\`pm list packages com.termux.api\` is the canonical way to detect the
Termux:API Android app, but on some devices it gives a false negative
even when the app is installed and \`termux-microphone-record\` runs
fine — the symptom reported in #31015 (\`/voice on\` complaining
"Termux:API Android app is not installed").
Replace the single probe with a graded strategy:
1. Try \`pm list packages com.termux.api\` (current behaviour).
2. If \`pm\` isn't on PATH or returns non-zero, fall back to
\`cmd package list packages com.termux.api\` — the modern Android
API 28+ equivalent that's present on devices where \`pm\` is gone.
3. If both probes are inconclusive (binary missing, permission
denied, timeout, or non-zero exit) and \`termux-microphone-record\`
is on PATH, trust the binary. The CLI ships in the \`termux-api\`
package which is essentially only useful with the Android app
installed; users who installed it deliberately almost always have
the app too.
Polarity matters: a false negative on this gate blocks \`/voice on\`
entirely (the user-reported symptom), while a false positive only
surfaces a precise runtime error from the binary itself when it
tries to talk to the missing app — strictly more actionable.
The clean-probe-but-no-package case still returns False, so the
existing "Termux:API CLI installed without the app" warning still
fires when the package manager *can* tell us 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.
WSLg RDP audio has two issues causing crackling:
1. systemd-timesyncd clock adjustments jitter PulseAudio timing
(microsoft/wslg#1257) — user action: stop the service
2. Cold-start RDP connection drops first ~100ms of audio packets
before the virtual channel stabilises
Fix (automated, WSL-only):
- Detect WSL via /proc/version 'microsoft' marker
- Prepend 100ms silence + apply 100ms fade-in to audio
- Append 50ms silence tail for clean stream teardown
- Set blocksize=4096 (default auto ~1024 is too small for RDP)
- All in a single continuous sd.play() buffer
Non-WSL paths unchanged.
Closes#38893
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
When voice continuous mode detects 3 consecutive no-speech cycles it sets
_voice_continuous = False and previously did an immediate
eturn. The
restart guard if self._voice_continuous and not submitted and not
self._voice_recording came *after* the early return, so the thread could
never restart. However a timing window existed: if _voice_start_recording was
already queued from a prior iteration, the
eturn bypassed the guard while
the thread was already in flight.
Replace the bare
eturn with a stop_continuous_restart boolean evaluated
in the same if that guards _restart_recording. This ensures both branches
read a consistent no-restart signal and no recording thread is spawned after
the user's session has been intentionally halted.
In continuous voice mode, pressing the record hotkey (Ctrl+B) was a
silent no-op while the agent was running or voice was being transcribed.
The hotkey only cleared _voice_continuous when _voice_recording was True,
leaving the user trapped in an auto-restart loop that only /voice off
could break.
Fix: when the agent is running or transcribing, still allow the hotkey
to clear _voice_continuous so the loop stops after the current turn.
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.
[[audio_as_voice]] is message-global but was applied to every media file in a
message. A non-audio file flagged is_voice is excluded from the embedded-photo
batch and falls through to send_document, so an image in a message that also
carries a voice note arrives as a file attachment instead of an inline photo.
Gate the voice flag on the file extension so one message can carry an inline
image AND a voice bubble. Also wrap the path append in try/except so a crafted
~\x00 path is skipped rather than aborting extraction of all attachments.
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
VoiceMixer duck-typed the discord.AudioSource interface (is_opus, read,
cleanup) but never inherited from it. discord.py's vc.play() does an
isinstance check and rejects non-AudioSource objects, causing the voice
fx mixer to silently fail with:
"Voice mixer failed to start: source must be an AudioSource not
VoiceMixer"
Added missing `import discord` and changed the class definition from
`class VoiceMixer:` to `class VoiceMixer(discord.AudioSource):`.
- 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.
_voice_input_callback was only set in _handle_voice_channel_join, not at adapter connect or reconnect. Voice transcription was logged but never forwarded as an inbound message without explicit /voice join.
Wire the callback at both connect and reconnect paths.
Fixes#60623
- 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).
Closes the #32029 gap: TelegramAdapter.send_voice passed captions raw with
no parse_mode, so auto-TTS captions (which carry the agent's markdown
reply) showed literal *asterisks*, backticks and [links](...). Captions
are now formatted to MarkdownV2 via the adapter's format_message when the
formatted text fits Telegram's 1024-char caption cap, with fallback to the
plain truncated caption when formatting overflows or the Bot API rejects
the entities (mirrors the text-send markdown fallback ladder).
Fixes#32029
Salvaged from PR #62040 (@giladbau), simplified per post-#73072 main: the
central _repair_ogg_container transcode makes an explicit .ogg output path
sufficient — no target_platform plumbing through the TTS tool needed.
Root cause (class-level): both gateway auto-TTS delivery call sites relied
on the TTS tool reading HERMES_SESSION_PLATFORM to pick Ogg/Opus vs MP3,
but that contextvar is cleared by _clear_session_env before the base
adapter's post-handler auto-TTS block runs, so want_opus was always False
on that path → MP3 → Telegram sent an audio attachment instead of a native
voice bubble (#57049, #36685). The runner's _send_voice_reply had the
sibling bug: it hardcoded .ogg for Telegram only, leaving Matrix and
Feishu runner voice replies as MP3 (#14841, #45557).
Fix: new build_auto_tts_output_path(platform) in gateway/platforms/base.py
hands an explicit .ogg temp path when the platform is in the TTS tool's
OPUS_VOICE_PLATFORMS set (single source of truth — telegram/matrix/feishu/
whatsapp/signal), .mp3 otherwise. Used by BOTH delivery call sites:
- BasePlatformAdapter auto-TTS block (also honors the tool's success flag
and cleans up requested + returned paths)
- GatewayRunner._send_voice_reply (replaces the telegram-only ternary)
Fixes#57049Fixes#36685
Refs #14841#45557
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 #53157 (@LLQWQ). Three changes for native Feishu voice
bubble delivery:
1. Include audio duration in the file-upload body when uploading opus
files — Feishu renders 0:00 bubbles without it. Duration is extracted
by parsing the OGG container's last granule position (pure Python,
no ffprobe dependency).
2. Thread routing fallback: Feishu's create-message API rejects
msg_type='audio' with receive_id_type='thread_id' (error 99992402);
retry via the reply API against the thread's last message, then fall
back to chat_id routing.
3. (dropped) tools/tts_tool.py want_opus hunk — superseded by main's
OPUS_VOICE_PLATFORMS, which already includes feishu. The PR's stray
scripts/release.py hunk was also dropped (frozen AUTHOR_MAP policy;
mapping added under contributors/emails/ instead).
Fixes#45557
Refs #18831#16524
Salvaged from PR #18861 (@shellybotmoyer). Switch the central Opus
transcoders from CBR 64k (-vbr off) to VBR 48k with -application voip and
-compression_level 10. The old CBR settings produced lower-quality speech
audio and occasionally failed to trigger Telegram's native voice-bubble
rendering. Applied to _ffmpeg_transcode_to_opus (the single transcoder
behind _convert_to_opus and _repair_ogg_container), the Gemini WAV→OGG
path, and the Matrix adapter boundary transcoder added in this branch.
Fixes#18818
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>
The 'language' config field in tts.openai.language was read but never passed
to the API. This caused Kokoro (and other OpenAI-compatible TTS backends that
support lang_code) to default to English phonemization regardless of the
configured language.
Now passes lang_code via extra_body when language is set in config.
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
Adds an optional provider parameter to the text_to_speech tool that lets the model select a TTS provider per-call instead of always using the globally configured tts.provider.
When provider is set, it bypasses the configured default and routes directly to the specified backend.
When omitted (the default), the tool behaves exactly as before.
Closes#47459
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.
When /reasoning show is enabled, the model's <think> blocks appear in
the final assistant message. TTS reads these aloud, which is unwanted —
users want to see reasoning but not hear it spoken.
- Add _THINK_BLOCK regex to _strip_markdown_for_tts() (tools/tts_tool.py)
— the general TTS text-preparation path used by all TTS providers
- Add <think> stripping to prepare_tts_text() (gateway/platforms/base.py)
— the gateway auto-TTS path
- Reuses the existing regex pattern from stream_tts_to_speaker()
Closes#34213
