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inference_s2st.py
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inference_s2st.py
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import time
import torch
try:
import torch_musa
except ImportError as e:
print("You should install torch_musa if you want to run on Moore Threads GPU")
import os
import argparse
import torchaudio
from torchaudio.transforms import Resample
import logging
from mooer.datasets.speech_processor import *
from mooer.configs import asr_config
from mooer.models import mooer_model
from mooer.utils.utils import *
from mooer.models.hifigan import save_wav, get_hifigan_model, get_speaker_encoder, encode_prompt_wav
parser = argparse.ArgumentParser()
parser.add_argument("--wav_path", default='demo/resources/demo.wav', type=str, help="decode one wav file")
parser.add_argument("--wav_scp", default=None, type=str, help="decode scp if you want")
parser.add_argument("--task", default='s2s_chat', choices=['asr', 'ast', 's2s_trans', 's2s_chat'],
type=str, help="task: asr or ast or s2s_trans or s2s_chat. "
"Please set ast if you choose a asr/ast/s2s_trans/s2s_chat multitask model")
parser.add_argument("--batch_size", default=1, type=int, help="decode batch for scp")
parser.add_argument("--cmvn_path", default='', type=str, help="cmvn path.")
parser.add_argument("--encoder_path", default='', type=str, help="encoder path.")
parser.add_argument("--llm_path", default='', type=str, help="llm path.")
parser.add_argument("--adapter_path", default='', type=str, help="adapter path.")
parser.add_argument("--lora_dir", default='', type=str, help="lora path.")
parser.add_argument("--vocoder_path", default='', type=str, help="vocoder path")
parser.add_argument("--spk_encoder_path", default='', type=str, help="spk encoder path")
parser.add_argument("--prompt_wav_path", default='', type=str, help="prompt wav path")
parser.add_argument("--output_dir", default="response_wavs_dir", type=str, help="path to save wav generated")
args = parser.parse_args()
assert args.batch_size == 1, "Only support bsz=1 for S2ST task now. We will support batch inference soon."
logging.basicConfig(
level=logging.INFO,
format="%(asctime)s | %(levelname)s | %(name)s | %(message)s",
datefmt="%Y-%m-%d %H:%M:%S",
filemode='w'
)
PROMPT_TEMPLATE_DICT = {
'qwen': "<|im_start|>system\nYou are a helpful assistant.<|im_end|>\n<|im_start|>user\n{}<|im_end|>\n<|im_start|>assistant\n",
}
PROMPT_DICT = {
'asr': "Transcribe speech to text. ",
'ast': "Translate speech to english text. ",
's2s_trans': "Translate speech to english speech. ",
's2s_chat': "Answer my question with speech. "
}
model_config = asr_config.ModelConfig()
logger = logging.getLogger()
logger.setLevel(logging.INFO)
# replace path
if args.llm_path and os.path.exists(args.llm_path):
model_config.llm_path = args.llm_path
if args.encoder_path and os.path.exists(args.encoder_path):
model_config.encoder_path = args.encoder_path
if args.adapter_path and os.path.exists(args.adapter_path):
model_config.adapter_path = args.adapter_path
if args.lora_dir and os.path.exists(args.lora_dir):
model_config.lora_dir = args.lora_dir
if args.cmvn_path and os.path.exists(args.cmvn_path):
model_config.cmvn_path = args.cmvn_path
if args.task:
model_config.prompt_key = args.task
device = str(get_device())
logger.info("This demo will run on {}".format(device.upper()))
logger.info(model_config)
os.makedirs(args.output_dir, exist_ok=True)
logger.info("Response wav will save in {}".format(args.output_dir))
model, tokenizer = mooer_model.init_model(
model_config=model_config)
AUDIO_START_TOKEN_INDEX = tokenizer.get_vocab()['<|audio_start|>']
model.to(device)
model.eval()
# data process
prompt_template_key = model_config.get('prompt_template_key', 'qwen')
prompt_template = PROMPT_TEMPLATE_DICT[prompt_template_key]
prompt_key = model_config.get('prompt_key', 'asr')
prompt_org = PROMPT_DICT[prompt_key]
logger.info(f"Use LLM Type {prompt_template_key}, "
f"Prompt template {prompt_template}, "
f"Use task type {prompt_key}, "
f"Prompt {prompt_org}")
cmvn = load_cmvn(model_config.get('cmvn_path'))
adapter_downsample_rate = model_config.get('adapter_downsample_rate')
hifigan_generator = get_hifigan_model(args.vocoder_path, device, decoder_dim=3584)
spk_encoder = get_speaker_encoder(args.spk_encoder_path, device)
spk_embedding = encode_prompt_wav(spk_encoder, args.prompt_wav_path, device)
def process_wav(wav_path):
audio_raw, sample_rate = torchaudio.load(wav_path)
if sample_rate != 16000:
# resample the data
resampler = Resample(orig_freq=sample_rate, new_freq=16000)
audio_raw = resampler(audio_raw)
if audio_raw.shape[0] > 1:
# convert to mono
audio_raw = audio_raw.mean(dim=0, keepdim=True)
audio_raw = audio_raw[0]
prompt = prompt_template.format(prompt_org)
audio_mel = compute_fbank(waveform=audio_raw)
audio_mel = apply_lfr(inputs=audio_mel, lfr_m=7, lfr_n=6)
audio_mel = apply_cmvn(audio_mel, cmvn=cmvn)
audio_length = audio_mel.shape[0]
audio_length = audio_length // adapter_downsample_rate
audio_pseudo = torch.full((audio_length,), -1)
prompt_ids = tokenizer.encode(prompt)
prompt_length = len(prompt_ids)
prompt_ids = torch.tensor(prompt_ids, dtype=torch.int64)
example_ids = torch.cat((audio_pseudo, prompt_ids)) # [audio, prompt]
example_mask = example_ids.ge(-1)
items = {
"input_ids": example_ids,
"attention_mask": example_mask,
"audio_mel": audio_mel,
"audio_length": audio_length,
"prompt_length": prompt_length,
}
return items
load_dtype = model_config.get('load_dtype', 'bfloat16')
dtype = torch.float32
if load_dtype == 'float16':
dtype = torch.float16
elif load_dtype == 'bfloat16':
dtype = torch.bfloat16
logging.info(f"Input data type: {dtype}")
context_scope = torch.musa.amp.autocast if 'musa' in device else torch.cuda.amp.autocast
with torch.no_grad():
if args.wav_scp is not None and os.path.exists(args.wav_scp):
batch_size = args.batch_size
infer_time = []
items = parse_key_text(args.wav_scp)
uttids = list(items.keys())
num_batches = len(uttids) // batch_size + (0 if len(uttids) % batch_size == 0 else 1)
for i in range(num_batches):
try:
batch_uttids = uttids[i * batch_size:(i + 1) * batch_size]
batch_wav_paths = [items[uttid] for uttid in batch_uttids]
samples = []
for wav_path in batch_wav_paths:
samples.append(process_wav(wav_path))
batch = process_batch(samples, tokenizer=tokenizer)
for key in batch.keys():
batch[key] = batch[key].to(device) if isinstance(batch[key], torch.Tensor) else batch[key]
with context_scope(dtype=dtype):
ss = time.perf_counter()
inputs_embeds, attention_mask, kwargs = model.generate(**batch, compute_llm=False)
prompt_and_encoding_length = inputs_embeds.shape[1]
model_outputs = model.llm.generate(
inputs_embeds=inputs_embeds,
max_new_tokens=kwargs.get("max_new_tokens", 1000),
num_beams=kwargs.get("num_beams", 4),
do_sample=True,
min_length=kwargs.get("min_length", 1),
top_p=0.85,
repetition_penalty=kwargs.get("repetition_penalty", 1.0),
length_penalty=kwargs.get("length_penalty", 1.0),
temperature=kwargs.get("temperature", 1.0),
attention_mask=attention_mask,
bos_token_id=model.tokenizer.bos_token_id,
eos_token_id=model.tokenizer.eos_token_id,
pad_token_id=model.tokenizer.pad_token_id,
)
infer_time.append(time.perf_counter() - ss)
logging.info(f"Infer time: {time.perf_counter() - ss}")
output_text = model.tokenizer.batch_decode(model_outputs, add_special_tokens=False,
skip_special_tokens=True)
if hasattr(model.llm.model, "embed_tokens"):
teacher_forcing_input_embeds = model.llm.model.embed_tokens(model_outputs)
teacher_forcing_input_att_mask = torch.ones((1, teacher_forcing_input_embeds.shape[1]),
dtype=torch.bool).to(device)
else:
raise NotImplementedError
inputs_embeds = torch.concat([inputs_embeds, teacher_forcing_input_embeds], dim=-2)
attention_mask = torch.concat([attention_mask, teacher_forcing_input_att_mask], dim=-1)
llm_output = model.llm(inputs_embeds=inputs_embeds, attention_mask=attention_mask,
output_hidden_states=True)
audio_start_index = prompt_and_encoding_length + model_outputs[0].tolist().index(AUDIO_START_TOKEN_INDEX)
audio_latents = llm_output.hidden_states[-1][:, audio_start_index:-6, :]
for idx, text in enumerate(output_text):
logger.info(f"uttid: {batch_uttids[idx]}")
audio_file_out_tts = os.path.join(args.output_dir, f"{batch_uttids[idx]}.tts.wav")
text_ast = text.split("<|audio_start|>")[0]
text_ast = text_ast.replace('\\n', '\n')
logger.info(f"AST: {text_ast}")
save_wav(hifigan_generator, spk_embedding, audio_latents.float(), audio_file_out_tts)
logger.info(f"Finished writing: {audio_file_out_tts}")
except Exception as e:
logging.error(e)
logging.info("Total inference cost")
logging.info(sum(infer_time))
elif args.wav_path != '' and os.path.exists(args.wav_path):
try:
wav_path = args.wav_path
items = process_wav(wav_path)
batch = process_batch([items], tokenizer=tokenizer)
for key in batch.keys():
batch[key] = batch[key].to(device) if isinstance(batch[key], torch.Tensor) else batch[key]
with context_scope(dtype=dtype):
ss = time.perf_counter()
inputs_embeds, attention_mask, kwargs = model.generate(**batch, compute_llm=False)
prompt_and_encoding_length = inputs_embeds.shape[1]
model_outputs = model.llm.generate(
inputs_embeds=inputs_embeds,
max_new_tokens=kwargs.get("max_new_tokens", 1000),
num_beams=kwargs.get("num_beams", 4),
do_sample=True,
min_length=kwargs.get("min_length", 1),
top_p=0.85,
repetition_penalty=kwargs.get("repetition_penalty", 1.0),
length_penalty=kwargs.get("length_penalty", 1.0),
temperature=kwargs.get("temperature", 1.0),
attention_mask=attention_mask,
bos_token_id=model.tokenizer.bos_token_id,
eos_token_id=model.tokenizer.eos_token_id,
pad_token_id=model.tokenizer.pad_token_id,
)
logging.info(f"Infer time: {time.perf_counter() - ss}")
output_text = model.tokenizer.batch_decode(model_outputs, add_special_tokens=False,
skip_special_tokens=True)
if hasattr(model.llm.model, "embed_tokens"):
teacher_forcing_input_embeds = model.llm.model.embed_tokens(model_outputs)
teacher_forcing_input_att_mask = torch.ones((1, teacher_forcing_input_embeds.shape[1]),
dtype=torch.bool).to(device)
else:
raise NotImplementedError
inputs_embeds = torch.concat([inputs_embeds, teacher_forcing_input_embeds], dim=-2)
attention_mask = torch.concat([attention_mask, teacher_forcing_input_att_mask], dim=-1)
llm_output = model.llm(inputs_embeds=inputs_embeds, attention_mask=attention_mask,
output_hidden_states=True)
audio_start_index = prompt_and_encoding_length + model_outputs[0].tolist().index(AUDIO_START_TOKEN_INDEX)
audio_latents = llm_output.hidden_states[-1][:, audio_start_index:-6, :]
for text in output_text:
uttid = os.path.basename(wav_path).replace(".wav", "")
audio_file_out_tts = os.path.join(args.output_dir, f"{uttid}.tts.wav")
text_ast = text.split("<|audio_start|>")[0]
text_ast = text_ast.replace('\\n', '\n')
logger.info(f"Text: {text_ast}")
save_wav(hifigan_generator, spk_embedding, audio_latents.float(), audio_file_out_tts)
logger.info(f"Finished writing: {audio_file_out_tts}")
except Exception as e:
logging.error(e)
else:
raise IOError("You should specify --wav_scp or --wav_path as the input")