leehack · Sep 24, 2026 · Sep 23, 2026 · Sep 23, 2026 · Sep 23, 2026 · Sep 23, 2026
@@ -43,11 +43,16 @@
 - Extend the GGUF speech-to-text validation pack with four synthetic edge
   fixtures built in-process, so no extra audio is stored: generated digital
   silence, plus a truncated RIFF, a stereo 44.1 kHz re-encode and a 33-second
-  concatenation crossing the 30-second preprocessing boundary, the last three
-  derived from the locked `jfk.wav`. Every GGUF STT pack run executes the four
-  checks and each one gates `functional_pass`; the LiteRT-ASR and TTS packs
-  pass no edge fixtures and keep their existing check count
+  concatenation, the last three derived from the locked `jfk.wav`. Every GGUF
+  STT pack run executes the four checks and each one gates `functional_pass`;
+  the LiteRT-ASR and TTS packs pass no edge fixtures
   ([#325](https://github.com/leehack/llamadart/issues/325)).
+- Raise every `stt`, `tts` and `litert-asr` speech validation pack run from 8
+  to 15 lifecycle checks: an immediate cancel, three
+  cancel/dispose/load/generate cycles, and bounds that fail the run when a
+  cancellation takes over 500 ms to end its task or the peak resident set
+  exceeds 1.10x the one sampled after the first generation
+  ([#594](https://github.com/leehack/llamadart/pull/594)).
 - Force greedy `topK: 1` for zero-temperature LiteRT-LM Web generation, matching
   the native clamp
   ([#548](https://github.com/leehack/llamadart/issues/548)).

diff --git a/doc/cross_platform_validation.md b/doc/cross_platform_validation.md
@@ -993,7 +993,12 @@ had not seen the task finish when it cancelled; it cannot show that the
 generation had begun. Exceeding any budget fails the run; if resident memory
 cannot be sampled, the memory bound records `SKIP` with a reason, and that is
 the only check a passing run may leave unmeasured.
-GGUF STT additionally compares file and bytes inputs.
+GGUF STT additionally compares file and bytes inputs and runs four generated
+edge fixtures: digital silence must fail with the typed empty-transcript
+`LlamaSpeechException`, a truncated RIFF must yield an inexact non-empty
+transcript or `LlamaAudioFormatException`, a 44.1 kHz stereo copy of
+`jfk.wav` must yield the reference, and three concatenated copies (33 s) must
+yield it three times.
 TTS rejects silent, nonfinite or truncated output; playability is not a
 listening-quality assertion. Its first playable audio is
 the final buffer, never a progress callback. The voice report preserves the

diff --git a/example/chat_app/README.md b/example/chat_app/README.md
@@ -11,9 +11,10 @@ A Flutter chat application demonstrating real-world usage of llamadart with UI.
 - 📋 **Clipboard attachments**: Paste screenshots or copied image/audio files
   with `Cmd/Ctrl+V`, or use **Paste attachment** from the attachment menu on
   touch devices. Plain-text paste continues to work normally.
-- 🎙️ **Whole-file transcription**: Compatible native GGUF ASR models can
-  transcribe a selected file or capture a foreground microphone recording,
-  then transcribe it after **Stop & transcribe**.
+- 🎙️ **Whole-file transcription**: Compatible GGUF ASR models, on native and
+  on Web with bridge assets `v0.1.30+`, can transcribe a selected file or
+  capture a foreground microphone recording, then transcribe it after
+  **Stop & transcribe**.
 - 📝 **Live dictation**: Native chat models, including generic audio-chat
   models, can use a separately installed LiteRT sidecar to show confirmed and
   pending English text while the user speaks, then place the final text in the
@@ -119,7 +120,12 @@ flutter test --run-skipped -t local-only \
      for up to five minutes; **Stop & transcribe** finalizes it, runs whole-file
      STT, and deletes the native file or revokes the browser blob.
      Capture is foreground-only and cancelling discards the temporary
-     recording. This Qwen path remains whole-file rather than live. For native chat
+     recording. Real-model checks cover only WAV input of at most 33 seconds.
+     On native macOS arm64 CPU, 297 seconds of audio filled a 4,096-token
+     context, the preset's size, and transcription completed with a truncated
+     transcript and no error
+     ([#636](https://github.com/leehack/llamadart/issues/636)).
+     This Qwen path remains whole-file rather than live. For native chat
      models, including generic audio-chat models, the composer can separately
      install a live-dictation model/tokenizer and expose **Live transcription**
      on Android, iOS, macOS, and Windows. Moonshine Tiny is the recommended
@@ -159,6 +165,7 @@ flutter test --run-skipped -t local-only \
      supported. It remains hidden on Linux because the recorder plugin can
      report startup before its required external tools are ready; selected-file
      transcription still works there.
+   - The complete Qwen3-ASR model/projector, microphone, and final-transcript
      flow has passed on a physical Pixel using CPU inference and in the iOS
      Simulator. This is not yet physical-iPhone or Windows validation.
    - The native Gemma 4 E2B presets expose **Ask with voice** when either the
@@ -167,7 +174,7 @@ flutter test --run-skipped -t local-only \
      30 seconds; **Stop & ask** sends the encoded WAV bytes through normal
      multimodal chat and asks the model to answer the spoken request. This is not
      `SpeechToTextEngine`: it has no transcript, timestamp, confidence, or live
-     partial-text contract. Qwen3-ASR continues to use the separate five-minute
+     partial-text contract. Qwen3-ASR continues to use the separate
      **Stop & transcribe** flow, and takes precedence for models declared as
      ASR profiles.
    - The voice-question UI is code-supported on Android, iOS, macOS, and

diff --git a/lib/src/core/speech/speech_platform_stub.dart b/lib/src/core/speech/speech_platform_stub.dart
@@ -13,7 +13,7 @@ bool get speechToTextSupportsFileInput => true;
 /// Whether byte inputs require explicit encoding metadata on this platform.
 bool get speechToTextRequiresEncodedAudioFormat => false;
 
-/// Encoded formats validated for prompt-adapted speech input.
+/// Encoded formats accepted for prompt-adapted speech input.
 Set<String> get speechToTextEncodedAudioFormats => const <String>{
   'wav',
   'mp3',

diff --git a/packages/llamadart_validation/assets/speech/README.md b/packages/llamadart_validation/assets/speech/README.md
@@ -17,10 +17,10 @@ reference or threshold to match the output of a failing run.
 The `stt` pack also builds four edge fixtures in process, so no extra audio is
 stored: digital silence, plus three derived from `jfk.wav` — a truncated RIFF
 whose header declares more audio than the bytes carry, a stereo 44.1 kHz
-re-encode, and a 33-second concatenation crossing the projector's 30-second
-`audio_chunk_len`. Their required outcomes are measured behavior, not
-aspiration, except that the truncated RIFF may equally be rejected with a typed
-`LlamaAudioFormatException` should decoding start validating the header.
+re-encode, and a 33-second concatenation. Their required outcomes are measured
+behavior, not aspiration, except that the truncated RIFF may equally be
+rejected with a typed `LlamaAudioFormatException` should decoding start
+validating the header.
 `speech-results.json` records each generated fixture's SHA-256 under
 `edge_fixtures`, keyed by the same id as its check.
 

diff --git a/packages/llamadart_validation/lib/src/speech_edge_fixtures.dart b/packages/llamadart_validation/lib/src/speech_edge_fixtures.dart
@@ -129,7 +129,7 @@ List<SpeechEdgeFixture> buildSpeechEdgeFixtures(Uint8List fixture) {
       seconds: concatenated.length / 16000,
       contract: SpeechEdgeContract.repeatedReference,
       referenceRepeats: repeats,
-      rationale: 'Audio crossing the 30-second preprocessing boundary.',
+      rationale: 'Audio over 30 seconds must be transcribed in full.',
     ),
   ];
   for (final entry in fixtures) {

diff --git a/packages/llamadart_validation/lib/src/speech_runner.dart b/packages/llamadart_validation/lib/src/speech_runner.dart
@@ -9,7 +9,8 @@ import 'runner.dart' show redactDiagnostic;
 import 'speech_edge_fixtures.dart';
 
 /// Word edit distance divided by reference words; insertions can exceed 1.0.
-/// Normalization ignores case and ASCII punctuation, preserving Unicode words.
+/// Normalization lowercases, replaces `.,!?:;"—–` with spaces, then splits on
+/// whitespace.
 double speechWordErrorRate(String reference, String actual) {
   List<String> words(String text) => text
       .toLowerCase()

@@ -71,7 +71,9 @@ const List<TestMatrixRow> testMatrixRows = <TestMatrixRow>[
     tier: 'platform',
     mode: 'opt-in; local model-backed',
     covers:
-        'Locked Qwen3-ASR/Qwen3-TTS and dedicated LiteRT CPU ASR, cancellation, reload, typed guards, WER and audio timing',
+        'Locked Qwen3-ASR/Qwen3-TTS and dedicated LiteRT CPU ASR, cancellation, '
+        'reload, typed guards, WER, audio timing, Qwen3-ASR edge fixtures, and '
+        'cancel-latency and peak-memory bounds',
     command:
         'dart run tool/testing/run_local_e2e.dart --scenario validation-speech-stt; '
         'dart run tool/testing/run_local_e2e.dart --scenario validation-speech-tts; '

@@ -50,11 +50,16 @@ For canonical full release notes, use:
 - Extend the GGUF speech-to-text validation pack with four synthetic edge
   fixtures built in-process, so no extra audio is stored: generated digital
   silence, plus a truncated RIFF, a stereo 44.1 kHz re-encode and a 33-second
-  concatenation crossing the 30-second preprocessing boundary, the last three
-  derived from the locked `jfk.wav`. Every GGUF STT pack run executes the four
-  checks and each one gates `functional_pass`; the LiteRT-ASR and TTS packs
-  pass no edge fixtures and keep their existing check count
+  concatenation, the last three derived from the locked `jfk.wav`. Every GGUF
+  STT pack run executes the four checks and each one gates `functional_pass`;
+  the LiteRT-ASR and TTS packs pass no edge fixtures
   ([#325](https://github.com/leehack/llamadart/issues/325)).
+- Raise every `stt`, `tts` and `litert-asr` speech validation pack run from 8
+  to 15 lifecycle checks: an immediate cancel, three
+  cancel/dispose/load/generate cycles, and bounds that fail the run when a
+  cancellation takes over 500 ms to end its task or the peak resident set
+  exceeds 1.10x the one sampled after the first generation
+  ([#594](https://github.com/leehack/llamadart/pull/594)).
 - Force greedy `topK: 1` for zero-temperature LiteRT-LM Web generation, matching
   the native clamp
   ([#548](https://github.com/leehack/llamadart/issues/548)).

diff --git a/website/docs/examples/chat-app.md b/website/docs/examples/chat-app.md
@@ -74,7 +74,11 @@ flutter test
   current LiteRT-LM chat bundles. Microphone capture is enabled on Android,
   iOS, macOS, Windows, and supported secure browser origins; Linux capture
   remains disabled pending a safe external-recorder preflight, while
-  selected-file transcription remains available there.
+  selected-file transcription remains available there. Recordings stop at five
+  minutes, but real-model checks cover only WAV input of at most 33 seconds. On
+  native macOS arm64 CPU, 297 seconds of audio filled a 4,096-token context, the
+  preset's size, and transcription completed with a truncated transcript and no
+  error ([#636](https://github.com/leehack/llamadart/issues/636)).
 - Experimental live English dictation for native chat models, including
   generic audio-chat models, through independently installed, checksum-pinned
   LiteRT sidecars. Moonshine Tiny is the recommended 54 MB default; Parakeet
@@ -95,8 +99,7 @@ flutter test
   bytes through ordinary multimodal chat so the model can answer the spoken
   request. It does not use `SpeechToTextEngine` and provides no transcript,
   timestamp, confidence, or live-partial contract. Qwen3-ASR keeps the separate
-  five-minute **Stop & transcribe** workflow and takes precedence for ASR
-  profiles.
+  **Stop & transcribe** workflow and takes precedence for ASR profiles.
 - A dedicated cross-platform Qwen3-TTS mode backed by `TextToSpeechEngine`.
   Type an utterance, optionally choose a language and select or record
   speaker-reference audio, then cancel synthesis, automatically play the

diff --git a/website/docs/guides/speech-to-text.md b/website/docs/guides/speech-to-text.md
@@ -20,7 +20,7 @@ multimodal models.
 
 | Runtime | Generic audio-input chat | Typed `SpeechToTextEngine` | Text to speech |
 | --- | --- | --- | --- |
-| Native llama.cpp / GGUF | Model + projector dependent | Experimental Qwen3-ASR adapter; complete WAV/MP3/FLAC file or bytes, final transcript only | Experimental Qwen3-TTS adapter; see [Text to Speech](./text-to-speech) |
+| Native llama.cpp / GGUF | Model + projector dependent | Experimental Qwen3-ASR adapter; complete WAV/MP3/FLAC file or bytes (real-model validation covers WAV only), final transcript only | Experimental Qwen3-TTS adapter; see [Text to Speech](./text-to-speech) |
 | WebGPU / GGUF | Bridge + model dependent | Experimental Qwen3-ASR adapter with bridge assets `v0.1.30+`; complete WAV bytes, final transcript only | Unsupported |
 | Native LiteRT-LM | Separate `.litertlm` audio chat remains bundle dependent | Experimental dedicated CPU ASR through `SpeechToTextEngine.liteRtLm`; mono 16 kHz float PCM, partial/final text, streaming input | Unsupported |
 | LiteRT-LM Web | Unsupported | Unsupported | Unsupported |
@@ -51,7 +51,7 @@ final recognizer = SpeechToTextEngine.liteRtLm(
 
 final capabilities = await recognizer.capabilities;
 if (!capabilities.isSupported) {
-  throw StateError(capabilities.unsupportedReason);
+  throw StateError(capabilities.unsupportedReason!);
 }
 
 final session = await recognizer.startStream();
@@ -69,6 +69,7 @@ for (final chunk in mono16KhzFloatPcmChunks) {
 await session.finish();
 final completion = await session.done;
 await events.cancel();
+print(completion.state);
 ```
 
 The input contract is mono 16 kHz normalized `Float32List` PCM. The public
@@ -158,10 +159,12 @@ preflight fails before a task can start. After startup, `events` is a
 single-subscription stream: runtime failure is emitted as a stream error and
 the same terminal condition is available through `task.done`.
 
-Native llama.cpp currently decodes WAV, MP3, and FLAC file or byte inputs. Raw
-PCM remains unsupported for that prompt adapter because projector sample rates
-are model-specific. Dedicated LiteRT-LM accepts `SpeechAudioPcmInput` for a
-complete mono 16 kHz float buffer, or the incremental session shown above.
+Native llama.cpp accepts WAV, MP3, and FLAC file or byte inputs. Only WAV is
+validated with a real model; native tests check only that the adapter accepts
+MP3 and FLAC, and no test decodes them. Raw PCM remains unsupported for
+that prompt adapter because projector sample rates are model-specific.
+Dedicated LiteRT-LM accepts `SpeechAudioPcmInput` for a complete mono 16 kHz
+float buffer, or the incremental session shown above.
 
 WebGPU accepts encoded WAV bytes only. Browser file pickers must read the
 selected file into memory and use `SpeechAudioBytesInput`; local filesystem
@@ -242,11 +245,12 @@ LiteRT-LM chat bundles. Live dictation uses the native LiteRT-LM streaming STT
 API with an app-managed sidecar; it is not a capability of the selected chat
 bundle and remains unavailable on Web.
 ASR microphone recordings are capped at five minutes, cancelled when the app is
-backgrounded, and deleted on native or revoked on Web after transcription. This
-remains a whole-file workflow: it does not produce live partial transcripts
-while the user speaks. The recorder requests 16 kHz mono WAV, but hardware or
-the browser may choose another valid sample rate; the downstream decoder reads
-the WAV metadata.
+backgrounded, and deleted on native or revoked on Web after transcription.
+The repository's real-model checks cover at most 33 seconds of audio; see
+[Known limits](#known-limits). This remains a whole-file workflow: it does not
+produce live partial transcripts while the user speaks. The recorder requests
+16 kHz mono WAV, but hardware or the browser may choose another valid sample
+rate; the downstream decoder reads the WAV metadata.
 The live sidecar path instead requests PCM16 mono 16 kHz streaming, preserves
 samples split across arbitrary byte-chunk boundaries, applies one in-flight
 worker push at a time, and caps each session at five minutes. It is currently
@@ -284,6 +288,37 @@ native and Web. Its immutable artifact revision, byte sizes, and SHA-256
 digests are pinned. Native downloads verify both files before selection; Web
 uses origin-scoped Cache Storage and the pinned source metadata.
 
+## Validated behavior
+
+The `validation-speech-stt` pack
+(`dart run tool/testing/run_local_e2e.dart --scenario validation-speech-stt`)
+runs the checksum-locked Qwen3-ASR 0.6B Q8_0 model and projector on `jfk.wav`,
+an 11-second English WAV, as a file and as bytes. It scores each transcript
+against the reference by word error rate, after lowercasing both and replacing
+`.,!?:;"—–` with spaces. It also sends four generated WAV inputs as bytes:
+
+| Input | Required outcome |
+| --- | --- |
+| 3 s of digital silence | `LlamaSpeechException` with the message `Speech recognition produced an empty transcript.` |
+| The first 20,044 bytes of `jfk.wav`, whose RIFF header declares more audio than the bytes carry | A non-empty transcript that is not the reference, or `LlamaAudioFormatException`. Every recorded run returned a short transcript and no error. |
+| `jfk.wav` resampled to 44.1 kHz stereo | The reference transcript |
+| `jfk.wav` three times (33 s) | The reference three times |
+
+Each run then repeats three cancel/dispose/load/generate cycles. It fails if
+either budget is exceeded:
+
+- **Cancellation, 500 ms**: from `cancel()` to the task's terminal state, for
+  every cancel issued as soon as `transcribe` returns and every cancel issued
+  after half the duration of the most recent completed generation. This bounds
+  when the task ends for its caller, not when native work stops.
+- **Memory, 1.10x**: the largest whole-process resident set sampled after the
+  checks that follow the first generation, as a multiple of the one sampled
+  right after that generation.
+
+With native `v0.4.1-1`, the pack has passed on macOS arm64 with CPU and with
+Metal, and on Linux x64 with CPU (AMD EPYC 7B12). The Metal runs report the
+Metal backend; the pack does not verify GPU execution.
+
 ## Known limits
 
 - Qwen3-ASR may emit a leading `language English<asr_text>` marker. llamadart
@@ -293,12 +328,21 @@ uses origin-scoped Cache Storage and the pinned source metadata.
   diarization. Incremental audio and partial text are LiteRT-LM-only.
 - Inference backend correctness and performance remain device dependent;
   establish a CPU baseline before claiming GPU support for a deployment.
-- The local real-model smoke has passed on macOS arm64 CPU, and a separate
-  chat-app/manual smoke has passed on Metal. The full chat-app
-  model/projector, microphone, and final-transcript flow has also passed on a
-  physical Pixel using CPU inference and in the iOS Simulator. A physical
-  iPhone and Windows remain unverified; Linux keeps selected-file STT but not
-  microphone capture. Web requires `v0.1.30+`, a browser with enough memory for
+- The repository's real-model checks use WAV input of at most 33 s; none uses
+  MP3 or FLAC. Longer input is unvalidated. The audio prompt grows with
+  duration (3,890 tokens for 297 s in
+  [#636](https://github.com/leehack/llamadart/issues/636)). On native, if it
+  fits in `contextSize` but leaves too little room for the transcript, the task
+  completes with a truncated transcript and no error.
+- Beyond [Validated behavior](#validated-behavior), chat-app microphone
+  transcription on CPU has passed on a physical Pixel and in the iOS Simulator
+  ([#328](https://github.com/leehack/llamadart/pull/328)), and CPU file
+  transcription plus chat-app microphone transcription have passed on a
+  physical iPad ([#462](https://github.com/leehack/llamadart/pull/462)). No
+  Qwen3-ASR run covers Android x64, Linux arm64, Windows arm64 or x64, macOS
+  x86_64, or a physical iPhone, nor the Vulkan, CUDA, HIP, OpenCL, or BLAS
+  backends. Linux keeps selected-file STT but not microphone capture.
+- Web requires `v0.1.30+`, a browser with enough memory for
   the roughly 1.02 GB model/projector pair, and the targeted
   `web-speech-to-text-smoke` validation row. That row verifies both browser
   file selection and Chromium fake-device microphone capture with the same WAV