completes the JACK output on top of the foundation (fbc03f41):
- StreamType::Jack + the player dispatch (backend==Jack -> JackStream).
- PlaybackEngine::Jack: a feeder thread (mirrors AlsaDirect) that resamples each
source to the JACK graph rate (UniformSourceIterator, stereo) and writes it to
a lock-free f32 SPSC ring (ringbuf); wired at the 4 engine-creation sites.
- jack_backend.rs: f32-native ring (fixes a byte-alignment white-noise bug) plus
auto-connect of qbz:out_FL/FR to the physical playback on activation, so audio
reaches the hardware without a patchbay (a user may re-patch freely).
- Re-enabled the available_backends push (JACK is now offered in the selector).
Verified on hardware: select JACK -> QBZ registers as client 'qbz' with stable
ports qbz:out_FL/FR (client.api=jack) -> music plays through the DAC, resampled
to the graph rate (NOT bit-perfect; BitPerfectMode::Disabled, as designed). The
ALSA-exclusive / DAC-passthrough bit-perfect paths are untouched.
Slice 3.2 (pending, separate): JACK<->passthrough/exclusive mutual-exclusion UI,
the bit-perfect-OFF note, runtime graph-rate-change callback, mono/multichannel.
cc
Exclusive-mode device acquisition suspends the default PipeWire sink
(pactl suspend-sink @DEFAULT_SINK@ 1) on the EBUSY-retry and CPAL-exclusive
paths, but never resumed it — leaving the rest of the system on a suspended
sink after exclusive playback.
Add suspend_default_sink_for_exclusive() (resolves + records the suspended
sink name) and resume_suspended_sink() (resumes that exact name; PipeWire sink
names are deterministic, so resume-by-name survives a vacate/re-create). The
player resumes on the pause-suspend teardown — after the exclusive stream is
actually dropped, never while the device is still held. Self-gating: a no-op
unless QBZ suspended a sink, so it is safe to call on every stop.
Does not touch the bit-perfect path. Verified no regression on hardware:
exclusive mode plays through the DAC (LED-confirmed) and system audio uses the
onboard sink (not stuck suspended). The suspend/resume paths themselves
(EBUSY race / CPAL-exclusive fallback) are rare and were not exercised in that
test; the fix is correct-by-construction and cannot affect bit-perfect.
cc
QBZ forces the global PipeWire clock for any non-locked PipeWire stream, but the
stop/pause-suspend reset was gated on pw_force_bitperfect — which is only set in
DAC-passthrough mode. So a plain PipeWire user (no passthrough) forced the graph
clock on play and never reset it, leaving the whole graph force-clocked after
stopping (a stale clock.force-rate that resamples every other app).
Track whether QBZ actually wrote the force (CLOCK_FORCE_APPLIED) and make
reset_pipewire_clock self-gating: it only resets a force WE applied, never
clobbering another app's forced rate. The player's two reset call sites (Stop,
pause-suspend) now call it unconditionally.
Verified on hardware (Cambridge USB DAC, PipeWire + passthrough off): a 192kHz
track forces clock.force-rate=192000; on pause-suspend the clock returns to 0
(previously stayed stuck at 192000).
cc
A Local Library album, or the Tracks list, now plays through with auto-advance
across mixed local/offline sources.
The one change to the shared playback path is additive and provably safe for
Qobuz: play_audible gains a guard at the top — when the CURRENT queue track's
source is "local" (and its id matches), it plays the file from disk via
play_local_file_audible (read -> player.play_data + CUE seek); otherwise it
falls through to the EXISTING offline/Qobuz tier-walk, byte-for-byte unchanged.
Qobuz/offline queue tracks never carry source "local", so their behavior cannot
change. Auto-advance, next/previous and play-all all route through play_audible,
so they become source-aware for free. current_track() releases the queue read
lock before play (no nested lock / deadlock).
- qbz-player: QueueManager::current() (additive).
- qbz-core: QbzCore::current_track() (additive).
- qbz-slint playback.rs: source-aware play_audible + play_local_file_audible
(extracted) + play_local_tracks_now + play_local_album
(get_album_tracks_metadata) + play_local_tracks_from (search_with_filter);
replaces the Stage-1 single-track helper.
- main.rs: album card "play" -> play the album; Tracks row "play" -> play from
that row down the searched list. (play-next / queue still stubbed.)
Built green (cargo check). Local-file gapless isn't wired (local tracks aren't
prefetched, so local/mixed transitions use the simple track-ended -> next path);
Qobuz gapless is unchanged. Runtime smoke-test pending: (1) Qobuz streaming
still plays + advances, (2) an offline album advances, (3) a local album/list
advances.
cc
OOTB playback froze (seekbar advancing, no audio) on hosts where the
configured backend was not usable — e.g. PipeWire as the Linux default but
`pactl` absent (minimal Debian/Ubuntu). "Auto" (a null backend_type) fell
through to a legacy CPAL path that forced the track rate onto the shared
default device, producing a suspended/silent stream and leaving a
process-wide stuck audio handle that survived Reset.
- Add a cross-platform "System" backend (CpalDefaultBackend now on Linux
too): opens the OS default output at its negotiated rate with rodio
resampling and an explicit ~100ms CPAL buffer (MixerDeviceSink has zero
internal buffering, so the bare default underran and stayed suspended).
Shared, no exclusivity, no `pactl` — mixes with other apps like any
normal player.
- Make System the OOTB default and what Reset lands on; seed it on first
run. Drop the per-startup #375 NULL->PipeWire backfill migration.
- Resolve a null backend_type to System in the player so the legacy CPAL
path is never reached (covers legacy installs and headless callers).
- "Auto" in Settings is now a resolve-and-set action: it picks the best
available backend (PipeWire if available, else System), stores that
concrete value and updates the dropdown + device picker, so backend_type
is never null.
- Explicit PipeWire (DAC passthrough) and ALSA Direct (exclusive) configs
are untouched and remain bit-perfect.
cc
The stream-recreate decision only forced a rebuild on a decoded-format change
for bit-perfect backends (DAC passthrough, ALSA Direct, CoreAudio exclusive).
The default/PipeWire path relied on Stop dropping the stream to get a fresh one
at the new rate; once that drop was deferred to avoid a track-change click
(e93fcaec), the PipeWire node stayed locked to the first track's rate for all
subsequent tracks — a regression vs 1.2.10. Make any decoded-format change
rebuild the stream on every backend; same-rate tracks still reuse it, so the
click fix is preserved.
Verified on hardware via pw-top: PipeWire without passthrough now switches rate
per track; PipeWire with passthrough and ALSA Direct unchanged (no regression).
cc
Recovered from 63ecc324 (authored on the native-port branch, never merged to
pre-release). Backend-only, no Slint. Scales the prefetch cap down on slow
links and during audio underruns, and exposes the per-segment download signal
the offline detector reuses as a positive liveness override (#467).
cc
Add three public Player methods that drive gapless playback without
touching the audio thread:
- prefetch_into_cache: download an upcoming track fully into L1/L2
cache so it later plays via play_data (cache hit) rather than the
streaming path. CMAF download_full first, legacy full download as
fallback. No-ops on streaming_only / already-cached / in-flight.
- is_track_cached: cheap L1/L2 membership check for the controller.
- fetch_for_gapless: resolve a track's bytes for a gapless handoff
via L1 memory then L2 disk then CMAF download_full. Ports the
L1/L2/CMAF tiers of Tauri's v2_play_next_gapless.
All three reuse existing primitives (audio_cache, download_audio);
the protected audio-thread command handlers are untouched.
cc
QueueManager::get_state_full() returns the upcoming and history lists
without the 20/10 caps that get_state() applies, so clients can
paginate the upcoming list. QbzCore exposes get_queue_state_full()
plus favorite_track_ids() and set_track_favorite() for reflecting and
toggling per-track favorite state.
cc
play_track now checks the qbz-cache two-level cache (L1 memory ~400 MB
+ L2 disk ~800 MB) before fetching: a hit replays through play_data,
skipped only when the cached copy is a lower quality than requested
(cached_quality_below_requested, ported from Tauri). After a play the
track is stored — the CMAF path accumulates the assembled FLAC while
streaming, the legacy path stores the download. streaming_only
suppresses the write.
cc
The CMAF path passed duration_secs: 0 into play_streaming_dynamic, and
the streaming position timer clamps current_position to the duration
it was given — so elapsed, remaining, progress and auto-advance all
froze at zero. Duration is now derived from the CMAF segment table's
per-segment sample counts.
PlaybackEvent also gained buffer_progress (the streaming buffer fill
the audio thread already tracked but never surfaced); the poll loop
feeds it to the seek-bar cache overlay.
cc
Port the CMAF streaming playback path from the Tauri v2_play_track /
v2_cmaf_stream reference into the qbz-player core crate so any shell
gets CMAF transparently.
play_track now tries qbz_qobuz::cmaf::setup_streaming first: it fetches
only the init segment, derives sample rate / bit depth / total FLAC
size, estimates speed from init_fetch_ms, starts playback via the
existing play_streaming_dynamic, and spawns a background task that
fetches + decrypts + pushes audio segments. On setup failure it falls
back to the legacy get_stream_url_with_fallback + download_audio +
play_data path, which is kept unchanged.
The new private cmaf_stream_segments method ports the v2_cmaf_stream
fetch/decrypt/push loop: FLAC header first, per-segment reqwest get,
parse_segment_crypto, decrypt_frame for encrypted frames, push_chunk
to the streaming buffer, complete() at the end. The L1/L2 cache
parameters and cache calls from the reference are intentionally
omitted (separate later phase).
play_track's public signature and the bit-perfect output path are
unchanged.
cc
Three bugs were stacking up to make session resume drop the saved offset
for streamed Qobuz tracks:
1. Position save was debounced, not throttled. Each playback tick reset
the 5s timer, so the save never fired while a track played
continuously — the persisted position froze at whatever value got
written during the first quiet window after track start (typically
~2s). Switch to throttle: fire the first call immediately, then once
per 5s window, with a tail call to flush the latest value when the
window ends.
2. togglePlay tried to apply the restore offset via a separate cmdSeek
right after cmdPlayTrack. For Qobuz streaming the audio thread's
Seek handler bails when current_audio_data is unset — streaming
uses current_streaming_source instead — so the seek was silently
dropped and the track started at 0:00. Pass start_position_secs
directly to v2_play_track. The streaming backend (legacy + V2
player) waits for enough buffer to cover the offset plus 8s
headroom, then eagerly decode-and-discards decoder output up to
the offset before engaging the audio engine — no race, no
first-pull underrun. Plex and local tracks keep the old play+seek
path; they load full audio_data so Seek works as-is.
3. The seekbar showed 0:00 from app launch until the user hit play,
even when a saved position was queued. setPendingSessionRestore
now pre-positions currentTime when positionSecs > 0 so the UI
reflects "paused at X" the moment the app opens.
Other plumbing:
- AudioCommand::PlayStreaming carries start_position_secs and
content_length (used to estimate bytes-per-second for the resume
buffer target).
- Player::play_streaming{,_dynamic} grow a start_position_secs
parameter; QConnect and api_server callers pass 0 (remote-driven
playback always starts fresh).
- cmdPlayTrack forwards startPositionSecs as a 5th optional arg.
cc
Clearing the error flag also drops any pending stream_error_message.
A fast record_stream_error -> set_stream_error(false) -> polling-tick
sequence will silently swallow the toast — that is the intent (a
recovered error is not user-actionable), but it surprises future
readers, so document it inline.
Pairs the existing stream_error AtomicBool with an Option<String> that
the audio thread can populate when a stream init failure has a
meaningful cause to surface to the user (currently: the macOS
CoreAudio rate-mismatch path).
set_stream_error(false) also clears the message; record_stream_error
sets both atomically; take_stream_error_message drains so the UI
fires its notification exactly once per error.
evaluate_stream_recreate runs on every track change on the audio
thread, and each call issued two HAL syscalls
(resolve_output_device_id + get_nominal_sample_rate). Cache the result
per-thread for 750ms — long enough to absorb the bulk of repeats
within an album, short enough to notice a system rate change.
Cache invalidates on device-name change, so switching the output
device still triggers a fresh query.
The error strings flow up through stream_error and end up in user-
visible logs. Replace the internal-shaped messages ("opened at XHz
while device nominal rate is YHz") with text that names the cause,
the remediation, and the reason the fallback path is refused.
Extract the boolean decision out of evaluate_stream_recreate into a
pure function so the truth table can be tested without faking a
MixerDeviceSink or AudioSettings mutex. Eight cases cover the matrix:
no stream, unchanged format, format change with and without each
bit-perfect backend, and the CoreAudio shared-mode rate-mismatch path.
The rate-mismatch detection block was duplicated between the Play and
Stream handlers, and each handler acquired thread_settings.lock() three
times in a row. Extract into one helper that reads settings once and
returns every flag the recreate decision needs.
Also drops a dead else-branch that logged "Creating initial audio
stream" — unreachable inside if stream_opt.is_some() — and the
redundant info-log that re-announced the rate-mismatch reason at
recreate time (the warn-level log inside the helper already says
"Recreating stream to avoid wrong-speed playback").
The field tracks the decoded source format, not the output stream's
hardware rate. The names are now explicit so future readers don't
conflate the two — on macOS shared mode these can legitimately differ
since Rodio resamples internally.
Address review feedback from the macOS Exclusive Mode PR.
Pin the resolved CoreAudio output device before acquiring Hog Mode so
System Default cannot drift between Hog Mode acquisition, sample-rate
switching, and stream creation.
Fall back to software volume when CoreAudio hardware volume is unavailable,
which keeps the volume slider usable on knob-only DACs.
Snapshot and restore CoreAudio hardware volume around Exclusive Mode so
leaving Exclusive Mode does not leave the output device at QBZ's last
hardware volume.
Refuse shared-mode fallback when macOS Exclusive Mode stream creation
fails, so the app does not silently play in non-exclusive mode while the UI
still reports Exclusive Mode.
Clean up Linux unused-variable warnings from the macOS-only guard plumbing.
The audio-thread iteration that swaps in the next gapless track
mutates several atomics in qbz-player::player::SharedState. The
polling loop in lib.rs reads those atomics independently (one load
per field) at ~250 ms cadence and emits a PlaybackEvent that the
frontend handlePlaybackEvent then classifies. When the polling load
falls inside the swap window, it can observe an inconsistent mix:
current_track_id already advanced to the new track, but
gapless_next_track_id still pointing at that same track because the
reset hasn't run yet.
The frontend's isGaplessTransition predicate
(playerStore.ts) requires both `event.track_id !== currentTrack.id`
AND `event.gapless_next_track_id === 0`. The mid-swap event
satisfies the first and fails the second, so the event is mis-
classified as an external track change. The "Track changed
externally" branch then asks the queue manager for the current
queue track (still the previous track because the gapless flow
hasn't called v2_next_track yet) and prints the now-familiar
"Queue track mismatch or null: 212171 vs event: 216597" warn.
In 1.2.8 this race was self-recovering: subsequent ticks observed
the consistent post-swap state and isGaplessTransition fired one
tick later. dee987c1a tightened the gate with
`gaplessAttemptedForCurrent`, which gets reset to null by the
mid-swap event's external-track path; subsequent ticks then fail
the new gate and the misclassification becomes permanent for the
remainder of the track. UI stays on the previous title while the
audio plays the new one, and the queue cursor never advances —
visible to the user as "the player says it's still on Viento but
Afuera is playing".
Fix the race at the producer side: in both transition swap paths
(mainline pos>=dur detection and ALSA Direct take_source_transition),
clear gapless_next_track_id and gapless_ready BEFORE storing the
new current_track_id. With SeqCst ordering, any reader that
observes the new track_id is guaranteed to also observe the
cleared gapless slot — eliminating the inconsistent intermediate
state by construction. Cheaper than coordinating reads on the
consumer side and keeps the polling loop's lock-free design.
cc
Two regressions ride together when gapless plays a chain of more than
two tracks:
- The pos/dur snapshot at the top of the audio-thread iteration is
captured BEFORE the gapless transition swap. After the swap
updates current_track_id and duration, the local pos/dur still
point at the outgoing track, and the "approaching end" check a
few lines below fires immediately for the new track with stale
299/299-style values.
- That spurious firing arms gapless_request_armed=true and sets
gapless_ready=true. gapless_request_armed only gets cleared again
on a future transition (which will never happen because the new
track has no prefetch queued) or on engine empty (i.e. when
playback has already stopped). gapless_ready also stays sticky-
true because nothing else clears it once the false-positive fires.
- End result: the new track plays through with the gate stuck
closed, the real "approaching end" check at ~T-10s never fires,
the frontend never gets prompted to call play_next_gapless for
track #N+1, and the audio thread reaches engine empty and stops.
Fix:
- Track whether a transition swap consumed pending in this
iteration via a local transition_consumed_pending bool. Skip the
"approaching end" check while that flag is set — the next
iteration will run with fresh pos/dur for the new track and
fire the trigger correctly.
- Reset gapless_ready=false in both transition swap paths
(mainline and ALSA Direct), mirroring the existing
gapless_request_armed=false reset, so neither flag survives a
transition.
cc
The marker becomes part of the queue state snapshot the frontend
already consumes via syncQueueState(). Single source of truth — no
parallel event channel, no separate store on the frontend.
Refs #356.
cc
Removes all tracks at indices > pivot, preserving the pivot itself.
Returns the count removed. Updates shuffle_order and history to
drop dangling indices. Auto-invalidates the stop-after marker if it
referenced any removed track.
Refs #356.
cc
set_queue, set_queue_with_order, and clear unconditionally drop the
marker (queue replaced/emptied → marked track may not exist anymore).
remove_track and remove_upcoming_track conditionally clear the marker
when the removed track matches it. move_track and add_* are unchanged
— a moved track is still the same track ID, and adding doesn't touch
existing tracks.
Refs #356.
cc
Auto-advance drivers (frontend natural-end handlers) call this with
the just-finished track ID. If it matches the marker, returns true
and clears the marker; the caller pauses instead of advancing.
Otherwise returns false — caller proceeds with normal advance.
Manual skip paths must NOT call this method.
Refs #356.
cc
Add `stop_after_track_id: Option<u64>` to InternalState plus three
methods on QueueManager: `set_stop_after(id)`, `clear_stop_after()`,
`get_stop_after()`. Setting on an ID not present in the queue is a
silent no-op so the marker can never reference a phantom track.
Refs #356.
cc
set_queue and set_queue_with_order unconditionally cleared history, which
wiped legitimate plays whenever QConnect echoed a queue version bump
(reorder, shuffle toggle, peer-driven snapshot) without changing track
identity. That is the missing-tracks half of issue #316.
Instead, remap history entries by track id: each pre-existing index is
resolved to its track id via the old tracks list, then looked up in the
new tracks list. Entries whose track id is no longer present are dropped.
When the new queue has zero overlap with the old, history clears
naturally; when the queue is a pure reorder, history survives intact.
Six unit tests cover pure reorder, additions, partial removals,
disjoint replacement, and verify the remapped indices (not just rendered
ids) point at the right new tracks.
cc
StreamingConfig::from_speed_mbps inflates the initial buffer to 2 MB
on slow connections (< 1 MB/s) to absorb network jitter. On a healthy
desktop that's fine, but on a memory-constrained host the slow link
is itself a symptom: when QBZ on a Pi 3B starts allocating ~300 MB
of Hi-Res prefetch the kernel swaps, networking stalls, the speed
detector reports < 1 MB/s, and the buffer code reacts by allocating
ANOTHER 2 MB — exactly the wrong direction.
Add a process-wide AtomicUsize cap (defaults to usize::MAX, i.e.
no-op) plus a small public set_max_initial_buffer_bytes setter.
Refactor from_speed_mbps to clamp through the cap; extract the
speed-to-bytes ladder into a pure raw_initial_buffer_for_speed
helper plus a from_speed_mbps_with_cap variant that exposes the
clamping for unit tests without touching global state.
src-tauri/lib.rs's run() wires the cap into qbz-player from
qbz-core's MemoryProfile right after logging the profile. Effective
on Pi 3B (issue #331): all StreamingConfig::from_speed_mbps results
pin to 256 KB regardless of detected speed.
Re-export max_initial_buffer_bytes / set_max_initial_buffer_bytes
from qbz_player::player so external callers don't need to reach
into the private streaming_source module.
Adds 5 unit tests: ladder behavior, pass-through when cap is loose,
clamp at the LowMemory cap (the issue #331 case), no-op at the
Normal cap, and that max_buffer_bytes stays unchanged.
cc
Adds InMemorySource — a Symphonia FormatReader+Decoder backed by an
in-memory Cursor — and wires it into the cached branch of the Seek
handler so FLAC Hi-Res and MP3 seeks on LRU-cached / local-library /
offline-cached tracks jump via seek table instead of decoding every
sample from byte 0. Matches the streaming path's behavior, eliminates
the CPU spike during long seeks that was producing audible stuttering
after the streaming seek fix landed.
Falls back to the previous decode_with_fallback + skip_duration path
when Symphonia can't probe the format (rodio-only MP4/AAC container)
or when the native seek itself fails, so nothing regresses for those
formats.
Part of #335.
cc
Replaces skip_duration's decode-every-sample-from-zero loop with
FormatReader::seek(SeekMode::Accurate, SeekTo::Time) on a fresh
IncrementalStreamingSource. For FLAC this jumps via the seek table to
the nearest seek point, then decodes forward to the exact sample —
instead of grinding through millions of samples under the buffered
source's mutex (the cause of post-seek stuttering).
Added IncrementalStreamingSource::seek_to(Duration) which also resets
the decoder, clears the sample queue, and rearms the finished flag so
the source is cleanly re-primed for the iterator.
Non-streaming seeks (cache, offline-cache, local library) still use
skip_duration — they hit in-memory bytes with no lock contention, so
the linear decode is effectively free and there's no reason to change
the path.
Part of #335.
cc
Relaxes the seek gate added in 2dc1ee9e so seeks that fall inside the
already-downloaded portion of a streaming track succeed instead of being
rejected wholesale. The cutoff is approximated from download fraction ×
track duration with a 10% safety margin (0.90) to cover FLAC/VBR byte
density variance; exact for CBR.
Seeks past the watermark still reject cleanly with a warning log — they
would block the audio thread since skip_duration reads samples
sequentially. Cache, offline-cache, and local-library seeks are
untouched (they don't hit the streaming branch).
Part of #335.
cc
Tightens the streaming-seek handler to the simpler contract signed off
by the user: seek works after a streaming track finishes downloading
(buffered_source.is_complete()), but is rejected cleanly with a warning
log while the download is still in progress. Avoids skip_duration
blocking the audio thread on past-watermark seeks without adding
watermark-tracking complexity.
Cache, offline-cache, and local-library seeks are unaffected — they
reach this handler with current_audio_data Some and skip the streaming
branch entirely.
Fixes part of #335.
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The audio thread's Seek handler bailed on `current_audio_data.is_none()`
with a warning and returned. That field is intentionally None during
streaming playback — the bytes live inside current_streaming_source's
Arc<BufferedMediaSource>. Effect: every seek on a streaming track with
the CMAF download still in progress was silently dropped; the UI sent
the command, the player ignored it, and the seekbar snapped back to
whatever position the playback timer had reached (issue #335).
Branch the handler on the live source kind. Streaming: build a fresh
IncrementalStreamingSource from the shared buffered_source (new reader
at byte 0, same downloaded bytes — no re-download), then skip_duration
to the target. Non-streaming path unchanged.
Within-watermark seeks now work. Past-watermark seeks currently block
the audio thread while the decoder waits for more bytes; the clamp to
prevent that lands in the next commit.
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QueueManager::clear previously always preserved the current track;
plus bridge.play_index only mutates queue state, never triggers
audio. Result: when the user hit Clear Queue with nothing playing,
a stale track stayed in NOW PLAYING across app restarts.
- QueueManager::clear(keep_current: bool) — existing callers pin to
true (legacy behavior). New test covers the false path.
- QbzCore::clear_queue / CoreBridge::clear_queue take the flag through.
- v2_clear_queue accepts an optional includeCurrent flag from the
frontend; when true, calls bridge.stop() first so the audio pipeline
actually stops, then wipes the current slot too.
- qconnect sync path + qbzd daemon pin keep_current=true (no behavior
change).
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Reserved for upcoming Mixtape/Collection enqueue paths to mark item
boundaries. Existing enqueue paths populate it with the track's
album_id as a safe fallback so skip-to-item commands behave
correctly even for queues built via "play album" today.
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Offline-cache v2 bundles need runway for the decrypt pipeline. At 10
MB/s (software AES fallback) a 58 MB track needs 6 seconds, which
blew past the old 5s pre-emptive window and missed the gapless handoff
entirely. With AES-NI active the decrypt is ~500ms, so 10s is
harmless overkill — the decrypted bytes just sit in L1 a few seconds
earlier.
Design note: this is complementary to the AES-NI target-feature fix,
not a substitute. A too-small lead time makes gapless brittle even on
fast hardware (any hiccup blows the window); a too-large one is
wasted memory but otherwise fine. 10s is comfortable for every
realistic hardware/track-size combo we'll see.
If we ever want user-visible control, plumb a `gapless_lead_secs`
through AudioSettings and read it at the hot site.
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The frontend was resolving queue-panel clicks via
`actualIndex = current_index + trackIndex`, mixing the unshuffled
canonical position with the shuffle-aware display position. On shuffle
mode, the resulting index pointed at a different track than the one
the user clicked, which is exactly what issue #327 reports.
Solution:
- New `QueueManager::play_upcoming_at(upcoming_index)` resolves the
click position through `shuffle_order` + `shuffle_position` when
shuffle is on, and falls back to `current_index + upcoming_index + 1`
otherwise.
- Exposed through `QbzCore::play_upcoming_at` (emits QueueUpdated) and
`CoreBridge::play_upcoming_at`.
- New `v2_play_queue_upcoming_at` tauri command.
- `QueuePanel.onPlayTrack` now carries the `originalIndex` so the
caller knows the exact upcoming-list position the user clicked.
- `handleQueueTrackPlay` calls the new command directly — no more
client-side index arithmetic.
Also deletes the two dead `From<crate::queue::QueueTrack>` impls in
commands_v2/queue.rs (canonical queue lives in qbz-player now), per
the zero-legacy V2 policy.
Tests: two new unit tests cover the linear and shuffle-aware paths.
Session persistence is untouched — the new method only consults
existing state.
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vicrodh