sync: v3 < dev - #217
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sync: v3 < dev#217
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this is the gap between heapTotal and the v8 heap space, this revels v8 internal memory.
fixes the npm build error.
Added support for playing tiktok videos.
After some researching and testing, aead_xchacha20_poly1305_rtpsize runs natively on libsodum + has support for encrypt_into apis, that on my machine, gave an 300% speed boost in decrypt/encrypt packets, since it runs on sodium-native or libsodium-wrappers, only by running on these packages makes it faster. Also tested with bun, sodium-native is still faster than it by ~80%.
Fixed fadeTo/tapeTo/setFadeVolume silently no-oping on connection.audioStream (missing from _assignStream proxy) Fixed 'audioStream' event listener being dead code (voice lib never emits it) Fixed 'reconnecting' connection status never firing (voice lib uses connecting state, not reconnecting) Fixed pause detection broken (voice lib uses idle+reason='paused', not status='paused') Fixed double-destroy race in _cleanupCurrentAudioStream Fixed stuck detection conflict (set stuckTimeout above player threshold) Fixed force=true reconnect being no-op when token+endpoint unchanged Fixed ping -1 not caught by nullish coalescing Implement setLoudnessNormalizer runtime toggle via VolumeTransformer.setAGCEnabled Handle circuit breaker disconnect reason Update type declarations to match actual voice lib behavior Remove 'seamless_bridge' from playing reason check, add 'unpaused' Formmated the entire code with biome.
Replaced the circular ring buffer (RING_SIZE, writePos/readPos tracking, swap buffer for wrap-around copies) with a straightforward leftover strategy. Incoming chunks are concatenated with any leftover bytes using bufferPool, full frames are encoded in a single pass, and remaining bytes are saved for the next transform call. This reduces memory overhead (no fixed 512KB ring allocation), eliminates wrap-around complexity, and properly releases pooled buffers on error and flush. After testing this for 2d, managed to get a 97% buffer reuse, which alones gets an ~200/400mb saving in memory.
since its gonna be an LTS in october, and can run TS natively, i have updated the experimental flag and fixed an error on config validator causing it to crash on startup.
Since tidal does not seem to work now, will be default to qobuz.
Replaced the fragile NODE_UNIQUE_ID environment variable approach with
an explicit two-step identification protocol:
Master sends clusterId via IPC on fork
Worker confirms identity with HELLO packet (incl. PID) over event socket
Changed eventSockets from Set to Map keyed by worker ID. This enables
reliable mapping, stale socket cleanup on reconnect, and correct stats
attribution
Fixes an issue where /v4/workers showed complete stats only for worker
ID 1 all workers reported workerId: 1, so their stats overwrote each
other, leaving other worker IDs with only the initial { players: 0,
playingPlayers: 0 } stub.
resampler.destroy() was being called twice.
Lazily allocate tape/scratch effect buffers only when those effects are used, instead of retaining them for every playback pipeline. Lowered oversized AAC and WebM demux ring buffer defaults that were retained per active stream. Added teardown cleanup for playback filters and mixers so ended streams release native buffers promptly. From my tests on the playback worker, with 19 active players, the JSArrayBufferData went from ~146MB to ~19.5MB.
This should help with client compatibility.
* fix: dynamicly spelling to dynamically Signed-off-by: Ramkrishna <144999854+ramsquishna@users.noreply.github.com> * fix: exaustive spelling to exhaustive Signed-off-by: Ramkrishna <144999854+ramsquishna@users.noreply.github.com> --------- Signed-off-by: Ramkrishna <144999854+ramsquishna@users.noreply.github.com>
Symphonia v0.6.0 Rubato v2.0.0 And fixed FLAC playback, ig thats good. New repo: https://github.com/ToddyTheNoobDud/symphonia-decoder
Pass abort signal into DASH segment fetches so stop() cancels in-flight downloads. Refactor DASHHandler backpressure from data-event drain detection to _read()-driven read-waiters. Replace .pipe() with pipeline() for proper error propagation and cleanup. Remove removeAllListeners() from streamProcessor and AudioMixer in favor of owned listener cleanup. Delete unreachable duplicate writableNeedDrain early-return in YouTube recovery. Also fixed an 403 error happening on bandcamp (by adding an Referer, and fixed getTrackUrl capturing an trailing JSON.)
This seems to be no longer experimental and can be used in prod, since bun got some major stability updates. Also fix a crash ocurring during startup.
request shim was still using the old arrayBuffer-only path and treating content-length: 0 as body (truthy string), also Object.fromEntries collapsed set-cookie. now it checks len>0 || transfer-encoding and short-circuits for GET/HEAD, body handling is a single arrayBuffer + emitEnd instead of the streaming reader mess. ipc tracker was doing Buffer.byteLength(JSON.stringify(payload)) then v8.serialize(payload) right after for process.send - double work for every cluster message (kinda dumb). now its just v8.serialize(payload).byteLength single pass. added a small bun fast-path in makeRequest for stateless GET/HEAD (no proxy/localAddress/streamOnly) - uses native fetch with abort signal, falls back to http2/http1 on any error.
Now it should pages that return x-cache MISS,MISS, which og:url/html doesn't contain the requested identifier.
Should fix clients not finding any player script to use hopefully.
trackEndSchedule used raw track-position ms as the setTimeout delay, ignoring the active timescale/rate filter, so fades and stops fired too early with timescale speed 0.75. also reschedule it after setFilters, since a pending timer wasn't updated on speed change
fix: recompute trackEnd fade delay for playback speed changes
Enable keepMdatData on MP4Box to support non-faststart MP4 containers where mdat precedes moov metadata. Add eager WASM cleanup and backpressure draining across audio decoders to prevent memory accumulation on stream teardown. This should fix the mass accumulation of WASM/libsamplerate allocations per-stream and JSArrayBufferData.
Remove all event listeners from CrossfadeController and pipeline transforms on stream destruction to prevent closure retention.
Added Lavalink.lua to supported clients list Signed-off-by: filispeen <72784386+filispeen@users.noreply.github.com>
HTML matching for shazam has been breaking a lot, switching to apple's music api for resolving it seems to be better for now. Its faster, will consume less recourses hoppefully and more realiable. (Removed all the html code since i don't pretend to use it)
I din't test it much, but seems to be working for now.
Re-supply identical voice state and explicitly reconnect when the connection is disconnected/destroyed or the voice library lost its voice state. Leave connecting handshakes alone so client retries do not tear down an in-flight connection. Reset the voice connection after the reconnect circuit breaker, since reusing the same connection keeps the old reconnect counters and trips the breaker again immediately. Report breaker and voice timeouts as suspicious rather than generic faults. Repair missing cluster players by recreating the worker player and retrying mutating commands once. Also repair player-state reads and fix stale worker cleanup to use session keys. This should fix players staying silent after a successful voice reconnect without requiring destroy/play again, and reduce Player not found errors after worker failure.
payload.endTime ?? undefined collapsed an explicit null into undefined before reaching seek(), so an explicit clear couldn't be told apart from the field being omitted
fix: preserve explicit null endTime through seek pipeline
Arruma o erro ' Uncaught Exception: Error [ERR_MODULE_NOT_FOUND]: Cannot find module'
add: missing file
fix(spotify): add GraphQL fallback for recommendations
Add Lavalink.lua to supported clients list
if you are testing a web player (locally with no server, file:// pages, which sends a null origin) you can now configure cors to be disabled/enabled. (true = allow browser web players from any origin)
Equalizer only runs bands with non-unity gain instead of all 15 biquads per sample. AudioMixer reuses the shared silence frame as mix base instead of allocating per tick. Opus encoder/decoder copy scratch output without the intermediate subarray view. BufferPool bucket depth 4 to 16 for hot frame sizes. in my machine, this is the results i got (can vary): EQ, 1 band active: ~154us -> ~50us per chunk. Saves 5.2ms core/s per EQ-active player. EQ, 3 bands: ~56% faster. Opus scratch copy: ~20% faster per op, just one less object per frame.
Route all cross-stream error emits through a guarded helper that skips destroyed or listener-less sinks, since emitting with zero listeners throws and escapes as uncaught. Tolerate pipeline premature-close in the master uncaught handler the same way the playback worker already does. (Example: this should fix a crash when you had a audiomixer instance and used stop [not destroy], it would crash the worker.)
…rity fix: resolve ratelimit, worker eviction, body hang, and path traversal
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Changes
Release v3.9 with new features, improvements, bug fixes, performance optimizations, and dependency updates.
Why
Merges all changes from the dev branch into v3 to prepare the v3.9 release.
Additional information
This pull request contains the complete set of changes planned for the v3.9 release.