VoiceAssistance / web_demo.py
StevenChen16's picture
first commit
31ba7c5
raw
history blame
10.8 kB
import json
import os.path
import tempfile
import sys
import re
import uuid
import requests
from argparse import ArgumentParser
import torchaudio
from transformers import WhisperFeatureExtractor, AutoTokenizer
from speech_tokenizer.modeling_whisper import WhisperVQEncoder
sys.path.insert(0, "./cosyvoice")
sys.path.insert(0, "./third_party/Matcha-TTS")
from speech_tokenizer.utils import extract_speech_token
import gradio as gr
import torch
audio_token_pattern = re.compile(r"<\|audio_(\d+)\|>")
from flow_inference import AudioDecoder
from audio_process import AudioStreamProcessor
if __name__ == "__main__":
parser = ArgumentParser()
parser.add_argument("--host", type=str, default="0.0.0.0")
parser.add_argument("--port", type=int, default="8888")
parser.add_argument("--flow-path", type=str, default="./glm-4-voice-decoder")
parser.add_argument("--model-path", type=str, default="THUDM/glm-4-voice-9b")
parser.add_argument("--tokenizer-path", type= str, default="THUDM/glm-4-voice-tokenizer")
args = parser.parse_args()
flow_config = os.path.join(args.flow_path, "config.yaml")
flow_checkpoint = os.path.join(args.flow_path, 'flow.pt')
hift_checkpoint = os.path.join(args.flow_path, 'hift.pt')
glm_tokenizer = None
device = "cuda"
audio_decoder: AudioDecoder = None
whisper_model, feature_extractor = None, None
def initialize_fn():
global audio_decoder, feature_extractor, whisper_model, glm_model, glm_tokenizer
if audio_decoder is not None:
return
# GLM
glm_tokenizer = AutoTokenizer.from_pretrained(args.model_path, trust_remote_code=True)
# Flow & Hift
audio_decoder = AudioDecoder(config_path=flow_config, flow_ckpt_path=flow_checkpoint,
hift_ckpt_path=hift_checkpoint,
device=device)
# Speech tokenizer
whisper_model = WhisperVQEncoder.from_pretrained(args.tokenizer_path).eval().to(device)
feature_extractor = WhisperFeatureExtractor.from_pretrained(args.tokenizer_path)
def clear_fn():
return [], [], '', '', '', None, None
def inference_fn(
temperature: float,
top_p: float,
max_new_token: int,
input_mode,
audio_path: str | None,
input_text: str | None,
history: list[dict],
previous_input_tokens: str,
previous_completion_tokens: str,
):
if input_mode == "audio":
assert audio_path is not None
history.append({"role": "user", "content": {"path": audio_path}})
audio_tokens = extract_speech_token(
whisper_model, feature_extractor, [audio_path]
)[0]
if len(audio_tokens) == 0:
raise gr.Error("No audio tokens extracted")
audio_tokens = "".join([f"<|audio_{x}|>" for x in audio_tokens])
audio_tokens = "<|begin_of_audio|>" + audio_tokens + "<|end_of_audio|>"
user_input = audio_tokens
system_prompt = "User will provide you with a speech instruction. Do it step by step. First, think about the instruction and respond in a interleaved manner, with 13 text token followed by 26 audio tokens. "
else:
assert input_text is not None
history.append({"role": "user", "content": input_text})
user_input = input_text
system_prompt = "User will provide you with a text instruction. Do it step by step. First, think about the instruction and respond in a interleaved manner, with 13 text token followed by 26 audio tokens."
# Gather history
inputs = previous_input_tokens + previous_completion_tokens
inputs = inputs.strip()
if "<|system|>" not in inputs:
inputs += f"<|system|>\n{system_prompt}"
inputs += f"<|user|>\n{user_input}<|assistant|>streaming_transcription\n"
with torch.no_grad():
response = requests.post(
"http://localhost:10000/generate_stream",
data=json.dumps({
"prompt": inputs,
"temperature": temperature,
"top_p": top_p,
"max_new_tokens": max_new_token,
}),
stream=True
)
text_tokens, audio_tokens = [], []
audio_offset = glm_tokenizer.convert_tokens_to_ids('<|audio_0|>')
end_token_id = glm_tokenizer.convert_tokens_to_ids('<|user|>')
complete_tokens = []
prompt_speech_feat = torch.zeros(1, 0, 80).to(device)
flow_prompt_speech_token = torch.zeros(1, 0, dtype=torch.int64).to(device)
this_uuid = str(uuid.uuid4())
tts_speechs = []
tts_mels = []
prev_mel = None
is_finalize = False
block_size_list = [25,50,100,150,200]
block_size_idx = 0
block_size = block_size_list[block_size_idx]
audio_processor = AudioStreamProcessor()
for chunk in response.iter_lines():
token_id = json.loads(chunk)["token_id"]
if token_id == end_token_id:
is_finalize = True
if len(audio_tokens) >= block_size or (is_finalize and audio_tokens):
if block_size_idx < len(block_size_list) - 1:
block_size_idx += 1
block_size = block_size_list[block_size_idx]
tts_token = torch.tensor(audio_tokens, device=device).unsqueeze(0)
if prev_mel is not None:
prompt_speech_feat = torch.cat(tts_mels, dim=-1).transpose(1, 2)
tts_speech, tts_mel = audio_decoder.token2wav(tts_token, uuid=this_uuid,
prompt_token=flow_prompt_speech_token.to(device),
prompt_feat=prompt_speech_feat.to(device),
finalize=is_finalize)
prev_mel = tts_mel
audio_bytes = audio_processor.process(tts_speech.clone().cpu().numpy()[0], last=is_finalize)
tts_speechs.append(tts_speech.squeeze())
tts_mels.append(tts_mel)
if audio_bytes:
yield history, inputs, '', '', audio_bytes, None
flow_prompt_speech_token = torch.cat((flow_prompt_speech_token, tts_token), dim=-1)
audio_tokens = []
if not is_finalize:
complete_tokens.append(token_id)
if token_id >= audio_offset:
audio_tokens.append(token_id - audio_offset)
else:
text_tokens.append(token_id)
tts_speech = torch.cat(tts_speechs, dim=-1).cpu()
complete_text = glm_tokenizer.decode(complete_tokens, spaces_between_special_tokens=False)
with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as f:
torchaudio.save(f, tts_speech.unsqueeze(0), 22050, format="wav")
history.append({"role": "assistant", "content": {"path": f.name, "type": "audio/wav"}})
history.append({"role": "assistant", "content": glm_tokenizer.decode(text_tokens, ignore_special_tokens=False)})
yield history, inputs, complete_text, '', None, (22050, tts_speech.numpy())
def update_input_interface(input_mode):
if input_mode == "audio":
return [gr.update(visible=True), gr.update(visible=False)]
else:
return [gr.update(visible=False), gr.update(visible=True)]
# Create the Gradio interface
with gr.Blocks(title="GLM-4-Voice Demo", fill_height=True) as demo:
with gr.Row():
temperature = gr.Number(
label="Temperature",
value=0.2
)
top_p = gr.Number(
label="Top p",
value=0.8
)
max_new_token = gr.Number(
label="Max new tokens",
value=2000,
)
chatbot = gr.Chatbot(
elem_id="chatbot",
bubble_full_width=False,
type="messages",
scale=1,
)
with gr.Row():
with gr.Column():
input_mode = gr.Radio(["audio", "text"], label="Input Mode", value="audio")
audio = gr.Audio(label="Input audio", type='filepath', show_download_button=True, visible=True)
text_input = gr.Textbox(label="Input text", placeholder="Enter your text here...", lines=2, visible=False)
with gr.Column():
submit_btn = gr.Button("Submit")
reset_btn = gr.Button("Clear")
output_audio = gr.Audio(label="Play", streaming=True,
autoplay=True, show_download_button=False)
complete_audio = gr.Audio(label="Last Output Audio (If Any)", show_download_button=True)
gr.Markdown("""## Debug Info""")
with gr.Row():
input_tokens = gr.Textbox(
label=f"Input Tokens",
interactive=False,
)
completion_tokens = gr.Textbox(
label=f"Completion Tokens",
interactive=False,
)
detailed_error = gr.Textbox(
label=f"Detailed Error",
interactive=False,
)
history_state = gr.State([])
respond = submit_btn.click(
inference_fn,
inputs=[
temperature,
top_p,
max_new_token,
input_mode,
audio,
text_input,
history_state,
input_tokens,
completion_tokens,
],
outputs=[history_state, input_tokens, completion_tokens, detailed_error, output_audio, complete_audio]
)
respond.then(lambda s: s, [history_state], chatbot)
reset_btn.click(clear_fn, outputs=[chatbot, history_state, input_tokens, completion_tokens, detailed_error, output_audio, complete_audio])
input_mode.input(clear_fn, outputs=[chatbot, history_state, input_tokens, completion_tokens, detailed_error, output_audio, complete_audio]).then(update_input_interface, inputs=[input_mode], outputs=[audio, text_input])
initialize_fn()
# Launch the interface
demo.launch(
server_port=args.port,
server_name=args.host
)