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.
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 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)
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).
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).
Follow-up for salvaged #65023/#53020: _prepare_busy_steer_text now calls
_transcribe_and_echo_pending_voice (the same helper the interrupt monitor
and pending-drain paths use) instead of a private transcription+echo copy,
so out-of-band voice pays one STT call per platform message and the echo
respects the count-based ledger from #67281. can_steer now accepts events
whose attachments are all STT-eligible voice media, completing the steer
half of #58780. Adds extract_media gating tests for #44826 and the
contributor mapping for chefboyrdave21.
_invalidate_pending_stt_cache() clears the gateway-side transcription
cache when merge_pending_message_event() folds a follow-up message into a
still-pending event, so the next transcription picks up the merged text
and attachments. It also cleared _gateway_pending_stt_echo_sent, but that
flag is not derived state — it records that the transcript was already
delivered to the user.
Dropping it makes the re-run transcription echo the earlier notes a second
time. Both merge branches are affected, including the text-only follow-up
case where no new audio arrived at all: there the cache is invalidated,
the same voice note is transcribed again (a second paid STT call) and the
same line is echoed again.
Sequence:
1. voice note arrives, interrupt monitor transcribes it and echoes
'🎙️ "hello"'
2. user sends a follow-up while the turn is still pending, so it merges
3. drain path re-transcribes and echoes '🎙️ "hello"' a second time
Keep the ledger out of the invalidation set and track it as a count of
already-echoed transcripts instead of a single boolean. A count is what
the merge case actually needs: re-running transcription over the extended
media list returns the earlier transcripts as a prefix of the new one, so
echoing only the unsent tail suppresses the repeat while still surfacing a
newly merged voice note. A count rather than a set of seen values, so two
separate notes that transcribe identically stay two distinct deliveries —
covered by test_pending_stt_merge_echoes_two_identical_transcripts.
The guard stays within the 12-line window that
test_all_gateway_transcript_echo_sends_are_gated enforces over run.py.
Two hand-written fixes in the voice input path:
- _handle_voice_channel_input now resolves the bound text channel's
channel_prompt via the adapter's _resolve_channel_prompt so voice input
gets the same per-channel context as typed messages (fixes#50149).
- _enrich_message_with_transcription now guards success=True results whose
transcript is empty/whitespace-only (silence, cut-off, inaudible audio):
instead of emitting empty quotes the agent gets a clear sentinel note.
Reimplemented against the current plain-quoted note wording; original
concept and tests by @deacon-botdoctor in PR #41603 (fixes#41603).
Keep one owner for PATH/local-prefix ffmpeg discovery: ffmpeg_utils now
delegates to tools.transcription_tools._find_ffmpeg_binary and only adds
the Discord-specific FFMPEG_PATH override and Windows winget fallback on
top (follow-up to PR #60627 by @LauraGPT, fixes#60624).
Discord's voice socket needs a brief warm-up before receiving clients
actually hear audio; the first ~100-200ms is lost, clipping the first
word/syllable of TTS playback. Prepend a configurable lead of silence to
speech on both playback paths:
- Mixer path: new _lead_silence_bytes() helper prepends PCM silence
(BYTES_PER_MS constant added to voice_mixer.py) before play_speech, on
both the reply and the pre-tool ack.
- Legacy FFmpegPCMAudio path: apply -af adelay=<ms>:all=1.
Tunable via discord.voice_fx.lead_silence_ms (default 200, 0 disables).
Fixes#66827
- Wire adapter._voice_input_callback at connect and reconnect so voice
transcription is forwarded without requiring /voice join (#60623).
- Add optional text_channel_id and source params to DiscordAdapter
.join_voice_channel() so automatic/programmatic voice joins can
establish the text-channel binding needed by _handle_voice_channel_input.
- Add TestVoiceInputCallbackWiring: asserts callback wiring on startup
and reconnect for Discord adapters with voice attributes.
PR #61407 accesses self._handle_voice_channel_input in _platform_reconnect_watcher. The test mock runner created via _make_runner() must have this attribute.
- tests/gateway/test_base_auto_tts_output_format.py (new): the base
adapter auto-TTS block passes an explicit .ogg output path on every
OPUS_VOICE_PLATFORMS member (parametrized from the tts_tool set — the
single source of truth), keeps .mp3 on non-opus platforms, honors the
tool's success flag, and stays unique/uuid-based.
- tests/gateway/test_auto_voice_reply_format.py: runner _send_voice_reply
parametrized across Matrix/Feishu/WhatsApp/Signal (.ogg) alongside the
existing Telegram/Slack cases; streamed+global-auto-TTS gate regression.
- tests/gateway/test_telegram_voice_caption_markdown.py (new): caption
MarkdownV2 formatting, entity-rejection plain fallback, overflow skip,
and no-caption passthrough (#32029).
- test_voice_command.py filename-uuid contract test updated to point at
build_auto_tts_output_path (construction moved there).
Salvaged from PR #51196 (@55nx954gn6-debug). _should_send_voice_reply only
consulted the runner's _voice_mode dict (/voice on|voice_only|all), so the
global voice.auto_tts config default — which is synced into each adapter's
_auto_tts_default on gateway connect — was invisible to the runner path.
Net effect: with streaming enabled and only global auto-TTS configured
(no per-chat /voice opt-in), the streamed reply consumed the text, the
base adapter's auto-TTS got text_content=None, and no voice reply was
ever sent (#51867/#23983 remainder).
The runner now also asks the adapter's _should_auto_tts_for_chat(chat_id)
(which encodes per-chat /voice on|off overrides over the global default);
an explicit /voice off chat mode remains a hard override.
Refs #51867#23983#51282#13126
Salvaged from PR #68063 (@malaiwah). MatrixAdapter.send_voice now transcodes
any non-Ogg audio to Ogg/Opus at the adapter boundary (best-effort — the
original file is sent unchanged when ffmpeg is unavailable), so MSC3245
voice bubbles render even when a caller hands the adapter MP3/WAV audio.
_matrix_voice_metadata_for_file probing now runs via asyncio.to_thread so
ffprobe/ffmpeg subprocess timeouts can't stall the adapter event loop.
The PR's tools/tts_tool.py want_opus hunk was dropped: main's
OPUS_VOICE_PLATFORMS set (PR #73072) already includes matrix; the Matrix
opus-routing test is kept.
Refs #14841
Salvaged from PR #68063 (base commit, runner-path hunks superseded by the
platform-aware OPUS_VOICE_PLATFORMS fix in this branch). Element and other
Matrix clients render voice bubbles more reliably when m.audio events carry
duration and waveform metadata; probe both best-effort via ffprobe/ffmpeg.
Review follow-up: the spoken script is for synthesis only. Caption
eligibility and payload stay on the original reply, so a long reply
whose normalized script fits the 1024 char limit is still delivered
in full as its own message. Adds the long-original/short-normalized
regression case to the auto-TTS caption tests.
iOS Signal delivers voice notes as MP4-container AAC carrying an audio
ftyp brand ("M4A "). `_guess_extension()` returned ".mp4" for every
`ftyp` file regardless of brand, so those attachments were cached as
documents instead of audio and STT rejected the upload:
API error: Error code: 400 - Invalid file format.
Supported formats: ['flac','m4a','mp3','mp4','mpeg','mpga','oga',
'ogg','wav','webm']
Read the 4-byte brand at offset 8 and return ".m4a" for audio brands
("M4A ", "M4B ") so they satisfy `_is_audio_ext()` and route to
`cache_audio_from_bytes()`. Video brands (isom/mp42/avc1/qt) still
return ".mp4".
This mirrors the existing brand/form-type disambiguation already used
in this function for RIFF (WEBP vs WAVE) and for ADTS AAC vs MP3.
Verified against a real iOS Signal voice note: pre-fix the upload was
rejected with the 400 above; post-fix the same bytes transcribe
successfully.
The grant_spent notice fired for every subscription user with top-up
funds the moment their cap was reached and camped in the CLI/TUI status
bar and desktop toasts with no action to take — the account keeps
working off top-up. Remove it everywhere:
- agent/credits_tracker.py: drop grant_cond + the emit/clear block;
the dev fixture state now (correctly) produces no notice
- TUI: keep the turn-start clear of credits.grant_spent as back-compat
for older backends that still emit the key
- Desktop: drop the demo step and stale comment references
- Docs/config comments: remove grant-spent from credits_notices text
- Tests updated: policy/cold-start now assert the key never fires
Usage bands, depleted, and restored notices are unchanged; /usage still
reports the full balance breakdown.
The structural AST walker in test_compression_session_id_persistence.py
only recursed into control-flow children found via iter_child_nodes(stmt).
When a session_entry.session_id assignment lives inside an `else` block
whose statements are all assigns (no nested If/Try/etc to trigger
_walk_node), the assignment was invisible to the walker and the test
florped with 'No assignments found'.
Walk the stmt itself when it is a control-flow node so its
body/orelse/finalbody (and Try handlers) are always expanded, regardless
of whether iter_child_nodes yields an inner control-flow child.
Manual /compress already persists the rotated child transcript first and
only then repoints the live session_entry; a False rewrite_transcript
return keeps the entry on the original session_id so the conversation
stays reachable. Session hygiene auto-compress did the opposite: it
rebound session_id (and lease/topic) first, then called rewrite_transcript
without checking the return value. On a failed write the live entry
already pointed at an empty child SID and the turn continued — permanent
silent conversation loss. Persist first; rebind only after success.
Critical fixes to salvaged PR #73020:
- Use SessionDB.append_message instead of raw INSERT INTO messages.
The original used wrong column names (session_key/created_at vs
session_id/timestamp) and bypassed FTS indexing, session metadata
updates, display_kind, and all other columns append_message handles.
- Use get_hermes_home() instead of hardcoded Path.home()/'.hermes'.
Profile-aware path resolution under HERMES_HOME override and active
profile isolation.
- Add 11 tests covering flush, recovery, serialisation, edge cases.
- Change await self._send_read_receipt to asyncio.create_task to avoid
blocking message dispatch on slow bridge responses (matches BlueBubbles
pattern). Up to 5s per-message latency eliminated.
- Update tests: assert_called_once_with instead of assert_awaited_once_with
since the receipt is now scheduled, not directly awaited.
- Add send_read_receipts documentation to whatsapp.md following the
BlueBubbles docs pattern.
`@/Desktop` returned nothing while `@Desktop` worked. The leading slash
was always read literally, so the lookup went to the absolute `/Desktop`
— which doesn't exist — and dead-ended instead of finding the folder one
level down.
The `@` has already announced "this is a path", so the slash people type
next reads as a separator out of habit rather than a filesystem root.
Take the absolute meaning only when it resolves: the parent directory has
to exist, and a partially-typed segment has to match something in it.
Otherwise drop the slash and resolve from the cwd.
Real absolute paths are unaffected — `/usr`, `/etc/hos`, `/Users/...` all
pass the existence probe and keep their current behaviour. The guard
matters: stripping the slash unconditionally would let a repo-local
`etc/` shadow the real `/etc`.
The fuzzy branch of `complete.path` ranked basenames from
`_list_repo_files`, which lists files only, so a directory was only ever
reachable by typing a `/` — `@Desktop` returned nothing at all. Rank each
ancestor directory alongside the files, and break same-tier ties toward
the folder so `@docs` leads with `docs/` rather than `docs.md`.
Outside a git repo the fallback `os.walk` compounded this: it can spend
the whole `_FUZZY_CACHE_MAX_FILES` budget inside one deep subtree before
reaching a sibling, hiding top-level folders entirely. Seed the scan with
a `listdir` of the root so immediate children are always candidates.
Three mechanisms to detect and notify when gateway sessions stall silently:
1. Mid-turn activity heartbeats stamped to SessionDB so hermes sessions list
and hermes status show progress during long turns without new message rows.
2. Stall watchdog: when a busy session has pending inbound and the shared
activity clock is idle past agent.session_stall_timeout (default 300),
log a WARNING and notify the user once to try /new. Notify-only; does
not kill the turn.
3. Compaction timeout: fenceless compress_context callers get a progress-aware
host budget (compression.context_timeout_seconds default 120 idle,
compression.context_total_ceiling_seconds default 600 ceiling). On timeout,
cancel via commit fence, skip compaction without dropping messages, and
continue the turn.
Closes#72016 (slices 1-3; slice 4 cumulative SSE stream-retry deadline
remains a follow-up).
Cherry-picked from PR #72424 by @fangliquanflq.
