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import sys
import os
sys.path.append(os.path.dirname(os.path.dirname(os.path.abspath(__file__))))
from utils.vad import get_speech_timestamps, collect_chunks, VadOptions
import streamlit as st
import wave
# from ASR import recognize
import requests
import pyaudio
import numpy as np
import base64
import io
from typing import List
import av
import os
import time
import tempfile
import librosa
import traceback
from pydub import AudioSegment
from datetime import datetime
from PIL import Image
import streamlit_webrtc
from io import BytesIO
# set wide mode
# st.set_page_config(layout="wide")
last_video_frame = None
last_video_frame_ts = time.time()
API_URL = os.getenv("API_URL", "http://127.0.0.1:60808/chat")
API_URL = None if API_URL == "" else API_URL
# recording parameters
IN_FORMAT = pyaudio.paInt16
IN_CHANNELS = 1
IN_RATE = 24000
IN_CHUNK = 1024
IN_SAMPLE_WIDTH = 2
VAD_STRIDE = 0.5
# playing parameters
OUT_FORMAT = pyaudio.paInt16
OUT_CHANNELS = 1
OUT_RATE = 24000
OUT_SAMPLE_WIDTH = 2
OUT_CHUNK = 5760
# Initialize chat history
if "messages" not in st.session_state:
st.session_state.messages = []
def run_vad(ori_audio, sr):
_st = time.time()
try:
audio = np.frombuffer(ori_audio, dtype=np.int16)
audio = audio.astype(np.float32) / 32768.0
sampling_rate = 16000
if sr != sampling_rate:
audio = librosa.resample(audio, orig_sr=sr, target_sr=sampling_rate)
vad_parameters = {}
vad_parameters = VadOptions(**vad_parameters)
speech_chunks = get_speech_timestamps(audio, vad_parameters)
audio = collect_chunks(audio, speech_chunks)
duration_after_vad = audio.shape[0] / sampling_rate
if sr != sampling_rate:
# resample to original sampling rate
vad_audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=sr)
else:
vad_audio = audio
vad_audio = np.round(vad_audio * 32768.0).astype(np.int16)
vad_audio_bytes = vad_audio.tobytes()
return duration_after_vad, vad_audio_bytes, round(time.time() - _st, 4)
except Exception as e:
msg = f"[asr vad error] audio_len: {len(ori_audio)/(sr*2):.3f} s, trace: {traceback.format_exc()}"
print(msg)
return -1, ori_audio, round(time.time() - _st, 4)
def warm_up():
frames = b"\x00\x00" * 1024 * 2 # 1024 frames of 2 bytes each
dur, frames, tcost = run_vad(frames, 16000)
print(f"warm up done, time_cost: {tcost:.3f} s")
def save_tmp_audio(audio_bytes):
with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tmpfile:
file_name = tmpfile.name
audio = AudioSegment(
data=audio_bytes,
sample_width=OUT_SAMPLE_WIDTH,
frame_rate=OUT_RATE,
channels=OUT_CHANNELS,
)
audio.export(file_name, format="wav")
return file_name
def speaking(status, resp_text_holder=None, encoded_img=None):
# Initialize PyAudio
p = pyaudio.PyAudio()
# Open PyAudio stream
stream = p.open(
format=OUT_FORMAT, channels=OUT_CHANNELS, rate=OUT_RATE, output=True
)
audio_buffer = io.BytesIO()
wf = wave.open(audio_buffer, "wb")
wf.setnchannels(IN_CHANNELS)
wf.setsampwidth(IN_SAMPLE_WIDTH)
wf.setframerate(IN_RATE)
total_frames = b"".join(st.session_state.frames)
dur = len(total_frames) / (IN_RATE * IN_CHANNELS * IN_SAMPLE_WIDTH)
status.warning(f"Speaking... recorded audio duration: {dur:.3f} s")
wf.writeframes(total_frames)
with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tmpfile:
with open(tmpfile.name, "wb") as f:
f.write(audio_buffer.getvalue())
with open("input_audio.wav", "wb") as f:
f.write(audio_buffer.getvalue())
file_name = tmpfile.name
with st.chat_message("user"):
st.audio(file_name, format="audio/wav", loop=False, autoplay=False)
st.session_state.messages.append(
{"role": "assistant", "content": file_name, "type": "audio"}
)
st.session_state.frames = []
audio_bytes = audio_buffer.getvalue()
base64_encoded = str(base64.b64encode(audio_bytes), encoding="utf-8")
if API_URL is not None:
output_audio_bytes = b""
files = {"audio": base64_encoded}
if encoded_img is not None:
files["image"] = encoded_img
print("sending request to server")
resp_text_holder.empty()
resp_text = ""
with requests.post(API_URL, json=files, stream=True) as response:
try:
buffer = b''
for chunk in response.iter_content(chunk_size=2048):
buffer += chunk
while b'\r\n--frame\r\n' in buffer:
frame, buffer = buffer.split(b'\r\n--frame\r\n', 1)
if b'Content-Type: audio/wav' in frame:
audio_data = frame.split(b'\r\n\r\n', 1)[1]
# audio_data = base64.b64decode(audio_data)
output_audio_bytes += audio_data
audio_array = np.frombuffer(audio_data, dtype=np.int8)
stream.write(audio_array)
elif b'Content-Type: text/plain' in frame:
text_data = frame.split(b'\r\n\r\n', 1)[1].decode()
resp_text += text_data
if len(text_data) > 0:
print(resp_text, end='\r')
resp_text_holder.write(resp_text)
except Exception as e:
st.error(f"Error during audio streaming: {e}")
out_file = save_tmp_audio(output_audio_bytes)
with st.chat_message("assistant"):
st.write(resp_text)
with st.chat_message("assistant"):
st.audio(out_file, format="audio/wav", loop=False, autoplay=False)
st.session_state.messages.append(
{"role": "assistant", "content": resp_text, "type": "text"}
)
st.session_state.messages.append(
{"role": "assistant", "content": out_file, "type": "audio"}
)
else:
st.error("API_URL is not set. Please set the API_URL environment variable.")
time.sleep(1)
wf.close()
# Close PyAudio stream and terminate PyAudio
stream.stop_stream()
stream.close()
p.terminate()
st.session_state.speaking = False
st.session_state.recording = True
def recording(status):
audio = pyaudio.PyAudio()
stream = audio.open(
format=IN_FORMAT,
channels=IN_CHANNELS,
rate=IN_RATE,
input=True,
frames_per_buffer=IN_CHUNK,
)
temp_audio = b""
vad_audio = b""
start_talking = False
last_temp_audio = None
st.session_state.frames = []
while st.session_state.recording:
status.success("Listening...")
audio_bytes = stream.read(IN_CHUNK)
temp_audio += audio_bytes
if len(temp_audio) > IN_SAMPLE_WIDTH * IN_RATE * IN_CHANNELS * VAD_STRIDE:
dur_vad, vad_audio_bytes, time_vad = run_vad(temp_audio, IN_RATE)
print(f"duration_after_vad: {dur_vad:.3f} s, time_vad: {time_vad:.3f} s")
if dur_vad > 0.2 and not start_talking:
if last_temp_audio is not None:
st.session_state.frames.append(last_temp_audio)
start_talking = True
if start_talking:
st.session_state.frames.append(temp_audio)
if dur_vad < 0.1 and start_talking:
st.session_state.recording = False
print(f"speech end detected. excit")
last_temp_audio = temp_audio
temp_audio = b""
stream.stop_stream()
stream.close()
audio.terminate()
async def queued_video_frames_callback(frames: List[av.VideoFrame]) -> List[av.VideoFrame]:
# print(f"test-------queued_video_frames_callback")
global last_video_frame
global last_video_frame_ts
if len(frames) != 0:
if time.time() - last_video_frame_ts > 1:
last_frame = frames[-1]
# with video_frame_lock:
# last_video_frame[0] = last_frame.to_image()
# last_video_frame_ts[0] = time.time()
last_video_frame = last_frame.to_image()
last_video_frame_ts = time.time()
return frames
def main():
st.title("Chat Mini-Omni2 Demo")
status = st.empty()
# Mode selection
mode = st.radio(
"Select mode:",
("Audio-only", "Audio-vision"),
key="mode_selection",
horizontal=True
)
if mode == "Audio-only":
st.session_state.use_vision = False
st.info("Audio-only mode selected. The system will process only audio input.")
else: # Audio-vision
st.session_state.use_vision = True
st.info("Audio-vision mode selected. The system will process both audio and video input.")
if "warm_up" not in st.session_state:
warm_up()
st.session_state.warm_up = True
if "start" not in st.session_state:
st.session_state.start = False
if "recording" not in st.session_state:
st.session_state.recording = False
if "speaking" not in st.session_state:
st.session_state.speaking = False
if "frames" not in st.session_state:
st.session_state.frames = []
if not st.session_state.start:
status.warning("Click Start to chat")
start_col, stop_col, _ = st.columns([0.2, 0.2, 0.6])
start_button = start_col.button("Start", key="start_button")
# stop_button = stop_col.button("Stop", key="stop_button")
if start_button:
time.sleep(1)
st.session_state.recording = True
st.session_state.start = True
if st.session_state.use_vision:
with st.sidebar:
webrtc_ctx = streamlit_webrtc.webrtc_streamer(
key="speech-w-video",
mode=streamlit_webrtc.WebRtcMode.SENDRECV,
# rtc_configuration={"iceServers": get_ice_servers()},
media_stream_constraints={"video": True, "audio": False},
# video_receiver_size=10, # Increased from default 4 to 10
queued_video_frames_callback=queued_video_frames_callback,
)
if not webrtc_ctx.state.playing:
st.warning("Please allow camera access and try again.")
return
resp_text_holder = st.empty()
for message in st.session_state.messages:
with st.chat_message(message["role"]):
if message["type"] == "text":
st.markdown(message["content"])
elif message["type"] == "img":
st.image(message["content"], width=300)
elif message["type"] == "audio":
st.audio(
message["content"], format="audio/wav", loop=False, autoplay=False
)
while st.session_state.start:
if st.session_state.recording:
recording(status)
if not st.session_state.recording and st.session_state.start:
encoded_img = None
if st.session_state.use_vision:
# last_img = webrtc_ctx.video_receiver.get_frame(timeout=5).to_image()
last_img = last_video_frame
if last_img:
with st.chat_message("user"):
st.image(last_img, width=300)
st.session_state.messages.append({"role": "user", "content": last_img, "type": "img"})
input_img = last_img
buffer = BytesIO()
input_img.save(buffer, format="JPEG")
with open("input_image.jpg", "wb") as f:
f.write(buffer.getvalue())
encoded_img = str(base64.b64encode(buffer.getvalue()), encoding="utf-8")
else:
st.error("No image captured. Please allow camera access and try again.")
return
st.session_state.speaking = True
speaking(status, resp_text_holder, encoded_img)
# if stop_button:
# status.warning("Stopped, click Start to chat")
# st.session_state.start = False
# st.session_state.recording = False
# st.session_state.frames = []
# break
if __name__ == "__main__":
main()
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