d5767050f2d9698c6e6dcd8a219a0df668ab7d33
24
Commits
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8a7c4ddf2b |
Multi-output fan-out, offline-marker fix, #19, and per-app send engine core
Local checkpoint - NOT for public release. Fan-out (every received stream to every selected output, no added latency): - SessionPlayout mirror replicas fed the same decoded bytes; delicate ReadFloats untouched. - PlayoutEngine reconciles replicas per active output lane; per-route tuner aggregation keeps lanes from disturbing each other. Recording dispatch gated to the recording route only. - PeerDspChain.Clone() gives each output lane independent biquad state. - ReceiverSelfChecks.FanOutToBothOutputs proves both lanes get audio (self-test). Offline-marker pile-up fix: - ResolvePeerDisplayName strips the " (offline)" marker before reuse, so a ghost peer no longer compounds the suffix hundreds of times in the status line. Issue #19 (Use-Windows-default follower in the loopback send list): - DefaultLoopbackSendFollower resolves to the current default render device's loopback spec, re-applied when the default changes. Per-application send engine core (issue #20) - WASAPI-only, Win10 19041+ gated: - CaptureKind.ProcessLoopback + ProcessLoopbackId ("proc:<pid>"). - AudioAppEnumerator: snapshots apps with audio sessions, tracked by process name, releasing every session object each pass so nothing piles up. - ProcessLoopbackCapture: IWaveIn over the process-loopback activation API (hand-rolled COM interop; NAudio has no binding). Fixed 48k/float/stereo. - CaptureSource IWaveIn overload; MixingEngine opens "proc:<pid>" sources with no MMDevice and no render keepalive. ASIO path untouched. - Self-test enumerated real apps on hardware; support gate verified. UI (Preferences device/app mode + app checklist), ApplySendSources app specs, and the reconcile timer are still to come. Gate: 15/15. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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2fb9274a95 |
v5.2: stability and polish — deep-audit bug fixes + install-flow fixes
Verified findings from a multi-dimension code audit, plus the two install-flow bugs: - Fix Opus encoder use-after-free on a codec/rate change while streaming (guard swap vs encode). - Fix "both" single-file recording dropping audio + drifting (drain both directions in lockstep). - Fix broken clip counter, UPnP teardown on exit, auto-update-restart foreground grant, and a malformed-Opus-format packet orphaning a playout session forever. - Post-install relaunch now respects start-minimised; uninstall is path-aware so it won't clear a different copy's run-at-startup. - Perf/hygiene: cache AppConfig off UI hot paths, fold per-peer EQ+gain into one pass, deterministic disposal (tray menu, timers, COM shortcut, Process handles, process meter), ring-buffer overflow guard, receiver session-lock fix, remote-control allow-list moved onto the UI thread. - Remove dead code (two IsAsioBackend, SessionPlayout.Reset, IsSameEndpoint, RemSoundUpdater IDisposable); several stale-doc fixes. Deferred (not in this release): drift-estimator tweak, peer-discovery pruning, uninstall retry-loop, encryption nonce. Wire format unchanged (interops v3.3-v5.1). Version -> 5.2. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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aeca24ae17 |
v5 pre-release: multi-track drift fix, more self-tests, logging, CLI, version bump
Multi-track recording drift fix (Ed's question — can the separate tracks drift over an hour?): * Root: FlushPeerTracks skipped a peer that produced no samples in a render block, so a peer that went quiet long enough for its session to be pruned (>4 s idle) would have its track fall behind and desync. Now every peer track is padded to a full render block each cycle (silence when the peer produced nothing), so all peer tracks stay sample-locked to the single render clock — they can't drift apart however long the recording runs, and all end the same length. Same padding for the single-file bypass path. OnRecordBlockComplete now carries the block's float count. (The peer tracks are already resampled to the render clock per peer, so this makes peer-to-peer sync exact; your own "me" track is capture-clocked — same soundcard for capture+playback = same clock = no drift, different interfaces can drift slightly.) * Self-test: two new steps — "Per-peer shaping DSP" (PeerDspChain unity/master-off/volume/parametric + ParametricToPeaking) and "v5 settings and shaping round-trip" (AppConfig defaults, NamedPeers, MainTabOrder, parametric PeerShaping, recording default = Both). * Logging (gated by the logging checkbox): master shaping switch, EQ-mode change, parametric band add/delete, peer rename/clear/delete, and the applied Appearance settings after Preferences close. * CLI: --list-profiles and --list-named-peers (read-only), in --help. * Version bumped to 5.0; About-box changelog, RELEASE_NOTES.md and README updated for v5. Build clean; --selftest passes. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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f35f6c3c28 |
Peer rename + details box on Connectivity tab; Add band dialog fixes
Add band dialog (from testing feedback): * Tab order is now Start, End, Gain, OK, Cancel (OK was after Cancel). * Gain box takes decimals like 1.5 (half-dB steps); parametric list shows one decimal. * The two frequency boxes start empty — nothing prepopulated to mislead; both required on OK. New Connectivity-tab feature (held for next release): * Rename peer (Alt+M) opens a dialog to give a peer a friendly name, with a Clear custom name button. Names are keyed by the peer's MACHINE NAME (stable across restarts, IP changes and networks), stored machine-wide in AppConfig.PeerFriendlyNames, and resolved everywhere a peer shows: both peer lists (via PeerListItem.DisplayNameProvider), the volume/pan/EQ list, the status line and split-recording filenames. Manual-by-IP peers with no announced name fall back to keying by address. * Peer details (Alt+E): a read-only box for the highlighted connected peer showing name, machine name, IP, connected-for, link health + ping, what they're sending, and whether they're receiving our audio. "Sending: 2 devices on ASIO at 48 kHz, Opus" is derived from the live receive streams — each stream is one device and its lane gives WASAPI vs ASIO — so NO protocol change and no new privacy exposure. AudioReceiver.ActiveFormatsFromAddress added for this. Manual updated (readme.html + MANUAL.md). Build clean; --selftest passes; deployed to both test folders. Held for next release. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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033edd776f |
Volume/pan/EQ tab overhaul: one master switch, peer checklist, 16-band parametric EQ
Reworks the per-peer shaping tab (held for next release):
* Renamed the tab to "Volume, pan and EQ for peers"; the Preferences toggle now defaults ON.
* Collapsed the two master switches (Enable EQ / Enable pan) into ONE: "Enable volume, pan and
EQ for all peers" (Alt+E). Volume now obeys it too. PeerDspChain.Build takes a single enabled
flag; Profile.EnableAllPeerShaping replaces the two bools (old ones kept for load-migration).
* Peer picker is now a CheckedListBox: ticking a peer shapes them (per-peer bypass via new
PeerShaping.Enabled, default true); the focused row is the one the controls edit. Effective
shaping = master switch AND that peer's tick. Letter-nav suppressed so keys never toggle a tick.
* Three EQ modes, renamed: "3 band simple EQ", "12 band advanced graphic EQ", and the new
"16 band parametric EQ" (PeerEqMode.Parametric16Band).
* Parametric EQ: up to 16 user bands, each a boost/cut across a start->end range (PeerShaping
.ParametricBands; ParametricToPeaking maps range -> peaking centre+Q, shared by DSP and curve).
Add band dialog (spin-or-type, numeric-only, live preview, OK/Escape); Bands list sorted
bass->treble reading "X Hz to Y Hz, plus/minus N dB"; Delete key / Delete button, multi-select.
Set peer EQ to default clears the parametric list too.
* dB now spoken as words ("plus 3 dB" / "minus 6 dB" / "flat") on the graphic sliders and the
parametric list, since NVDA users typically have punctuation off and never hear a "+".
* New unbound machine-wide global shortcut "Toggle volume, pan and EQ for all peers" (not stored
in any profile) via the hotkey controller + settings store.
* Renamed the Inputs/outputs "Set volume for all received audio" to "Master receive volume".
* Added EqCurveControl: a purely-visual EQ response graph (not focusable, invisible to NVDA).
* Full manual sweep (readme.html + regenerated MANUAL.md).
Build clean; --selftest passes. Deployed to both test folders. Held for next release.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
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6b1e65f418 |
Pan/EQ: add a per-peer volume slider (before pan)
Ed's request — an individual level fader per peer, sitting just before the pan control on the Pan and EQ tab. New PeerShaping.Volume (0..1, default 1.0 = 100% = transparent), always applied (no master switch — unity does nothing). It folds into PeerDspChain's L/R gain alongside pan (gainL = panL*vol, gainR = panR*vol), so it's another per-sample multiply, zero added latency, and multiplies with the global volume (per-peer fader -> mix -> master). Slider is 0-100%, saved per profile, announces "Volume: N percent". Shaped recording captures it; raw (bypass) recording doesn't. Held for next release. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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de55230048 |
Multi-track + shaped/raw recording (per-peer split tracks) — held for next release
Recording Settings gains two per-profile toggles (Ed's request): * "Split recording into separate tracks" — each recording becomes a folder with one file per connected peer (their received audio only) plus one for your own send. * "Bypass pan and EQ when recording" — record the RAW audio (before pan/EQ) instead of the shaped audio you hear; applies to single and split. Engine: a per-peer record tap in SessionPlayout hands each peer's block to the recorder RAW (before pan/EQ) or SHAPED (after) per the bypass flag; PlayoutEngine propagates the tap to all sessions (+ inherits on reconnect) and fires OnRecordBlockComplete each render; AudioReceiver exposes SetPeerRecordTap / OnRecordBlockComplete. AudioRecorder now accepts an explicit path and exposes ExtensionFor. RecordingController composes one AudioRecorder per track: multi-track = a recorder per connected peer + a "me" recorder; single-track shaped = the existing mixed tap; single-track raw (bypass) = sum each peer's raw block per render, flushed on the block boundary. Naming (Ed's scheme, sortable): <recordings>/<yyyy-MM-dd>/ then, single-track, "<HH-mm-ss> RemSound recording.<ext>"; multi-track, a folder "<HH-mm-ss> RemSound recording multi track/" containing "<machine name> <HH-mm-ss>.<ext>" per peer (name or IP) and for your own send. Known edge (noted): a peer using sender-side BothIndependent (two lanes) records both lanes to one file in a split recording; single-track bypass sums per render in the Mixed path. Off by default (both toggles unticked = today's behaviour). Builds clean; pending Ed's hands-on test. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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1c1bf5a5cb |
Per-peer pan + EQ ("Pan and EQ" tab) — off by default, held for next release
New feature (Ed's jam-mixing request): pan and EQ each peer's signal independently. Engine (zero added latency — per-sample, applied to each peer's isolated block just before the mix): PeerDspChain (balance pan + RBJ biquad EQ) built on the UI thread and swapped onto SessionPlayout via a volatile reference; PlayoutEngine remembers it per address so a reconnecting peer keeps its shaping; AudioReceiver.SetPeerDsp facade. Model: per-profile PeerShaping dict (keyed by peer address) + EnablePan/EnableEqForPeers master switches; machine-wide AppConfig.ShowPanEqTab. Fixed band layouts in PeerEqBands (3-band tone control: bass/mids/treble shelves+bell; 10-band ISO graphic EQ), +/-12 dB. UI: a "Pan and EQ" tab (before Audio profile, shown only when ShowPanEqTab is on) with the two enables, a connected-peer picker, a pan slider (balance, keeps stereo), a "reset EQ" button (clears both modes' bands, leaves pan), a 3/10-band mode picker and its band sliders — all TrackBars (arrow + page-up/down), updating in real time and saved per profile. Sliders set a friendly AccessibleName on change (pan centre/left/right %, band dB). "Show the Pan and EQ tab" checkbox added to Preferences > General. Everything is off by default (tab hidden, both enables off), so this is dormant for all users until switched on. Builds clean. Pending Ed's hands-on testing before release. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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034a7de632 |
Release v4.2: fix connect-freeze (#10) and new-profile minimize (#12); smoother WASAPI timing
- Connect no longer freezes: PushDiscoveryUnicastHints resolved remembered hostnames with synchronous Dns.GetHostAddresses on the UI thread, blocking the whole window for the DNS timeout on an unresolvable name. A screen-reader user experiences that as the entire machine locking up. Resolution now runs off the UI thread. Same class of bug as the v3.0.1 UPnP-on-the-UI-thread hang. (#10) - New profile / profile switch no longer hides the window: OnShown ORed the global StartMinimised into every instance, so creating a new profile while Start minimised was on dropped the window to the tray and looked like a crash. StartMinimised now applies only to a genuine cold launch; relaunches honour the explicit per-instance flag. (#12) - Smoother WASAPI audio: the receive producer loop and sender mix loop pace themselves with WaitHandle.WaitOne, bound by the system timer (~15.6ms default). Without a fine timer the 10ms feed slips to ~16-31ms and delivers audio in chunky bursts (the desktop-render chunkiness behind Andre's dropouts/lag). New SystemTimerResolution holds a 1ms timer whenever a stream is live, independent of the opt-in Priority mode. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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a408d2b56e |
v4.0: full audio-cue system, cause-aware auto-tune, four-tab Preferences, install-side default sounds
Audio cues - Cues for send/receive on-off, minimise/restore, checkbox tick/untick, and tab switch - Soft keyboard clicks while typing, with a distinct passkey sound on password fields - Per-cue "Choose sound" variant picker; "(none)" silences a cue; front-most missing-sound warning - Send/receive cues take priority over the generic checkbox sound; programmatic ticks stay silent Preferences - Redesigned into four tabs (General, Audio cues, Startup behaviour, Update settings) - Startup behaviour moved in from the Options menu - NVDA now announces the dialog on open (focus a real named control, not the quiet tab control) Auto-tune - Cause-aware: tells device render-callback stalls (more buffer can't fix) apart from genuine network/buffer starvation, so it no longer pins latency high on chunky onboard cards - Lowering the target eases the buffer down (glide) instead of trimming it, so no clicks while tuning Sounds layout - Shipped defaults moved out of the per-user folder into an install-side "default sounds" folder, so updates can refresh them; user customs are Browse-picked file paths and are left untouched - Startup migration removes both legacy sound folders; verified from oldest (v1.0-v3.3) and v3.4 layouts Quiet automated launches - New --silent launch flag mutes all cue sounds and suppresses the startup dialogs (migration notice, update check, Realtek/mic/missing-sound warnings) so test launches never disturb the user - run-tests / build-release / SelfTest repointed to the new "default sounds" layout Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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84c4a47411 |
v3.9.1: plain/WASAPI streams now play on an ASIO-mode receiver (silent-mic fix)
Receiver: a Mixed (plain) session is rendered on an active lane in BothIndependent mode instead of being skipped, so a WASAPI-only sender is no longer silent to a receiver that has an ASIO driver selected. Also port the per-session buffer depth-drain (stops the receive jitter buffer bloating). Sender: add sndAudFr meter (audio frames actually sent) to localise capture vs send. App: startup sound cue (machine-wide, Preferences); stop sending audio when no peer is reachable (issue #8). Version 3.9.1. Server (relay): fix updater version-compare for multi-dot tags, guard the main loop against crashes, reject spoofed BYE from a mismatched endpoint. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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4dbe9a47a0 |
Release v3.5: USB-card recovery, per-card adaptive buffer, one user folder, audit fixes
- Recover an unplugged/replugged output sound card automatically (issue #5): detect the dead WASAPI device and remember the receive-output selection so it re-ticks and re-opens when the card returns. - Adaptive per-card WASAPI buffer target sized to each card's pull chunk, held stable so it never flits about under CPU/network load. - Consolidate all per-user data (config, profiles, logs, sounds) into one "user settings and logs" folder; migrate every older layout; exclude it from the updater so custom cue sounds now survive updates. - Mic-privacy detector: warn once when a Windows-blocked mic is switched on, or a profile loads with one already on. - All warning/notice dialogs now come to the foreground even when minimised. - Apply volume + mute on profile load (were saved but not restored). - Crash-safe (atomic) profile/config saves. - Fix two resource leaks (push-mode capture MMDevice; UPnP DeviceFound handler). - Remove dead code (baseline-diff machinery, dead ASIO probes, no-op stubs). - Docs: readme.html, MANUAL.md, About-box changelog and release notes for v3.5. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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946e6f4be4 |
v3.4 refinements: WASAPI drift correction, startup-dialog sequencing, quick-switch polish, docs
Builds on the v3.4 freeze (
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dd70613017 |
v3.4: handle-leak fix, Realtek ASIO block, device notifications, quick profile switch, hotkey announcements
Freezes the v3.4 feature set (everything since the v3.3 public release): - Fix the receiver handle leak: the 3-second device-refresh timer reopened the configured ASIO driver every tick, and Realtek's ASIO driver leaks Event+Mutant handles on every open. Cache the ASIO probe per driver so it is opened once. - Realtek ASIO block: detect a Realtek ASIO driver, offer once to disable it, and never touch it again if disabled; Options-menu toggle to reverse. Global config. - Device hot-plug is event-driven (AudioDeviceChangeNotifier) instead of a 3s poll; debounced refresh, falls back to polling if registration fails. - Quick profile switch: new global hotkey opens an NVDA-friendly popup of all profiles (current marked); Enter/click switches; plays a new "profile menu open" cue with Preferences mute + custom-sound. - Announce assigned global hotkeys on the controls/menu items they drive (NVDA reads "press X anywhere"). File > Open already had Ctrl+O. - Held-back changes folded in: config-folder migration, codec-column fix, Tailscale endpoint network-prune, empty-password guard, and the handle-leak diagnostics (ProcessSelfMeter, HandleTypeProbe). Version bumped to 3.4.0. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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959720f54d |
Release v3.3: end-to-end encrypted audio, plus cue and reliability fixes
Headline: all audio is now encrypted (AES-256-GCM), keyed by a per-profile password. Mandatory — v3.3 only interoperates with v3.3+. Encryption - RemSoundCrypto (Core): PBKDF2 key derivation, AES-GCM encrypt/decrypt (low-alloc, into-span), password fingerprint, light on-disk obfuscation. - Wire: SenderLane encrypts the audio payload (PCM split across parts when the +28 overhead crosses MTU); AudioReceiver/StreamSession decrypt via a shared single-thread AudioDecryptor. Fingerprint piggybacks on the Format packet (offset 36, backward-compatible) so a peer can detect a password mismatch. - Profile.Password (scrambled), carried through BuildCurrentProfile; MainForm derives + pushes the key/fingerprint to sender + receiver (RecomputeAudioCrypto). - UX: ask-for-password on profile create; File -> Change this profile's password (ProfilePasswordDialog); Options -> Profile passwords (manager); a gate that prompts before streaming without a password; and a clear "passwords don't match" / "peer needs to update" message driven by the fingerprint. Cue fixes - CuePlayer (NAudio) replaces System.Media.SoundPlayer, which silently failed on the 96 kHz/24-bit cue WAVs (and any custom file) — cues now play reliably, resampled to 48 kHz/16-bit. Also fixes the Preferences preview button. - Connect/disconnect cues now audio-gated with hysteresis: connected when audio flows OR heartbeat healthy; lost only when audio stops AND heartbeat unreachable. Kills false disconnects and the receive-only "no cues" case. - Honest cue logging (played / muted / not loaded). Smaller - Endpoint stickiness: keep the audio target pinned to the heartbeat-proven address instead of chasing a multi-homed peer's other (unreachable) address. - "Online/offline" label now audio+heartbeat aware, not discovery-only. - "Show what's new after each update" preference (on by default). Docs: About v3.3 block, RELEASE_NOTES, README (encryption as a headline), manual section 12 "Passwords and encryption" (+ renumber), MANUAL.md regenerated. Version 3.2.0 -> 3.3.0. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com> |
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8aa8d0c3bd |
Bump to v3.0.2: fix slow native-memory leak on the receive side
Bug: Andre reported audio latency feeling laggier after long sessions
on his Win10 desktop receiving Opus from his laptop. His 23-hour log
showed working set climbing from 83 MB at startup to 3.5 GB at the
end, with the managed heap staying tiny (~5-7 MB) the whole time.
CPU climbed alongside (from steady-state ~7% mid-session to peaks of
~60% by the end) and audio threads ended up doing ~4x the work they
did at the start. Andre's perception of latency drift was the CPU
pressure showing up in audio scheduling, not the buffer itself
growing (bufAvg stayed roughly stable at 25-28 ms).
Root cause: Concentus.Native (introduced in v2.2 / shipped in v3.0)
returns concrete NativeOpusDecoder / NativeOpusEncoder objects that
implement IDisposable and own native libopus state. Three call sites
were taking the IOpusDecoder / IOpusEncoder interface reference and
never calling Dispose:
* StreamSession.Dispose — comment literally said "IOpusDecoder has
no Dispose; nothing else to free", which was correct for the
pure-managed Concentus.OpusDecoder pre-v2.2 but stopped being
correct the moment we added the native binding
* OpusEncoderState.Dispose — same misleading comment, same bug
* SenderLane.OnCodecChanged — overwrote the existing encoder field
without disposing the old instance on codec change
Compounding factor: Program.Main sets GCSettings.LatencyMode =
GCLatencyMode.SustainedLowLatency to keep audio scheduling smooth
(it suppresses gen2 collections). That's correct for the hot path
but it ALSO suppresses the finalizer pass that would have released
the leaked native handles as a backstop. Because the managed heap
stayed tiny, the GC never saw enough pressure to force a gen2 pass
on its own, and the native state piled up indefinitely. Multi-output
receive multiplied the per-output growth.
Fix is in two parts:
1. Call (... as IDisposable)?.Dispose() at every release point —
StreamSession.Dispose, OpusEncoderState.Dispose,
SenderLane.OnCodecChanged before overwrite, AudioRecorder's
Concentus.Oggfile-backed OpusOggFileWriter.Dispose. The
as-IDisposable cast handles both the native and the pure-managed
path transparently (managed-only IOpusDecoder isn't IDisposable;
the as-cast yields null and the null-conditional is a no-op).
2. Periodic native-memory reaper in MainForm.SnapshotLogIfDue — once
every 300 snapshot ticks (~5 min), run
GC.Collect(2, Optimized, blocking, !compacting) +
WaitForPendingFinalizers on a background Task.Run so the gen2
work doesn't hitch the UI thread. Audio threads are separate and
unaffected. Serves as belt-and-braces for any future code path we
forget to wire and for cleaning up any per-call native scratch
the underlying library might accumulate that isn't owned by a
single .NET wrapper.
Expected behaviour after fix: working set settles around 100-200 MB
on a typical receive session and holds roughly flat for as long as
the app stays running. CPU stays at its early-session baseline
across multi-hour sessions. Andre's "latency drift" symptom should
disappear.
Wire format unchanged; same codec list, same UI, same defaults.
v3.0.2 talks to other v3.0.x peers exactly as v3.0 / v3.0.1 do.
Co-Authored-By: Claude Opus 4.7 <noreply@anthropic.com>
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af0e7c3fff |
Bump to v3.0.0: Opus live latency, codec rename, profile-resume after update, save-through-lock
* New "Opus, live latency" codec mode: 2.5 ms frames (120 samples/ch at 48 kHz)
via the float-input encode path. End-to-end codec delay drops to ~5 ms (vs
~12.5 ms at standard 10 ms Opus). Test on LAN: 400 pps/lane, zero missed /
reordered / duplicate packets, ~15 ms one-way saved end-to-end.
* Wire-format change: AudioFormatInfo.FrameDurationMilliseconds renamed to
FrameSamplesPerChannel (int sample-count at announced sample rate). Removes
the lossy 48000*ms/1000 conversion that couldn't represent 2.5 ms. v3 <-> v3
exact; v3 <-> v2 still passes audio (Opus decoder is self-describing from
packet TOC) but v2 side over-sizes its buffer wildly. v2.x profiles auto-
migrate via <120 sentinel rule in RemSoundSettingsStore (anything below 120
is treated as legacy ms and multiplied by 48). Profile JSON key kept as
OpusFrameMilliseconds via [JsonPropertyName] so old profile files still load.
* Codec dropdown rebuilt with use-case names: "PCM 48K 24 bit - uncompressed",
"Opus, broadcast quality - loss tolerant", "Opus, live latency - for jamming
and monitoring". Middle 10 ms option retired; saved 480-sample profiles
collapse to broadcast quality (safer-side default).
* Profile auto-resume after self-update: RemSoundUpdater writes a one-shot
_resume-after-update.txt sentinel containing the active profile title before
exit; Program.Main reads + deletes it on next start and silently loads that
profile, skipping the picker. Helper batch's robocopy /XF excludes the
sentinel and the failure-branch cleans it up if the install aborts. Falls
through to normal startup behaviour (StartWithProfileTitle or picker) if the
sentinel is missing, empty, or names a profile that no longer exists.
* Read-only profile saves now go through on explicit Ctrl+S / File -> Save
with a one-time TaskDialog warning ("Save anyway" / "Cancel" + Do-not-show-
again). Lock continues to suppress the automatic unsaved-changes prompt on
close / profile switch (its main job). AppConfig.SaveOnReadOnlyMessageSuppressed
renamed to SaveOnReadOnlyWarningSuppressed; v2.x suppression flag is silently
discarded since the behaviour changed and the user needs to see the warning
once on each machine.
* Manual (readme.html) updated: codec table rewritten with the three new
choices and corrected bandwidth figures, send-rate description updated, new
sections "The same profile picks up automatically after an update" and
"Saving on purpose while a profile is locked".
* Subsumes the never-separately-released v2.2 work: native Opus encoder
(~97% less per-second memory churn on Opus send path via Concentus.Native),
efficiency tidy-ups (item 4 ASIO probe rate, item 6 WaitHandle, item 7
snapshot cache, items 14/16 heartbeat + discovery), legacy cleanup
(items 30/34/35/36: KeepAlive infrastructure, drift drop/repeat/accumulator
fields, fan-out cache stat). New diagnostic columns cpu/memMB/wsMB/
allocKBps/captureMs/sendMs/recvMs/renderMs gated on Enable-logs.
About dialog updated with v3.0 block at top; v2.2 block retained for the
subsumed work.
Co-Authored-By: Claude Opus 4.7 <noreply@anthropic.com>
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6d6d6897e4 |
Bump to v2.2.0: Opus native binding, efficiency tidy-up, diag self-meter
Single biggest change: added the Concentus.Native NuGet package. Concentus 2.0+ auto-detects native libopus at runtime and routes encode calls through it; encoder state lives on the C side and is reused across calls rather than `new`ing ~15 working buffers per call (Concentus issue #22, open since 2018). Measured on the desktop test at 15:36:55 — Opus 10 ms allocation rate dropped from 4,625 KB/s to 108 KB/s, a 97.7% reduction. Process CPU dropped from 4.7% to 1.6% in the same config. Audio is bit- for-bit identical (it's literally the same encoder, just better packaged). `OpusEncoderState.cs` itself unchanged on the call site. Diagnostic / measurement layer (gated on Enable-logs, zero cost when off): * ProcessSelfMeter: CPU%, managed heap MB, working set MB, allocation rate per second, GC counts per generation * Per-thread work-time counters: captureMs / sendMs / recvMs / renderMs expressed as milliseconds of CPU consumed by each audio thread per second * Inter-packet arrival gap measured at the user-space UDP socket (rxNetGapMs) — pinpoints whether arrival jitter is in the network or our own dispatch path Small efficiency wins (each one was small but cumulative): * deviceRefreshTimer interval 1s -> 3s (item 4) * WaitHandle array allocations eliminated in MixingEngine.MixLoop and MultiOutputPlayout.ProduceLoop (item 6) * MultiOutputPlayout caches its output-buffer snapshot and only rebuilds on SetOutputDevices, instead of rebuilding every 10 ms (item 7) * HeartbeatService reuses an outbound ping byte[] instead of allocating per send (item 14) * PeerDiscoveryService caches broadcast addresses and invalidates on Windows' NetworkChange event instead of walking all NICs every 1.5 s (item 16) Legacy / dead-code removal: * KeepAlive packet's implementation (struct, enums, writer, reader, size constant) — all dead since HeartbeatService landed 2026-05-06. Kept the RemPacketType.KeepAlive enum value and silent-drop dispatch for wire compat with any pre-2026-05-06 build still in the wild (item 30) * driftDropFramesTotal / driftRepeatFramesTotal fields and accessors — Phase-2 splice corrector relics, never incremented since Phase-4 resampler design landed; backed five always-zero diag log columns (items 34 + 35) * DriftAccumulator (always returned 0) — same shape, removed alongside the driftAcc= column (item 35) * TakeMaxFanOutCacheBytes / Ms + fanCacheMs column — FanOutSource was retired in May (item 36) Project documentation: * RemSoundefficiency.md added as the canonical record of the efficiency analysis, every item's status, and the measured wins from this round * Honest item-by-item review of the original 50-item list — several items I had sized optimistically in the original analysis turned out to be already-done (item 20), already-optimal (item 22), or below the meter floor (items 9, 15, 17, 25). Recorded so future passes don't re-investigate. Wire format and audio pipeline unchanged from v1.5 onward — v1.5 through v2.2 peers interoperate. |
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79b28b6c02 |
Bump to v2.1.0: UPnP, read-only profile lock, sleep/hibernate audio fix
Headline features: * Automatic router port opening (UPnP / NAT-PMP / PCP). Opt-in via Preferences; surfaces external address + carrier-grade NAT detection. * Lock profile (read-only). New File-menu tick that makes a profile load-only — session changes don't persist, no save prompt on close. Unblocks unattended shutdowns (NVDA gone, remote dropped, hibernate) where the existing save prompt could deadlock. * Check for updates on startup (default on) + brief countdown notice before silent updates install, so a launch-time update doesn't make the app silently vanish. * "Cue sounds" -> "Audio cue sounds" label clarification. Bug fixes: * No sound after the computer wakes from sleep. PowerResumeHandler rebuilds the audio backend automatically on resume; brief "Reconnecting to audio driver" splash during the rebuild. * Receiver audio silent after waking from hibernate. RefreshAudioDeviceLists now treats a transient ASIO probe failure (returns -1/-1 because the driver is mid-teardown / mid-reinit) as "retry next tick" instead of clearing the user's tick selection. Diagnostic-only changes (gated on the existing Enable-logs checkbox, zero cost when off): * AudioStepProbe split into cross-buffer vs within-buffer maxes so log inspection can tell a real-content sharp transient apart from a pipeline-boundary glitch. Plumbed through every probe owner. * New rxNetGapMs + gc0/gc1/gc2 delta columns in the diag log to split receive-side jitter into network-layer vs managed-runtime causes. Files touched: RELEASE_NOTES.md + readme.html + 24 source files across RemSound.Core / RemSound.Sender / RemSound.Receiver / RemSound.App. Three new app files: PowerResumeHandler, RouterPortMapper, UpdateInstallNoticeDialog. Wire format and audio pipeline unchanged from v1.5 onward — v1.5 through v2.1 peers interoperate. |
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918ca6cac0 |
Bump to v1.6.0: peer-address recovery, reconnect crash, long-run memory/CPU leak
Three reliability fixes. Wire format and audio pipeline unchanged from v1.4 / v1.5 — all interoperate. Peer address recovery: * When a tracked peer goes Unreachable (its resolved address — often a stale DNS / Pi-hole record, or a peer that rebooted onto a new IP) but the same peer is still heartbeat-pinging us from a different address, RemSound now adopts the live address instead of transmitting to a dead one. HeartbeatService records untracked ping sources; MainForm's TryAdoptLiveHeartbeatAddress (1 Hz) re-points the sender, heartbeat tracking and receiver allow-list. Conservative: fires only on the unambiguous one-unreachable-and-one-source case, private-range (RFC1918) addresses only so a relay can't hijack the sender, 10 s cooldown. Reconnect crash: * Fixed IndexOutOfRangeException in MainForm.SyncConnectedList. A churny peer-list rebuild (peer reboot) left SelectedIndex pointing past the rebuilt item array; the 1 Hz status timer read SelectedItem and crashed the app. New SafeSelectedItem bounds-checks the index; applied to all three timer-driven sync methods. The status tick is also wrapped in try/catch so a transient WinForms hiccup logs instead of crashing. Long-run memory / CPU leak: * A receiver left running for hours grew to gigabytes and climbing CPU. Decoder sessions orphaned by peer reconnects were not reaped — every reconnect mints a fresh (endpoint, streamId) key, and PruneIdleSessions silently skipped sessions whose PlayoutEngine lookup missed. Rewrote it to reap on each session's own LastWriteUtc (no cross-dictionary lookup), added a hard MaxLiveSessions cap as a backstop, and a "stream sessions live: N" diagnostic line. Bounds both memory and render-thread CPU. Co-Authored-By: Claude Opus 4.7 <noreply@anthropic.com> |
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4915a9afe1 |
Bump to v1.5.0: cross-lane routing fix, recording double-tap fix, menu reorg
Bug fixes: * BothIndependent recording was double-tapped — when both WASAPI and ASIO outputs were ticked, recordings came out garbled and ~2x the expected duration. AudioRecorder now uses four per-lane rings (sent-wasapi, sent-asio, recv-wasapi, recv-asio) plus a writer- thread mix step that drains min(wasapi, asio) frames and sum-mixes with the soft-tanh limiter. Tap signatures gained a RenderRoute parameter; Mixed maps to the wasapi slot. * A peer announcing on the WASAPI lane was inaudible when the receiver only had an ASIO output ticked (and vice versa). The session opened, samples flowed into the SessionPlayout ring, but ReadForRoute(AsioLane) skipped any session whose Route was WasapiLane so nothing drained the ring. PlayoutEngine now tracks per-lane "is this lane backed by a device" via volatile bools settable through SetLaneActive(RenderRoute, bool), called from CompositeRenderBackend.SetOutputDevices whenever the device split changes. ReadForRoute admits orphan sessions when the other lane is inactive. Menu reorganisation: * New Options menu (Alt+O) holds Recording settings (Alt+S), Keyboard shortcuts (Alt+K, Ctrl+K), Startup behaviour (Alt+T), Preferences (Alt+P, Ctrl+P). Pre-v1.5 these were scattered across File menu, Record menu, and inside the Preferences dialog itself. * Record menu mnemonic moved from Alt+O to Alt+K, rendered as "Record (Alt+K)" so the chord is visible despite K not being a letter in "Record". Alt+R is taken by Receive audio. * File menu — new Recent profiles submenu (Alt+F, R) listing the five most-recently-opened profiles. Press 1..5 inside the submenu to jump to a slot. Missing files are skipped from the menu but kept in storage. * Rename current profile moves to Alt+M (was R), Minimise to tray moves to Alt+N (was M). * Lock to audio clock was Alt+K, now Alt+D. UX additions: * Ctrl+O = Open profile (matches the menu chord). * New global hotkey: Start / Stop recording. Pickable from Options -> Keyboard shortcuts. Unbound by default. Works system-wide. Wire format and audio pipeline unchanged from v1.4 — v1.4 and v1.5 peers interoperate. Co-Authored-By: Claude Opus 4.7 <noreply@anthropic.com> |
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c59e413c1f |
Bump to v1.2.0: recording, sound cues, drift compensation, diagnostics
New user-facing features:
* Recording. Dedicated Record menu (Alt+O — moved from Alt+R to
avoid clashing with the Receive audio checkbox), Start/Stop on
Ctrl+R, settings dialog, per-profile source / format / bit-depth /
channel-mode / folder. Three source modes (received only, sent
only, both). Files are crash-resilient — a process crash
mid-recording leaves a playable file containing everything up to
the last header refresh (~5 seconds).
* Four output formats, all functional:
- WAV: 16/24-bit PCM or 32-bit float, custom writer with
periodic RIFF re-patching.
- MP3: LAME 128–320 kbps CBR (via NAudio.Lame).
- OGG-Opus: 96–256 kbps VBR (via Concentus.Oggfile, reusing the
Concentus encoder from the wire path).
- FLAC: 16/24-bit lossless (via CUETools.Codecs.FLAKE — pure
managed, no native DLL).
* Recording start/stop sound cues. record start.wav and
record stop.wav play around the recording transition. Played via
System.Media.SoundPlayer to the default Windows output, separate
from the recording pipeline so a normal recording does not contain
the cue.
* Per-cue Preferences. The old single "Mute connect/disconnect
sounds" checkbox is replaced by a CheckedListBox: Connect /
Disconnect / Recording start / Recording stop. Old profiles with
the legacy MuteConnectionCues=true are honoured on first load via
a migration path in the new Load* helpers.
* Receiver-side drift compensation switched from discrete
single-frame splices to a continuous WdlResampler at a smoothed
rate ratio. SessionPlayout.cs rewrite.
Diagnostics (only active with Enable logs ticked):
* Per-stage discontinuity probes — sender raw capture (per backend,
PushModeWasapi + Asio both wired), sender pre-encode (now per
lane in BothIndependent, fixing a cross-stream artefact), receiver
post-decode, post-ring, post-resampler.
* Wire-level packet sequence tracking on each PCM stream — in-order
/ missed / reordered / duplicated counts in the diag log.
* Clipped-sample delta in the diag log.
* New AudioStepProbe in RemSound.Core with per-channel scan helper.
UI changes:
* Record menu uses Alt+O (Rec&ord). Inside the menu, item mnemonics
unchanged (S / T / O / C).
* Auto-tune interval combo label is mode-aware: "Auto-tune latency
interval" in classic modes, "Auto-tune interval — WASAPI and ASIO"
in BothIndependent. The combo's Enabled state now follows EITHER
lane's auto-tune checkbox (was only the WASAPI one — bug).
Wire format and audio pipeline unchanged from v1.1 — v1.1 and v1.2
peers interoperate.
Co-Authored-By: Claude Opus 4.7 <noreply@anthropic.com>
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768875cebb |
Add priority mode + always-on network packet priority
Priority mode (per-profile toggle, Audio profile tab, Alt+U): - Bundled PROCESS_POWER_THROTTLING_STATE with EXECUTION_SPEED and IGNORE_TIMER_RESOLUTION flags (keeps CPU at full clock + 1 ms quantum pinned regardless of OS idle decisions) - PowerSetRequest(EXECUTION_REQUIRED) + SetThreadExecutionState so the system can't decide we're idle and downclock - timeBeginPeriod(1) for 1 ms scheduler quantum - ProcessPriorityClass.High, ProcessMemoryPriority normal, working-set minimum locked via SetProcessWorkingSetSizeEx(HARDWS_MIN_ENABLE) - Fully reversed on toggle-off and on form close - Off by default (battery cost on laptops); explicit opt-in per profile MMCSS thread priority on hot threads bumped from High to Critical (network listener, mix loop, render thread) — always on. Network packet priority (always on, no toggle): - qWAVE flow on the outbound UDP socket at QOS_TRAFFIC_TYPE_VOICE with QOS_NON_ADAPTIVE_FLOW (Voice priority, DSCP 46/EF, WMM Voice on Wi-Fi) - Silent fallback if qwave.dll missing or QoS service disabled - Sender + receiver kernel UDP buffers bumped to 1 MB each (was 256 KB send / 512 KB receive) for resilience against ~30 ms stalls Bug fix: PreferencesDialog logging toggle no longer flags the profile as dirty. Logging is a machine-local AppConfig setting; the spurious ChangedAnyProfileSetting flag was triggering the unsaved-changes prompt on exit after toggling logs alone. Manual updated: new "Use CPU and Windows performance settings in high priority mode (Alt+U)" subsection on the Audio profile tab, new "Network packet priority (qWAVE / Voice / WMM)" subsection in the network chapter, Alt+U row added to keyboard-shortcuts table. Co-Authored-By: Claude Opus 4.7 <noreply@anthropic.com> |
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17259438c6 | Initial commit: RemSound v1.0 |