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).
_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).
/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.
Two in-house micro-fixes from issue triage:
- #49883: voice.beep_enabled gates in cli.py and hermes_cli/voice.py used
bool() on the config value, so a quoted YAML string like "false" or
"off" kept beeps on. Route through utils.is_truthy_value.
- #18432: AudioRecorder.start() collapsed OSError from _import_audio into
the 'pip install sounddevice numpy' hint — but OSError means the
PortAudio SHARED LIBRARY is missing, which pip cannot fix. Mirror
detect_audio_environment's system-package hint (libportaudio2 /
brew portaudio / Termux pkg install portaudio) on that path.
Fixes#49883Fixes#18432
Addresses review on #62601. Applies a single rule — no sounddevice for
audio OUTPUT on macOS (PortAudio/CoreAudio init triggers a
kTCCServiceMediaLibrary prompt) — consistently at all three output sites:
- play_audio_file: WAV playback (already routed to afplay) now uses the
shared _sounddevice_output_allowed() helper.
- play_beep: synthesize the tone with numpy only, then on macOS play it
via a temp WAV through afplay instead of sounddevice.
- stream_tts_to_speaker (tts_tool): on macOS, skip the sounddevice
OutputStream so playback falls through to the existing tempfile/afplay path.
Audio INPUT (recording) is untouched — it legitimately needs mic permission.
Tests: TestMacOSAudioOutputPolicy (voice_mode) proves WAV + beep routing
does not import sounddevice on Darwin and still uses it off Darwin;
test_tts_macos_output proves streaming TTS skips the OutputStream on Darwin.
Existing test_play_wav_via_sounddevice pinned to non-Darwin for determinism.
Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
Two test classes lock in the new detection contract:
1. TestTermuxApiAppInstalledProbeLadder — drives _termux_api_app_installed
with a fake subprocess.run dispatcher and walks each rung of the
ladder:
- non-Termux env returns False (no probes run),
- pm-confirms returns True (back-compat),
- pm-clean-miss → cmd-confirms returns True,
- pm-FileNotFoundError → cmd-confirms returns True,
- pm-TimeoutExpired → cmd-confirms returns True,
- pm-nonzero-exit → cmd-confirms returns True,
- both probes inconclusive + binary on PATH returns True (the
core #31015 case),
- both probes inconclusive + no binary returns False,
- both probes clean-miss returns False (the genuine "CLI without
app" case keeps the existing warning),
- case-insensitive package match for ROMs that capitalise differently.
2. TestDetectAudioEnvironmentTermuxFallback — end-to-end through
detect_audio_environment, asserting the misleading "Termux:API
Android app is not installed" warning no longer fires when probes
are inconclusive but the binary is on PATH (the user-reported #31015
symptom), AND that the warning still fires when probes can
conclusively report the app is missing.
Refs: NousResearch/hermes-agent#31015
Ports #63768 forward onto current main per teknium1's review.
On WSL2 without a PulseAudio bridge, ffplay and aplay have no audio
device and TTS playback silently fails (issue #17608). When
powershell.exe and ffmpeg are available, convert the audio to a
uniquely-named WAV in the Windows %TEMP% directory and play it via
Media.SoundPlayer.
Per review, this fixes two gaps in the original port:
1. Exit-status masking: the cleanup subshell was
'( ffmpeg && powershell ); rm -f wav' -- the shell's exit status is
the LAST command's (rm -f, which is always 0), so a real
ffmpeg/PowerShell failure could never be detected by the rc-checking
fallback logic added to the player loop. Now captures the real
status before cleanup and re-exits with it:
'( ffmpeg && powershell ); rc=0; rm -f wav; exit '.
2. The no-Pulse WSL gate in detect_audio_environment() still hard-blocked
voice mode entirely (input AND output) even when the PowerShell
fallback made TTS output viable. Added _wsl_powershell_tts_available()
and use it to downgrade the WSL-without-Pulse case from a hard
'warnings' block to a non-blocking 'notices' entry when the fallback
is available -- the same PulseAudio-bridge recording guidance is still
surfaced (mic capture genuinely still needs it), it just no longer
blocks /voice on for TTS-only usage. cli.py's existing
env_check['available'] gate needed no changes since it already
respects this flag.
Also fixed the flaky uniqueness test (the original asserted
len(filenames) >= 2, which passed trivially on zero captured
filenames) and added a real fallback-triggering regression test for
the exit-status fix.
10 new/fixed tests pass in TestWSL2PowerShellFallback and the new
TestWSLAudioEnvironmentGate; 80/80 in the full tests/tools/test_voice_mode.py file.
AudioRecorder hard-codes SAMPLE_RATE (16 kHz) when opening the input
stream, but some capture devices (e.g. USB microphones exposed through
ALSA hw) reject 16 kHz outright — sd.InputStream fails with
PaErrorCode -9997 (Invalid sample rate) and voice recording is broken.
Query the default input device for its native default_samplerate at
recording start and open the stream / write the WAV at that rate,
falling back to the Whisper-friendly 16 kHz constant when the backend
does not expose a usable rate. STT providers accept standard WAV rates,
so downstream transcription is unaffected.
Closes#55908. The CLI voice-mode beep amplitude is hardcoded at 0.3 inside
tools.voice_mode:play_beep(), which makes the record start/stop cues too
quiet on low-volume systems and headphones. Users couldn't adjust it
without editing source.
Move the literal into a configurable voice.beep_volume setting (clamped to
0.0-1.0, default 0.3 to preserve prior behaviour). The new
_get_beep_volume() helper reads via the same load_config() pattern used by
cli.py's _voice_beeps_enabled() and hermes_cli/voice.py's _beeps_enabled(),
keeps bools / out-of-range / non-numeric / NaN values safely on the default,
and falls back silently if config can't load so the audio cue never breaks
the voice loop on a degenerate config.yaml.
Covered by tests/tools/test_voice_mode.py:
- TestGetBeepVolume (12 cases: missing key, custom value, boundary 0.0/1.0,
out-of-range clamp, type coercion, bool guard, NaN guard, exception
guard, dict-typed voice section)
- TestPlayBeepVolumeWiring (guards against re-introducing a hardcoded 0.3
literal in play_beep)
Docs: website/docs/user-guide/configuration.md mentions the new key.
Other locale translations (zh-Hans etc.) intentionally untouched —
handled by the regular i18n sync pipeline as a separate change.
No change in default behaviour: existing users hear exactly the same beep.
Review follow-up: the sweeper is right that the first commit only cured
half the dead config — the TUI gateway builds its recorder params
explicitly, so the cap never reached recordings started from the TUI.
- start_continuous() grows a max_recording_seconds param (default 0.0 =
disabled, so existing callers keep today's behaviour) and applies it to
the shared recorder next to the silence params
- tui_gateway voice.record start forwards the validated cap; corruption
semantics now mirror the silence params everywhere: non-numeric/bool
falls back to the documented 120 default (a hand-edited
`max_recording_seconds: true` must not become a 1-second cap), while
an explicit numeric <= 0 disables the cap
- cli.py wiring updated to the same corruption semantics
New coverage, per review:
- TestMaxRecordingCap drives the mocked InputStream callback past the
cap during continuous loud speech (the silence branch physically can't
fire there) and asserts the one-shot callback fires exactly once; plus
the disabled-cap negative
- TestMaxRecordingSecondsConfigReal pins the CLI config assignment for
the valid / zero / bool / garbage cases via _voice_start_recording
- test_voice_record_start_forwards_max_recording_seconds pins the TUI
forwarding for the same matrix
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.
Consolidates all TTS text-preparation paths onto
tools/tts_text_normalize.prepare_spoken_text:
- strip_nonspoken_blocks: removes <think> reasoning blocks (#34213,
incl. unterminated streaming blocks) and the end-of-turn
file-mutation verifier footer emitted by run_agent.py (#40772).
- flatten_newlines_for_payload: collapses newlines into sentence
breaks so newline-sensitive OpenAI-compatible providers (Kokoro)
speak the whole script instead of truncating at the first newline
(#9004).
- tools/tts_tool._strip_markdown_for_tts (voice-mode streaming + web
dashboard path) now delegates to the shared cleaner, with the legacy
regex pipeline kept as a best-effort fallback.
- hermes_cli/voice.py speak_text and cli.py _voice_speak_response now
use the shared cleaner instead of their own duplicated regex
pipelines.
- gateway auto-TTS fallback also strips think blocks.
Tests: tests/tools/test_tts_prepare_spoken.py covers think blocks,
verifier footer, emoji, newline flattening, and the shared-cleaner
wiring on the tool/streaming/gateway paths. Updated the header
expectation in test_voice_cli_integration.py for the heading-fold
behavior of the shared cleaner.
Closes#34213, #9004, #40772
Address review feedback on #31693:
- Regression tests assert extra_body == {"lang_code": ...} is forwarded
when tts.openai.language is configured, and omitted when unset/empty
- Document tts.openai.language as intended for OpenAI-compatible
endpoints that support lang_code (e.g. Kokoro-FastAPI)
Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
