* feat(voice): unified mobile voice bar with gesture-isolated PTT row
Replace separate VoiceStatusChip + VoicePttButton with a single
CompactVoiceBar widget that combines both into a two-row layout:
- Control row (tap): status text, mute, deafen, settings chevron
- PTT row (hold): full-width hold-to-talk, shown only in PTT mode
Gesture isolation prevents mis-touch between rows: the control row
uses tap-only InkWell/IconButton while the PTT row uses a raw
Listener for pointer-down/up events.
Key changes:
- Add CompactVoiceBar widget with state-colored container (normal,
muted, talk-power-blocked)
- Remove mute/deafen IconButtons from AppBar headerActions
- Restructure voice details sheet into primary section + collapsible
ExpansionTiles (audio processing, PTT capability, debug)
- Optimistic state updates for mute/deafen to eliminate tap delay
- Instant PTT visual feedback (no AnimatedContainer fade)
- Constant geometry across all states (no layout shift on toggle)
* fix(voice): preserve current PTT button format
* feat(voice): move mute/deafen controls into VoiceStatusChip
* fix(voice): ensure consistent chip height across mute states
Remove isSelected/selectedIcon from IconButtons inside VoiceStatusChip.
Material 3 toggle IconButtons (_SelectableIconButton) can vary in height
when the selected state changes due to tap target sizing. Use simple
conditional icons instead and set shrinkWrap tap target size with tight
constraints for stable 40x40 buttons regardless of state.
* fix(voice): remove leftover duplicate mute/deafen buttons in VoiceStatusChip
* fix(voice): replace unsafe stereo cast with bytemuck and localise talk-power tooltip
Replace the raw-pointer `&mut [(f32, f32)]` to `&mut [f32]` cast in
the oboe output callback with `bytemuck::cast_slice_mut`, eliminating
the unsafe block and relying on bytemuck compile-time NoUninit
verification instead.
Add voiceTalkPowerBlocked l10n key (en + zh) and replace the only
remaining hard-coded English tooltip in VoiceStatusChip with it.
* feat(voice): add real-time mic input level metering at 30 Hz
Expose input RMS from the audio engine through the bridge as a
dedicated Rust→Dart Stream<double>, replacing the binary on/off
indicator with a proportional dBFS level meter.
Rust side:
- chanora_audio: add set_input_dbfs/input_dbfs accessors to
SharedAudioProcessingStats; restructure CaptureState::ingest()
to compute dBFS from mono buffer before the PTT guard so the
meter shows mic activity even when not transmitting.
- chanora_core: widen audio_stats() return to include f32 input
level.
- chanora_bridge: add input_level: f32 to BridgeAudioStats and
new input_level_stream(sink: StreamSink<f32>) that pushes at
~30 Hz via tokio interval task.
- Update frb_generated.rs serialization for the new field.
Flutter side:
- VoiceLevelMeter: accept optional double level (dBFS), map
-60..0 dBFS to 0..1 fill fraction, animate with
TweenAnimationBuilder for smooth transitions.
- voice_compact.dart: subscribe to inputLevelStream in the voice
details sheet for 30 Hz meter updates, keeping 250 ms poll for
TX/RX counters.
- voice_bar.dart: accept optional inputLevel from the stream.
- main.dart: subscribe to inputLevelStream, pass to VoiceBar.
* chore: sync Flutter build config and dependency updates
- Add Flutter migrator flags to gradle.properties (builtInKotlin, newDsl)
- Add FlutterGeneratedPluginSwiftPackage to iOS/macOS Xcode projects
- Update meta 1.17→1.18, test_api 0.7.10→0.7.11
- Rebuild chanora_bridge framework for macOS
- Update Podfile.lock for iOS and macOS
* fix(voice): correct meter animation, pre-gain dBFS, stream lifecycle, and protocol warnings
B1: Convert VoiceLevelMeter to StatefulWidget tracking previous fill
as Tween begin so the meter animates smoothly instead of resetting
to zero on every frame.
B2: Compute dBFS from pre-gain mono samples in CaptureState::ingest()
so the level meter reflects raw mic input, matching mobile paths.
B4: End input_level_stream after 10 consecutive session errors instead
of emitting -120 dBFS forever when the session is gone.
Also fixes all 13 clippy warnings in chanora_protocol: collapsed
nested if-let patterns, replaced .ok() + Some matching with Ok, used
? operator, and introduced EventChannels struct to reduce the four
helper functions below the 7-argument threshold.
* fix(voice): use MissedTickBehavior::Skip for level meter stream and align dBFS doc
Set MissedTickBehavior::Skip on the input_level_stream tokio interval
so slow audio_stats() calls skip missed ticks instead of bursting,
preventing CPU spikes on the UI meter thread.
Align VoiceLevelMeter class doc: the mapping floors at -60 dBFS
(via dbfsToFraction), not the full -120 range.
* fix(audio): eliminate Android output stutter via Oboe config + lock-free callback
Phase 1 — Oboe configuration:
- Change output stream from Usage::VoiceCommunication to Usage::Game with
ContentType::Sonification to avoid forcing the Legacy (OpenSL ES) data
path on most devices (Oboe issue #2075)
- Switch output format from i16 Mono to f32 Stereo, matching Qint's proven
configuration and eliminating per-callback downmix conversion
- Set buffer size to 2x burst after stream open, reducing default buffer
from 8-20x burst to 2x burst for lower latency
- Remove scratch Mutex<Vec<f32>>; callback writes directly to Oboe buffer
Phase 2 — Lock-free output callback:
- Add audio_event_queue.rs: lock-free SPSC bridge using crossbeam ArrayQueue
with separate packet (lossy) and control (reliable) channels
- OutputCallback now owns AudioHandler directly (no Arc<Mutex<>> on Android)
- Inbound forwarder pushes packets via AudioEventProducer (no mutex)
- set_client_volume pushes control commands via event queue on Android
- iOS/desktop Arc<Mutex<AudioHandler>> path unchanged
* fix(audio): address PR #20 review findings
- Store AudioEventConsumer directly in OutputCallback to eliminate
per-callback Arc clone on the real-time audio thread
- Add SAFETY comment for the unsafe from_raw_parts_mut transmute
- Bound set_client_volume spin-loop to 64 retries with warn log
- Remove redundant crossbeam-utils direct dependency
- Regenerate license inventory for new crossbeam deps (CI fix)
* fix(audio): use ASCII TODO punctuation
* docs(rust): add doc comments to delta enums and fix dead_code warnings
Add missing documentation to ProtocolDelta, CoreDelta, and BridgeDelta
enum variants and their struct fields across the protocol, core, and
bridge crates. Document the ChannelId::ROOT constant and the
take_delta_rx adapter method.
Fix dead_code warnings:
- keyring_disabled: add #[cfg] gate matching its callers
- snapshot_signature: add #[cfg(test)] for future test use
* fix(rust): correct `order` field docs to predecessor channel ID, narrow audio engine cfg gates
- Correct `order` field documentation in ProtocolDelta, SessionEvent,
and BridgeEvent from 'sort order' to 'predecessor channel ID
(TeamSpeak linked-list ordering hint)' per Copilot review feedback.
- Narrow AudioEngine voice_out_tx, voice_activity_selector, and mic_gain
cfg gates from ios+macos+android to android-only, since these fields
are only read from self in android_restart_voice_unit. On iOS/macOS the
values are passed directly to the voice backend at construction time.
* fix(ui): restore speaking status indicators
Speaking state (isSpeaking) is computed from voice activity timestamps
in the protocol layer and cannot be represented as a discrete delta.
The event-driven refactor removed periodic snapshot refreshes, causing
speaking indicators to go stale.
Adds a 750ms periodic snapshot refresh (matching SPEAKING_ACTIVITY_WINDOW)
while the audio stats timer is active (in-channel only). Structural
changes (moves, joins, leaves) are still handled by instant deltas.
* fix(ui): hide server query clients from delta joins
When a ServerQuery client sends a message, a ClientJoined delta fires.
Before PR#15 the periodic snapshot rebuild would include the SQ client
but the snapshot_view filter hid it. With deltas, the client persisted
in the local snapshot. Now ClientJoined deltas skip SQ clients entirely.
* fix(proto): log getconnectioninfo errors instead of silently discarding
Ping and packet loss showing 'Unknown' in the client info sheet is
caused by getconnectioninfo failures being silently swallowed. Now
logs the error with the client_id so the root cause can be diagnosed
(e.g. missing b_client_connectioninfo_view permission on the server).
Also logs clientgetvariables failures.
* fix(proto): refresh non-self client profiles before mapping
* feat(protocol): add ping deviation to client profiles
* chore(ui): regenerate Flutter bridge bindings for ping deviation
* fix(l10n): add ping deviation labels to client info
* feat(ui): show ping deviation in client info sheet
* chore: regenerate Cargo.lock after rebase
* fix(ui): add 1s cool-down to prevent double-tap channel join
voiceJoin returns instantly (fire-and-forget protocol), so the
pending-join guard clears before a second tap lands. The cool-down
prevents the rapid channel oscillation and ClientIsFlooding (524)
that results from double-tapping.
* fix(ui): handle ChannelAlreadyIn as success, ClientIsFlooding with backoff
- ChannelAlreadyIn (0x0302): treat as silent success, update UI state
- ClientIsFlooding (0x020c): show localized snackbar, extend cooldown 5s
- Add l10n strings for flooding error (en + zh)
* fix(proto): use Windows TS3 client version for broadest compatibility
Matches Qint's default (Windows_3_X_X__1). Avoids server-side
behavioral differences with TS5 version strings.
* fix(proto): patch tsproto-types to handle short P-256 coordinates
BigInt::to_bytes_be() strips leading zeros, causing WrongPublicKeyLength
when a server's ephemeral key coordinate starts with 0x00. Patch from
EdisonJwa/tsclientlib fix/p256-short-coordinate-pad branch left-pads
coordinates to the P-256 field size instead of rejecting them.
* refactor(core): stop watchdog from emitting SnapshotChanged
The watchdog now serves only as a liveness probe (miss counting for
reconnection). UI updates are handled entirely by the event-driven
delta path (ProtocolDelta → SessionEvent → BridgeEvent → Flutter).
Removes signature tracking and SnapshotChanged emission from the
supervisor loop. The initial snapshot is still fetched via the
Connected event handler in Flutter.
* refactor(ui): remove channel-join cooldown guard
With event-driven deltas the UI updates instantly on channel moves,
so the 1-second cooldown is no longer needed. Double-taps are handled
by the server (ChannelAlreadyIn → success) and the pending-channel-id
guard prevents overlapping requests.
Also removes the _lastJoinCompletedAt field entirely.
* fix(core): reattach event forwarders after reconnect
The reconnect path swapped in a new ProtocolClient but never took
chat_rx, activity_rx, or delta_rx from it. After the first reconnect,
the event-driven UI pipeline was dead.
Fix by extracting spawn_event_forwarders() helper called on both
initial connect and reconnect. Also replaces lossy try_recv+sleep
polling with proper recv().await for push-based delivery.
* feat(protocol): enrich delta schema with all snapshot-visible fields
ClientJoined now carries input_muted, output_muted, is_server_query,
talk_power, talk_power_granted. ChannelAdded/ChannelUpdated now carry
has_password and needed_talk_power. ClientUpdated also carries
is_server_query, talk_power, talk_power_granted.
This prevents local snapshot drift where fabricated defaults could
hide password requirements, talk-power restrictions, or client type.
* refactor: remove dead SnapshotChanged variant end-to-end
SnapshotChanged is no longer emitted since the watchdog was refactored
to liveness-only. Removes the variant from SessionEvent, BridgeEvent,
and the Flutter switch statement. FRB bindings regenerated.
* fix(ci): regenerate license inventory and fix iOS submodule fetch
- Regenerate docs/security/license-inventory.md to match current lockfile
- Remove submodules: true from checkout (causes hard fail on private submodule)
- Add explicit git submodule update --init --depth=1 with || true fallback
- Check silero-coreml/Package.swift instead of directory existence
* fix(diag): record bridge events in diagnostics page
Connection lost/reconnecting/disconnected and iOS audio interruption
events now appear in the diagnostics dialog alongside existing error
snackbar entries.
* fix(diag): reduce audio callback sample verbosity to debug
The render callback diagnostic sample logged every 100 callbacks
(~2s) at INFO level, flooding the 256-entry release log buffer and
pushing out useful events. Changed to DEBUG so it only appears in
debug builds with the larger 4096-entry buffer.
* fix(diag): add timestamps to Rust diagnostic log entries
Clarify ONNX Runtime guidance with direct-open install hints, restore desktop WebRTC VAD visibility, map mouse side buttons through focused PTT capture/runtime paths, and wait for server acks before showing chat sends as successful.
Constraint: Linux release UX must stay functional when ONNX Runtime is optional and GNOME portal availability varies
Rejected: Keep desktop VAD locked to Silero only | misleads users when ONNX Runtime is skipped
Confidence: medium
Scope-risk: moderate
Directive: Preserve the protocol send-ack wait path for chat so UI success always tracks real server acceptance
Tested: flutter analyze lib/main.dart lib/widgets/chat_views.dart lib/widgets/input_dialogs.dart lib/widgets/startup_dependency_screen.dart; flutter test test/widgets/input_dialogs_test.dart test/widgets/chat_views_test.dart test/services/startup_dependency_check_test.dart test/widgets/startup_dependency_screen_test.dart test/widgets/voice_settings_controls_test.dart test/widgets/audio_processing_config_state_test.dart; cargo test -p chanora_protocol --lib; cargo test -p chanora_audio ptt_backends --lib
Not-tested: Live manual GNOME portal rebind/global PTT on a real desktop session; observer-bot chat against a live server after the sender-name fallback change
Implementation of SDD-120 §1-§8:
Bench harness (crates/chanora_audio/benches/):
- common.rs: deterministic synthetic audio (440 Hz sine, no RNG).
- realtime_capture.rs: bench_capture_alloc_count (dhat) +
bench_capture_callback_wall_clock (criterion).
- opus_codec.rs: bench_opus_encode_latency + bench_opus_decode_latency
(direct audiopus, not AudioHandler — SDD-120 §3 item 4).
- resampler.rs: bench_resampler_throughput across 44.1->48 /
16->48 / 48->48 passthrough.
CI tooling (crates/chanora_audio/examples/):
- emit_baseline.rs: aggregates criterion estimates.json outputs
into the SRS-217 baseline schema.
- compare_baseline.rs: applies SRS-219 tolerance, renders markdown
table with 🟢/🟡/🔴 markers + yellow simpler-form realization per
SDD-120 §8.
Deviation from SDD-120 §2 / §5 / §7 placement: these tools live
under examples/, not benches/ or src/bin/. Rationale: they must
consume serde_json (a dev-only dep — production builds must not
pull it). Cargo only resolves dev-dependencies for [[test]],
[[bench]], and [[example]] targets; [[bin]] targets under
src/bin/ see only regular [dependencies]. examples/ keeps the
binaries out of the production dep tree while still giving them
cargo run --example invocation. An SDD-120 amendment should
reflect this.
Workflows (.github/workflows/):
- bench-advisory.yml: PR + push triggers; runs benches; posts a
sticky PR comment via actions/github-script@v7; job status is
always success (SRS-218 clause 4 — non-blocking).
- bench-baseline-update.yml: workflow_dispatch only; runs benches;
opens PR via peter-evans/create-pull-request@v6 (sole writer of
the SAD-089 baseline JSON).
Cargo.toml additions ([dev-dependencies] only — verified excluded
from --release builds): criterion 0.5, dhat 0.3, serde_json 1.
Source-code seam: minimal pub-but-#[doc(hidden)] bench_seam module
in chanora_audio (engine.rs + lib.rs re-export) so the criterion
bench harness can construct a CaptureState and drive
CaptureState::ingest without re-implementing the engine (SDD-120
§3). Non-iOS targets only — CaptureState itself is iOS-gated.
Initial baseline seed: crates/chanora_audio/benches/baselines/
x86_64-unknown-linux-gnu.json = {}. compare_baseline handles the
missing-baseline case gracefully and emits a 'no red markers'
report; the first manual dispatch of bench-baseline-update.yml
after merge establishes the real values.
Out of scope per SDD-120 §10: production telemetry export,
build-failing hard CI gate, multi-host benchmarking, IDE
integration, Dart-side bridge round-trip bench.
Verification:
- cargo check --workspace --all-targets: PASS.
- cargo bench --bench realtime_capture --no-run: PASS.
- cargo bench --bench opus_codec --no-run: PASS.
- cargo bench --bench resampler --no-run: PASS.
- cargo build --example emit_baseline --example compare_baseline
-p chanora_audio: PASS.
- cargo test --workspace: 106 passed, 0 failed, 3 ignored — no
regression from prior count.
The capture cpal callback (CaptureState::ingest) ran two heap
allocations per callback on the realtime audio thread:
1. engine.rs:1196-1202 — fresh `mono: Vec<f32>` for the downmix
output, once per cpal callback (50–100 Hz).
2. engine.rs:1217-1218 — `pcm_accum.drain(..FRAME_SAMPLES).collect()`
building a fresh Vec<f32> of 960 samples per Opus frame.
Both sites mirror the pattern already fixed for the output side at
engine.rs:1389-1397, where allocating per callback on glibc malloc
was correlated with user-perceptible audio popping. The output-side
fix replaced the per-callback allocation with a pre-allocated
`scratch` Vec that is cleared and resized in place; this commit
applies the same template to the capture side.
Changes:
- Add `mono_scratch: Vec<f32>` and `frame_scratch: Vec<f32>` to
CaptureState. Initialised with Vec::with_capacity(4096) and
Vec::with_capacity(FRAME_SAMPLES=960) respectively in
CaptureState::new.
- Replace the downmix Vec construction with in-place push into
`self.mono_scratch`; `clear()` retains capacity across callbacks.
- Replace the drain().collect() with `self.frame_scratch.extend(
self.pcm_accum.drain(..FRAME_SAMPLES))`; same capacity-retention.
- The resampler call uses std::mem::take to swap the scratch buffer
out for the duration of the &mut self call, then moves it back —
the backing allocation is preserved across callbacks.
Algorithm semantics are unchanged: same downmix arithmetic, same
clamp loop, same Opus encode call sequence. Only the storage
strategy differs.
Out of scope (intentionally not touched):
- Android audio path (android_voice_unit.rs, mobile_voice_backend.rs):
researcher constraint C-4 — the Android cpal data path is mid-
migration and being replaced.
- Output callback at engine.rs:1409+: the only obvious per-callback
allocation there (`scratch`) was already fixed; a fuller audit
is a separate scope decision.
- The `scratch` buffer at engine.rs:1389-1397 — already correct.
Verification:
- cargo check --workspace --all-targets: passes.
- cargo test --workspace: 106 passed / 0 failed / 3 ignored.
- cargo clippy --workspace --all-targets: no new lints introduced;
the one warning inside the edited region (clamp-like pattern at
line 1266) was pre-existing on the copied clamp loop.
Per SDD-106 §5 add BridgeEvent::PermissionState{permission, state}
with the PermissionStateKind enum (Granted, Denied, PermanentlyDenied,
Unknown). The Kotlin side publishes mid-session permission changes
through a new JNI entry point Java_app_chanora_chanora_1flutter
_MainActivity_publishPermissionState routed by the new
permission_jni.rs module; the Rust audio engine subscribes and
authoritatively clamps the transmit gate (see SDD-106 §6).
Adds crates/chanora_bridge/build.rs to emit
cargo:rustc-link-lib=dylib=c++_shared on Android so libchanora_bridge
.so carries DT_NEEDED libc++_shared.so; this is required by Android
API 24+ per-library linker namespaces to resolve __cxa_pure_virtual
and friends at System.loadLibrary time.
Includes the FRB-regenerated Dart counterparts so each commit is
independently buildable.
Trace: SDD-105, SDD-106 §5, SDD-118 item 6 (extended).
Per SDD-106 §6 add a permission-state clamp to TransmitModeSelector.
When RECORD_AUDIO is Denied or PermanentlyDenied the transmit gate
is forced false regardless of PTT or voice-activity state; on
Granted the clamp releases and normal transmit decisions resume.
The clamp takes precedence over PTT and hard_mute in the decision
ordering documented inline.
Three new tests cover the clamp behavior, the release-on-grant
transition, and the non-RECORD_AUDIO ignore path.
Trace: SDD-106 §6, SRS-209.
Replace the prior one-shot android_engage_voice_communication call
with a ModeStack-mediated acquire/release pair. AudioEngine snapshots
the system audio mode on first acquire via android_get_audio_mode()
and restores it on last release via android_set_audio_mode(prior).
MODE_IN_COMMUNICATION (3) is engaged across the voice-session lifetime
per SDD-108.
Includes the Android AudioManager getMode/setMode JNI helpers
(placed in chanora_audio::engine alongside the existing JNI surface)
and the small ptt.rs touch needed for the SDD-108 ID-tag on the
existing tests.
Trace: SDD-108, SDD-115.
Introduce ModeStack, a pure-Rust refcount-composable wrapper for
Android audio-mode acquire/release with prior-mode snapshot. Per
SDD-108 §1/§2 the engine snapshots the system audio mode on first
acquire and restores it on last release; composed acquires are
no-ops while the mode is held.
ModeStack is panic-free; release-on-zero returns AlreadyReleased
rather than panicking. Six SWE4-UV-045-tagged unit tests cover the
acquire/release semantics on the host target.
Trace: SDD-108, SWE4-UV-045.
The ring-buffer architecture (rc.8+73..+74) was making playback
strictly worse. Diagnostic data at +74 conclusively showed:
* Producer task ran perfectly at 50 Hz (250 ticks per 5 s).
* AudioHandler returned silence on 65-84% of fill_buffer calls
even when window_peak_f32 reached 0.98 (full-scale audio).
* Ring buffer never accumulated beyond 30 ms because consumer
(VPIO render callback at 43.5 Hz, ~1440 samples per call)
drained samples faster than the 50 Hz producer could push
them, in net effect.
The producer drained AudioHandler at 50 Hz \u2014 slightly faster
than iOS VPIO actually consumes audio. Each fill_buffer call
asked for 20 ms but adjacent Opus packets hadn't arrived yet, so
fill_buffer returned mostly silence. Linux/SDL's same pattern
works because SDL calls fill_buffer at EXACTLY the device
callback rate (50 Hz = 20 ms per buffer); the rates match.
Fix: revert to direct fill_buffer call from the render callback
(the SDL pattern in tsclientlib's own reference example at
tsclientlib/examples/audio_utils/ts_to_audio.rs). The render
callback now:
1. Resizes scratch_stereo Vec to 2 * num_frames f32 if needed
2. Zeros the live slice (fill_buffer is additive, not clearing)
3. Locks AudioHandler, calls fill_buffer(scratch_stereo)
4. Downmixes L+R -> mono i16 with master gain into out[]
5. Applies output_muted bypass
6. Tracks peak_out + audio/silence ratios for diagnostic
The closure owns scratch_stereo across callbacks for stable
allocation. Same memory model as Linux/SDL.
Removed:
* tokio::spawn producer task
* rtrb dep + RingBuffer<i16> + Producer/Consumer split
* tokio::sync::oneshot shutdown channel
* producer_shutdown_tx field on IosVoiceUnit struct
* RING_BUFFER_SAMPLES / PRODUCER_TICK_MS constants
* Producer-side diagnostic counters
Diagnostic kept: cb / num_frames / frames_changes /
callbacks_with_audio / callbacks_with_silence / peak_out_i16 /
gain. Logged every 100 callbacks.
The choppy / clicks symptom is independent of the buffer
architecture \u2014 it's whatever AudioHandler is doing on iOS
that's different from Linux. Next investigation step is to
either (a) switch from VPIO to RemoteIO unit (lose Apple's
voice processing entirely), or (b) understand why AudioHandler
returns silence so often on iOS-arrival packet timing patterns.
Build counter 74 -> 75.
Edison Jwa