Review follow-up: test_micro_pause_tolerance_during_speech was left on real
sleeps, so it kept the wall-clock dependency the rest of this PR removes.
Its three sleeps (50 ms, 50 ms, 60 ms) feed the same monotonic gate at
tools/voice_mode.py:561 that the other two tests were converted for.
It was measured rather than assumed before being left out: 0 failures in 40
runs, because on Windows sleep() rounds up to the timer tick, so sleep(0.05)
really costs ~62.5 ms and the three sleeps clear the 150 ms gate by ~37.5 ms
-- wider than the 15.625 ms GetTickCount64() quantum that made the other two
flake. That margin is an accident of sleep granularity rather than anything
the test guarantees, so it is not worth keeping the bet.
Driving it from fake_clock makes the timeline exact (0.05 + 0.05 + 0.06 =
0.16 against the 0.15 gate, dip 0.05 under the 0.1 tolerance) and drops the
real sleeping from the suite. Mutation-checked: widening the gate at :561
fails the test.
test_silence_callback_fires_after_speech_then_silence and
test_custom_threshold_and_duration space their audio frames with
time.sleep(0.06) and check the result against 50 ms thresholds. That
leaves a 10 ms margin, which only survives if the clock is finer-grained
than the margin — and on Windows it isn't. time.monotonic() there is
GetTickCount64() with 15.625 ms resolution until CPython 3.13 moved it to
QueryPerformanceCounter(), so a sleep that really lasts 62.5 ms measures
as 46.9 ms often enough to matter, landing under the threshold. Depending
on which sleep got clipped, either the speech-confirm gate misses or the
silence timer never matures — which is why the same flake shows up on two
different asserts.
Both tests now advance a hand-driven clock instead of sleeping, so the
arithmetic is exact on every platform and neither test waits on real time.
The fixture patches the `time` name inside tools.voice_mode rather than
time.monotonic itself: voice_mode.time IS the stdlib module, so patching
through it would swap the clock out from under every other importer for
the duration of the test.
Measured on Windows 11 / Python 3.11.9, 20 runs of each test: 6/20 and
3/20 failed before, 0/30 after. The same machine on Python 3.13, where
monotonic() is QueryPerformanceCounter(), never failed either test — that
comparison is what pinned the clock resolution as the cause rather than
the detection logic. Linux CI never sees this: clock_gettime is
nanosecond-resolution there.
These tests mock the actual download; stub is_safe_url so host DNS/proxy
mappings for cdn.discordapp.com can't decide whether document handling is
exercised.
Residual hunk salvaged from PR #25426 — the rest of that PR's hermeticity
fixes (bedrock botocore fakes, faster_whisper module fakes, kittentts
plain-sequence PCM, web-provider SSRF stubs) have since landed on main in
equivalent form.
Running the suite could take over the machine it ran on. Two real
effects, both now closed:
- **The suite spoke through the speakers.** Once any test drove the
`voice.toggle` RPC with `action="tts"`, the handler set
`HERMES_VOICE_TTS=1` in the *live process environment*, and the flag
outlived that test. Every later test that drove a turn to completion
then fed its final response text to `hermes_cli.voice.speak_text` on a
background thread - real synthesis, real playback, no API key needed
(the default `edge` provider is keyless). A developer heard the fixture
string "partial answer complete" out loud. Because the flag is set from
inside the process, `scripts/run_tests.sh`'s `env -i` never protected
against this.
`tests/conftest.py` now blanks `HERMES_VOICE`/`HERMES_VOICE_TTS` per
test, and a new autouse `_audio_playback_guard` stubs `speak_text` and
its playback binding outright, so the speakers stay shut even inside the
test that sets the flag itself. `@pytest.mark.real_audio_playback` opts
out.
- **The suite launched Chrome.** `tests/tools/test_browser_supervisor.py`
spawned a real browser on any machine with Chrome on `PATH`. Its
docstring promised a `HERMES_E2E_BROWSER=1` gate that existed nowhere in
the code. That gate is now real, and the file is marked `integration`
so the default marker filter excludes it. `scripts/run_tests.sh`
forwards `HERMES_E2E_BROWSER` so the documented manual run still works.
Miscellanea
- `tests/test_audio_playback_guard.py`: regression cover for both defences,
driving the real `voice.toggle` handler rather than a stand-in.
Address teknium1 review on PR #47125: the existing adapter-level tests cover
the prerequisites (_download_voice / _collect_media / _extract_text) but not
the gateway-runner handoff. Add TestWeixinVoiceGatewayHandoff covering the
final routing contract:
- An inbound Weixin voice item carrying Tencent Cloud text is surfaced as a
VOICE MessageEvent whose media is audio/silk (the shape the runner keys off
to enter Hermes' STT pipeline), exercised through the real _process_message.
- That VOICE event's body does NOT leak Tencent's STT text, so the central
transcript replaces it rather than being trusted.
- A VOICE/audio/silk event reaches the real GatewayRunner
_enrich_message_with_transcription (patched as a spy), proving the runner
handoff the adapter-only tests missed.
When WeChat (Weixin) returns a voice_item.text (Tencent Cloud's STT),
Hermes previously trusted that text as the user-visible message body and
skipped downloading the raw audio. For non-Chinese audio that text is
garbage — the original report was a Russian voice message that came
back as English phonemes — and the user sees nonsense as their own
message. International users on the WeChat gateway effectively can't
use voice.
Two short-circuits in gateway/platforms/weixin.py caused this:
- _download_voice() returned None whenever voice_item.text was set,
so the raw SILK/Opus audio was never fetched.
- _extract_text() returned voice_item.text verbatim as the body,
so even if the audio had been downloaded, the central STT
pipeline in gateway/run.py never had a chance to replace it.
Fix both: always download the raw audio (the central pipeline picks
it up via _collect_media()), and skip voice items in _extract_text()
so the body comes from Hermes' own mlx-whisper / whisper.cpp /
faster-whisper transcription instead of Tencent's. Behavior is
unchanged when voice_item.text is absent (the original happy path
where audio was already being downloaded).
Tests:
- 5 new tests in TestWeixinVoiceAlwaysDownloaded covering both
functions, the _collect_media integration path, and a
regression guard for the text-item path.
- 74/74 in tests/gateway/test_weixin.py pass.
- Move U+FFFC placeholder detection before _record_last_inbound() so the
placeholder message id is not recorded as the reaction target
- Cancel pending U+FFFC tasks in disconnect() to prevent task leaks
- Check mimeType audio/x-caf in addition to filename for CAF→VOICE promotion,
fixing unnamed attachments that default to '(unnamed)'
- Add 7 Photon adapter tests: CAF named/unnamed promotion, U+FFFC no-dispatch,
U+FFFC not recorded as last inbound, U+FFFC+attachment cancel, timeout fire,
disconnect cleanup
- Add 6 transcription tests: ffmpeg conversion, afconvert fallback, all-fail,
CAF→WAV before Groq, conversion failure error, local provider skip
Cloud STT APIs (Groq, OpenAI, etc.) cannot parse Apple CAF containers.
Add .caf to SUPPORTED_FORMATS and _convert_caf_to_wav() which tries
ffmpeg (cross-platform) then afconvert (macOS built-in).
iMessage's gRPC cloud push fires before CloudKit syncs the attachment
metadata, emitting a U+FFFC placeholder as text. Instead of dispatching
a spurious message, start a non-blocking 15s wait. If the real attachment
arrives, cancel the timeout and process normally. If not, log a warning.
spectrum-ts classifies all inbound attachments as type 'attachment',
never 'voice'. Without this, .caf voice notes are routed as AUDIO or
DOCUMENT and bypass the STT pipeline.
Minimal fix: add 'if ct == "file": return False' before extension
matching. The original fallback logic is preserved for empty/unknown
content_types. Only the bug case (file uploads with audio extensions)
is fixed.
Removed the over-engineered _looks_like_voice helper.
Refined the fix: instead of removing extension-based fallback entirely,
only skip it when content_type is explicitly 'file' (or image/video).
Empty or unknown content_types still fall back to extension matching
as a defensive measure.
- content_type='voice' or 'audio/*' → True (API signal)
- content_type='file' → False (file transfer, never voice)
- content_type='' → extension fallback (defensive)
- content_type=unknown → extension fallback (defensive)
Added _looks_like_voice() module-level helper and comprehensive tests.
Update TestIsVoiceContentType to match the new behavior:
- Empty content_type with audio extensions → False (no sniffing)
- File upload with audio extension → False
- Added test_file_upload_with_audio_extension for the reported bug case
The _is_voice_content_type() heuristic matched audio file extensions
(.wav, .mp3, .ogg, etc.) even when the QQ Bot API explicitly reported
content_type='file'. This caused files sent via QQ's file-transfer
feature to be routed through the speech-to-text pipeline instead of
being saved as regular attachments.
The QQ Bot API already distinguishes voice messages (content_type=
'voice') from file uploads (content_type='file'), so filename-based
extension sniffing is unnecessary and harmful.
Removed the _VOICE_EXTENSIONS fallback; now only content_type is
checked.
Closes #XXXX
The msg_type_str == "richText" branch reset msg_type to PHOTO/TEXT after
the rich-text item scan had already promoted it (e.g. a native voice item
→ VOICE), dropping voice notes from the auto-STT path. Only re-derive when
the scan left the type at TEXT.
Ports the root-cause analysis from PR #38276 (stale, targeted the deleted
gateway/platforms/dingtalk.py) onto the live plugin adapter, with
regression tests.
Refs #38211#38219#38276
- Promote EXT_MAP to a module-level constant for reuse
- Classify DingTalk image messages and image file attachments as PHOTO
- Extract DingTalk card/interactiveCard document link content with defensive parsing
- Handle None, empty, JSON, and plain-string card content fields
- Add coverage: 8 card + 4 interactiveCard + 5 _extract_media = 17 test cases
- Remove a shadowed duplicate TestExtractText class (pytest collected only the later one)
3 fixes for the dingtalk-platform plugin (plugins/platforms/dingtalk/adapter.py):
1. _extract_text: parse ASR recognition text from voice messages via
extensions['content']['recognition'], fall back to fileName for
file messages ("[文件] xxx")
2. _extract_media: handle audio/file/image msgtype_str with correct
MIME mapping; exclude audio from media_urls to preserve DingTalk's
built-in ASR result (avoids failed whisper re-transcription)
3. _resolve_media_codes: parse downloadCode from extensions['content']
for file/image message types
All tested with live DingTalk group chat: voice → recognition text,
PDF/Markdown file → filename + download.
Trimmed cherry-pick of PR #40592 (duration + dedup hunks only; the
voice-classification hunk duplicates #29235 and the send_voice Opus
rewrite is out of scope for this inbound-focused PR):
- adapter.py: ffprobe duration (off-loop) attached to Feishu voice
uploads via _build_file_upload_body(duration=...) and the audio
message payload (#16524, #8300)
- gateway/run.py: TTS dedup narrowed to the current turn;
_enrich_message_with_transcription dedups repeated audio paths
Refs #40592#16524#8300
Lark's native "audio" msg_type is an in-app voice recording — uploaded
audio files arrive as "file"/"media". But _resolve_normalized_message_type
resolved the "audio" preferred type to MessageType.AUDIO, which the gateway
treats as a non-transcribed file attachment (run.py: AUDIO -> audio_file_paths,
"never STT"; VOICE -> audio_paths, "always STT"). Result: a Feishu voice
note reached the agent as an untranscribable audio attachment and was
silently ignored — the user's spoken message never became text.
Every other platform that receives native voice notes (Telegram, Discord,
Slack, WhatsApp, Signal, Matrix, WeChat, WeCom, DingTalk, QQ, BlueBubbles,
Mattermost, Yuanbao) classifies them as MessageType.VOICE. Feishu was the
only one classifying them as AUDIO. This is the follow-up to #28993, which
added native voice-note transcription for Discord + DingTalk but did not
cover Feishu.
Return MessageType.VOICE for the "audio" branch. The branch is reached only
for Lark's top-level audio msg_type (set in the normalizer; file uploads map
to "document"), so VOICE is unconditionally correct here — no risk of
auto-transcribing an uploaded music/audio file.
- plugins/platforms/feishu/adapter.py: _resolve_normalized_message_type audio
branch returns VOICE instead of resolving to AUDIO via mime.
- tests/gateway/test_feishu.py: test_extract_audio_message_downloads_and_caches
asserted the old AUDIO behavior on a fixture literally named voice.ogg —
updated to expect VOICE (the corrected classification).
- tests/gateway/test_feishu_voice_message_type.py: new focused regression
tests (audio->VOICE with and without mime; photo/document/text unaffected).
Rebased onto current main: the Feishu platform moved from the single-file
gateway/platforms/feishu.py into the plugins/platforms/feishu/ package; the
fix applies to the same _resolve_normalized_message_type logic at its new home.
Verified the new voice tests fail when the branch resolves to AUDIO and pass
with VOICE, while the photo/document/text cases are unaffected either way.
Note: classification is mock-tested here; the downstream STT pipeline is the
shared, already-proven path (#28993). End-to-end verification on a live
Feishu account would be a welcome confirmation.
Refs #28993 (sibling-gap: Feishu was the platform left uncovered).
Salvage of PR #52188. The original PR raised RuntimeError when LM Studio
load was rejected or unverifiable, which would abort agent startup on
transient network failures. Replace with logger.warning + fallback to
configured context length, preserving the old graceful-degradation behavior.
Bare-login noreply emails (no NNN+ numeric prefix) do not match the
check-attribution auto-resolve rule, so they need an explicit
contributors/emails/ file. Unblocks PR #73592.
* fix(desktop): allow camera capture through the permission handlers
The session permission hooks were written for the voice composer and denied
video outright, so any renderer getUserMedia({video}) failed with
NotAllowedError before the OS was ever consulted.
Rename isAudioCapturePermission to isMediaCapturePermission and accept video
alongside audio in both the request and check handlers. The OS capture
permission still applies, so the user keeps a real allow/deny.
* build(desktop): declare camera usage for signed macOS builds
A hardened-runtime build needs the camera entitlement and an
NSCameraUsageDescription string, or the packaged app is killed on first
camera access instead of prompting.
_generate_xai_tts passes stream=True to requests.post since the TTS
speaker pipeline moved to the streaming core. Four fake_post mocks in
test_tts_xai_speech_tags.py still had the pre-streaming signature and
raised TypeError: unexpected keyword argument 'stream'. The other 20+
mocks in the same file already accept stream=False; this aligns the
stragglers.
Fixes the 4 failures in Python tests slice 5/8 on main.
The codex app-server namespaces MCP tool call ids as
codex_mcp__<server>__<tool>_<codex_call_id>. With an exec-<uuid> component the
built-in hermes-tools server alone overflows the Responses API's 64-char
call_id limit, so the request 400s with a non-retryable "string too long".
The offending item sits near the head of the transcript and replays every
turn, permanently bricking the session — the only recovery is /reset.
Sibling defect to #10788, which clamped input[*].id via
_MAX_RESPONSES_ITEM_ID_LENGTH. Apply the same treatment to call_id at both
Responses emit sites in _chat_messages_to_responses_input: a deterministic
surrogate (call_ + sha256[:32]) for ids over the limit, short ids unchanged.
Because the surrogate is a pure function of the original id, a function_call
and its matching function_call_output — which carry the same original id — map
to the same surrogate and stay paired without correlating the two items.
Two log-spam bugs found in live gateway logs:
1. gateway_routing UNIQUE-constraint spam (261 warnings in one errors.log):
early builds of the #59203 routing-index migration created
gateway_routing with 'session_key TEXT PRIMARY KEY' and no scope
column. _reconcile_columns() ADDs the missing scope column but SQLite
cannot ALTER a primary key, so the shipped composite
PRIMARY KEY (scope, session_key) never lands on those databases. Both
write paths then fail on every save:
- save_gateway_routing_entry: 'ON CONFLICT clause does not match any
PRIMARY KEY or UNIQUE constraint'
- replace_gateway_routing_entries: 'UNIQUE constraint failed:
gateway_routing.session_key' whenever the same session_key exists
under another scope (e.g. test-suite scopes leaked into a live DB).
New _heal_gateway_routing_pk() rebuilds the table once with the
composite key, preserving rows (newest wins on collisions, NULL scope
coalesced to ''). Same one-time-heal pattern as the #51646 active-
column repair. Verified E2E against a copy of a real affected state.db.
2. pre_gateway_dispatch warned ''GatewayRunner' object has no attribute
'session_store'' and silently dropped the hook for every message on
partially-initialized runners (bare object.__new__ runners in tests,
and any future init-order change). Pass
getattr(self, 'session_store', None) so the hook always fires
(pitfall #17 pattern).
Both regression tests fail without their fixes (sabotage-verified).
The /api/audio/speak-stream WebSocket URL was always minted for the primary
backend with no profile param, so streaming read-aloud used the default
profile's TTS provider even when another profile was active. Mint the
ticket for the active profile's connection and carry ?profile= (same seam
as /api/pty), via a read-only getApiRequestProfile() accessor.
/api/audio/transcribe, /api/audio/speak, /api/audio/elevenlabs/voices, and
the /api/audio/speak-stream WebSocket resolved TTS/STT config from the
dashboard's own HERMES_HOME regardless of the active profile, so a
non-default profile's voice settings were silently ignored. Give all four
the same optional profile param as the rest of the dashboard surface,
entering _config_profile_scope (await-safe, config-only — the audio paths
touch no skills globals) inside their worker threads.
Backend half of the desktop fix; completes the renderer-side profileScoped()
threading. Fixes#53441#45506#66012#64057.
Generated media often arrives as image markdown (). The chat
markdown renderer maps the img element to MarkdownImage, which renders a raw
<img>; for a video/audio source the browser cannot paint it and shows a
broken-image icon even though the file is valid and plays fine. Images worked
only because <img src=...png> is valid markup.
Route video/audio sources in MarkdownImage to the existing MediaAttachment,
which already picks the correct <video>/<audio> element (streaming protocol +
open-externally fallback) by media kind. The intended MEDIA:-tag -> #media:
link path is unaffected; this only fixes the bare image-markdown case.
Since #57944 the image path is built on hooks (async src resolution, failure
and loading states), so the routing check cannot be a plain early return
inside it -- it would have to sit after every hook call and would still fire
an image resolve for media never rendered as an image. MarkdownImage is
therefore a thin hookless router in front of MarkdownImageContent, which is
the previous body unchanged.
The regression tests join the existing markdown-text.media.test.tsx added by
#57944 rather than replacing it.
Fixes#40896
Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_018vU45SEvcxLkVeyyEE9gJ2
