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>
xAI TTS supports a text_normalization boolean that normalizes
written-form text (numbers, abbreviations, symbols) into spoken-form
before synthesis. This parameter was not being sent, leaving users
with literal number/symbol pronunciation.
- Add DEFAULT_XAI_TEXT_NORMALIZATION_DEFAULT = False
- Read tts.xai.text_normalization from config (via _xai_bool_config)
- Attach text_normalization: true to the POST payload when enabled
- Omit the field entirely when unset or explicitly false (API default)
- Add 3 tests: default omission, enabled passthrough, explicit false
The `text_to_speech` tool schema accepted only `text` and `output_path`,
so style direction (tone, emotion, pacing, whispering) could never reach
the OpenAI backend — even though `gpt-4o-mini-tts` (Hermes's OpenAI
provider default) treats `instructions` as its primary voice-design
control.
This plumbs an optional `instructions` argument through the tool schema,
the handler lambda, and `text_to_speech_tool()` into
`_generate_openai_tts`, where it is forwarded to
`client.audio.speech.create()` only when truthy. Empty/None values still
omit the key entirely, preserving behavior on `tts-1`/`tts-1-hd` and
strict OpenAI-compatible servers.
The same passthrough unblocks self-hosted OpenAI-compatible voice-design
servers (Qwen3-TTS-VoiceDesign on oMLX, etc.) that are already wired in
via `tts.openai.base_url` — the established convention per #9004 and the
TTS config docs — without inventing a new provider backend.
Tests: `tests/tools/test_tts_instructions.py` covers backend passthrough,
tool-level threading, schema declaration, and the empty-string/absent
omission cases. `tests/tools/test_tts_max_text_length.py` fake_openai
signature widened to accept the new kwarg.
Refs NousResearch/hermes-agent#14196
Add optional 'speed' parameter (0.25-4.0) to the text_to_speech tool
schema and handler. When provided by the model, it overrides the
config-level tts.speed setting, enabling per-request speed control
without config changes.
Use cases:
- Language learning: slow playback (0.5x) for pronunciation practice
- Accessibility: adjustable speed for hearing preferences
- Content review: accelerated playback for long text
The speed value is clamped to [0.25, 4.0] and injected into tts_config
before dispatching to any provider (Edge, OpenAI, MiniMax, etc.), so
all existing provider-level speed handling works transparently.
Includes 3 new tests for tool-level speed injection, clamping, and
config preservation when speed is not specified.
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.
Auto-TTS previously fed raw chat Markdown and compact symbols straight to the
speech provider, so units were read as stray letters and headings or bullets ran
together. This routes spoken text through a new normalizer,
tools/tts_text_normalize.prepare_spoken_text, that expands units (for example a
temperature written with the degree symbol becomes "degrees Celsius") and
flattens Markdown into a transcript-like script with sentence pauses.
The normalizer is best-effort: if it ever fails the code falls back to the
previous markdown-strip behavior, so auto-TTS keeps working. The Telegram voice
caption uses the same normalized text. Includes a unit test.
The native Gemini provider profile's default_aux_model and the curated
model picker catalog were still pinned to gemini-3.5-flash, a stale
generation now superseded by gemini-3.6-flash (documented GA). Bump
both so the auxiliary-task default and the picker stay in sync with
the current model.
Contract test asserts the durable lockstep invariant only
(default_aux_model is a member of _PROVIDER_MODELS["gemini"]) rather
than pinning either side to a frozen model-name string, so it doesn't
need updating on the next model-generation bump.
Salvaged from PR #62977 (@Vissirexa) — TTS model-cache half only (the
hindsight turn-buffer half is a different subsystem and was dropped).
_piper_voice_cache and _kittentts_model_cache were keyed by voice/model
with no eviction, and each entry is a whole loaded model (tens of MB).
A surface that sweeps voices pinned one model per voice for the process
lifetime. New _tts_cache_get_or_load() get-or-loads through a small LRU
(_TTS_MODEL_CACHE_MAX=3), refreshing recency on a hit and evicting the
least-recently-used model on a cold miss.
Class-level sweep following the ElevenLabs salvage (#66311): every cloud
TTS provider section now honors tts.<provider>.base_url. xAI, MiniMax,
Gemini, OpenAI and DeepInfra already did; Mistral (SDK server_url) was
the remaining gap. Adds per-provider config tests locking in the
ElevenLabs environment plumbing and the Mistral server_url passthrough.
Follow-up to salvaged PR #70307 (@aml1973): OpenAIStreamer now checks
tts.openai.api_key (config.yaml) ahead of the env resolver in both
available() and stream(), completing config parity between the sync
and streaming OpenAI TTS paths.
Use the shared OpenAI audio key resolver and prefer tts.openai.base_url over the global environment fallback in the Desktop streaming path. Add focused regression coverage for credential and endpoint propagation.
Salvaged from PR #26233 (@LeonSGP43), rebased onto the current 3-tuple
_resolve_openai_audio_client_config (is_managed flag, post-#73072 layout).
Same fix independently submitted earlier in PR #26209 (@zccyman) — credit
to both.
Resolution order now mirrors the STT resolver: tts.openai.api_key/base_url
from config.yaml -> VOICE_TOOLS_OPENAI_KEY/OPENAI_API_KEY env (still
honoring config base_url) -> managed gateway. _has_openai_audio_backend
also counts a config api_key as an available backend.
Fixes#26175
