Rebrand to FastRTC (#60)

* Add code * add code * add code * Rename messages * rename * add code * Add demo * docs + demos + bug fixes * add code * styles * user guide * Styles * Add code * misc docs updates * print nit * whisper + pr * url for images * whsiper update * Fix bugs * remove demo files * version number * Fix pypi readme * Fix * demos * Add llama code editor * Update llama code editor and object detection cookbook * Add more cookbook demos * add code * Fix links for PR deploys * add code * Fix the install * add tts * TTS docs * Typo * Pending bubbles for reply on pause * Stream redesign (#63) * better error handling * Websocket error handling * add code --------- Co-authored-by: Freddy Boulton <freddyboulton@hf-freddy.local> * remove docs from dist * Some docs typos * more typos * upload changes + docs * docs * better phone * update docs * add code * Make demos better * fix docs + websocket start_up * remove mention of FastAPI app * fastphone tweaks * add code * ReplyOnStopWord fixes * Fix cookbook * Fix pypi readme * add code * bump versions * sambanova cookbook * Fix tags * Llm voice chat * kyutai tag * Add error message to all index.html * STT module uses Moonshine * Not required from typing extensions * fix llm voice chat * Add vpn warning * demo fixes * demos * Add more ui args and gemini audio-video * update cookbook * version 9 --------- Co-authored-by: Freddy Boulton <freddyboulton@hf-freddy.local>
2026-02-05 18:09:23 +08:00 · 2025-02-24 01:13:42 -05:00
parent 36190066ec
commit 853d6a06b5
131 changed files with 12349 additions and 4741 deletions
--- a/demo/whisper_realtime/app.py
+++ b/demo/whisper_realtime/app.py
@@ -0,0 +1,86 @@
+import json
+from pathlib import Path
+
+import gradio as gr
+import numpy as np
+from dotenv import load_dotenv
+from fastapi import FastAPI
+from fastapi.responses import HTMLResponse, StreamingResponse
+from fastrtc import (
+    AdditionalOutputs,
+    ReplyOnPause,
+    Stream,
+    WebRTCError,
+    audio_to_bytes,
+    get_twilio_turn_credentials,
+)
+from gradio.utils import get_space
+from groq import AsyncClient
+
+cur_dir = Path(__file__).parent
+
+load_dotenv()
+
+
+groq_client = AsyncClient()
+
+
+async def transcribe(audio: tuple[int, np.ndarray]):
+    try:
+        transcript = await groq_client.audio.transcriptions.create(
+            file=("audio-file.mp3", audio_to_bytes(audio)),
+            model="whisper-large-v3-turbo",
+            response_format="verbose_json",
+        )
+        yield AdditionalOutputs(transcript.text)
+    except Exception as e:
+        raise WebRTCError(str(e))
+
+
+stream = Stream(
+    ReplyOnPause(transcribe),
+    modality="audio",
+    mode="send",
+    additional_outputs=[
+        gr.Textbox(label="Transcript"),
+    ],
+    additional_outputs_handler=lambda a, b: a + " " + b,
+    rtc_configuration=get_twilio_turn_credentials() if get_space() else None,
+    concurrency_limit=5 if get_space() else None,
+    time_limit=90 if get_space() else None,
+)
+
+app = FastAPI()
+
+stream.mount(app)
+
+
+@app.get("/transcript")
+def _(webrtc_id: str):
+    async def output_stream():
+        async for output in stream.output_stream(webrtc_id):
+            transcript = output.args[0]
+            yield f"event: output\ndata: {transcript}\n\n"
+
+    return StreamingResponse(output_stream(), media_type="text/event-stream")
+
+
+@app.get("/")
+def index():
+    rtc_config = get_twilio_turn_credentials() if get_space() else None
+    html_content = (cur_dir / "index.html").read_text()
+    html_content = html_content.replace("__RTC_CONFIGURATION__", json.dumps(rtc_config))
+    return HTMLResponse(content=html_content)
+
+
+if __name__ == "__main__":
+    import os
+
+    if (mode := os.getenv("MODE")) == "UI":
+        stream.ui.launch(server_port=7860, server_name="0.0.0.0")
+    elif mode == "PHONE":
+        stream.fastphone(host="0.0.0.0", port=7860)
+    else:
+        import uvicorn
+
+        uvicorn.run(app, host="0.0.0.0", port=7860)