livekit-plugins-munsit 0.3.0__tar.gz

livekit_plugins_munsit-0.3.0/.gitignore

@@ -0,0 +1,156 @@
+# Byte-compiled / optimized / DLL files
+__pycache__/
+*.py[cod]
+*$py.class
+
+# C extensions
+*.so
+
+# Distribution / packaging
+.Python
+build/
+develop-eggs/
+dist/
+downloads/
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+lib/
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+parts/
+sdist/
+var/
+wheels/
+share/python-wheels/
+*.egg-info/
+.installed.cfg
+*.egg
+MANIFEST
+
+# PyInstaller
+*.manifest
+*.spec
+
+# Installer logs
+pip-log.txt
+pip-delete-this-directory.txt
+
+# Unit test / coverage reports
+htmlcov/
+.tox/
+.nox/
+.coverage
+.coverage.*
+.cache
+nosetests.xml
+coverage.xml
+*.cover
+*.py,cover
+.hypothesis/
+.pytest_cache/
+cover/
+
+# Translations
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+*.pot
+
+# Django stuff:
+*.log
+local_settings.py
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+db.sqlite3-journal
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+
+# pyenv
+.python-version
+
+# pipenv
+Pipfile.lock
+
+# poetry
+poetry.lock
+
+# pdm
+.pdm.toml
+
+# PEP 582
+__pypackages__/
+
+# Celery stuff
+celerybeat-schedule
+celerybeat.pid
+
+# SageMath parsed files
+*.sage.py
+
+# Environments
+.env
+.venv
+env/
+venv/
+ENV/
+env.bak/
+venv.bak/
+
+# Spyder project settings
+.spyderproject
+.spyproject
+
+# Rope project settings
+.ropeproject
+
+# mkdocs documentation
+/site
+
+# mypy
+.mypy_cache/
+.dmypy.json
+dmypy.json
+
+# Pyre type checker
+.pyre/
+
+# pytype static type analyzer
+.pytype/
+
+# Cython debug symbols
+cython_debug/
+
+# IDEs
+.vscode/
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+*.swp
+*.swo
+*~
+
+# OS
+.DS_Store
+Thumbs.db
+
+# Project specific
+*.pcm
+*.wav
+*.mp3
+test_output/
+
+# Test fixtures we DO want tracked (must come after *.wav above).
+!tests/sample_arabic.wav

livekit_plugins_munsit-0.3.0/LICENSE

@@ -0,0 +1,190 @@
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livekit_plugins_munsit-0.3.0/PKG-INFO

@@ -0,0 +1,102 @@
+Metadata-Version: 2.4
+Name: livekit-plugins-munsit
+Version: 0.3.0
+Summary: LiveKit Agents plugin for Munsit speech-to-text (Arabic + Arabic/English code-switching). Successor to livekit-plugins-faseeh.
+Project-URL: Homepage, https://github.com/CNTXTFZCO0/livekit-plugins-munsit
+Project-URL: Documentation, https://github.com/CNTXTFZCO0/livekit-plugins-munsit#readme
+Project-URL: Repository, https://github.com/CNTXTFZCO0/livekit-plugins-munsit
+Project-URL: Bug Tracker, https://github.com/CNTXTFZCO0/livekit-plugins-munsit/issues
+Project-URL: Munsit, https://munsit.com
+Author-email: Mohan Raj Palanisamy <mohanraj.palanisamy@cntxt.tech>
+Maintainer-email: Mohan Raj Palanisamy <mohanraj.palanisamy@cntxt.tech>
+License-Expression: Apache-2.0
+License-File: LICENSE
+Keywords: ai,arabic,audio,faseeh,livekit,munsit,realtime,speech-to-text,stt,voice
+Classifier: Development Status :: 4 - Beta
+Classifier: Intended Audience :: Developers
+Classifier: License :: OSI Approved :: Apache Software License
+Classifier: Programming Language :: Python :: 3
+Classifier: Programming Language :: Python :: 3 :: Only
+Classifier: Programming Language :: Python :: 3.10
+Classifier: Programming Language :: Python :: 3.11
+Classifier: Programming Language :: Python :: 3.12
+Classifier: Topic :: Multimedia :: Sound/Audio
+Classifier: Topic :: Scientific/Engineering :: Artificial Intelligence
+Requires-Python: >=3.10.0
+Requires-Dist: aiohttp>=3.8.0
+Requires-Dist: livekit-agents[codecs]>=1.4.4
+Requires-Dist: numpy>=1.26
+Description-Content-Type: text/markdown
+
+# LiveKit Plugins Munsit
+
+STT plugin for [Munsit](https://munsit.com), the merged Faseeh + Munsit ASR product. Optimized for Arabic with optional Arabic/English code-switching (`munsit-en-ar` model).
+
+## Install
+
+```bash
+pip install livekit-plugins-munsit
+```
+
+(While developing inside this monorepo, use `uv sync --all-extras --dev`.)
+
+## Quickstart
+
+```python
+from livekit.agents import AgentSession
+from livekit.plugins import munsit, silero
+
+session = AgentSession(
+    stt=munsit.STT(),                # batch mode is the default
+    vad=silero.VAD.load(),
+    # ... llm, tts ...
+)
+```
+
+`MUNSIT_API_KEY` is read from the environment by default. Munsit's batch HTTP endpoint is more accurate and returns word-level timestamps; we use it by default. The `AgentSession`'s VAD signals end-of-speech via `flush()`, which causes the plugin to POST the buffered utterance and emit a single `FINAL_TRANSCRIPT` with per-word timings.
+
+## Modes
+
+| Mode | When to use | Endpoint | Latency | Word timestamps | Interim events |
+|---|---|---|---|---|---|
+| `"batch"` (default) | Production with `AgentSession` + VAD; transcribing recorded audio. | `POST /api/v1/audio/transcribe` | VAD detection + upload + server processing (~1–2 s for short utterances) | yes — populated on `SpeechData.words` | no |
+| `"streaming"` | You need live captions / on-the-fly UI updates while the user is still speaking. | WebSocket `/api/v1/websocket/speech-to-text` | ~700 ms idle threshold by default | no | yes (`INTERIM_TRANSCRIPT` flow as cumulatives arrive) |
+
+```python
+munsit.STT()                       # batch (default)
+munsit.STT(mode="streaming")       # opt-in for low-latency interims
+```
+
+`STT.recognize(audio_buffer)` always uses the batch HTTP endpoint regardless of `mode`.
+
+## Parameters
+
+| Param | Default | Notes |
+|---|---|---|
+| `mode` | `"batch"` | `"batch"` (HTTP) or `"streaming"` (WS). |
+| `model` | `"munsit"` | `"munsit"` (Arabic) or `"munsit-en-ar"` (code-switching). |
+| `api_key` | `MUNSIT_API_KEY` env | Required. |
+| `base_url` | Munsit prod WSS | Override the streaming WebSocket URL. |
+| `batch_base_url` | Munsit prod HTTPS | Override the batch HTTP URL. |
+| `auth_method` | `"header"` | `"header"` / `"bearer"` / `"query"` — Munsit accepts all three on both endpoints. |
+| `sample_rate` | `16000` | Used for the synthesized WAV header. |
+| `num_channels` | `1` | |
+| `interim_results` | `True` | Emit `INTERIM_TRANSCRIPT` events. Only meaningful in streaming mode. |
+| `endpointing` | `"server_diff"` | Streaming-mode only: `"server_diff"` (idle timer) or `"client_vad"` (energy-based). |
+| `finalize_after_silence_ms` | `700` | Streaming-mode `server_diff` idle threshold. |
+| `energy_filter` | `False` | Streaming-mode `client_vad` only. |
+| `vad_silence_ms` | `1500` | Streaming-mode `client_vad` silence-end duration. |
+| `language` | `None` | Label attached to `SpeechData.language`. Defaults to `"ar"`. |
+| `http_session` | `None` | Custom `aiohttp.ClientSession`. |
+| `extra_query_params` | `None` | Forward-compat for new Munsit query params (streaming WS only). |
+
+## Streaming mode endpointing
+
+When using `mode="streaming"`, the plugin offers two end-of-utterance strategies:
+
+- **`server_diff`** (default) — keep one long-lived WS open, emit `INTERIM_TRANSCRIPT` on each transcript update, emit `FINAL_TRANSCRIPT` + `END_OF_SPEECH` after `finalize_after_silence_ms` of server silence. Best when `AgentSession` already has its own VAD / turn detector.
+- **`client_vad`** — open WS on local energy onset, close on silence end. Each utterance gets a fresh WS connection. Slightly higher per-utterance latency but harder utterance boundaries.
+
+## Examples
+
+See [`examples/README.md`](examples/README.md). Three demos cover file-based, mic-based, and full-agent scenarios. The file demo prints per-word timings when batch mode is active.

livekit_plugins_munsit-0.3.0/README.md

@@ -0,0 +1,72 @@
+# LiveKit Plugins Munsit
+
+STT plugin for [Munsit](https://munsit.com), the merged Faseeh + Munsit ASR product. Optimized for Arabic with optional Arabic/English code-switching (`munsit-en-ar` model).
+
+## Install
+
+```bash
+pip install livekit-plugins-munsit
+```
+
+(While developing inside this monorepo, use `uv sync --all-extras --dev`.)
+
+## Quickstart
+
+```python
+from livekit.agents import AgentSession
+from livekit.plugins import munsit, silero
+
+session = AgentSession(
+    stt=munsit.STT(),                # batch mode is the default
+    vad=silero.VAD.load(),
+    # ... llm, tts ...
+)
+```
+
+`MUNSIT_API_KEY` is read from the environment by default. Munsit's batch HTTP endpoint is more accurate and returns word-level timestamps; we use it by default. The `AgentSession`'s VAD signals end-of-speech via `flush()`, which causes the plugin to POST the buffered utterance and emit a single `FINAL_TRANSCRIPT` with per-word timings.
+
+## Modes
+
+| Mode | When to use | Endpoint | Latency | Word timestamps | Interim events |
+|---|---|---|---|---|---|
+| `"batch"` (default) | Production with `AgentSession` + VAD; transcribing recorded audio. | `POST /api/v1/audio/transcribe` | VAD detection + upload + server processing (~1–2 s for short utterances) | yes — populated on `SpeechData.words` | no |
+| `"streaming"` | You need live captions / on-the-fly UI updates while the user is still speaking. | WebSocket `/api/v1/websocket/speech-to-text` | ~700 ms idle threshold by default | no | yes (`INTERIM_TRANSCRIPT` flow as cumulatives arrive) |
+
+```python
+munsit.STT()                       # batch (default)
+munsit.STT(mode="streaming")       # opt-in for low-latency interims
+```
+
+`STT.recognize(audio_buffer)` always uses the batch HTTP endpoint regardless of `mode`.
+
+## Parameters
+
+| Param | Default | Notes |
+|---|---|---|
+| `mode` | `"batch"` | `"batch"` (HTTP) or `"streaming"` (WS). |
+| `model` | `"munsit"` | `"munsit"` (Arabic) or `"munsit-en-ar"` (code-switching). |
+| `api_key` | `MUNSIT_API_KEY` env | Required. |
+| `base_url` | Munsit prod WSS | Override the streaming WebSocket URL. |
+| `batch_base_url` | Munsit prod HTTPS | Override the batch HTTP URL. |
+| `auth_method` | `"header"` | `"header"` / `"bearer"` / `"query"` — Munsit accepts all three on both endpoints. |
+| `sample_rate` | `16000` | Used for the synthesized WAV header. |
+| `num_channels` | `1` | |
+| `interim_results` | `True` | Emit `INTERIM_TRANSCRIPT` events. Only meaningful in streaming mode. |
+| `endpointing` | `"server_diff"` | Streaming-mode only: `"server_diff"` (idle timer) or `"client_vad"` (energy-based). |
+| `finalize_after_silence_ms` | `700` | Streaming-mode `server_diff` idle threshold. |
+| `energy_filter` | `False` | Streaming-mode `client_vad` only. |
+| `vad_silence_ms` | `1500` | Streaming-mode `client_vad` silence-end duration. |
+| `language` | `None` | Label attached to `SpeechData.language`. Defaults to `"ar"`. |
+| `http_session` | `None` | Custom `aiohttp.ClientSession`. |
+| `extra_query_params` | `None` | Forward-compat for new Munsit query params (streaming WS only). |
+
+## Streaming mode endpointing
+
+When using `mode="streaming"`, the plugin offers two end-of-utterance strategies:
+
+- **`server_diff`** (default) — keep one long-lived WS open, emit `INTERIM_TRANSCRIPT` on each transcript update, emit `FINAL_TRANSCRIPT` + `END_OF_SPEECH` after `finalize_after_silence_ms` of server silence. Best when `AgentSession` already has its own VAD / turn detector.
+- **`client_vad`** — open WS on local energy onset, close on silence end. Each utterance gets a fresh WS connection. Slightly higher per-utterance latency but harder utterance boundaries.
+
+## Examples
+
+See [`examples/README.md`](examples/README.md). Three demos cover file-based, mic-based, and full-agent scenarios. The file demo prints per-word timings when batch mode is active.

livekit_plugins_munsit-0.3.0/livekit/plugins/munsit/__init__.py

@@ -0,0 +1,33 @@
+"""Munsit plugin for LiveKit Agents.
+
+See https://docs.munsit.com for the upstream API documentation.
+"""
+
+from ._utils import AudioEnergyFilter
+from .models import MunsitModels
+from .stt import STT, SpeechStream
+from .version import __version__
+
+__all__ = ["STT", "SpeechStream", "AudioEnergyFilter", "MunsitModels", "__version__"]
+
+
+from livekit.agents import Plugin
+
+from .log import logger
+
+
+class MunsitPlugin(Plugin):
+    def __init__(self) -> None:
+        super().__init__(__name__, __version__, __package__, logger)
+
+
+Plugin.register_plugin(MunsitPlugin())
+
+# Cleanup docs of unexported modules
+_module = dir()
+NOT_IN_ALL = [m for m in _module if m not in __all__]
+
+__pdoc__ = {}
+
+for n in NOT_IN_ALL:
+    __pdoc__[n] = False

livekit_plugins_munsit-0.3.0/livekit/plugins/munsit/_utils.py

@@ -0,0 +1,129 @@
+# Copyright 2026 LiveKit, Inc.
+from __future__ import annotations
+
+import struct
+import time
+from collections.abc import Callable
+from enum import Enum
+from typing import Generic, TypeVar
+
+import numpy as np
+
+from livekit import rtc
+
+
+def build_wav_header(
+    *, sample_rate: int, num_channels: int = 1, bits_per_sample: int = 16
+) -> bytes:
+    """Build a 44-byte PCM WAV header.
+
+    The data chunk size is set to a sentinel max value (0xFFFFFFFF - 44) because we are streaming
+    and don't know the total size in advance. Munsit only validates the format fields.
+    """
+    if sample_rate <= 0:
+        raise ValueError("sample_rate must be positive")
+    if num_channels <= 0:
+        raise ValueError("num_channels must be positive")
+    if bits_per_sample not in (8, 16, 24, 32):
+        raise ValueError("bits_per_sample must be 8, 16, 24, or 32")
+
+    byte_rate = sample_rate * num_channels * bits_per_sample // 8
+    block_align = num_channels * bits_per_sample // 8
+    data_size = 0xFFFFFFFF - 44
+    riff_size = data_size + 36
+
+    return (
+        b"RIFF"
+        + struct.pack("<I", riff_size)
+        + b"WAVE"
+        + b"fmt "
+        + struct.pack("<I", 16)  # fmt chunk size
+        + struct.pack("<H", 1)  # PCM format
+        + struct.pack("<H", num_channels)
+        + struct.pack("<I", sample_rate)
+        + struct.pack("<I", byte_rate)
+        + struct.pack("<H", block_align)
+        + struct.pack("<H", bits_per_sample)
+        + b"data"
+        + struct.pack("<I", data_size)
+    )
+
+
+def pcm_to_audiobuffer(data: bytes) -> list[int]:
+    """Encode raw PCM bytes as Munsit's `audioBuffer` JSON int array (0-255 per byte)."""
+    return list(data)
+
+
+T = TypeVar("T")
+
+
+class PeriodicCollector(Generic[T]):
+    """Accumulate values and call ``callback`` once per ``duration`` seconds.
+
+    Used by SpeechStream to batch audio-duration reports so RECOGNITION_USAGE
+    events don't fire on every push.
+    """
+
+    def __init__(self, callback: Callable[[T], None], *, duration: float) -> None:
+        self._duration = duration
+        self._callback = callback
+        self._last_flush_time = time.monotonic()
+        self._total: T | None = None
+
+    def push(self, value: T) -> None:
+        if self._total is None:
+            self._total = value
+        else:
+            self._total += value  # type: ignore[operator]
+        if time.monotonic() - self._last_flush_time >= self._duration:
+            self.flush()
+
+    def flush(self) -> None:
+        if self._total is not None:
+            self._callback(self._total)
+            self._total = None
+        self._last_flush_time = time.monotonic()
+
+
+_DEFAULT_RMS_THRESHOLD = 0.004**2
+
+
+class AudioEnergyFilter:
+    """Simple RMS-based VAD copied from the Gladia plugin pattern."""
+
+    class State(Enum):
+        START = 0
+        SPEAKING = 1
+        SILENCE = 2
+        END = 3
+
+    def __init__(
+        self, *, min_silence: float = 1.5, rms_threshold: float = _DEFAULT_RMS_THRESHOLD
+    ) -> None:
+        self._cooldown_seconds = min_silence
+        self._cooldown = min_silence
+        self._state = self.State.SILENCE
+        self._rms_threshold = rms_threshold
+
+    def update(self, frame: rtc.AudioFrame) -> AudioEnergyFilter.State:
+        arr = np.frombuffer(frame.data, dtype=np.int16)
+        float_arr = arr.astype(np.float32) / 32768.0
+        rms = float(np.mean(np.square(float_arr)))
+
+        if rms > self._rms_threshold:
+            self._cooldown = self._cooldown_seconds
+            if self._state in (self.State.SILENCE, self.State.END):
+                self._state = self.State.START
+            else:
+                self._state = self.State.SPEAKING
+        else:
+            if self._cooldown <= 0:
+                if self._state in (self.State.SPEAKING, self.State.START):
+                    self._state = self.State.END
+                elif self._state == self.State.END:
+                    self._state = self.State.SILENCE
+            else:
+                self._cooldown -= frame.duration
+                self._state = self.State.SPEAKING
+
+        return self._state

livekit_plugins_munsit-0.3.0/livekit/plugins/munsit/log.py

@@ -0,0 +1,3 @@
+import logging
+
+logger = logging.getLogger("livekit.plugins.munsit")