4c13b2be5c6ced13f4ddca8a9ffc6194b4872bfa
16 Commits
| Author | SHA1 | Message | Date | |
|---|---|---|---|---|
 Claude
|
a96ec9db00 |
chore(nests): standardise Log import in audio-pipeline files
Audit-12 cleanup. The audio-pipeline sources mixed two import styles:
some files used `com.vitorpamplona.quartz.utils.Log.d/w/e` fully-
qualified (NestPlayer's 13 sites, AudioTrackPlayer's 8, two each in
the encoder/decoder, three in NestMoqLiteBroadcaster) while
AudioRecordCapture had already migrated to a top-level
`import com.vitorpamplona.quartz.utils.Log`. The lambda overload
(`Log.d(tag) { lazyMessage }`) is what the find-non-lambda-logs
skill enforces and is the only style the rest of the codebase uses;
the fully-qualified form is verbose noise on every call site.
Add the `import com.vitorpamplona.quartz.utils.Log` line to each
file and rewrite call sites to bare `Log.d` / `Log.w` / `Log.e`. No
behaviour change; logs still go through PlatformLog with the same
tag and lazy-message contract. Five files updated, ~30 call sites.
https://claude.ai/code/session_014JfZJHSTvyYYWJbC9VbB47
|
||
|
Claude
|
8a49486b37 |
fix(nests): G4+G5 plumb catalog sampleRate through decoder + AudioTrack
G4: Audit-9 (catalog-driven decoder reconfig) plumbed numberOfChannels
end-to-end but left sampleRate hardcoded at AudioFormat.SAMPLE_RATE_HZ
in MediaCodecOpusDecoder.buildOpusIdHeader / buildFormat and in
AudioTrackPlayer's MinBufferSize / 250 ms target / setSampleRate
calls. For Opus this is benign in practice — Codec2 always emits
48 kHz PCM regardless of OpusHead inputSampleRate — but hardcoding
the constant means a future codec or container variant whose
decoder DOES respect input sample rate would mis-clock playback.
And the OpusHead identification header should match what the
catalog declares either way.
Thread sampleRate alongside channelCount through every layer:
- MediaCodecOpusDecoder constructor takes
`sampleRate: Int = AudioFormat.SAMPLE_RATE_HZ`. buildFormat
and buildOpusIdHeader both take it as a parameter.
- AudioTrackPlayer constructor takes the same. Used in
AudioTrack.getMinBufferSize, the 250 ms-equivalent target
bytes calculation (`(sampleRate / 4) * BYTES_PER_SAMPLE *
channelCount`), AndroidAudioFormat.Builder.setSampleRate, and
the diagnostic log.
- decoderFactory and playerFactory in NestViewModel become
`(channelCount: Int, sampleRate: Int) -> ...`.
- awaitDecoderChannelCount → awaitAudioPipelineConfig, returning
a private `AudioPipelineConfig(channelCount, sampleRate)`
struct so openSubscription handles both fields uniformly.
`sampleRate` falls back to AudioFormat.SAMPLE_RATE_HZ on
timeout / non-positive declaration with a warning log.
- NestViewModelFactory + NestViewModelTest updated to the
two-arg factory shape.
G5: documented the SUBSCRIBE_BUFFER safety budget on
CATALOG_AWAIT_TIMEOUT_MS's kdoc. With the current 500 ms timeout +
SUBSCRIBE_BUFFER = 64-frame DROP_OLDEST flow + 50 fps Opus, at
most 25 frames buffer during the wait — leaving ≥ 39 frames of
margin (≈ 780 ms) before the oldest frame would be evicted. Even
at the production framesPerGroup = 50 (1 group/sec) cadence this
never trips during normal startup. No code change; just pinning
the rationale so a future timeout bump is checked against the
buffer size.
https://claude.ai/code/session_014JfZJHSTvyYYWJbC9VbB47
|
||
|
Claude
|
a94d12638f |
perf(nests): bound encoder CSD loop, single-alloc audio framing, cache catalog JSON
Three audit follow-ups, all in the audio publish hot path: Audit-3: MediaCodecOpusEncoder.encode could busy-loop on a buggy encoder that emits BUFFER_FLAG_CODEC_CONFIG on every dequeue. The existing format-change path has a one-shot `formatChangeAbsorbed` guard for exactly this reason; the new CSD-skip path didn't. Cap consecutive CSD skips per encode call at MAX_CSD_SKIPS_PER_CALL=4 (generous: Codec2 emits 1 OpusHead or 2 OpusHead+OpusTags in practice). On overshoot, log a warning and return ByteArray(0) — the broadcaster's `if (opus.isEmpty()) continue` already handles that contract. Audit-6: NestMoqLiteBroadcaster's per-frame send path allocated twice per Opus packet — once for the timestamp Varint and once for the concatenated `varint+opus` payload. At 50 fps × N speakers that's measurable young-gen pressure. Switch to the same single-allocation pattern PublisherStateImpl.send already uses: compute Varint.size(timestampUs), allocate the final payload buffer once, write the varint directly into it via Varint.writeTo, then copy the opus bytes. Drops audio-thread allocations from 3/frame to 1/frame (the third was the wrap in PublisherStateImpl.send, already optimised). Audit-7: MoqLiteNestsSpeaker.startBroadcasting and ReconnectingNestsSpeaker.runHotSwapIteration both encode the same fixed catalog JSON on every call — once per broadcast start and once per JWT-refresh hot-swap iteration. Cache the encoded bytes on MoqLiteHangCatalog.Companion.OPUS_MONO_48K_AUDIO_DATA_JSON_BYTES so the kotlinx.serialization run happens at class-init time and both call sites read a static reference. Trivial perf, but co-locates the "default Amethyst publish shape" in one place. https://claude.ai/code/session_014JfZJHSTvyYYWJbC9VbB47 |
||
|
Claude
|
96cfa1235a |
fix(nests): T8 skip BUFFER_FLAG_CODEC_CONFIG outputs in MediaCodecOpusEncoder
Android's `audio/opus` MediaCodec encoder emits `BUFFER_FLAG_CODEC_CONFIG` output buffers BEFORE any audio buffer — typically the 19-byte OpusHead identification header per RFC 7845, and (on some Codec2 stacks) the OpusTags comment header. These are decoder-config blobs, NOT audio frames. We weren't filtering them, so the first 1–2 wire frames every encoder lifetime were OpusHead bytes wrapped in our legacy-container varint(timestamp_us) prefix. The web watcher's WebCodecs `AudioDecoder` handles this by burning warmup slots (`@moq/watch/src/audio/decoder.ts`'s `warmed <= 3` policy) — so the listener mostly recovers — but on Android Codec2 stacks that emit BOTH OpusHead AND OpusTags as separate CSD buffers, two of the three warmup slots get absorbed by metadata and the listener hears a tiny click on the next group rollover. The hot-swap path (`ReconnectingNestsSpeaker`) repeats the warmup on every JWT refresh, so this fired once every 9 minutes in production. Filter via `bufferInfo.flags and MediaCodec.BUFFER_FLAG_CODEC_CONFIG` inside `encode`'s output-drain loop. CSD buffers are released and the loop continues to the next dequeue rather than returning the bytes. Logged once per encoder lifetime via `loggedCsdSkip` so we have proof the path fired without flooding logcat — a stack that emits CSD on every frame would otherwise be noisy. Returning `ByteArray(0)` for the first call (because the only output was a CSD + format-change pair) is unchanged behaviour and the broadcaster's `if (opus.isEmpty()) continue` already handles it. https://claude.ai/code/session_014JfZJHSTvyYYWJbC9VbB47 |
||
|
Claude
|
1c47fd2403 |
feat(nests): drive Opus decoder + AudioTrack channel count from catalog
Web speakers (kixelated/hang reference) emit stereo Opus when the
user's AudioContext picks a stereo input — and our catalog now lets
us know that ahead of decoder construction. Previously the decoder
+ AudioTrack were both hardcoded to mono via AudioFormat.CHANNELS, so
a stereo web publisher's frames either threw on the Opus TOC byte
mismatch, decoded with downmix artifacts, or output left-channel-only
depending on the device's MediaCodec implementation.
Wire the catalog-discovered numberOfChannels through:
- MediaCodecOpusDecoder accepts `channelCount: Int = 1`. Drives
MediaFormat.createAudioFormat's channel count, the OpusHead CSD-0
channel byte, and the per-frame output buffer size (stereo frame =
2× shorts vs mono). Validates 1..2 — RFC 7845 mapping family 0
covers both mono and stereo without an explicit channel mapping
table; multichannel needs family 1 which we don't support.
- AudioTrackPlayer accepts `channelCount: Int = 1`. Drives the
AudioTrack channel mask (CHANNEL_OUT_MONO vs CHANNEL_OUT_STEREO)
and the 250 ms wall-clock buffer-size target (stereo doubles bytes
per sample).
- NestViewModel.openSubscription now starts the catalog fetch
BEFORE waiting for it, then awaits `_speakerCatalogs[pubkey]` for
CATALOG_AWAIT_TIMEOUT_MS (500 ms) before constructing the decoder
+ player. With the speaker-side emit-on-subscribe hook in place
(
|
||
|
Claude
|
be8dd0a3bc |
debug(nests): add playback-path logs to localise receiver silence
The wire is healthy: drainOneGroup logs show every group reaching the
receiver with 5 frames each, no drops, no parse errors, no pump exits,
no announce-ended teardown. Yet the green ring on the receiver still
blinks-and-stops. The bug must be in the playback pipeline below
drainOneGroup. Add NestPlay-tagged logs there so the next repro
identifies whether the failure is decoder, AudioTrack allocation,
beginPlayback, or write-blocking.
NestPlayer.play (`NestPlay` tag):
- log start with prerollFrames
- log preroll flush + beginPlayback transition
- throttled per-N counters: receivedObjects / decodedFrames /
emptyDecodes / enqueued
- log decoder.decode throws explicitly
- log empty-pcm decoder returns (separate from throws)
- log enqueue duration when > 50 ms (catches AudioTrack.write-blocking)
- log flow COMPLETED / cancelled / pipeline threw with final counts
AudioTrackPlayer (`NestPlay` tag):
- log start() + the constructed AudioTrack's state / playState /
bufferSizeBytes / minBuffer
- log beginPlayback's t.play() return + post-call state
- log AudioTrack.write partial / error returns
- log setMuted / setVolume changes (silent mute is a known
suspect for "audio not playing")
MediaCodecOpusDecoder (`NestPlay` tag):
- log codec name on successful allocation
- log allocation failure with exception type / message
All log calls use the lambda overload so format work only runs when
the tag is enabled. Throttled counters keep logcat at < 1 line/sec
per speaker even at 50 fps capture cadence.
|
||
|
Claude
|
6237c02c6f |
fix(nests): platform-side audio robustness — focus, AEC, route obs, network handover
Four follow-up fixes from the post-audit review (#4 / #5 / #6 / #7). #6 AcousticEchoCanceler / NoiseSuppressor / AGC on the AudioRecord session. The VOICE_COMMUNICATION input source engages the platform echo canceller automatically on most modern Android devices, but a small set of older / OEM-customised devices only attach AEC under MODE_IN_COMMUNICATION — which an audio-room app deliberately avoids. Attaching the standalone audiofx effects to the AudioRecord's session id covers those devices without rerouting through the call audio path. All three are best-effort and a no-op on devices where the source already engages them. #4 Real audio focus handling. The previous OnAudioFocusChangeListener was a no-op based on the assumption that the OS would auto-duck us; it doesn't (CONTENT_TYPE_SPEECH streams aren't auto-ducked). Inbound phone calls were mixing on top of room audio. - New `NestAudioFocusBus` (commons) — process-wide enum signal, decoupled from android.media so commons stays platform-free. - NestForegroundService translates AUDIOFOCUS_GAIN / LOSS_TRANSIENT* / LOSS into the bus enum. - NestViewModel observes the bus and silences both the listener playback (effective listen-mute = user OR focus) and the broadcast mic (effective mic-mute = user OR focus). User-visible mute states stay the user's choice so a focus regain restores them automatically. - The pipeline keeps running while focus is lost (decoder, capture, network) so unmute is sample-accurate when the call ends. #5 AudioDeviceCallback observability. Registers a callback in NestForegroundService that logs Bluetooth / wired / USB headset attach + detach with device type + name. Doesn't drive playback decisions — Android's auto-routing handles route swaps — but makes "audio cut out when I plugged in headphones" reports correlatable with a concrete event for the first time. #7 Network-change → fast reconnect. Without this, a Wi-Fi → cellular handover left the QUIC connection sitting on a now-dead socket until its PTO fired (~30 s) before the wrapper noticed. Now: - New `NestNetworkChangeBus` (commons) — collapses bursts of onLost/onAvailable into a single recycle event. - NestsListener + NestsSpeaker grow `recycleSession()` (default no-op); the reconnecting wrappers override to close the inner session so their orchestrator opens a fresh one. - NestForegroundService registers a default-network callback; suppresses the first onAvailable (registration callback) and only publishes on actual default-network changes. - NestViewModel observes the bus and calls recycleSession on both wrappers. The SubscribeHandle re-issuance pump (listener) and the hot-swap publisher pump (speaker) cut existing subscriptions / broadcasts onto the new session as soon as it lands — same paths the JWT-refresh recycle uses. - Manifest gains ACCESS_NETWORK_STATE for registerDefaultNetworkCallback. |
||
|
Claude
|
e4e55d1df6 |
fix(nests): three more dropout sources from the post-audit review
1. Split AudioPlayer.start() into allocate + beginPlayback to restore
the synchronous DeviceUnavailable error path that the previous
pre-roll fix collapsed.
- AudioPlayer gains `beginPlayback()` (default no-op) so test
fakes / desktop sinks aren't forced to grow a method they
don't need.
- AudioTrackPlayer.start() now allocates the AudioTrack + audio-
priority writer thread but does NOT call AudioTrack.play();
beginPlayback() flips the device into the playing state.
AudioTrack in MODE_STREAM explicitly supports write() before
play() per the platform docs, which is exactly the contract
pre-roll wants.
- NestPlayer.play() now calls player.start() synchronously
(caller catches DeviceUnavailable + rolls back the slot like
before) and defers beginPlayback() until pre-roll fills.
2. MediaCodecOpusDecoder: drain output + retry input dequeue before
dropping a frame. The prior 10 ms input-buffer dequeue followed
by an unconditional `return ShortArray(0)` turned every transient
stall (thermal throttling, GC pause, output not yet pulled) into
a 20 ms audio gap. Now we drain whatever output is ready —
freeing input slots — then retry input dequeue with a 5 ms
timeout. Only after both misses do we drop the packet, and even
then we return any output samples that the drain produced so
the player doesn't underrun on the back of one tight cycle. The
decoder's drain logic is extracted into a `drainAvailableOutput`
helper so both the pressure-relief path and the post-queue main
drain share it.
3. ReconnectingNestsListener: exponential backoff for the opener-
throws retry path. Replaces the flat 1 000 ms `SUBSCRIBE_RETRY_BACKOFF_MS`
with 250 → 500 → 1 000 ms, reset on first successful subscribe.
The 250 ms floor is well under moq-rs's typical announce-
propagation latency (< 200 ms), so a subscribe-before-announce
miss usually recovers fast enough that the wrapper SharedFlow's
~1.3 s buffer hides the gap entirely.
|
||
|
Claude
|
076b301d84 |
fix(nests): close 4 audio-dropout sources across listener + speaker
1. Listener-side pre-roll + bigger playback buffer + audio-priority thread.
- NestPlayer buffers 5 decoded frames (~100 ms) before starting
the AudioPlayer, masking the first-frame underrun that fires
whenever Compose / GC briefly stalls Main.
- AudioTrackPlayer sizes the AudioTrack at max(minBuffer*16, 250 ms)
instead of minBuffer*4 (~80 ms) and writes via a per-instance
audio-priority single-thread executor (Process.THREAD_PRIORITY_AUDIO)
instead of Dispatchers.IO, so WRITE_BLOCKING never contends with
unrelated IO work.
2. MoqLiteSession.subscribe: hoist response typeCode out of the
collect lambda. readVarint advances pos permanently while
readSizePrefixed only rolls back its own length-varint, so a
chunk-split between type and body would re-read the body's size
prefix as the type code on the next chunk and misframe the
response. Mirrors the same fix already in handleInboundBidi.
3. MoqLiteSession.subscribe: register the subscription in the map
BEFORE writing the SUBSCRIBE bytes on the wire. The relay can
open the first group's uni stream before our continuation
re-enters [state] to register; if so, drainOneGroup looked the
id up against an empty map and silently dropped the frame
(~1 frame / 20 ms gap on first attach). Wrap the post-register
writes so a transport-failure unwinds the orphan registration.
4. Hot-swap moq-lite publisher across JWT-refresh boundaries.
- NestMoqLiteBroadcaster: publisher is now @Volatile + supports
swapPublisher(); capture loop snapshots the reference per frame
and resets framesInCurrentGroup on swap.
- MoqLiteNestsSpeaker implements a new internal
HotSwappablePublisherSource interface that exposes
openPublisherForHotSwap(track) without spinning up a broadcaster.
- ReissuingBroadcastHandle keeps a single long-lived broadcaster
across session recycles when the speaker supports hot swap;
legacy IETF / fake speakers fall back to close-then-restart.
- connectReconnectingNestsSpeaker.orchestrator hoists the old-
session close onto a sibling launch so the wrapper can swap
the publisher into the broadcaster on the new session before
the old session's WebTransport drops. Eliminates the previously-
accepted 50–150 ms audible silence at every JWT refresh.
|
||
|
Claude
|
523e8a92ec |
fix(nests): play audio-room output through STREAM_MUSIC, not STREAM_VOICE_CALL
`AudioTrackPlayer` was constructing its `AudioTrack` with `USAGE_VOICE_COMMUNICATION` + `CONTENT_TYPE_SPEECH` so the room would behave like a phone call (volume rocker controls call volume, ducks notifications). The `AudioTrack` then renders on `STREAM_VOICE_CALL`, which Android only services audibly while the device is in `MODE_IN_COMMUNICATION`. Nests never drives `AudioManager.mode` (only NIP-100's `CallAudioManager` does), so on most devices the playback either dropped to the earpiece at near-zero volume or produced no audio at all — making rooms appear silent on both phones. Fix: route through `USAGE_MEDIA` + `CONTENT_TYPE_SPEECH` (= `STREAM_MUSIC`) so audio-room playback comes out of the loudspeaker by default and the volume rocker controls it any time. Same approach Twitter/X Spaces and Clubhouse take for hands-free audio rooms. Echo cancellation still works on the capture side via `MediaRecorder.AudioSource.VOICE_COMMUNICATION` regardless of the playback usage. Also update `NestForegroundService.requestAudioFocus` to request focus under the matching `USAGE_MEDIA` attributes so the focus claim and the playback attributes line up. |
||
|
Claude
|
702885f4cb |
fix(nestsClient): fix race + leaks + Android encoder/decoder bugs from second audit
A fresh audit caught a real race in the round-1 subscribe() rewrite plus several distinct bugs in the Android MediaCodec layer. **MoqLiteSession.subscribe() race regression** — the launched collector that reads the SubscribeResponse and watches for peer FIN ran cleanup (remove from subscriptionsBySubscribeId, close frames) before subscribe() itself reached the post-await registration. If the publisher FIN'd immediately after sending Ok, the collector exited first against an empty map; subscribe() then registered the subscription, leaving frames in the map with no live collector to ever close it on transport drop. Consumer hung forever — exactly the failure mode round-1 fix #3 was supposed to prevent. Fix: pre-register the subscription BEFORE launching the collector. Drop unused `ok` field from ListenerSubscription. **MoqLiteNestsSpeaker.startBroadcasting publisher leak** — if session.publish() succeeded but broadcaster.start() then threw (mic permission denied, AudioRecord allocation failure), the publisher was never closed and stayed registered as session.activePublisher, permanently blocking subsequent startBroadcasting calls. **runCatching{suspend close} swallowing CancellationException** — MoqLiteNestsSpeaker.close, MoqLiteBroadcastHandle.close, MoqLiteNestsListener.close all wrapped suspending closes in runCatching, breaking structured cancellation when the parent scope cancelled teardown. Replaced with explicit cancel-rethrowing try/catch. **MediaCodecOpusDecoder ArrayList<Short> boxing on every audio frame** — at 50 fps × 960 samples × N speakers ≈ 48 000 boxed Short allocations/sec/speaker on the audio hot path. Rewritten to write directly into a pre-sized ShortArray via ShortBuffer.get(dst, off, len). **MediaCodecOpusEncoder bugs** - KEY_AAC_PROFILE = AACObjectLC was being set on an audio/opus encoder. Meaningless for Opus; stricter Codec2 stacks on Android 13+ reject the configure() call with IllegalArgumentException and surface as DeviceUnavailable. Removed. - The drain loop's INFO_OUTPUT_FORMAT_CHANGED branch had no progress guard. A buggy encoder re-emitting FORMAT_CHANGED without producing output would busy-spin against the 10 ms dequeue timeout. Now absorbed at most once per encode call. Same guard added to the decoder. Adds regression test: - frames_flow_completes_when_peer_FINs_immediately_after_Ok 224 tests pass, 0 failures. Android target compiles clean. |
||
|
Claude
|
eb20ac2c2a |
fix(nestsClient): bug + perf fixes from Nests implementation review
- send() now nulls currentGroup and returns false on uni-stream write failure (was: silently returned true while reusing the dead stream) - Subscribe bidi gets a single long-running collector that closes the frames channel on peer FIN / transport tear-down (was: consumer could hang forever on a black-holed UDP path with no Announce(Ended)) - AudioException.Kind gains EncoderError; mic-side encode failures no longer mis-route through DecoderError handlers - MoqCodec.decode bounds the payload-length varint before .toInt() so a peer-driven absurd length can't wedge the parser forever - Publish hot-path framing collapses to a single ByteArray allocation (was: Varint.encode + plus operator, two allocs per Opus frame at 50 fps) - Drops the now-unused readSubscribeResponseFromBidi / readSizePrefixedFromBidiInto / EarlyExit helpers Adds a regression test that asserts the frames flow completes when the peer FINs the subscribe bidi. 213 tests pass, 0 failures. |
||
|
Claude
|
d41a24f945 |
feat(nests): empty-stage hint, local hush, moderator/force-mute moderation
Three improvements stacked into one commit since they share files: #3 Empty-stage hint: StageGrid no longer disappears when nobody is on stage. The "Stage" label stays and a quiet "Waiting for speakers…" line keeps the strip visible so the room doesn't look broken before the first speaker arrives. #5 Local per-speaker hush: AudioPlayer gains a setVolume(Float) default-no-op method; AudioTrackPlayer composes mute and volume multiplicatively into AudioTrack.setVolume so a hushed stream stays silent regardless of mute state. NestViewModel exposes locallyHushed: ImmutableSet<String> in NestUiState plus setLocalHushed(pubkey, hushed) — applied at attach time so a re-subscribe of an already-hushed speaker stays silent. New "Hush this speaker" / "Restore this speaker" row in ParticipantHostActionsSheet, available to anyone (it affects only our own playback, nothing on the wire). #4 Host moderation gaps: wires Promote-to-Moderator using the existing RoomParticipantActions.setRole(ROLE.MODERATOR) builder — UI gap only, no protocol change. Adds a new AdminCommandEvent.Action.MUTE variant + AdminCommandEvent.forceMute(room, target) builder; the sheet emits it on "Force-mute speaker" and the AdminCommandsCollector dispatches incoming MUTE actions to a new NestViewModel.onForceMuted() that routes through the existing setMicMuted(true) path. Honor-based, same trust model as KICK — relays don't enforce signer authority, the client checks the signer is host or moderator on the active kind-30312. |
||
|
Claude
|
f9aa762d9b |
feat(audio-rooms): M1 listener-only wire-up — Connect button + Connected chip + mute
Wires `connectNestsListener` into the existing NIP-53 audio-room "stage" so
a user can tap Connect, see the live connection state, and hear hosts
speaking. Per the audio-rooms completion plan
(`nestsClient/plans/2026-04-26-audio-rooms-completion.md`) phase M1.
UI (amethyst):
- AudioRoomStage gets a state-chip / Connect / Disconnect / Mute row
above the existing hand-raise. State chip walks ResolvingRoom →
OpeningTransport → MoQ-handshake → Connected; Failed surfaces the
reason inline with a Retry button. The 30312 `service` tag drives
visibility — rooms hosted on non-nests servers show "Audio not
available" and the rest of the stage UI continues to work.
ViewModel (commons):
- New `AudioRoomViewModel` in commons/.../viewmodels/ owns one
`NestsListener` and one `AudioRoomPlayer` per active speaker, exposes
a single `StateFlow<AudioRoomUiState>`, and reconciles subscriptions
whenever the screen pushes a new speaker set via `updateSpeakers`.
Audio-pipeline construction is injected via `decoderFactory` /
`playerFactory` so a future desktop port can reuse the orchestration
once Compose Desktop has WebTransport. Unit-tested with a fake
`NestsListenerConnector` driving state transitions directly.
nestsClient:
- `AudioPlayer.setMuted(Boolean)` joins the interface; `AudioTrackPlayer`
routes it through `AudioTrack.setVolume(0f/1f)`. Mute keeps the
decode + network pipeline running so unmute is sample-accurate.
Build:
- `:commons` commonMain now `implementation(project(":nestsClient"))` so
the shared VM can see `NestsListener` / `AudioRoomPlayer` / interfaces.
Concrete OkHttp / Quic / MediaCodec / AudioTrack actuals stay in
`:nestsClient`'s platform source sets and are bound by an
Android-side `AudioRoomViewModelFactory` co-located with the screen.
Verified: `./gradlew spotlessApply :commons:jvmTest :nestsClient:jvmTest
:quic:jvmTest :amethyst:compilePlayDebugKotlin` all green.
|
||
|
Claude
|
b62e3dd0ec |
feat(nestsClient): MoQ ANNOUNCE family + Opus encoder + AudioRecord capture
Phase 4 (codec + audio capture slice). Adds the publisher-side MoQ
control messages and the Opus encode + microphone capture pieces a
speaker needs. The host-grants-speaker UI flow is deferred — that's
multi-screen UX that should be designed before being implemented.
commonMain (nestsclient.moq):
- 5 new MoqMessageType entries with codec round-trip + tests:
* Announce (0x06): publisher offers a track namespace.
* AnnounceOk (0x07): subscriber acknowledges.
* AnnounceError (0x08): subscriber rejects with error code +
reason phrase.
* Unannounce (0x09): publisher withdraws a previously-announced
namespace.
* SubscribeDone (0x0B): publisher tells the subscriber no more
objects are coming for this subscription, with stream count and
reason.
commonMain (nestsclient.audio):
- New `OpusEncoder` interface — symmetric to `OpusDecoder`, one
instance per outgoing track since Opus state is per-stream.
- New `AudioCapture` interface — `start()`, `readFrame()` returns a
PCM frame or null when stopped, `stop()` releases the mic.
androidMain (nestsclient.audio):
- `MediaCodecOpusEncoder` — wraps `MediaCodec("audio/opus")` encoder
variant (API 29+). 48 kHz mono in, 32 kbit/s VBR Opus out, 20 ms
frames. Drains output queue per encode call.
- `AudioRecordCapture` — wraps `AudioRecord` from
`MediaRecorder.AudioSource.VOICE_COMMUNICATION` so the platform's
echo-cancellation + noise-suppression filters apply when available.
Reads exactly one PCM frame per readFrame() call, retries on
underrun, throws AudioException(DeviceUnavailable) on permission/
resource failures.
commonTest:
- `AnnounceCodecTest` — 7 cases covering each message round-trip,
concatenated decode in sequence, and a guard against accidentally
reordering MoqMessageType enum codes.
Permission already declared:
- `RECORD_AUDIO` is already in amethyst/AndroidManifest.xml — no
manifest change needed.
What this does NOT include:
- A publisher loop class (analogous to AudioRoomPlayer for publish
direction) — can be added when the speak-button wiring lands.
- The host-grants-speaker UI — needs design input.
- STREAM_HEADER_SUBGROUP for stream-based object delivery — datagrams
cover the listener happy-path; streams add reliability that nests
may or may not require.
https://claude.ai/code/session_013nVLALALKaHVgHm9u5Cg8D
|
||
|
Claude
|
6e6fc4cabb |
feat(nestsClient): listener audio pipeline (Opus decode + AudioTrack)
Phase 3d-1 of the Clubhouse/nests integration. Adds the listener-side
audio pipeline that turns a SubscribeHandle's Flow<MoqObject> into
audible PCM through Android's MediaCodec + AudioTrack. Encoder /
AudioRecord (speaker publishing) lands in Phase 4.
commonMain (nestsclient.audio):
- `AudioFormat` constants — 48 kHz mono signed-16-bit, 20 ms frames
(960 samples), matching the nests Opus profile.
- `OpusDecoder` interface — stateful per-track decoder.
- `AudioPlayer` interface — start/enqueue/stop, suspend on backpressure.
- `AudioRoomPlayer` — wires a Flow<MoqObject> through OpusDecoder into
AudioPlayer. Decoder errors are surfaced via an `onError` callback
but don't tear down the loop (one bad packet shouldn't kill the
room); player errors are fatal. play()/stop() are single-shot per
instance, with stop() idempotent and double-release-safe.
- `AudioException` with three canonical kinds (DecoderError,
DeviceUnavailable, PlaybackFailed).
androidMain:
- `MediaCodecOpusDecoder` — wraps `MediaCodec("audio/opus")` with the
RFC 7845 §5.1 Opus identification header in csd-0 and zeroed
pre-skip / seek-pre-roll in csd-1 / csd-2. Drains the output queue
per-packet, handles INFO_OUTPUT_FORMAT_CHANGED gracefully.
- `AudioTrackPlayer` — wraps `AudioTrack` in MODE_STREAM with USAGE_
VOICE_COMMUNICATION / CONTENT_TYPE_SPEECH so the OS treats audio-
room playback like a phone call (call-volume rocker, ducks
notifications). Buffer = 4× minimum so the producer can fall behind
~80 ms (roughly the WebTransport datagram jitter on mobile).
Tests (commonTest, fake-based):
- `AudioRoomPlayerTest` — 7 cases covering happy-path decode +
enqueue, decoder failure with continued loop, stop idempotency,
double-play rejection, play-after-stop rejection, empty-PCM
skip-enqueue, and live channel-backed flow.
The real MediaCodec / AudioTrack code is thin glue against Android
APIs and is validated manually on device — no Robolectric here, the
seams are the Fake* doubles.
Next: Phase 3d-2 wires NestsClient end-to-end (HTTP auth → MoqSession
listener subscribe → AudioRoomPlayer) plus the Amethyst-side
AudioRoomViewModel. After that the only remaining pure-MoQ work is
the Kwik handshake (Phase 3b-2).
https://claude.ai/code/session_013nVLALALKaHVgHm9u5Cg8D
|