Feat: STT & TTS Gemini, Creds Gemini Cloud Speech, Generate Buffer Audio, Testing

ticket:https://bukittechnology.atlassian.net/browse/KM-500

- develop feat STT using gemini
- debugging gemini speech to text and text to speech credential using
personal account
- testing functionality STT and TTS
- generate 3 buffer audio for buffering response

.env.example

@@ -15,19 +15,24 @@ GEMINI_TTS_MODEL=gemini-2.5-flash-preview-tts
 GEMINI_TTS_VOICE=Autonoe
 GEMINI_TTS_LANGUAGE=id-ID
 
-# Chatbot agent service
-CHATBOT_BASE_URL=http://localhost:8000/chatbot-knowledgemanagement
-CHATBOT_TIMEOUT=30
 
-# Google Cloud Project (untuk referensi Vertex AI di masa depan)
+# Google Cloud Project
 GOOGLE_PROJECT_NAME=
+GOOGLE_PROJECT_ID=
 GOOGLE_PROJECT_NUMBER=
+GOOGLE_CLOUD_LOCATION=
 
 # Gemini STT
 GEMINI_STT_MODEL=gemini-2.0-flash
 GEMINI_LIVE_MODEL=gemini-live-2.5-flash-preview
 GEMINI_STT_LANGUAGE=id-ID
 
-# Provider default untuk WebSocket /ws/voice: "deepgram" atau "gemini"
-STT_PROVIDER=gemini
+# Chirp 3 STT (Google Cloud Speech-to-Text V2)
+# Membutuhkan GOOGLE_PROJECT_NAME dan Application Default Credentials (ADC)
+# Set GOOGLE_APPLICATION_CREDENTIALS=/path/to/service-account.json
+CHIRP3_REGION=us
+CHIRP3_LANGUAGE=id-ID
+
+# Provider default untuk WebSocket /ws/voice: "deepgram", "gemini", atau "chirp3"
+STT_PROVIDER=chirp3
 TTS_PROVIDER=gemini

.gitignore

@@ -23,4 +23,5 @@ test_client.py
 convert_audio.py
 
 API_CONTRACT_CHATBOT.md
-API_CONTRACT_VOICE.md
+API_CONTRACT_VOICE.md
+HIGHLIGHT_VOICE.md

README.md

@@ -52,7 +52,10 @@ WAKE_WORD=Hai EMA                    # Default: "Hai EMA"
 ## Run
 
 ```bash
-uv run uvicorn main:app --host 0.0.0.0 --port 7861
+`uv run uvicorn main:app --host 0.0.0.0 --port 7861`
+or
+`uv run uvicorn main:app --host 0.0.0.0 --port 7861 --reload`
+
 ```
 
 Server akan berjalan di `http://localhost:7861`.

main.py

@@ -8,7 +8,7 @@ from src.pipeline import VoicePipeline
 from typing import Literal
 from src.config import (
     DEEPGRAM_API_KEY, CARTESIA_API_KEY, CARTESIA_VOICE_ID,
-    SAMPLE_RATE, GOOGLE_API_KEY, CHATBOT_BASE_URL, STT_PROVIDER, TTS_PROVIDER, WAKE_WORD_ENABLED,
+    SAMPLE_RATE, GOOGLE_API_KEY, STT_PROVIDER, TTS_PROVIDER, WAKE_WORD_ENABLED,
 )
 from src.stt.deepgram_rest import transcribe_audio as deepgram_transcribe
 from src.stt.gemini_stt import transcribe_audio as gemini_stt_transcribe
@@ -23,13 +23,20 @@ VERSION = "1.2.0"
 app = FastAPI(title="Voice Agent Service", version=VERSION)
 
 
+@app.get("/")
+async def health() -> JSONResponse:
+    body: dict = {
+        "message": "Welcome to the Voice Agent Service! Please use the /health endpoint to check service status.",
+        "version": VERSION,
+    }
+    return JSONResponse(status_code=200, content=body)
+
 @app.get("/health")
 async def health() -> JSONResponse:
     stt_ready = bool(DEEPGRAM_API_KEY)
     tts_ready = bool(CARTESIA_API_KEY and CARTESIA_VOICE_ID)
     gemini_ready = bool(GOOGLE_API_KEY)
-    chatbot_ready = bool(CHATBOT_BASE_URL)
-    all_ready = stt_ready and tts_ready and chatbot_ready
+    all_ready = stt_ready and tts_ready
 
     body: dict = {
         "status": "ok" if all_ready else "degraded",
@@ -38,7 +45,6 @@ async def health() -> JSONResponse:
         "tts_ready": tts_ready,
         "gemini_tts_ready": gemini_ready,
         "gemini_stt_ready": gemini_ready,
-        "chatbot_ready": chatbot_ready,
     }
     if not all_ready:
         body["message"] = "One or more required configurations are missing. Check your .env file."
@@ -99,21 +105,14 @@ async def text_to_speech(req: TTSRequest) -> StreamingResponse:
 @app.websocket("/ws/voice")
 async def voice_ws(
     ws: WebSocket,
-    user_id: str = Query(default="anonymous"),
-    fullname: str = Query(default=""),
-    company: str = Query(default=""),
-    function: str = Query(default=""),
-    site: str = Query(default="HO"),
-    role: str = Query(default="engineer"),
-    agent: str = Query(default="analysis"),
     stt_provider: str = Query(default=STT_PROVIDER),
     tts_provider: str = Query(default=TTS_PROVIDER),
     wake_word_enabled: bool = Query(default=WAKE_WORD_ENABLED),
 ) -> None:
     await ws.accept()
     logger.info(
-        "Client connected: %s (user_id=%s, stt=%s, tts=%s)",
-        ws.client, user_id, stt_provider, tts_provider,
+        "Client connected: %s (stt=%s, tts=%s)",
+        ws.client, stt_provider, tts_provider,
     )
 
     async def send_audio(chunk: bytes) -> None:
@@ -135,13 +134,6 @@ async def voice_ws(
     pipeline = VoicePipeline(
         send_audio=send_audio,
         send_event=send_event,
-        user_id=user_id,
-        fullname=fullname,
-        company=company,
-        function=function,
-        site=site,
-        role=role,
-        agent=agent,
         stt_provider=stt_provider,
         tts_provider=tts_provider,
         wake_word_enabled=wake_word_enabled,
@@ -163,12 +155,19 @@ async def voice_ws(
                         await ws.send_text(json.dumps({"event": "pong"}))
                     elif action == "interrupt":
                         await pipeline.interrupt()
+                    elif action == "speak":
+                        text = msg.get("text", "").strip()
+                        if text:
+                            await pipeline.speak(text)
                 except json.JSONDecodeError:
                     pass
     except WebSocketDisconnect:
         logger.info("Client disconnected: %s", ws.client)
     finally:
-        pipeline.stop()
+        try:
+            await pipeline.stop_async()
+        except Exception:
+            logger.exception("Error during pipeline stop")
 
 
 if __name__ == "__main__":

pyproject.toml

@@ -13,6 +13,10 @@ dependencies = [
     "deepgram-sdk>=6.1.1",
     "python-multipart>=0.0.26",
     "google-genai>=1.0.0",
+    "google-generativeai>=0.8.6",
+    "google-cloud-texttospeech>=2.29.0",
+    "google-cloud-speech>=2.0.0",
+    "numpy>=1.26.0",
 ]
 
 [project.optional-dependencies]
@@ -26,3 +30,8 @@ build-backend = "hatchling.build"
 
 [tool.hatch.build.targets.wheel]
 packages = ["src"]
+
+[dependency-groups]
+dev = [
+    "sounddevice>=0.5.5",
+]

src/chatbot/client.py

@@ -1,60 +0,0 @@
-import uuid
-import logging
-import httpx
-from src.config import CHATBOT_BASE_URL, CHATBOT_TIMEOUT
-
-logger = logging.getLogger(__name__)
-
-
-async def call_agent(
-    question: str,
-    chat_history: list[dict],
-    user_id: str = "anonymous",
-    fullname: str = "",
-    company: str = "",
-    function: str = "",
-    site: str = "HO",
-    role: str = "engineer",
-    agent: str = "analysis",
-    room_id: str | None = None,
-    chat_id: str | None = None,
-) -> str:
-    """
-    Call POST /api/agents/call and return audio_text (fallback to content).
-
-    Raises RuntimeError if CHATBOT_BASE_URL is not configured or the agent
-    returns a failed status.
-    """
-    if not CHATBOT_BASE_URL:
-        raise RuntimeError("CHATBOT_BASE_URL is not configured.")
-
-    history = chat_history + [{"role": "human", "content": question}]
-
-    payload = {
-        "user_id": user_id,
-        "fullname": fullname,
-        "company": company,
-        "function": function,
-        "site": site,
-        "role": role,
-        "agent": agent,
-        "room_id": room_id,
-        "chat_id": chat_id or str(uuid.uuid4()),
-        "req_plot_object": False,
-        "chat_history": history,
-    }
-
-    async with httpx.AsyncClient(timeout=CHATBOT_TIMEOUT) as client:
-        resp = await client.post(f"{CHATBOT_BASE_URL}/api/agents/call", json=payload)
-        resp.raise_for_status()
-        data = resp.json()
-
-    status = data.get("status") or {}
-    if status.get("status") == "failed":
-        raise RuntimeError(status.get("message", "Agent call failed"))
-
-    answer = data.get("answer") or {}
-    audio_text: str | None = answer.get("audio_text")
-    content: str = answer.get("content", "")
-
-    return audio_text or content

src/config.py

@@ -14,9 +14,11 @@ CARTESIA_MODEL: str = os.getenv("CARTESIA_MODEL", "sonic-3")
 
 GOOGLE_API_KEY: str = os.getenv("GOOGLE_API_KEY", "")
 GOOGLE_PROJECT_NAME: str = os.getenv("GOOGLE_PROJECT_NAME", "")
+GOOGLE_PROJECT_ID: str = os.getenv("GOOGLE_PROJECT_ID", "")
 GOOGLE_PROJECT_NUMBER: str = os.getenv("GOOGLE_PROJECT_NUMBER", "")
+GOOGLE_CLOUD_LOCATION: str = os.getenv("GOOGLE_CLOUD_LOCATION", "us-central1")
 
-GEMINI_TTS_MODEL: str = os.getenv("GEMINI_TTS_MODEL", "gemini-2.5-flash-preview-tts")
+GEMINI_TTS_MODEL: str = os.getenv("GEMINI_TTS_MODEL", "gemini-2.5-flash-tts")
 GEMINI_TTS_VOICE: str = os.getenv("GEMINI_TTS_VOICE", "Autonoe")
 GEMINI_TTS_LANGUAGE: str = os.getenv("GEMINI_TTS_LANGUAGE", "id-ID")
 
@@ -29,5 +31,5 @@ DEEPGRAM_LANGUAGE: str = os.getenv("DEEPGRAM_LANGUAGE", "id")
 DEEPGRAM_ENDPOINTING_MS: int = int(os.getenv("DEEPGRAM_ENDPOINTING_MS", "300"))
 DEEPGRAM_UTTERANCE_END_MS: int = int(os.getenv("DEEPGRAM_UTTERANCE_END_MS", "2000"))
 
-CHATBOT_BASE_URL: str = os.getenv("CHATBOT_BASE_URL", "")
-CHATBOT_TIMEOUT: int = int(os.getenv("CHATBOT_TIMEOUT", "30"))
+CHIRP3_REGION: str = os.getenv("CHIRP3_REGION", "us")
+CHIRP3_LANGUAGE: str = os.getenv("CHIRP3_LANGUAGE", "id-ID")

src/pipeline.py

@@ -1,4 +1,3 @@
-import uuid
 import asyncio
 import logging
 from typing import Callable, Awaitable
@@ -6,7 +5,6 @@ from src.config import WAKE_WORDS, WAKE_WORD_ENABLED
 from src.stt.deepgram_client import DeepgramStreamer
 from src.tts.cartesia_client import synthesize_stream as cartesia_synthesize_stream
 from src.tts.gemini_client import synthesize_stream as gemini_synthesize_stream, GEMINI_SAMPLE_RATE
-from src.chatbot.client import call_agent
 
 logger = logging.getLogger(__name__)
 
@@ -27,16 +25,10 @@ class VoicePipeline:
         self,
         send_audio: SendAudioCallback,
         send_event: SendEventCallback,
-        user_id: str = "anonymous",
-        fullname: str = "",
-        company: str = "",
-        function: str = "",
-        site: str = "HO",
-        role: str = "engineer",
-        agent: str = "analysis",
         stt_provider: str = "deepgram",
         tts_provider: str = "cartesia",
         wake_word_enabled: bool = WAKE_WORD_ENABLED,
+        **kwargs,
     ):
         self._send_audio = send_audio
         self._send_event = send_event
@@ -44,27 +36,18 @@ class VoicePipeline:
         self._tts_provider = tts_provider
         self._wake_word_enabled = wake_word_enabled
 
-        self._user_id = user_id
-        self._fullname = fullname
-        self._company = company
-        self._function = function
-        self._site = site
-        self._role = role
-        self._agent = agent
-
-        # Stable identifiers for this session
-        self._room_id = str(uuid.uuid4())
-        self._chat_id = str(uuid.uuid4())
-
-        # Running chat history for multi-turn context
-        self._chat_history: list[dict] = []
-
         if stt_provider == "gemini":
             from src.stt.gemini_stt import GeminiSTTStreamer
             self._stt = GeminiSTTStreamer(
                 on_final_transcript=self._on_final_transcript,
                 loop=self._loop,
             )
+        elif stt_provider == "chirp3":
+            from src.stt.chirp3_client import Chirp3STTStreamer
+            self._stt = Chirp3STTStreamer(
+                on_final_transcript=self._on_final_transcript,
+                loop=self._loop,
+            )
         else:
             self._stt = DeepgramStreamer(
                 on_final_transcript=self._on_final_transcript,
@@ -80,21 +63,18 @@ class VoicePipeline:
         self._stt.send_audio(chunk)
 
     async def _on_final_transcript(self, text: str) -> None:
-        await self._send_event({"event": "transcript", "text": text})
-
         if self._wake_word_enabled:
             question = self._extract_question(text)
             if question is None:
                 logger.debug("No wake word detected, ignoring: %s", text)
                 return
             logger.info("Wake word detected, question: %s", question)
+            await self._send_event({"event": "transcript", "text": question})
         else:
-            question = text.strip()
-            if not question:
+            if not text.strip():
                 return
-            logger.info("Processing transcript (wake word disabled): %s", question)
-
-        self._tts_task = asyncio.create_task(self._answer_and_speak(question))
+            logger.info("Transcript: %s", text)
+            await self._send_event({"event": "transcript", "text": text})
 
     def _extract_question(self, transcript: str) -> str | None:
         lower = transcript.lower()
@@ -109,36 +89,9 @@ class VoicePipeline:
         question = transcript[idx:].strip(" ,.")
         return question if question else None
 
-    async def _answer_and_speak(self, question: str) -> None:
-        try:
-            reply_text = await call_agent(
-                question=question,
-                chat_history=self._chat_history,
-                user_id=self._user_id,
-                fullname=self._fullname,
-                company=self._company,
-                function=self._function,
-                site=self._site,
-                role=self._role,
-                agent=self._agent,
-                room_id=self._room_id,
-                chat_id=self._chat_id,
-            )
-        except Exception:
-            logger.exception("Chatbot call failed for question: %s", question)
-            await self._send_event({
-                "event": "error",
-                "code": "CHATBOT_ERROR",
-                "message": "Failed to get response from chatbot agent.",
-            })
-            return
-
-        # Update history for multi-turn context
-        self._chat_history.append({"role": "human", "content": question})
-        self._chat_history.append({"role": "ai", "content": reply_text})
-
-        await self._send_event({"event": "reply", "text": reply_text})
-        await self._speak(reply_text)
+    async def speak(self, text: str) -> None:
+        """Called by the WebSocket handler when the FE sends a speak action."""
+        self._tts_task = asyncio.create_task(self._speak(text))
 
     async def _speak(self, text: str) -> None:
         async with self._tts_lock:
@@ -176,6 +129,14 @@ class VoicePipeline:
             self._tts_task.cancel()
         self._stt.stop()
 
+    async def stop_async(self) -> None:
+        if self._tts_task and not self._tts_task.done():
+            self._tts_task.cancel()
+        if hasattr(self._stt, "stop_and_transcribe"):
+            await self._stt.stop_and_transcribe()
+        else:
+            self._stt.stop()
+
 
 # Backward-compatible alias
 EchoPipeline = VoicePipeline

src/stt/chirp3_client.py

@@ -0,0 +1,119 @@
+import io
+import wave
+import asyncio
+import logging
+from typing import Callable, Awaitable
+
+from google.cloud.speech_v2 import SpeechClient
+from google.cloud.speech_v2.types import cloud_speech as cloud_speech_types
+from google.api_core.client_options import ClientOptions
+
+from src.config import GOOGLE_PROJECT_ID, CHIRP3_REGION, CHIRP3_LANGUAGE
+
+logger = logging.getLogger(__name__)
+
+OnTranscriptCallback = Callable[[str], Awaitable[None]]
+
+_CHUNK_SIZE = 24 * 1024  # stay under the 25 KB gRPC streaming limit
+
+
+class Chirp3STTStreamer:
+    """
+    Push-to-talk STT using Google Cloud Speech-to-Text V2 streaming with Chirp 3 model.
+
+    Interface matches GeminiSTTStreamer: start(), send_audio(chunk), stop(), stop_and_transcribe().
+    Audio chunks (PCM Linear16 16kHz mono) are buffered, then streamed to Chirp 3 in 24 KB
+    pieces via streaming_recognize() on stop_and_transcribe().
+
+    Authentication: Application Default Credentials (ADC).
+    Set GOOGLE_APPLICATION_CREDENTIALS or run: gcloud auth application-default login
+    """
+
+    def __init__(self, on_final_transcript: OnTranscriptCallback, loop: asyncio.AbstractEventLoop):
+        self._on_final_transcript = on_final_transcript
+        self._loop = loop
+        self._audio_chunks: list[bytes] = []
+
+    def start(self) -> None:
+        self._audio_chunks = []
+        logger.info("Chirp 3 STT ready (region=%s, language=%s)", CHIRP3_REGION, CHIRP3_LANGUAGE)
+
+    def send_audio(self, chunk: bytes) -> None:
+        self._audio_chunks.append(chunk)
+
+    def stop(self) -> None:
+        logger.info("Chirp 3 STT stopped")
+
+    async def stop_and_transcribe(self) -> None:
+        audio_data = b"".join(self._audio_chunks)
+        self._audio_chunks = []
+
+        if not audio_data:
+            logger.info("Chirp 3 STT: no audio received, skipping transcription")
+            return
+
+        logger.info("Chirp 3 STT transcribing %d bytes of audio...", len(audio_data))
+        wav_data = self._pcm_to_wav(audio_data)
+
+        try:
+            text = await asyncio.get_event_loop().run_in_executor(
+                None, self._recognize, wav_data
+            )
+            if text:
+                await self._on_final_transcript(text)
+        except Exception:
+            logger.exception("Chirp 3 STT transcription failed")
+
+    def _recognize(self, wav_data: bytes) -> str:
+        client = SpeechClient(
+            client_options=ClientOptions(
+                api_endpoint=f"{CHIRP3_REGION}-speech.googleapis.com"
+            )
+        )
+
+        chunks = [
+            wav_data[i : i + _CHUNK_SIZE]
+            for i in range(0, len(wav_data), _CHUNK_SIZE)
+        ]
+        audio_requests = (
+            cloud_speech_types.StreamingRecognizeRequest(audio=chunk)
+            for chunk in chunks
+        )
+
+        recognition_config = cloud_speech_types.RecognitionConfig(
+            auto_decoding_config=cloud_speech_types.AutoDetectDecodingConfig(),
+            language_codes=[CHIRP3_LANGUAGE],
+            model="chirp_3",
+        )
+        streaming_config = cloud_speech_types.StreamingRecognitionConfig(
+            config=recognition_config
+        )
+        config_request = cloud_speech_types.StreamingRecognizeRequest(
+            recognizer=f"projects/{GOOGLE_PROJECT_ID}/locations/{CHIRP3_REGION}/recognizers/_",
+            streaming_config=streaming_config,
+        )
+
+        def requests():
+            yield config_request
+            yield from audio_requests
+
+        transcripts: list[str] = []
+        for response in client.streaming_recognize(requests=requests()):
+            for result in response.results:
+                if result.alternatives:
+                    transcripts.append(result.alternatives[0].transcript)
+
+        transcript = " ".join(transcripts).strip()
+        if transcript:
+            logger.info("Chirp 3 STT transcript: %s", transcript[:80])
+        return transcript
+
+    @staticmethod
+    def _pcm_to_wav(pcm_data: bytes) -> bytes:
+        buf = io.BytesIO()
+        with wave.open(buf, "wb") as wf:
+            wf.setnchannels(1)
+            wf.setsampwidth(2)
+            wf.setframerate(16000)
+            wf.writeframes(pcm_data)
+        return buf.getvalue()

src/stt/gemini_stt.py

@@ -1,3 +1,5 @@
+import io
+import wave
 import asyncio
 import logging
 from typing import Callable, Awaitable
@@ -7,10 +9,10 @@ from google.genai import types
 
 from src.config import (
     GOOGLE_API_KEY,
+    GOOGLE_PROJECT_NAME,
+    GOOGLE_CLOUD_LOCATION,
     GEMINI_STT_MODEL,
-    GEMINI_LIVE_MODEL,
     GEMINI_STT_LANGUAGE,
-    DEEPGRAM_UTTERANCE_END_MS,
 )
 
 logger = logging.getLogger(__name__)
@@ -23,7 +25,10 @@ async def transcribe_audio(data: bytes, mimetype: str = "audio/wav") -> dict:
     Transcribes a full audio file using Gemini multimodal API.
     Returns dict with 'text', 'language', and 'duration'.
     """
-    client = genai.Client(api_key=GOOGLE_API_KEY)
+    if GOOGLE_API_KEY:
+        client = genai.Client(api_key=GOOGLE_API_KEY)
+    else:
+        client = genai.Client(vertexai=True, project=GOOGLE_PROJECT_NAME, location=GOOGLE_CLOUD_LOCATION)
     response = await client.aio.models.generate_content(
         model=GEMINI_STT_MODEL,
         contents=[
@@ -39,99 +44,53 @@ async def transcribe_audio(data: bytes, mimetype: str = "audio/wav") -> dict:
 
 class GeminiSTTStreamer:
     """
-    Real-time streaming STT using the Gemini Live API.
-
-    Interface matches DeepgramStreamer: start(), send_audio(chunk), stop().
-    Audio chunks must be PCM Linear16 16kHz mono bytes.
+    Push-to-talk STT using Gemini batch transcription.
 
-    Final transcript segments are buffered and flushed after DEEPGRAM_UTTERANCE_END_MS
-    of silence (reusing the same configurable delay).
+    Interface matches DeepgramStreamer: start(), send_audio(chunk), stop_and_transcribe().
+    Audio chunks (PCM Linear16 16kHz mono) are buffered during the session.
+    On stop_and_transcribe(), all buffered audio is sent to Gemini as a single WAV request.
     """
 
     def __init__(self, on_final_transcript: OnTranscriptCallback, loop: asyncio.AbstractEventLoop):
         self._on_final_transcript = on_final_transcript
         self._loop = loop
-        self._audio_queue: asyncio.Queue[bytes | None] = None  # type: ignore[assignment]
-        self._session_task: asyncio.Task | None = None
-        self._transcript_buffer: list[str] = []
-        self._flush_handle: asyncio.TimerHandle | None = None
+        self._audio_chunks: list[bytes] = []
 
     def start(self) -> None:
-        self._audio_queue = asyncio.Queue()
-        self._session_task = asyncio.ensure_future(self._run_session(), loop=self._loop)
-        logger.info("Gemini Live STT starting...")
+        self._audio_chunks = []
+        logger.info("Gemini STT ready")
 
     def send_audio(self, chunk: bytes) -> None:
-        if self._audio_queue is not None:
-            self._audio_queue.put_nowait(chunk)
+        self._audio_chunks.append(chunk)
 
     def stop(self) -> None:
-        if self._session_task and not self._session_task.done():
-            self._session_task.cancel()
-        if self._audio_queue is not None:
-            # Unblock the send loop if it's waiting on the queue
-            self._audio_queue.put_nowait(None)
-        if self._flush_handle is not None:
-            self._flush_handle.cancel()
-            self._flush_handle = None
-        logger.info("Gemini Live STT stopped")
-
-    async def _run_session(self) -> None:
-        client = genai.Client(api_key=GOOGLE_API_KEY)
-        config = types.LiveConnectConfig(
-            response_modalities=["TEXT"],
-            input_audio_transcription=types.AudioTranscriptionConfig(),
-        )
-        try:
-            async with client.aio.live.connect(model=GEMINI_LIVE_MODEL, config=config) as session:
-                logger.info("Gemini Live STT connected")
-                await asyncio.gather(
-                    self._send_loop(session),
-                    self._receive_loop(session),
-                )
-        except asyncio.CancelledError:
-            pass
-        except Exception:
-            logger.exception("Gemini Live session error")
+        logger.info("Gemini STT stopped")
+
+    async def stop_and_transcribe(self) -> None:
+        audio_data = b"".join(self._audio_chunks)
+        self._audio_chunks = []
+
+        if not audio_data:
+            logger.info("Gemini STT: no audio received, skipping transcription")
+            return
+
+        logger.info("Gemini STT transcribing %d bytes of audio...", len(audio_data))
+        wav_data = self._pcm_to_wav(audio_data)
 
-    async def _send_loop(self, session) -> None:
-        try:
-            while True:
-                chunk = await self._audio_queue.get()
-                if chunk is None:
-                    break
-                await session.send_realtime_input(
-                    audio=types.Blob(data=chunk, mime_type="audio/pcm;rate=16000")
-                )
-        except asyncio.CancelledError:
-            raise
-
-    async def _receive_loop(self, session) -> None:
         try:
-            async for response in session.receive():
-                if response.server_content is None:
-                    continue
-                transcription = response.server_content.input_transcription
-                if transcription and transcription.text:
-                    self._buffer_transcript(transcription.text)
-        except asyncio.CancelledError:
-            raise
+            result = await transcribe_audio(wav_data, mimetype="audio/wav")
+            text = result.get("text", "").strip()
+            if text:
+                await self._on_final_transcript(text)
         except Exception:
-            logger.exception("Gemini Live receive error")
-
-    def _buffer_transcript(self, text: str) -> None:
-        self._transcript_buffer.append(text)
-        logger.debug("Buffered segment: %s", text)
-        if self._flush_handle is not None:
-            self._flush_handle.cancel()
-        delay = DEEPGRAM_UTTERANCE_END_MS / 1000.0
-        self._flush_handle = self._loop.call_later(delay, self._flush_buffer)
-
-    def _flush_buffer(self) -> None:
-        if not self._transcript_buffer:
-            return
-        full_text = " ".join(self._transcript_buffer)
-        self._transcript_buffer.clear()
-        self._flush_handle = None
-        logger.info("Final transcript: %s", full_text)
-        asyncio.ensure_future(self._on_final_transcript(full_text), loop=self._loop)
+            logger.exception("Gemini STT batch transcription failed")
+
+    @staticmethod
+    def _pcm_to_wav(pcm_data: bytes) -> bytes:
+        buf = io.BytesIO()
+        with wave.open(buf, "wb") as wf:
+            wf.setnchannels(1)
+            wf.setsampwidth(2)
+            wf.setframerate(16000)
+            wf.writeframes(pcm_data)
+        return buf.getvalue()

src/tts/gemini_client.py

@@ -1,39 +1,39 @@
+import asyncio
 import logging
 from typing import AsyncIterator
-from google import genai
-from google.genai import types
-from src.config import GOOGLE_API_KEY, GEMINI_TTS_MODEL, GEMINI_TTS_VOICE, GEMINI_TTS_LANGUAGE
+
+from google.cloud import texttospeech
+
+from src.config import GEMINI_TTS_MODEL, GEMINI_TTS_VOICE, GEMINI_TTS_LANGUAGE
 
 logger = logging.getLogger(__name__)
 
-GEMINI_SAMPLE_RATE = 24000  # Gemini outputs PCM Linear16 at 24kHz
+GEMINI_SAMPLE_RATE = 24000  # Cloud TTS Gemini model outputs PCM Linear16 at 24kHz
 
 
 async def synthesize_stream(text: str) -> AsyncIterator[bytes]:
-    """Calls Gemini TTS and yields raw PCM Linear16 chunks at 24kHz."""
-    client = genai.Client(api_key=GOOGLE_API_KEY)
+    """Calls Cloud TTS Gemini model and yields raw PCM Linear16 chunks at 24kHz."""
     logger.info("Gemini TTS: synthesizing '%s'", text[:60])
 
-    response_iter = await client.aio.models.generate_content_stream(
-        model=GEMINI_TTS_MODEL,
-        contents=text,
-        config=types.GenerateContentConfig(
-            response_modalities=["AUDIO"],
-            speech_config=types.SpeechConfig(
-                language_code=GEMINI_TTS_LANGUAGE,
-                voice_config=types.VoiceConfig(
-                    prebuilt_voice_config=types.PrebuiltVoiceConfig(
-                        voice_name=GEMINI_TTS_VOICE,
-                    )
-                ),
-            ),
-        ),
-    )
-
-    async for chunk in response_iter:
-        if (
-            chunk.candidates
-            and chunk.candidates[0].content.parts
-            and chunk.candidates[0].content.parts[0].inline_data
-        ):
-            yield chunk.candidates[0].content.parts[0].inline_data.data
+    def _collect() -> list[bytes]:
+        client = texttospeech.TextToSpeechClient()
+        config_req = texttospeech.StreamingSynthesizeRequest(
+            streaming_config=texttospeech.StreamingSynthesizeConfig(
+                voice=texttospeech.VoiceSelectionParams(
+                    name=GEMINI_TTS_VOICE,
+                    language_code=GEMINI_TTS_LANGUAGE,
+                    model_name=GEMINI_TTS_MODEL,
+                )
+            )
+        )
+
+        def _gen():
+            yield config_req
+            yield texttospeech.StreamingSynthesizeRequest(
+                input=texttospeech.StreamingSynthesisInput(text=text)
+            )
+
+        return [r.audio_content for r in client.streaming_synthesize(_gen())]
+
+    for chunk in await asyncio.to_thread(_collect):
+        yield chunk

uv.lock

@@ -1,6 +1,11 @@
 version = 1
 revision = 2
 requires-python = ">=3.11"
+resolution-markers = [
+    "python_full_version >= '3.14'",
+    "python_full_version == '3.13.*'",
+    "python_full_version < '3.13'",
+]
 
 [[package]]
 name = "aiohappyeyeballs"
@@ -571,6 +576,90 @@ wheels = [
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 ]
 
+[[package]]
+name = "google-ai-generativelanguage"
+version = "0.6.15"
+source = { registry = "https://pypi.org/simple" }
+dependencies = [
+    { name = "google-api-core", version = "2.25.2", source = { registry = "https://pypi.org/simple" }, extra = ["grpc"], marker = "python_full_version >= '3.14'" },
+    { name = "google-api-core", version = "2.30.3", source = { registry = "https://pypi.org/simple" }, extra = ["grpc"], marker = "python_full_version < '3.14'" },
+    { name = "google-auth" },
+    { name = "proto-plus" },
+    { name = "protobuf" },
+]
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+wheels = [
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+    { name = "google-auth" },
+    { name = "google-auth-httplib2" },
+    { name = "httplib2" },
+    { name = "uritemplate" },
+]
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 version = "2.49.2"
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 ]
 
+[[package]]
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+version = "0.3.1"
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+dependencies = [
+    { name = "google-auth" },
+    { name = "httplib2" },
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 version = "1.2.0"
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+sdist = { url = "https://files.pythonhosted.org/packages/09/a9/6ba95a270c6f1fbcd8dac228323f2777d886cb206987444e4bce66338dd4/tqdm-4.67.3.tar.gz", hash = "sha256:7d825f03f89244ef73f1d4ce193cb1774a8179fd96f31d7e1dcde62092b960bb", size = 169598, upload-time = "2026-02-03T17:35:53.048Z" }
+wheels = [
+    { url = "https://files.pythonhosted.org/packages/16/e1/3079a9ff9b8e11b846c6ac5c8b5bfb7ff225eee721825310c91b3b50304f/tqdm-4.67.3-py3-none-any.whl", hash = "sha256:ee1e4c0e59148062281c49d80b25b67771a127c85fc9676d3be5f243206826bf", size = 78374, upload-time = "2026-02-03T17:35:50.982Z" },
+]
+
 [[package]]
 name = "typing-extensions"
 version = "4.15.0"
@@ -1184,6 +1570,15 @@ wheels = [
     { url = "https://files.pythonhosted.org/packages/dc/9b/47798a6c91d8bdb567fe2698fe81e0c6b7cb7ef4d13da4114b41d239f65d/typing_inspection-0.4.2-py3-none-any.whl", hash = "sha256:4ed1cacbdc298c220f1bd249ed5287caa16f34d44ef4e9c3d0cbad5b521545e7", size = 14611, upload-time = "2025-10-01T02:14:40.154Z" },
 ]
 
+[[package]]
+name = "uritemplate"
+version = "4.2.0"
+source = { registry = "https://pypi.org/simple" }
+sdist = { url = "https://files.pythonhosted.org/packages/98/60/f174043244c5306c9988380d2cb10009f91563fc4b31293d27e17201af56/uritemplate-4.2.0.tar.gz", hash = "sha256:480c2ed180878955863323eea31b0ede668795de182617fef9c6ca09e6ec9d0e", size = 33267, upload-time = "2025-06-02T15:12:06.318Z" }
+wheels = [
+    { url = "https://files.pythonhosted.org/packages/a9/99/3ae339466c9183ea5b8ae87b34c0b897eda475d2aec2307cae60e5cd4f29/uritemplate-4.2.0-py3-none-any.whl", hash = "sha256:962201ba1c4edcab02e60f9a0d3821e82dfc5d2d6662a21abd533879bdb8a686", size = 11488, upload-time = "2025-06-02T15:12:03.405Z" },
+]
+
 [[package]]
 name = "urllib3"
 version = "2.6.3"
@@ -1264,8 +1659,12 @@ dependencies = [
     { name = "cartesia" },
     { name = "deepgram-sdk" },
     { name = "fastapi" },
+    { name = "google-cloud-speech" },
+    { name = "google-cloud-texttospeech" },
     { name = "google-genai" },
+    { name = "google-generativeai" },
     { name = "httpx" },
+    { name = "numpy" },
     { name = "python-dotenv" },
     { name = "python-multipart" },
     { name = "uvicorn", extra = ["standard"] },
@@ -1277,14 +1676,23 @@ dev = [
     { name = "websockets" },
 ]
 
+[package.dev-dependencies]
+dev = [
+    { name = "sounddevice" },
+]
+
 [package.metadata]
 requires-dist = [
     { name = "assemblyai", specifier = "==0.33.0" },
     { name = "cartesia", specifier = "==1.3.1" },
     { name = "deepgram-sdk", specifier = ">=6.1.1" },
     { name = "fastapi", specifier = "==0.115.0" },
+    { name = "google-cloud-speech", specifier = ">=2.0.0" },
+    { name = "google-cloud-texttospeech", specifier = ">=2.29.0" },
     { name = "google-genai", specifier = ">=1.0.0" },
+    { name = "google-generativeai", specifier = ">=0.8.6" },
     { name = "httpx", specifier = "==0.27.2" },
+    { name = "numpy", specifier = ">=1.26.0" },
     { name = "python-dotenv", specifier = "==1.0.1" },
     { name = "python-multipart", specifier = ">=0.0.26" },
     { name = "uvicorn", extras = ["standard"], specifier = "==0.30.6" },
@@ -1293,6 +1701,9 @@ requires-dist = [
 ]
 provides-extras = ["dev"]
 
+[package.metadata.requires-dev]
+dev = [{ name = "sounddevice", specifier = ">=0.5.5" }]
+
 [[package]]
 name = "watchfiles"
 version = "1.1.1"