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| import streamlit as st | |
| import wave | |
| # from ASR import recognize | |
| import requests | |
| import pyaudio | |
| import numpy as np | |
| import base64 | |
| import io | |
| import os | |
| import time | |
| import tempfile | |
| import librosa | |
| import traceback | |
| from pydub import AudioSegment | |
| from utils.vad import get_speech_timestamps, collect_chunks, VadOptions | |
| API_URL = os.getenv("API_URL", "http://127.0.0.1:60808/chat") | |
| # 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): | |
| # 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()) | |
| 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") | |
| files = {"audio": base64_encoded} | |
| output_audio_bytes = b"" | |
| with requests.post(API_URL, json=files, stream=True) as response: | |
| try: | |
| for chunk in response.iter_content(chunk_size=OUT_CHUNK): | |
| if chunk: | |
| # Convert chunk to numpy array | |
| output_audio_bytes += chunk | |
| audio_data = np.frombuffer(chunk, dtype=np.int8) | |
| # Play audio | |
| stream.write(audio_data) | |
| 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.audio(out_file, format="audio/wav", loop=False, autoplay=False) | |
| st.session_state.messages.append( | |
| {"role": "assistant", "content": out_file, "type": "audio"} | |
| ) | |
| 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() | |
| def main(): | |
| st.title("Chat Mini-Omni Demo") | |
| status = st.empty() | |
| 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 | |
| for message in st.session_state.messages: | |
| with st.chat_message(message["role"]): | |
| if message["type"] == "msg": | |
| 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: | |
| st.session_state.speaking = True | |
| speaking(status) | |
| # 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() | |