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Upload Live_audio.py
Browse files- Live_audio.py +348 -0
Live_audio.py
ADDED
@@ -0,0 +1,348 @@
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1 |
+
import asyncio
|
2 |
+
import base64
|
3 |
+
import os
|
4 |
+
import time
|
5 |
+
from io import BytesIO
|
6 |
+
from google.genai import types
|
7 |
+
from google.genai.types import (
|
8 |
+
LiveConnectConfig,
|
9 |
+
SpeechConfig,
|
10 |
+
VoiceConfig,
|
11 |
+
PrebuiltVoiceConfig,
|
12 |
+
Content,
|
13 |
+
Part,
|
14 |
+
)
|
15 |
+
import gradio as gr
|
16 |
+
import numpy as np
|
17 |
+
import websockets
|
18 |
+
from dotenv import load_dotenv
|
19 |
+
from fastrtc import (
|
20 |
+
AsyncAudioVideoStreamHandler,
|
21 |
+
Stream,
|
22 |
+
WebRTC,
|
23 |
+
get_cloudflare_turn_credentials_async,
|
24 |
+
wait_for_item,
|
25 |
+
)
|
26 |
+
from google import genai
|
27 |
+
from gradio.utils import get_space
|
28 |
+
from PIL import Image
|
29 |
+
|
30 |
+
# ------------------------------------------
|
31 |
+
import asyncio
|
32 |
+
import base64
|
33 |
+
import json
|
34 |
+
import os
|
35 |
+
import pathlib
|
36 |
+
from typing import AsyncGenerator, Literal
|
37 |
+
|
38 |
+
import gradio as gr
|
39 |
+
import numpy as np
|
40 |
+
from dotenv import load_dotenv
|
41 |
+
from fastapi import FastAPI
|
42 |
+
from fastapi.responses import HTMLResponse
|
43 |
+
from fastrtc import (
|
44 |
+
AsyncStreamHandler,
|
45 |
+
Stream,
|
46 |
+
get_cloudflare_turn_credentials_async,
|
47 |
+
wait_for_item,
|
48 |
+
)
|
49 |
+
from google import genai
|
50 |
+
from google.genai.types import (
|
51 |
+
LiveConnectConfig,
|
52 |
+
PrebuiltVoiceConfig,
|
53 |
+
SpeechConfig,
|
54 |
+
VoiceConfig,
|
55 |
+
)
|
56 |
+
from gradio.utils import get_space
|
57 |
+
from pydantic import BaseModel
|
58 |
+
# ------------------------------------------------
|
59 |
+
from dotenv import load_dotenv
|
60 |
+
load_dotenv()
|
61 |
+
import os
|
62 |
+
import io
|
63 |
+
import asyncio
|
64 |
+
from pydub import AudioSegment
|
65 |
+
|
66 |
+
# Gemini: google-genai
|
67 |
+
from google import genai
|
68 |
+
# ---------------------------------------------------
|
69 |
+
# VAD imports from reference code
|
70 |
+
import collections
|
71 |
+
import webrtcvad
|
72 |
+
import time
|
73 |
+
|
74 |
+
# helper functions
|
75 |
+
GEMINI_API_KEY="AIzaSyCUCivstFpC9pq_jMHMYdlPrmh9Bx97dFo"
|
76 |
+
|
77 |
+
TAVILY_API_KEY="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"
|
78 |
+
|
79 |
+
OPENAI_API_KEY="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"
|
80 |
+
|
81 |
+
QDRANT_API_KEY="eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJhY2Nlc3MiOiJtIiwiZXhwIjoxNzUxMDUxNzg4fQ.I9J-K7OM0BtcNKgj2d4uVM8QYAHYfFCVAyP4rlZkK2E"
|
82 |
+
|
83 |
+
QDRANT_URL="https://6a3aade6-e8ad-4a6c-a579-21f5af90b7e8.us-east4-0.gcp.cloud.qdrant.io"
|
84 |
+
|
85 |
+
OPENAI_API_KEY="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"
|
86 |
+
|
87 |
+
WEAVIATE_URL="yorcqe2sqswhcaivxvt9a.c0.us-west3.gcp.weaviate.cloud"
|
88 |
+
|
89 |
+
WEAVIATE_API_KEY="d2d0VGdZQTBmdTFlOWdDZl9tT2h3WDVWd1NpT1dQWHdGK0xjR1hYeWxicUxHVnFRazRUSjY2VlRUVlkwPV92MjAw"
|
90 |
+
|
91 |
+
DEEPINFRA_API_KEY="285LUJulGIprqT6hcPhiXtcrphU04FG4"
|
92 |
+
|
93 |
+
DEEPINFRA_BASE_URL="https://api.deepinfra.com/v1/openai"
|
94 |
+
def encode_audio(data: np.ndarray) -> dict:
|
95 |
+
"""Encode Audio data to send to the server"""
|
96 |
+
return {
|
97 |
+
"mime_type": "audio/pcm",
|
98 |
+
"data": base64.b64encode(data.tobytes()).decode("UTF-8"),
|
99 |
+
}
|
100 |
+
def encode_audio2(data: np.ndarray) -> bytes:
|
101 |
+
"""Encode Audio data to send to the server"""
|
102 |
+
return data.tobytes()
|
103 |
+
|
104 |
+
import soundfile as sf
|
105 |
+
|
106 |
+
def numpy_array_to_wav_bytes(audio_array, sample_rate=16000):
|
107 |
+
buffer = io.BytesIO()
|
108 |
+
sf.write(buffer, audio_array, sample_rate, format='WAV')
|
109 |
+
return buffer.getvalue()
|
110 |
+
|
111 |
+
|
112 |
+
def numpy_array_to_wav_bytes(audio_array, sample_rate=16000):
|
113 |
+
"""
|
114 |
+
Convert a NumPy audio array to WAV bytes.
|
115 |
+
|
116 |
+
Args:
|
117 |
+
audio_array (np.ndarray): Audio signal (1D or 2D).
|
118 |
+
sample_rate (int): Sample rate in Hz.
|
119 |
+
|
120 |
+
Returns:
|
121 |
+
bytes: WAV-formatted audio data.
|
122 |
+
"""
|
123 |
+
buffer = io.BytesIO()
|
124 |
+
sf.write(buffer, audio_array, sample_rate, format='WAV')
|
125 |
+
buffer.seek(0)
|
126 |
+
return buffer.read()
|
127 |
+
# webrtc handler class
|
128 |
+
class GeminiHandler(AsyncStreamHandler):
|
129 |
+
"""Handler for the Gemini API with chained latency calculation."""
|
130 |
+
|
131 |
+
def __init__(
|
132 |
+
self,
|
133 |
+
expected_layout: Literal["mono"] = "mono",
|
134 |
+
output_sample_rate: int = 24000,prompt_dict: dict = {"prompt":"PHQ-9"},
|
135 |
+
) -> None:
|
136 |
+
super().__init__(
|
137 |
+
expected_layout,
|
138 |
+
output_sample_rate,
|
139 |
+
input_sample_rate=16000,
|
140 |
+
)
|
141 |
+
self.input_queue: asyncio.Queue = asyncio.Queue()
|
142 |
+
self.output_queue: asyncio.Queue = asyncio.Queue()
|
143 |
+
self.quit: asyncio.Event = asyncio.Event()
|
144 |
+
self.prompt_dict = prompt_dict
|
145 |
+
# self.model = "gemini-2.5-flash-preview-tts"
|
146 |
+
self.model = "gemini-2.0-flash-live-001"
|
147 |
+
self.t2t_model = "gemini-2.0-flash"
|
148 |
+
self.s2t_model = "gemini-2.0-flash"
|
149 |
+
|
150 |
+
# --- VAD Initialization ---
|
151 |
+
self.vad = webrtcvad.Vad(3)
|
152 |
+
self.VAD_RATE = 16000
|
153 |
+
self.VAD_FRAME_MS = 20
|
154 |
+
self.VAD_FRAME_SAMPLES = int(self.VAD_RATE * (self.VAD_FRAME_MS / 1000.0))
|
155 |
+
self.VAD_FRAME_BYTES = self.VAD_FRAME_SAMPLES * 2
|
156 |
+
padding_ms = 300
|
157 |
+
self.vad_padding_frames = padding_ms // self.VAD_FRAME_MS
|
158 |
+
self.vad_ring_buffer = collections.deque(maxlen=self.vad_padding_frames)
|
159 |
+
self.vad_ratio = 0.9
|
160 |
+
self.vad_triggered = False
|
161 |
+
self.wav_data = bytearray()
|
162 |
+
self.internal_buffer = bytearray()
|
163 |
+
|
164 |
+
self.end_of_speech_time: float | None = None
|
165 |
+
self.first_latency_calculated: bool = False
|
166 |
+
|
167 |
+
def copy(self) -> "GeminiHandler":
|
168 |
+
return GeminiHandler(
|
169 |
+
expected_layout="mono",
|
170 |
+
output_sample_rate=self.output_sample_rate,
|
171 |
+
prompt_dict=self.prompt_dict,
|
172 |
+
)
|
173 |
+
|
174 |
+
def t2t(self, text: str) -> str:
|
175 |
+
print(f"Sending text to Gemini: {text}")
|
176 |
+
response = self.chat.send_message(text)
|
177 |
+
print(f"Received response from Gemini: {response.text}")
|
178 |
+
return response.text
|
179 |
+
|
180 |
+
def s2t(self, audio) -> str:
|
181 |
+
response = self.s2t_client.models.generate_content(
|
182 |
+
model=self.s2t_model,
|
183 |
+
contents=[
|
184 |
+
types.Part.from_bytes(data=audio, mime_type='audio/wav'),
|
185 |
+
'Generate a transcript of the speech.'
|
186 |
+
]
|
187 |
+
)
|
188 |
+
return response.text
|
189 |
+
|
190 |
+
async def start_up(self):
|
191 |
+
# Flag for if we are using text-to-text in the middle of the chain or not.
|
192 |
+
self.t2t_bool = False
|
193 |
+
self.sys_prompt = None
|
194 |
+
|
195 |
+
self.t2t_client = genai.Client(api_key=os.getenv("GEMINI_API_KEY"))
|
196 |
+
self.s2t_client = genai.Client(api_key=os.getenv("GEMINI_API_KEY"))#, http_options={"api_version": "v1alpha"})
|
197 |
+
if self.sys_prompt is not None:
|
198 |
+
chat_config = types.GenerateContentConfig(system_instruction=self.sys_prompt)
|
199 |
+
else:
|
200 |
+
chat_config = types.GenerateContentConfig(system_instruction="You are a helpful assistant.")
|
201 |
+
self.chat = self.t2t_client.chats.create(model=self.t2t_model, config=chat_config)
|
202 |
+
self.t2s_client = genai.Client(api_key=os.environ.get("GEMINI_API_KEY"))
|
203 |
+
|
204 |
+
voice_name = "Puck"
|
205 |
+
if self.t2t_bool:
|
206 |
+
sys_instruction = f""" You are Wisal, an AI assistant developed by Compumacy AI , and a knowledgeable Autism .
|
207 |
+
Your sole purpose is to provide helpful, respectful, and easy-to-understand answers about Autism Spectrum Disorder (ASD).
|
208 |
+
Always be clear, non-judgmental, and supportive."""
|
209 |
+
else:
|
210 |
+
sys_instruction = self.sys_prompt
|
211 |
+
|
212 |
+
if sys_instruction is not None:
|
213 |
+
config = LiveConnectConfig(
|
214 |
+
response_modalities=["AUDIO"],
|
215 |
+
speech_config=SpeechConfig(
|
216 |
+
voice_config=VoiceConfig(
|
217 |
+
prebuilt_voice_config=PrebuiltVoiceConfig(voice_name=voice_name)
|
218 |
+
)
|
219 |
+
),
|
220 |
+
system_instruction=Content(parts=[Part.from_text(text=sys_instruction)])
|
221 |
+
)
|
222 |
+
else:
|
223 |
+
config = LiveConnectConfig(
|
224 |
+
response_modalities=["AUDIO"],
|
225 |
+
speech_config=SpeechConfig(
|
226 |
+
voice_config=VoiceConfig(
|
227 |
+
prebuilt_voice_config=PrebuiltVoiceConfig(voice_name=voice_name)
|
228 |
+
)
|
229 |
+
),
|
230 |
+
)
|
231 |
+
|
232 |
+
async with self.t2s_client.aio.live.connect(model=self.model, config=config) as session:
|
233 |
+
async for text_from_user in self.stream():
|
234 |
+
print("--------------------------------------------")
|
235 |
+
print(f"Received text from user and reading aloud: {text_from_user}")
|
236 |
+
print("--------------------------------------------")
|
237 |
+
if text_from_user and text_from_user.strip():
|
238 |
+
if self.t2t_bool:
|
239 |
+
prompt = f"""
|
240 |
+
You are Wisal, an AI assistant developed by Compumacy AI , and a knowledgeable Autism .
|
241 |
+
Your sole purpose is to provide helpful, respectful, and easy-to-understand answers about Autism Spectrum Disorder (ASD).
|
242 |
+
Always be clear, non-judgmental, and supportive.
|
243 |
+
|
244 |
+
{text_from_user}
|
245 |
+
"""
|
246 |
+
else:
|
247 |
+
prompt = text_from_user
|
248 |
+
await session.send_client_content(
|
249 |
+
turns=types.Content(
|
250 |
+
role='user', parts=[types.Part(text=prompt)]))
|
251 |
+
async for resp_chunk in session.receive():
|
252 |
+
if resp_chunk.data:
|
253 |
+
array = np.frombuffer(resp_chunk.data, dtype=np.int16)
|
254 |
+
self.output_queue.put_nowait((self.output_sample_rate, array))
|
255 |
+
|
256 |
+
|
257 |
+
async def stream(self) -> AsyncGenerator[bytes, None]:
|
258 |
+
while not self.quit.is_set():
|
259 |
+
try:
|
260 |
+
# Get the text message to be converted to speech
|
261 |
+
text_to_speak = await self.input_queue.get()
|
262 |
+
yield text_to_speak
|
263 |
+
except (asyncio.TimeoutError, TimeoutError):
|
264 |
+
pass
|
265 |
+
|
266 |
+
async def receive(self, frame: tuple[int, np.ndarray]) -> None:
|
267 |
+
sr, array = frame
|
268 |
+
audio_bytes = array.tobytes()
|
269 |
+
self.internal_buffer.extend(audio_bytes)
|
270 |
+
|
271 |
+
while len(self.internal_buffer) >= self.VAD_FRAME_BYTES:
|
272 |
+
vad_frame = self.internal_buffer[:self.VAD_FRAME_BYTES]
|
273 |
+
self.internal_buffer = self.internal_buffer[self.VAD_FRAME_BYTES:]
|
274 |
+
is_speech = self.vad.is_speech(vad_frame, self.VAD_RATE)
|
275 |
+
|
276 |
+
if not self.vad_triggered:
|
277 |
+
self.vad_ring_buffer.append((vad_frame, is_speech))
|
278 |
+
num_voiced = len([f for f, speech in self.vad_ring_buffer if speech])
|
279 |
+
if num_voiced > self.vad_ratio * self.vad_ring_buffer.maxlen:
|
280 |
+
print("Speech detected, starting to record...")
|
281 |
+
self.vad_triggered = True
|
282 |
+
for f, s in self.vad_ring_buffer:
|
283 |
+
self.wav_data.extend(f)
|
284 |
+
self.vad_ring_buffer.clear()
|
285 |
+
else:
|
286 |
+
self.wav_data.extend(vad_frame)
|
287 |
+
self.vad_ring_buffer.append((vad_frame, is_speech))
|
288 |
+
num_unvoiced = len([f for f, speech in self.vad_ring_buffer if not speech])
|
289 |
+
if num_unvoiced > self.vad_ratio * self.vad_ring_buffer.maxlen:
|
290 |
+
print("End of speech detected.")
|
291 |
+
|
292 |
+
|
293 |
+
self.end_of_speech_time = time.monotonic()
|
294 |
+
|
295 |
+
self.vad_triggered = False
|
296 |
+
full_utterance_np = np.frombuffer(self.wav_data, dtype=np.int16)
|
297 |
+
audio_input_wav = numpy_array_to_wav_bytes(full_utterance_np, sr)
|
298 |
+
|
299 |
+
text_input = self.s2t(audio_input_wav)
|
300 |
+
if text_input and text_input.strip():
|
301 |
+
if self.t2t_bool:
|
302 |
+
text_message = self.t2t(text_input)
|
303 |
+
else:
|
304 |
+
text_message = text_input
|
305 |
+
self.input_queue.put_nowait(text_message)
|
306 |
+
else:
|
307 |
+
print("STT returned empty transcript, skipping.")
|
308 |
+
|
309 |
+
self.vad_ring_buffer.clear()
|
310 |
+
self.wav_data = bytearray()
|
311 |
+
|
312 |
+
async def emit(self) -> tuple[int, np.ndarray] | None:
|
313 |
+
|
314 |
+
return await wait_for_item(self.output_queue)
|
315 |
+
|
316 |
+
def shutdown(self) -> None:
|
317 |
+
|
318 |
+
self.quit.set()
|
319 |
+
|
320 |
+
|
321 |
+
with gr.Blocks() as demo:
|
322 |
+
gr.Markdown("# Gemini Chained Speech-to-Speech Demo")
|
323 |
+
|
324 |
+
# for audio modality
|
325 |
+
# with gr.Row(visible=(modality_selector.value == "audio")) as row2:
|
326 |
+
with gr.Row() as row2:
|
327 |
+
with gr.Column(): # Optional, can be removed if not needed
|
328 |
+
webrtc2 = WebRTC(
|
329 |
+
label="Audio Chat",
|
330 |
+
modality="audio",
|
331 |
+
mode="send-receive",
|
332 |
+
elem_id="audio-source",
|
333 |
+
rtc_configuration=get_cloudflare_turn_credentials_async,
|
334 |
+
icon="https://www.gstatic.com/lamda/images/gemini_favicon_f069958c85030456e93de685481c559f160ea06b.png",
|
335 |
+
pulse_color="rgb(255, 255, 255)",
|
336 |
+
icon_button_color="rgb(255, 255, 255)",
|
337 |
+
)
|
338 |
+
# Corrected inputs and outputs for webrtc2.stream to use webrtc2
|
339 |
+
webrtc2.stream(
|
340 |
+
GeminiHandler(),
|
341 |
+
inputs=[webrtc2], # Was webrtc
|
342 |
+
outputs=[webrtc2],# Was webrtc
|
343 |
+
time_limit=180 if get_space() else None,
|
344 |
+
concurrency_limit=2 if get_space() else None,
|
345 |
+
)
|
346 |
+
|
347 |
+
if __name__ == "__main__":
|
348 |
+
demo.launch(server_port=7860)
|