@dialt/sdk 0.23.0
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- package/CHANGELOG.md +336 -0
- package/LICENSE +202 -0
- package/NOTICE +12 -0
- package/README.md +319 -0
- package/THIRD_PARTY_LICENSES/README.md +9 -0
- package/THIRD_PARTY_LICENSES/abseil-Apache-2.0.txt +203 -0
- package/THIRD_PARTY_LICENSES/emscripten.txt +102 -0
- package/THIRD_PARTY_LICENSES/fft-Mark-Olesen.txt +25 -0
- package/THIRD_PARTY_LICENSES/libcxxabi-Apache-2.0-WITH-LLVM-exception.txt +311 -0
- package/THIRD_PARTY_LICENSES/musl.txt +193 -0
- package/THIRD_PARTY_LICENSES/ooura.txt +8 -0
- package/THIRD_PARTY_LICENSES/pffft-FFTPACK.txt +45 -0
- package/THIRD_PARTY_LICENSES/rnnoise-BSD-3-Clause.txt +31 -0
- package/THIRD_PARTY_LICENSES/spl-sqrt-floor-public-domain.txt +27 -0
- package/THIRD_PARTY_LICENSES/webrtc-BSD-3-Clause.txt +29 -0
- package/THIRD_PARTY_LICENSES/webrtc-PATENTS.txt +24 -0
- package/THIRD_PARTY_LICENSES/webrtc-audio-processing-BSD-3-Clause.txt +29 -0
- package/package.json +41 -0
- package/src/aec.js +181 -0
- package/src/aec3-wasm.js +0 -0
- package/src/ambience.js +508 -0
- package/src/audio.js +79 -0
- package/src/index.js +1798 -0
- package/src/mic-worklet.js +75 -0
- package/src/mic.js +155 -0
- package/src/player.js +347 -0
- package/src/track-feeder-worklet.js +80 -0
- package/src/webrtc.js +194 -0
- package/src/worklet-url.js +18 -0
package/CHANGELOG.md
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# Changelog
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## 0.23.0
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- The package is now published as `@dialt/sdk`; `@trelis/converse` is deprecated.
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- Package metadata and documentation now point to Dialt and
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[`dialt-ai/dialt`](https://github.com/dialt-ai/dialt).
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- The runtime API and wire protocol are unchanged.
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## 0.22.1
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- `startMic()` and the WebRTC track feeder retry their packaged source worklet when a CDN-transformed
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SDK entry cannot load its sibling asset. This fixes esm.sh imports, where the transformed entry
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lives under `es2022/` while the worklets remain under `es2022/src/`.
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## 0.22.0
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- Playback-health telemetry: the player counts mid-reply underruns (queue drained at the
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speaker) and, at each reply's `done`, the client sends
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`{"type":"client_event","event":"playback_report", underruns, starved_ms, max_gap_ms, turn_id}`
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over WebSocket transports. No behaviour change: buffering is untouched; the report is
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record-only on the server and sizes the jitter buffer from real sessions. Not sent over
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WebRTC, where the browser owns the jitter buffer.
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## 0.21.1
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- Removed `playAcknowledgements` and the `ack` frame handler: the server does not send assistant
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backchannel clips yet (roadmap, not shipped), so the option was a no-op and the docs overstated
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it. Passing the old option is still silently accepted.
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## 0.21.0 - 2026-08-30
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- `mode.end_call` (default `false`) declares the managed `end_call(farewell)` tool so the agent can
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end the session. When it calls the tool, Dialt speaks only the farewell, sends
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`session_end_requested` with that `farewell`, and closes after a short grace unless the user
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speaks. Without the flag the agent cannot end the session; the host ends it with `wrap_up` or by
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closing. While enabled, the name `end_call` is reserved, like `web_search` (`invalid_tools`). Pass-through only:
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validation and the default are the only client changes.
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## 0.20.0 - 2026-08-28
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- `mode.modality` selects `"voice"` (the default) or `"text"`. Text sessions use the same
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instructions, tools, history and event lifecycle without microphone or playback setup.
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- `sendText(text)` commits one `input_text` user turn in text mode and returns whether it was
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written to the live connection. In voice sessions `sendText` is unchanged: still the user-role
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`injectContext` shorthand returning the acknowledgement promise, so existing voice
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integrations keep working. Text rejects microphone capture and the WebRTC transport so
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integrations cannot accidentally open an unused media pipeline.
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## 0.19.1 - 2026-08-22
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- Tool declarations: `expected_duration` (`"instant"` / `"seconds"` / `"long"`) replaces
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`wait_for_tool` in the docs. It says what the caller should hear on a tool turn (the answer
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directly, or an acknowledgement first); left out, Dialt learns from observed results.
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`wait_for_tool: true` still passes through as a deprecated alias of `"instant"` during alpha and
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will be removed at beta. Pass-through only: no client code changed.
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## 0.19.0 - 2026-08-22
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- `ambience` constructor option (`'thinking'` DEFAULT; `'off'`, `'continuous'`; or an object with
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`mode` plus `afterS`/`fadeInS`/`fadeOutS`/`level`) and `client.setAmbience(mode)`: a soft
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generative bed rendered in the SDK and mixed THROUGH the SDK player, so it sits in the echo
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canceller's far-end reference on every transport and on WebKit/iOS (where the old playground
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bed on its own AudioContext had to be disabled). `'continuous'` plays it under the whole call
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from the first reply; `'thinking'` plays it only while Dialt is blocking on a tool result with
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nothing to say - fading in after ~1.5 s of silence and out again under the reply's first
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syllables, a real crossfade - so a caller waiting on a slow backend hears "still working"
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instead of dead air. The thinking sound is ON BY DEFAULT from this release (it plays nothing
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unless a tool wait runs long); pass `ambience: 'off'` to keep the old silence. Bed-only audio
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is never queued ahead of a reply and never counts toward
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barge `discarded_ms`. Same musical design and constants as the server-mixed WebRTC
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`background_audio` bed (the two renderers share the score, not the samples). WebSocket
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transport only: over webrtc the SDK player is not in the audio path, so the local ambience
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stays silent and `mode.background_audio` is the option there.
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- New server event `working` (`active: true|false`): Dialt is blocking on a tool result
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(client tools, web_search, think_deeply) with nothing audible, or that wait ended. Drives the
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thinking sound; also usable for a "working..." UI state.
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- `StreamingPlayer.setUnderlay(bed)` / `resumeUnderlay()`: the underlay path the ambience uses.
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Audio is now resampled at schedule time rather than at enqueue (no wire or API change).
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## 0.18.0 - 2026-08-18
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- `mode.background_audio` (default `false`): a server-mixed background underscore that plays for
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the whole call, so the silence between turns feels connected. **Requires
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`transport: 'webrtc'`** — the bed rides the server's playout track, the only downlink that runs
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continuously between turns. The SDK drops the field with a console warning rather than sending
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it on any non-WebRTC session, which matters because WebKit downgrades `webrtc` to `ws` for you;
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without that, asking for both would cost the whole session on iOS instead of just the music.
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## 0.17.0 - 2026-08-17
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- **Breaking (inert):** removed the reversible-playback protocol. The broker stopped sending
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`playback_pause`/`playback_resume` when the ink-decider barge fallback was deleted, so this
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client half has been dead code since. `StreamingPlayer.pause()`, `.resume()` and `.paused` are
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gone, custom players no longer need them, and start frames no longer advertise
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`playback_pause_v1` (`client.capabilities` is now always `[]`). Barge handling is unchanged:
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`interrupted` still fade-clears and reports `discarded_ms`/`remaining_ms`.
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## 0.16.0 - 2026-08-16
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- Interactions accept a `resolver` binding (`{tool, args, option_args, answer_arg}`): once the ask
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has been voiced, subsequent user turns are constrained to an explicit transition
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(resolve / clarify / supersede / cancel) via the broker-managed `interaction_transition` tool,
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and `resolve` executes the bound client tool with host-declared arguments — the model picks the
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option, never the arguments. See docs/client-tool-protocol.md §3a.
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## 0.15.0 - 2026-08-16
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- `tool_choice` lands in `ConverseMode` and as `setToolChoice(choice, { oneShot })` — the familiar
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OpenAI/Gemini restriction vocabulary (`"auto"` | `"none"` | `"required"` | `{allowed: [...]}` |
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`{tool: "..."}`). `required`/`allowed`/`tool` constrain the first planning round of each user
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turn; `none` withholds declared client tools while broker protocol tools stay available;
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`oneShot` reverts after the next user turn. Unknown names are rejected server-side with the new
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`invalid_tool_choice` error code; `setTools` resets the choice to `"auto"`.
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## 0.14.1 - 2026-08-16
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- `sendToolInteractionUpdate` normalizes whitespace-padded interaction ids the same way the server
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does, so the ack (which echoes the stripped id) correlates instead of surfacing as a timeout;
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ack correlation is additionally state-matched, so a server answer arriving after a client-side
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timeout can no longer be handed to the next same-id caller.
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## 0.14.0 - 2026-08-16
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- Interactions now carry a stable identity: pass `interaction.id` in `sendToolPartialResult` (or
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read the broker-derived id from `tool_job_narration.interaction_ids`), then close an ask without
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completing its parent call via `sendToolInteractionUpdate(id, interactionId, state, { note })`
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(`state`: `resolved` | `cancelled` | `superseded`). Queued or actively-speaking narration for it
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stops, the model is told not to act on it, and the returned promise resolves with the server's
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deterministic `tool_interaction_update_ack` (late/duplicate/unknown updates come back
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`applied: false` with a stable `reason`). Add `interactionState(interactionId)`; narration and
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interaction state caches now reset on connection loss (a resumed session implicitly supersedes
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any open interaction — re-raise it if still needed).
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## 0.13.1 - 2026-08-15
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- `sendToolPartialResult(id, content, { interaction })` marks a partial as needing a user decision
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(docs/client-tool-protocol.md §3a): unlike `reply: true` alone, it is never silently dropped when
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the floor is busy — it queues and preempts pending completion narration, with a far more
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persistent delivery retry. Add `narrationState(jobId)` and `waitForNarrationState(jobId, states,
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{ timeoutMs })` to track its queued/started/superseded/cancelled lifecycle via the new
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`tool_job_narration` server frame.
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## 0.13.0 - 2026-08-13
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- `startMic()` now resolves only after the first AudioWorklet frame. If an opened capture produces
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no frames within the bounded startup window, the SDK fully releases it and reacquires once; a
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repeated stall rejects with the structured `capture_stalled` error. Silent frames remain valid.
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- Add `warming_up`, `listening`, `recovering`, and `failed` capture lifecycle events so host apps can
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render status without implementing their own retry or readiness fallback.
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- Add audio-input enumeration and selection with `getInputDevices()`, `setInputDevice(deviceId)`,
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the `inputDeviceId` option/property, `devices_changed`, and `input_device_changed`. Active capture
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follows relevant `devicechange` events through guarded track restart.
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- Make `stopMic()` a cancellation barrier and harden stop/retry, overlapping device-switch, and
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delayed WebRTC track-replacement races without leaking tracks, worklets, or AudioContexts.
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## 0.12.3 - 2026-08-11
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- Tool result, progress, deferred, partial-result, and cancellation methods now return whether a
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live transport accepted the frame, allowing durable bridges to retain controls across reconnects.
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## 0.12.2 - 2026-08-11
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- Keep a newer successful connection authoritative when an older failed WebSocket delivers its
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close event late, so explicit capacity retries cannot clear the live transport.
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## 0.12.1 - 2026-08-11
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- Bound correlated injection acknowledgement waits and reject with cleanup when an older or
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mismatched broker never returns `inject_context_ack`.
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## 0.12.0 - 2026-08-11
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- `injectContext(text, {messageId, role, reply})` now returns the broker's authoritative
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accepted/rejected acknowledgement. The client-generated ID and `input_source: "text"` are
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echoed on the canonical final `asr` event; spoken transcripts carry `input_source: "voice"`.
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- `sendText(text, {messageId})` exposes the same correlation and acknowledgement contract.
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## 0.11.0 - 2026-08-11
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- Add `sendText(text)` as the concise alias for a typed user turn that should receive a reply.
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Typed and spoken turns use the same `asr` event contract, including a stable `turn_id`.
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## 0.10.0 - 2026-08-10
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- Add a supported, versioned Browser SDK resume-state API for full page reloads:
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`resumeState`, `exportResumeState()`, `importResumeState()`, and the `resume_state` event. The
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state rotates with each server token and clears on terminal or rejected sessions, so apps can
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safely keep a tab-scoped `sessionStorage` copy without reading private SDK fields.
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- Add `injectContext(text, {role, reply})` as the supported Browser SDK surface for the existing
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`inject_context` protocol frame, including proactive host announcements with `reply: true`.
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## 0.9.0 - 2026-08-09
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- Add `mode.silence_nudge_s` / `mode.silence_end_s` to override the broker's two-stage idle
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policy (check-in nudge, then sign-off + end) per session — useful for benchmark harnesses or
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flows with long think-time. Both default to the broker's env-configured values (10s/20s) when
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omitted; the broker also falls back to those defaults if either value is non-positive or
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`silence_end_s` does not exceed `silence_nudge_s`.
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## 0.8.0 - 2026-08-08
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- License Dialt-authored Browser SDK code under Apache 2.0; bundled AEC components remain under
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the third-party terms reproduced in the package.
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- Keep assistant playback on the browser's unity-gain path, with no SDK limiter, software boost,
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output-route switch, or `navigator.audioSession` manipulation.
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- Document that device volume, physical routing, and mobile full-duplex attenuation are controlled
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by the browser and operating system; native integration is required for routing guarantees.
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- Document the Browser SDK support matrix across Chromium, Brave, Firefox and Safari, including
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the automatic WebKit fallback from experimental WebRTC to WebSocket.
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## 0.7.0 - 2026-08-07
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- Add `sendToolDeferred(id, {handle, statusLabel})` for jobs that outlive their originating voice
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turn. Eligible tools opt in with `deferred`, `deferred_timeout`, and `notify_on_complete`.
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- Deferred jobs remain addressable by their original call ID or host handle for progress,
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cancellation, and exactly one terminal result. The SDK forwards `tool_deferred_ack` and
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`tool_deferred_resume` events through the normal typed and catch-all event APIs.
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- Tool interruption is unchanged: barge-in never cancels deferred work; explicit cancellation does.
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## 0.6.0 - 2026-08-06
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- **Hidden-tab playback fix.** Browsers throttle `setInterval` to ~1 Hz in hidden tabs while the
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AudioContext keeps running, so the player's 0.2 s scheduling horizon starved playback into
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0.2 s bursts with ~0.8 s gaps whenever the user wasn't looking at the page. The player now
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commits a 2.5 s horizon while `document.visibilityState === 'hidden'` (and tops up immediately
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on `visibilitychange`), keeping playback gapless; the short barge-friendly horizon is unchanged
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in visible tabs, and barge/clear still stops committed-but-unplayed sources at any horizon.
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- **Tool replies from anywhere.** `ConverseClient` gains `sendToolResult(id, content)`,
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`sendToolProgress(id, note)`, `sendToolPartialResult(id, content, {reply})` and
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`sendToolCancel(id)`, mirroring the Python SDK. Listen for `tool_call` events and answer them
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from the page or relay them to a backend — no raw-socket sidecar needed.
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- `binaryToFloat32` and `toWebSocketUrl` are re-exported from the package root for raw-socket
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integrations.
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- Server-side (deployed independently): the per-tool `timeout` ceiling rises from 120 s to 600 s
|
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for long-running agentic tools.
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## 0.5.0 - 2026-08-06
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- **Downlink audio is PCM16 on the wire — required update.** The server's downlink default
|
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changed from Float32 to PCM16 (bandwidth-halving negotiation, `start.audio.output_encoding`).
|
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0.4.5 and earlier decode the downlink as Float32 and therefore produce noise or decode errors
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against current servers. This release requests `output_encoding: "pcm16"` explicitly in the
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start frame, decodes Int16, and fails `connect()` loudly if the server's `ready.audio` field
|
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announces a format other than pcm16 at 16 kHz instead of playing garbage.
|
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245
|
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- `transport: 'webrtc'` option (experimental): carries the session over a WebRTC peer connection
|
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|
+
with a native remote audio track; the WS protocol remains the default and recommended path.
|
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+
Includes bounded ICE gathering, the `webrtc_connect_failed` error, and mic dropout/resampler
|
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fixes from the field-test rounds.
|
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|
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## 0.4.5 - 2026-07-30
|
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|
|
252
|
+
- **Remove the speaker/earpiece output option entirely (`audioOutputMode`, `setAudioOutputMode`,
|
|
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|
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and the 0.4.1 `<audio>`-element sink)** — settled by an 11-configuration on-device experiment
|
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|
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(iPhone 17, Chrome + Safari): modern iOS routes web audio to the loudspeakers in every reachable
|
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|
+
configuration, "earpiece AND speaker at once" is normal iPhone stereo playback (the receiver is
|
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|
+
the second stereo speaker), and no configuration reaches earpiece-only. Web pages have no output
|
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routing to control, so the SDK ships none: playback is a plain
|
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258
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+
`AudioContext` → master gain → `destination` path on every platform. The
|
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+
never-touch-`navigator.audioSession` regression guard remains.
|
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## 0.4.4 - 2026-07-30
|
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|
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- **Revert 0.4.3's `navigator.audioSession` usage — field-falsified on real iPhones (Chrome and
|
|
264
|
+
Safari).** Setting `type = 'playback'` (speaker mode) before capture made `getUserMedia` fail
|
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with a mic-permission error — a broken mic in the DEFAULT mode — and `'play-and-record'`
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|
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(earpiece mode) still produced earpiece+loudspeaker dual output. Speaker routing is back on the
|
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0.4.2 `<audio>`-element sink (mic works; known-imperfect dual output remains the open bug). A
|
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regression test now pins that the SDK never touches `navigator.audioSession`.
|
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`setAudioOutputMode()` (live switching) and the speaker/earpiece option remain.
|
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## 0.4.3 - 2026-07-30
|
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|
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|
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- `ConverseClient.setAudioOutputMode(mode)`: the 0.4.2 speaker/earpiece choice is now live-switchable
|
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|
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mid-session (previously took effect on the next `Start` only) — re-wires `StreamingPlayer`'s output
|
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|
+
route and, on WebKit, sets `navigator.audioSession.type` (`'playback'` for speaker, `'play-and-record'`
|
|
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|
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for earpiece; Safari 16.4+, feature-detected, a no-op elsewhere). This directly targets the
|
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|
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documented WebKit routing decision the `<audio>`-element sink workaround (0.4.1) could only work
|
|
278
|
+
around indirectly — field testing on iPhone showed the 0.4.1 fix alone produced simultaneous
|
|
279
|
+
earpiece + loudspeaker output rather than a clean switch, and `navigator.audioSession.type` is the
|
|
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|
+
platform's own API for this exact "mic is live, but I want loudspeaker anyway" case.
|
|
281
|
+
- The two mechanisms are mutually exclusive, not stacked: where `navigator.audioSession` exists
|
|
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|
+
(Safari 16.4+), the 0.4.1 `<audio>`-element sink route is skipped entirely (`hasAudioSessionApi()`
|
|
283
|
+
in `aec.js`) — running both at once would leave two live output paths fighting over the same
|
|
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|
+
routing decision, the same shape that produced the dual-output field result. The element sink
|
|
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|
+
remains only as the pre-16.4 WebKit fallback.
|
|
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|
+
|
|
287
|
+
## 0.4.2 - 2026-07-30
|
|
288
|
+
|
|
289
|
+
- Add the `audioOutputMode` (`'speaker' | 'earpiece'`, default `'speaker'`) `StreamingPlayer`
|
|
290
|
+
constructor option so apps can explicitly choose between the loudspeaker route (0.4.1) and the
|
|
291
|
+
platform's own call-audio/earpiece routing on iOS/WebKit, instead of only ever getting one. No
|
|
292
|
+
effect on platforms without that fork (desktop, Android).
|
|
293
|
+
|
|
294
|
+
## 0.4.1 - 2026-07-30
|
|
295
|
+
|
|
296
|
+
- Route assistant playback through a sink `<audio>` element (instead of `AudioContext.destination`
|
|
297
|
+
directly) on iOS/WebKit, to avoid the call-audio session routing playback to the earpiece
|
|
298
|
+
receiver instead of the loudspeaker while a mic stream is active (WebKit bug 218012). Pending
|
|
299
|
+
on-device confirmation.
|
|
300
|
+
|
|
301
|
+
## 0.4.0 - 2026-07-29
|
|
302
|
+
|
|
303
|
+
- Publish the package publicly on npm with installation and authentication guidance.
|
|
304
|
+
|
|
305
|
+
- Advertise reversible playback only for players that implement pause and resume, and report the
|
|
306
|
+
actual SDK-owned microphone/AEC frontend across reconnects for echo diagnostics.
|
|
307
|
+
- Make web search opt-in by default.
|
|
308
|
+
- Add `setMicEnabled(enabled)` to gate SDK-owned microphone tracks without reopening capture.
|
|
309
|
+
- Add the default-on `playAcknowledgements` constructor option so half-duplex integrations can
|
|
310
|
+
suppress automatic backchannel playback while retaining `ack` and `audio` event dispatch.
|
|
311
|
+
- Add `setVoice(voice)` to switch character voice from the next reply and reassert it on reconnect.
|
|
312
|
+
- Add `sendAmbienceState(active)` to report the client's ambience state on the session timeline.
|
|
313
|
+
|
|
314
|
+
## 0.3.1 - 2026-07-16
|
|
315
|
+
|
|
316
|
+
- Enable web search by default while preserving an explicit `webSearch: false` opt-out.
|
|
317
|
+
|
|
318
|
+
## 0.3.0 - 2026-07-16
|
|
319
|
+
|
|
320
|
+
- Add the opt-in `webSearch` session capability.
|
|
321
|
+
|
|
322
|
+
## 0.2.1 - 2026-07-15
|
|
323
|
+
|
|
324
|
+
- Gate `playback_pause_v1` to desktop Chromium and Firefox until physical WebKit validation.
|
|
325
|
+
|
|
326
|
+
## 0.2.0 - 2026-07-15
|
|
327
|
+
|
|
328
|
+
- Add synchronized processed/raw uplink framing for raw-assisted barge detection.
|
|
329
|
+
- Keep WebKit on one physical raw capture teed through SDK WASM AEC.
|
|
330
|
+
- Add reversible `playback_pause` / `playback_resume` handling for backchannels.
|
|
331
|
+
- Add AEC configuration plumbing and explicit desktop/WebKit engine controls.
|
|
332
|
+
- Fail closed to processed-only audio when raw capture or classification is unavailable.
|
|
333
|
+
|
|
334
|
+
## 0.1.0
|
|
335
|
+
|
|
336
|
+
- Initial browser SDK extracted from the Dialt web client.
|
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To apply the Apache License to your work, attach the following
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boilerplate notice, with the fields enclosed by brackets "[]"
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183
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replaced with your own identifying information. (Don't include
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the brackets!) The text should be enclosed in the appropriate
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comment syntax for the file format. We also recommend that a
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file or class name and description of purpose be included on the
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same "printed page" as the copyright notice for easier
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identification within third-party archives.
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Copyright [yyyy] [name of copyright owner]
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Licensed under the Apache License, Version 2.0 (the "License");
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you may not use this file except in compliance with the License.
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You may obtain a copy of the License at
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http://www.apache.org/licenses/LICENSE-2.0
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Unless required by applicable law or agreed to in writing, software
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distributed under the License is distributed on an "AS IS" BASIS,
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WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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201
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See the License for the specific language governing permissions and
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limitations under the License.
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package/NOTICE
ADDED
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@@ -0,0 +1,12 @@
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1
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Dialt Browser SDK
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2
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Copyright 2026 Dialt
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3
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Dialt-authored portions of this SDK are licensed under the Apache License, Version 2.0.
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5
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6
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The bundled src/aec3-wasm.js includes third-party WebRTC audio-processing and Emscripten
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7
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components, including libc++abi. Their copyright, license, disclaimer, and patent texts are
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8
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reproduced under
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THIRD_PARTY_LICENSES/ and continue to govern those components.
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10
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11
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Neither the Apache license nor these third-party licenses grant rights to the hosted Dialt
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12
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service, its models, Dialt trademarks, or server-side software.
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