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ed216da cc4e668 ed216da 597df9b ed216da ef48650 ed216da b811820 ed216da b811820 ed216da c39bcb6 ed216da | 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 | import gradio as gr
from gradio import wasm_utils
from fastrtc import ReplyOnPause, AlgoOptions, SileroVadOptions, AdditionalOutputs, WebRTC, get_cloudflare_turn_credentials_async, get_cloudflare_turn_credentials #get_hf_turn_credentials,
import os
from dotenv import load_dotenv
import time
import numpy as np
import sys
sys.path.append(os.path.abspath(os.path.join(os.path.dirname(__file__), '..')))
from backend.tts import synthesize_text
from backend.asr import transcribe_audio
from backend.utils import preprocess_audio, is_valid_turn
from backend.main import stream_chat_response
from pydub import AudioSegment
phone_waiting_sound = AudioSegment.from_mp3("frontend/phone-ringing-382734.mp3")[:1000]
sound_samples = np.array(phone_waiting_sound.get_array_of_samples(), dtype=np.int16)
if phone_waiting_sound.channels > 1:
sound_samples = sound_samples.reshape((-1, phone_waiting_sound.channels)).mean(axis=1)
sound_samples = sound_samples.astype(np.float32) / 32768.0 # Normalize to [-1,
def startup(_):
yield (phone_waiting_sound.frame_rate, sound_samples)
STARTUP_MESSAGE = "สวัสดีค่ะณภัทร 1157Homeshopping ยินดีให้บริการค่ะ"
yield from synthesize_text(STARTUP_MESSAGE)
time.sleep(2)
yield AdditionalOutputs([{"role": "assistant", "content": STARTUP_MESSAGE}])
custom_css = """
/* Overall Gradio page styling: hot pink background */
body {
/* background-color: #ff69b4; /* Hot pink */
margin: 0;
padding: 0;
font-family: sans-serif;}
/* Title styling */
h1 {
color: #fff;
text-shadow: 1px 1px 2px #ff85a2;
font-size: 2.5em;
margin-bottom: 20px;
text-align: center;
}
/* Style the column holding the telephone interface */
.phone-column {
max-width: 350px !important; /* Limit the width of the phone column */
margin: 0 auto; /* Center the column */
border-radius: 20px;
background-color: #ff69b4; /* Lighter pink for telephone interface */
box-shadow: 0 0 15px rgba(0, 0, 0, 0.2);
padding: 20px;
}
/* Conversation history box styling */
#conversation-history-chatbot {
background-color: #ffc0cb; /* Lighter pink for conversation history */
border: 1px solid #ccc;
border-radius: 10px;
padding: 10px;
box-shadow: 0 0 15px rgba(0, 0, 0, 0.2);
}
"""
def response(audio: tuple[int, np.ndarray] | None, conversation_history):
"""
Handles user audio input, transcribes it, streams LLM text via backend.main,
and synthesizes chunks to audio while updating the conversation history.
"""
print(f"--- Latency Breakdown ---")
start_time = time.time()
if conversation_history is None:
conversation_history = []
previous_history = list(conversation_history)
if not audio or audio[1] is None or not np.any(audio[1]):
print("No audio input detected; skipping response generation.")
print(f"------------------------")
return
import soundfile as sf
sample_rate, audio_array = audio
try:
processed_audio = preprocess_audio((sample_rate, audio_array), target_frame_rate=16000)
except Exception as audio_err:
print(f"Audio preprocessing failed: {audio_err}")
print(f"------------------------")
return
t0 = time.time()
transcription = transcribe_audio( processed_audio)
t_asr = time.time() - t0
print(f"ASR: {t_asr:.4f}s")
if not transcription.strip():
print("No valid transcription; skipping response generation.")
print(f"------------------------")
return
user_turn = {"role": "user", "content": transcription}
print(f"User: {transcription}")
if is_valid_turn(user_turn):
conversation_history.append(user_turn)
yield AdditionalOutputs(conversation_history)
print("Conversation history:", conversation_history)
assistant_turn = {"role": "assistant", "content": ""}
conversation_history.append(assistant_turn)
history_for_stream = [dict(turn) for turn in previous_history if is_valid_turn(turn)]
text_buffer = ""
full_response = ""
delimiter_count = 0
n_threshold = 4
max_n_threshold = 7
lang = "th"
chunk_count = 0
first_chunk_sent = False
start_llm_stream = time.time()
try:
for text_chunk in stream_chat_response(history_for_stream, transcription):
if not isinstance(text_chunk, str):
text_chunk = str(text_chunk)
i = 0
while i < len(text_chunk):
char = text_chunk[i]
text_buffer += char
full_response += char
assistant_turn["content"] = full_response.strip()
is_delimiter = False
if char in {' ', '\n'}:
is_delimiter = True
delimiter_count += 1
if i + 1 < len(text_chunk) and text_chunk[i + 1] == 'ๆ':
text_buffer += text_chunk[i + 1]
full_response += text_chunk[i + 1]
i += 1
send_now = False
if not first_chunk_sent:
if is_delimiter and text_buffer.strip():
send_now = True
else:
if delimiter_count >= n_threshold and text_buffer.strip():
send_now = True
if n_threshold < max_n_threshold:
n_threshold += 1
if send_now:
buffer_to_send = text_buffer.strip()
try:
if buffer_to_send and buffer_to_send.endswith('วันที่'):
buffer_to_send = buffer_to_send[:-len('วันที่')]
if buffer_to_send and first_chunk_sent and buffer_to_send.endswith('ค่ะ'):
buffer_to_send = buffer_to_send[:-len('ค่ะ')]
except Exception:
buffer_to_send = buffer_to_send.replace('ค่ะ', '')
if buffer_to_send:
chunk_count += 1
if chunk_count == 1:
first_llm_chunk_time = time.time()
t_llm_first_token = first_llm_chunk_time - start_llm_stream
print(f"LLM TTFC: {t_llm_first_token:.4f}s (Time To First Chunk)")
yield from synthesize_text(buffer_to_send, lang=lang)
first_chunk_sent = True
text_buffer = ""
delimiter_count = 0
yield AdditionalOutputs(conversation_history)
i += 1
if text_buffer.strip():
buffer_to_send = text_buffer.strip()
try:
if buffer_to_send and buffer_to_send.endswith('วันที่'):
buffer_to_send = buffer_to_send[:-len('วันที่')]
if buffer_to_send and first_chunk_sent and buffer_to_send.endswith('ค่ะ'):
buffer_to_send = buffer_to_send[:-len('ค่ะ')]
except Exception:
buffer_to_send = buffer_to_send.replace('ค่ะ', '')
if buffer_to_send:
chunk_count += 1
if chunk_count == 1:
first_llm_chunk_time = time.time()
t_llm_first_token = first_llm_chunk_time - start_llm_stream
print(f"LLM TTFC: {t_llm_first_token:.4f}s (Time To First Chunk)")
yield from synthesize_text(buffer_to_send, lang=lang)
first_chunk_sent = True
text_buffer = ""
delimiter_count = 0
yield AdditionalOutputs(conversation_history)
except Exception as e:
print(f"An error occurred during response generation or synthesis: {e}")
error_message = "ขออภัยค่ะ เกิดข้อผิดพลาดบางอย่าง"
try:
yield from synthesize_text(error_message, lang=lang)
except Exception as synth_error:
print(f"Could not synthesize error message: {synth_error}")
assistant_turn["content"] = (assistant_turn.get("content", "") + f" [Error: {e}]").strip()
yield AdditionalOutputs(conversation_history)
total_latency = time.time() - start_time
print(f"Total: {total_latency:.4f}s")
print(f"------------------------")
async def get_credentials():
return await get_cloudflare_turn_credentials_async(hf_token=os.getenv('HF_TOKEN'))
with gr.Blocks(css=custom_css, theme=gr.themes.Soft(primary_hue="pink", secondary_hue="pink")) as demo:
gr.HTML("""<h1 style='text-align: center'>1157 Voicebot Demo</h1>""")
with gr.Row():
with gr.Column(scale=1, elem_classes=["phone-column"]):
audio = WebRTC(
mode="send-receive",
modality="audio",
track_constraints={
"echoCancellation": True,
"noiseSuppression": {"exact": True},
"autoGainControl": {"exact": True}
},
rtc_configuration=get_credentials,
server_rtc_configuration=get_cloudflare_turn_credentials(ttl=360_000),
icon="https://i.pinimg.com/originals/0c/67/5a/0c675a8e1061478d2b7b21b330093444.gif",
icon_button_color="#17dbaa",
pulse_color="#b0f83b",
button_labels={"start": "Call", "stop": "Hang up", "waiting": "Connecting…"},
icon_radius=45,
height="650px",
width="100%",
container=False,
elem_id="phone-call-webrtc"
)
with gr.Column():
conversation_history = gr.Chatbot(
label="Conversation History",
type="messages",
value=[],
height="675px",
resizable=True,
avatar_images=(None, "https://i.pinimg.com/originals/0c/67/5a/0c675a8e1061478d2b7b21b330093444.gif"),
)
gr.DeepLinkButton()
audio.stream(
fn=ReplyOnPause(
response,
algo_options=AlgoOptions(
audio_chunk_duration=1.0,
started_talking_threshold=0.4,
speech_threshold=0.6
),
model_options=SileroVadOptions(
threshold=0.5,
min_speech_duration_ms=300,
max_speech_duration_s=float("inf"),
min_silence_duration_ms=1200,
),
can_interrupt=True,
startup_fn=startup,
),
inputs=[audio, conversation_history],
outputs=[audio],
concurrency_limit=1000,
time_limit=8192
)
audio.on_additional_outputs(
lambda history: history,
outputs=[conversation_history],
queue=True,
show_progress="hidden"
)
demo.queue(default_concurrency_limit=1000)
demo.launch(debug=True, show_error=True, share=True)
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