This commit is contained in:
游雁 2024-06-19 10:26:40 +08:00
commit 45d7aa9004
17 changed files with 1275 additions and 153 deletions

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@ -6,8 +6,11 @@
import sys
from funasr import AutoModel
model_dir=sys.argv[1]
input_file=sys.argv[2]
model_dir = "/Users/zhifu/Downloads/modelscope_models/ctc_model"
input_file = (
"https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
)
model = AutoModel(
model=model_dir,

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@ -69,7 +69,7 @@ dataset_conf:
batch_size_scale_ratio_max: 2
num_workers: 4
audio_adaptor_downsample_rate: ${audio_adaptor_conf.downsample_rate}
audio_encoder_downsample_rate: 2
audio_encoder_downsample_rate: 4
data_split_num: 512
batch_size_sample_max: 15
retry: 20

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@ -28,6 +28,9 @@ model = AutoModel(
init_param=f"{os.path.join(ckpt_dir, ckpt_id)}",
output_dir=output_dir,
device=device,
fp16=False,
bf16=False,
llm_dtype="bf16",
)

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@ -12,6 +12,7 @@ jsonl_dir="/nfs/beinian.lzr/workspace/GPT-4o/Data/Speech2Text/TestData"
out_dir="${ckpt_dir}/inference-${ckpt_id}"
mkdir -p ${out_dir}
for data_set in "librispeech_test_clean_speech2text.jsonl" "librispeech_test_other_speech2text.jsonl"; do
{
jsonl=${jsonl_dir}/${data_set}
output_dir=${out_dir}/${data_set}
mkdir -p ${output_dir}
@ -22,10 +23,12 @@ for data_set in "librispeech_test_clean_speech2text.jsonl" "librispeech_test_oth
python /mnt/workspace/zhifu.gzf/codebase/FunASR/funasr/metrics/wer.py ++ref_file=${ref_file} ++hyp_file=${pred_file} ++cer_file=${pred_file}.cer ++cn_postprocess=false
}&
done
wait
for data_set in "aishell1_test_speech2text.jsonl" "aishell2_ios_test_speech2text.jsonl" "librispeech_test_other_speech2text.jsonl"; do
for data_set in "aishell1_test_speech2text.jsonl" "aishell2_ios_test_speech2text.jsonl"; do
{
jsonl=${jsonl_dir}/${data_set}
output_dir=${out_dir}/${data_set}
mkdir -p ${output_dir}
@ -36,9 +39,12 @@ for data_set in "aishell1_test_speech2text.jsonl" "aishell2_ios_test_speech2text
python /mnt/workspace/zhifu.gzf/codebase/FunASR/funasr/metrics/wer.py ++ref_file=${ref_file} ++hyp_file=${pred_file} ++cer_file=${pred_file}.cer ++cn_postprocess=true
}&
done
wait
for data_set in "s2tt_en2zh.v20240605.test.jsonl"; do
for data_set in "common_voice_zh-CN_speech2text.jsonl" "common_voice_en_speech2text.jsonl"; do
{
jsonl=${jsonl_dir}/${data_set}
output_dir=${out_dir}/${data_set}
mkdir -p ${output_dir}
@ -47,19 +53,13 @@ for data_set in "s2tt_en2zh.v20240605.test.jsonl"; do
python ./demo_speech2text.py ${ckpt_dir} ${ckpt_id} ${jsonl} ${output_dir} ${device}
python /mnt/workspace/zhifu.gzf/codebase/FunASR/funasr/metrics/wer.py ++ref_file=${ref_file} ++hyp_file=${pred_file} ++cer_file=${pred_file}.cer ++cn_postprocess=true
cn_postprocess=false
if [ $data_set = "common_voice_zh-CN_speech2text.jsonl" ];then
cn_postprocess=true
fi
python /mnt/workspace/zhifu.gzf/codebase/FunASR/funasr/metrics/wer.py ++ref_file=${ref_file} ++hyp_file=${pred_file} ++cer_file=${pred_file}.cer ++cn_postprocess=${cn_postprocess}
}&
done
for data_set in "s2tt_zh2en.v20240605.test.jsonl"; do
jsonl=${jsonl_dir}/${data_set}
output_dir=${out_dir}/${data_set}
mkdir -p ${output_dir}
pred_file=${output_dir}/1best_recog/text_tn
ref_file=${output_dir}/1best_recog/label
python ./demo_speech2text.py ${ckpt_dir} ${ckpt_id} ${jsonl} ${output_dir} ${device}
python /mnt/workspace/zhifu.gzf/codebase/FunASR/funasr/metrics/wer.py ++ref_file=${ref_file} ++hyp_file=${pred_file} ++cer_file=${pred_file}.cer ++cn_postprocess=false
done
wait

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@ -0,0 +1,25 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
import sys
from funasr import AutoModel
model_dir = "/Users/zhifu/Downloads/modelscope_models/SenseVoiceCTC"
input_file = (
"https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
)
model = AutoModel(
model=model_dir,
)
res = model.generate(
input=input_file,
cache={},
language="zh",
text_norm="wotextnorm",
)
print(res)

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@ -213,7 +213,6 @@ class AutoModel:
deep_update(model_conf, kwargs.get("model_conf", {}))
deep_update(model_conf, kwargs)
model = model_class(**model_conf, vocab_size=vocab_size)
model.to(device)
# init_param
init_param = kwargs.get("init_param", None)
@ -236,6 +235,7 @@ class AutoModel:
model.to(torch.float16)
elif kwargs.get("bf16", False):
model.to(torch.bfloat16)
model.to(device)
return model, kwargs
def __call__(self, *args, **cfg):
@ -324,7 +324,7 @@ class AutoModel:
input, input_len=input_len, model=self.vad_model, kwargs=self.vad_kwargs, **cfg
)
end_vad = time.time()
# FIX(gcf): concat the vad clips for sense vocie model for better aed
if kwargs.get("merge_vad", False):
for i in range(len(res)):
@ -466,7 +466,7 @@ class AutoModel:
result[k] = restored_data[j][k]
else:
result[k] += restored_data[j][k]
if not len(result["text"].strip()):
continue
return_raw_text = kwargs.get("return_raw_text", False)
@ -481,7 +481,7 @@ class AutoModel:
if return_raw_text:
result["raw_text"] = raw_text
result["text"] = punc_res[0]["text"]
# speaker embedding cluster after resorted
if self.spk_model is not None and kwargs.get("return_spk_res", True):
if raw_text is None:
@ -602,6 +602,6 @@ class AutoModel:
)
with torch.no_grad():
export_dir = export_utils.export(model=model, data_in=data_list, **kwargs)
export_dir = export_utils.export(model=model, data_in=data_list, **kwargs)
return export_dir

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@ -84,6 +84,8 @@ def main(**kwargs):
dist.init_process_group(backend=kwargs.get("backend", "nccl"), init_method="env://")
torch.cuda.set_device(local_rank)
# rank = dist.get_rank()
logging.info("Build model, frontend, tokenizer")
device = kwargs.get("device", "cuda")
kwargs["device"] = "cpu"
@ -124,6 +126,7 @@ def main(**kwargs):
use_ddp=use_ddp,
use_fsdp=use_fsdp,
device=kwargs["device"],
excludes=kwargs.get("excludes", None),
output_dir=kwargs.get("output_dir", "./exp"),
**kwargs.get("train_conf"),
)
@ -143,7 +146,7 @@ def main(**kwargs):
dataloader = dataloader_class(**kwargs)
# dataloader_tr, dataloader_val = dataloader_class(**kwargs)
scaler = GradScaler(enabled=trainer.use_fp16) if trainer.use_fp16 else None
scaler = GradScaler(enabled=True) if trainer.use_fp16 or trainer.use_bf16 else None
scaler = ShardedGradScaler(enabled=trainer.use_fp16) if trainer.use_fsdp else scaler
trainer.resume_checkpoint(
@ -182,7 +185,7 @@ def main(**kwargs):
time_escaped = (time.perf_counter() - time_slice_i) / 3600.0
logging.info(
f"rank: {local_rank}, "
f"\n\nrank: {local_rank}, "
f"time_escaped_epoch: {time_escaped:.3f} hours, "
f"estimated to finish {dataloader.data_split_num} data_slices, remaining: {dataloader.data_split_num-data_split_i} slices, {(dataloader.data_split_num-data_split_i)*time_escaped:.3f} hours, "
f"epoch: {trainer.max_epoch - epoch} epochs, {((trainer.max_epoch - epoch - 1)*dataloader.data_split_num + dataloader.data_split_num-data_split_i)*time_escaped:.3f} hours\n"
@ -199,7 +202,7 @@ def main(**kwargs):
time2 = time.perf_counter()
time_escaped = (time2 - time1) / 3600.0
logging.info(
f"rank: {local_rank}, "
f"\n\nrank: {local_rank}, "
f"time_escaped_epoch: {time_escaped:.3f} hours, "
f"estimated to finish {trainer.max_epoch} "
f"epoch: {(trainer.max_epoch - epoch) * time_escaped:.3f} hours\n"

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@ -64,6 +64,8 @@ class OpenAIDataset(torch.utils.data.Dataset):
self.max_token_length = kwargs.get("max_token_length", 1024)
self.batch_size_scale_ratio_max = kwargs.get("batch_size_scale_ratio_max", 1.5)
self.batch_size_token_max = kwargs.get("batch_size_token_max", 2500)
self.audio_adaptor_downsample_rate = kwargs.get("audio_adaptor_downsample_rate", 2)
self.audio_encoder_downsample_rate = kwargs.get("audio_encoder_downsample_rate", 4)
def get_source_len(self, index):
item = self.index_ds[index]
@ -136,10 +138,13 @@ class OpenAIDataset(torch.utils.data.Dataset):
speech = speech.permute(0, 2, 1)
# if speech_lengths > self.batch_size:
# continue
if self.audio_encoder_downsample_rate == 4:
olens = 1 + (speech_lengths[0].item() - 3 + 2 * 1) // 2
olens = 1 + (olens - 3 + 2 * 1) // 2
elif self.audio_encoder_downsample_rate == 1:
olens = speech_lengths[0].item()
olens = 1 + (speech_lengths[0].item() - 3 + 2 * 1) // 2
olens = 1 + (olens - 3 + 2 * 1) // 2
sub_token_len = (olens - 1) // 2 + 1
sub_token_len = (olens - 1) // self.audio_adaptor_downsample_rate + 1
sub_token = [0] * sub_token_len
fbank_beg_i = [len(source_ids)]
source_ids += sub_token
@ -222,3 +227,247 @@ class OpenAIDataset(torch.utils.data.Dataset):
break
return outputs
@tables.register("dataset_classes", "OpenAIDatasetMultiTurn")
class OpenAIDatasetMultiTurn(torch.utils.data.Dataset):
"""
SenseVoiceDataset
"""
def __init__(
self,
path,
index_ds: str = None,
frontend=None,
tokenizer=None,
int_pad_value: int = -1,
float_pad_value: float = 0.0,
**kwargs,
):
super().__init__()
index_ds_class = tables.index_ds_classes.get(index_ds)
self.index_ds = index_ds_class(path, **kwargs)
preprocessor_speech = kwargs.get("preprocessor_speech", None)
if preprocessor_speech:
preprocessor_speech_class = tables.preprocessor_classes.get(preprocessor_speech)
preprocessor_speech = preprocessor_speech_class(
**kwargs.get("preprocessor_speech_conf")
)
self.preprocessor_speech = preprocessor_speech
preprocessor_text = kwargs.get("preprocessor_text", None)
if preprocessor_text:
preprocessor_text_class = tables.preprocessor_classes.get(preprocessor_text)
preprocessor_text = preprocessor_text_class(**kwargs.get("preprocessor_text_conf"))
self.preprocessor_text = preprocessor_text
self.frontend = frontend
self.fs = 16000 if frontend is None else frontend.fs
self.data_type = "sound"
self.tokenizer = tokenizer
self.int_pad_value = int_pad_value
self.float_pad_value = float_pad_value
self.sos = kwargs.get("sos", "<|startoftranscript|>")
self.eos = kwargs.get("eos", "<|endoftext|>")
self.batch_size = kwargs.get("batch_size")
self.batch_type = kwargs.get("batch_type")
self.prompt_ids_len = 0
self.retry = kwargs.get("retry", 100)
self.permute = False
from funasr.frontends.whisper_frontend import WhisperFrontend
if isinstance(self.frontend, WhisperFrontend):
self.permute = True
self.pattern = re.compile(r"(<\|startofspeech\|>.*?<\|endofspeech\|>)")
# self.kwargs = kwargs
self.max_token_length = kwargs.get("max_token_length", 1024)
self.batch_size_scale_ratio_max = kwargs.get("batch_size_scale_ratio_max", 1.5)
self.batch_size_token_max = kwargs.get("batch_size_token_max", 2500)
self.multiturn_num_max = kwargs.get("multiturn_num_max", 5)
def get_source_len(self, index):
item = self.index_ds[index]
return self.index_ds.get_source_len(item)
def get_target_len(self, index):
item = self.index_ds[index]
return self.index_ds.get_target_len(item)
def __len__(self):
return len(self.index_ds)
def __getitem__(self, index):
# import pdb
#
# pdb.set_trace()
output = None
for idx in range(self.retry):
badcase_flag = False
if idx == 0:
index_cur = index
else:
index_cur = torch.randint(0, len(self.index_ds), ()).item()
item = self.index_ds[index_cur]
system = item["system"]
user = item["user"]
assistant = item["assistant"]
input_ids, labels, fbank, fbank_lens, fbank_mask, fbank_beg, fake_token_len = (
[],
[],
[],
[],
[],
[],
[],
)
for i, (system_prompt, user_prompt, target_out) in enumerate(
zip(system, user, assistant)
):
if i >= self.multiturn_num_max:
break
if i == 0:
source_input = f"<|im_start|>system\n{system_prompt}<|im_end|>\n<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n"
else:
source_input = (
f"<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n"
)
splits = self.pattern.split(source_input)
source_ids = []
fbank_i = []
fbank_mask_i = []
fake_token_len_i = 0
fbank_beg_i = -1
fbank_lens_i = []
for k, sub_str in enumerate(splits):
if not sub_str.startswith("<|startofspeech|>"):
sub_token = self.tokenizer.encode(sub_str)
source_ids += sub_token
fbank_mask_i += [0] * len(sub_token)
else:
sub_str = sub_str.replace("<|startofspeech|>", "").replace(
"<|endofspeech|>", ""
)
if sub_str.startswith("!"):
try:
data_src = load_audio_text_image_video(sub_str[1:], fs=self.fs)
except Exception as e:
logging.error(
f"Loading wav failed! {str(e)}, {traceback.format_exc()}"
)
badcase_flag = True
continue
speech, speech_lengths = extract_fbank(
data_src,
data_type=self.data_type,
frontend=self.frontend,
is_final=True,
) # speech: [b, T, d]
if self.permute:
speech = speech.permute(0, 2, 1)
# if speech_lengths > self.batch_size:
# continue
olens = 1 + (speech_lengths[0].item() - 3 + 2 * 1) // 2
olens = 1 + (olens - 3 + 2 * 1) // 2
fake_token_len_i = (olens - 1) // 2 + 1
fake_token = [0] * fake_token_len_i
fbank_beg_i = len(source_ids)
source_ids += fake_token
fbank_mask_i += [1] * len(fake_token)
if badcase_flag:
continue
fbank_beg += [fbank_beg_i + len(input_ids)]
fake_token_len += [fake_token_len_i]
source_mask = [-100] * len(source_ids)
target_out = f"{target_out}<|im_end|>"
target_ids = self.tokenizer.encode(target_out)
input_ids += source_ids + target_ids
labels += source_mask + target_ids
fbank.append(speech[0, :, :])
fbank_mask += fbank_mask_i
fbank_lens.append(speech_lengths)
if len(input_ids) > self.max_token_length:
logging.info(
f"input_ids > max_token_length: {len(input_ids)}>{self.max_token_length}, {item}"
)
badcase_flag = True
if badcase_flag:
continue
input_ids = torch.tensor(input_ids, dtype=torch.int64) # [: self.max_token_length]
attention_mask = torch.tensor([1] * len(input_ids), dtype=torch.int32)
labels = torch.tensor(labels, dtype=torch.int64) # [: self.max_token_length]
# fbank = speech[0, :, :]
# fbank_lens = torch.tensor(fbank_lens, dtype=torch.int32)
fbank_mask = torch.tensor(fbank_mask, dtype=torch.float32)
fbank_beg = torch.tensor(fbank_beg, dtype=torch.int32)
fake_token_len = torch.tensor(fake_token_len, dtype=torch.int32)
output = {
"speech": fbank,
"speech_lengths": fbank_lens,
"fbank_mask": fbank_mask,
"fbank_beg": fbank_beg,
"fake_token_len": fake_token_len,
"input_ids": input_ids,
"attention_mask": attention_mask,
"labels_ids": labels,
}
break
return output
def collator(self, samples: list = None):
for idx in range(self.retry):
badcase_flag = False
outputs = {}
for sample in samples:
if sample is None:
continue
for key in sample.keys():
if key not in outputs:
outputs[key] = []
if isinstance(sample[key], (list, tuple)):
outputs[key].extend(sample[key])
else:
outputs[key].append(sample[key])
for key, data_list in outputs.items():
if isinstance(data_list[0], torch.Tensor):
if data_list[0].dtype == torch.int64 or data_list[0].dtype == torch.int32:
pad_value = self.int_pad_value
else:
pad_value = self.float_pad_value
outputs[key] = torch.nn.utils.rnn.pad_sequence(
data_list, batch_first=True, padding_value=pad_value
)
if self.batch_type != "example":
b, t = outputs["input_ids"].shape
if b > 1 and b * t > self.batch_size_token_max:
logging.info(
f"Warning, {idx}th, b*t: {b}*{t}={b * t} > batch_size_sample_max: {self.batch_size_token_max}, drop last data"
)
samples = samples[:-1]
continue
break
return outputs

View File

@ -15,11 +15,6 @@ class OpenAIIndexDSJsonl(torch.utils.data.Dataset): # torch.utils.data.Dataset
def __init__(self, path: str, **kwargs):
super().__init__()
self.max_source_length = kwargs.get("max_source_length", 2048)
self.min_source_length = kwargs.get("min_source_length", 0)
self.max_target_length = kwargs.get("max_target_length", 2048)
self.min_target_length = kwargs.get("min_target_length", 0)
self.max_token_length = kwargs.get("max_token_length", 2200)
is_training = kwargs.get("is_training", True)
if not (path.endswith(".jsonl") or path.endswith(".json")):

View File

@ -56,13 +56,13 @@ def download_from_ms(**kwargs):
config = OmegaConf.load(cfg["config"])
kwargs = OmegaConf.merge(config, cfg)
kwargs["model"] = config["model"]
elif os.path.exists(os.path.join(model_or_path, "config.yaml")) and os.path.exists(
os.path.join(model_or_path, "model.pt")
):
elif os.path.exists(os.path.join(model_or_path, "config.yaml")):
config = OmegaConf.load(os.path.join(model_or_path, "config.yaml"))
kwargs = OmegaConf.merge(config, kwargs)
init_param = os.path.join(model_or_path, "model.pb")
kwargs["init_param"] = init_param
init_param = os.path.join(model_or_path, "model.pt")
if "init_param" not in kwargs or not os.path.exists(kwargs["init_param"]):
kwargs["init_param"] = init_param
assert os.path.exists(kwargs["init_param"]), "init_param does not exist"
if os.path.exists(os.path.join(model_or_path, "tokens.txt")):
kwargs["tokenizer_conf"]["token_list"] = os.path.join(model_or_path, "tokens.txt")
if os.path.exists(os.path.join(model_or_path, "tokens.json")):
@ -122,7 +122,7 @@ def download_from_hf(**kwargs):
):
config = OmegaConf.load(os.path.join(model_or_path, "config.yaml"))
kwargs = OmegaConf.merge(config, kwargs)
init_param = os.path.join(model_or_path, "model.pb")
init_param = os.path.join(model_or_path, "model.pt")
kwargs["init_param"] = init_param
if os.path.exists(os.path.join(model_or_path, "tokens.txt")):
kwargs["tokenizer_conf"]["token_list"] = os.path.join(model_or_path, "tokens.txt")

View File

@ -21,6 +21,8 @@ from funasr.register import tables
from funasr.train_utils.device_funcs import to_device
import traceback
dtype_map = {"bf16": torch.bfloat16, "fp16": torch.float16, "fp32": torch.float32}
@tables.register("model_classes", "LLMASR")
class LLMASR(nn.Module):
@ -394,7 +396,9 @@ class LLMASR2(nn.Module):
# frontend = model.kwargs.get("frontend")
audio_encoder_output_size = model.model.encoder_output_size
audio_encoder = model.model.model.encoder
audio_encoder = (
model.model.model.encoder if hasattr(model.model, "model") else model.model.encoder
)
# self.frontend = frontend
@ -405,38 +409,60 @@ class LLMASR2(nn.Module):
audio_encoder = encoder_class(input_size=input_size, **audio_encoder_conf)
audio_encoder_output_size = audio_encoder.output_size()
freeze = audio_encoder_conf.get("freeze", True)
freeze_layer_num = int(audio_encoder_conf.get("freeze_layer_num", -1))
# if freeze_layer_num > 0:
# freeze_layer_num = range(freeze_layer_num)
if freeze:
for name, param in audio_encoder.named_parameters():
param.requires_grad = False
if freeze_layer_num > 0:
idx = re.search(r"\.\d+\.", name)
if idx is not None:
beg, end = idx.regs[0]
layer_id = int(name[beg + 1 : end - 1])
if layer_id < freeze_layer_num:
param.requires_grad = False
elif "ln_post." not in name:
param.requires_grad = False
else:
param.requires_grad = False
audio_encoder.eval()
self.audio_encoder = audio_encoder
# llm
hub = llm_conf.get("hub", "hf")
self.llm = None
if hub == "hf":
from transformers import AutoModelForCausalLM, AutoTokenizer, AutoConfig
init_param_path = llm_conf.get("init_param_path", "vicuna-7b-v1.5")
from transformers import AutoModelForCausalLM, AutoTokenizer, AutoConfig
model = AutoModelForCausalLM.from_pretrained(
init_param_path,
load_in_8bit=None,
device_map=None,
use_cache=None,
)
freeze = llm_conf.get("freeze", True)
if freeze:
for name, param in model.named_parameters():
param.requires_grad = False
model.eval()
self.llm = model
init_param_path = llm_conf.get("init_param_path", "vicuna-7b-v1.5")
model = AutoModelForCausalLM.from_pretrained(
init_param_path,
load_in_8bit=None,
device_map=None,
use_cache=None,
)
freeze = llm_conf.get("freeze", True)
if freeze:
for name, param in model.named_parameters():
param.requires_grad = False
model.eval()
self.llm_dtype = llm_conf.get("llm_dtype", "fp32")
self.llm = model.to(dtype_map[self.llm_dtype])
llm_dim = model.get_input_embeddings().weight.shape[-1]
# adaptor
adaptor_class = tables.adaptor_classes.get(audio_adaptor)
audio_adaptor_conf["encoder_dim"] = audio_encoder_output_size
audio_adaptor_conf["llm_dim"] = llm_dim
audio_adaptor = adaptor_class(**audio_adaptor_conf)
init_param_path = audio_adaptor_conf.get("init_param_path", None)
if init_param_path is not None:
src_state = torch.load(init_param_path, map_location="cpu")
flag = audio_adaptor.load_state_dict(src_state, strict=False)
logging.info(f"Loading audio_adaptor ckpt: {init_param_path}, status: {flag}")
self.audio_adaptor = audio_adaptor
@ -470,11 +496,12 @@ class LLMASR2(nn.Module):
batch_size, frames, _ = speech.shape
# audio encoder
encoder_out, encoder_out_lens = self.audio_encoder(speech.permute(0, 2, 1), speech_lengths)
with torch.cuda.amp.autocast(enabled=False):
# audio encoder
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
# audio_adaptor
encoder_out, encoder_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
# audio_adaptor
encoder_out, encoder_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
input_ids[input_ids < 0] = 0
inputs_embeds = self.llm.model.get_input_embeddings()(input_ids)
@ -504,12 +531,17 @@ class LLMASR2(nn.Module):
batch_idx, :min_len, :
]
labels_ids[labels_ids == -1] = -100
model_outputs = self.llm(
inputs_embeds=inputs_embeds, attention_mask=attention_mask, labels=labels_ids
)
loss = model_outputs.loss
with torch.cuda.amp.autocast(
enabled=True if self.llm_dtype != "fp32" else False, dtype=dtype_map[self.llm_dtype]
):
labels_ids[labels_ids == -1] = -100
attention_mask[attention_mask < 0] = 0
model_outputs = self.llm(
inputs_embeds=inputs_embeds.to(dtype_map[self.llm_dtype]),
attention_mask=attention_mask,
labels=labels_ids,
)
loss = model_outputs.loss
stats = {}
with torch.no_grad():
@ -532,6 +564,519 @@ class LLMASR2(nn.Module):
loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device)
return loss, stats, weight
def encode(self, speech, speech_lengths):
# audio encoder
encoder_out, encoder_out_lens = self.audio_encoder(speech.permute(0, 2, 1), speech_lengths)
return encoder_out, encoder_out_lens
def data_template(self, data):
system, user, assistant = [], [], []
for i, item in enumerate(data):
role = item["role"]
content = item["content"]
if role == "system":
system.append(content)
elif role == "user":
user.append(content)
elif role == "assistant":
assistant.append(content)
system = system * len(user)
contents = {
"system": system,
"user": user,
"assistant": assistant,
}
return contents
def data_load_speech(self, contents: dict, tokenizer, frontend, meta_data={}, **kwargs):
system = contents["system"]
user = contents["user"]
assistant = contents["assistant"]
pattern = re.compile(r"(<\|startofspeech\|>.*?<\|endofspeech\|>)")
input_ids, labels, source_ids, target_ids, fbank, fbank_lens, fbank_mask, fbank_beg = (
[],
[],
[],
[],
[],
[],
[],
[],
)
for i, (system_prompt, user_prompt, target_out) in enumerate(zip(system, user, assistant)):
source_input = f"<|im_start|>system\n{system_prompt}<|im_end|>\n<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n"
splits = pattern.split(source_input)
source_ids_i = []
fbank_mask_i = []
fbank_beg_i = []
fbank_lens_i = []
# target_ids_i = []
for k, sub_str in enumerate(splits):
if not sub_str.startswith("<|startofspeech|>"):
sub_token = tokenizer.encode(sub_str)
source_ids_i += sub_token
fbank_mask_i += [0] * len(sub_token)
else:
sub_str = sub_str.replace("<|startofspeech|>", "").replace(
"<|endofspeech|>", ""
)
if sub_str.startswith("!"):
try:
time1 = time.perf_counter()
data_src = load_audio_text_image_video(sub_str[1:], fs=frontend.fs)
time2 = time.perf_counter()
meta_data["load_data"] = f"{time2 - time1:0.3f}"
except Exception as e:
logging.error(f"Loading wav failed! {str(e)}, {traceback.format_exc()}")
speech, speech_lengths = extract_fbank(
data_src,
data_type=kwargs.get("data_type", "sound"),
frontend=frontend,
is_final=True,
) # speech: [b, T, d]
time3 = time.perf_counter()
meta_data["extract_feat"] = f"{time3 - time2:0.3f}"
meta_data["batch_data_time"] = (
speech_lengths.sum().item()
* frontend.frame_shift
* frontend.lfr_n
/ 1000
)
if hasattr(frontend, "permute") and not frontend.permute:
# if kwargs.get("permute", True):
speech = speech.permute(0, 2, 1)
if (
kwargs.get("dataset_conf", {}).get("audio_encoder_downsample_rate", 1)
== 4
):
olens = 1 + (speech_lengths[0].item() - 3 + 2 * 1) // 2
olens = 1 + (olens - 3 + 2 * 1) // 2
elif (
kwargs.get("dataset_conf", {}).get("audio_encoder_downsample_rate", 1)
== 1
):
olens = speech_lengths[0].item()
sub_token_len = (olens - 1) // kwargs.get("dataset_conf", {}).get(
"audio_adaptor_downsample_rate", 1
) + 1
sub_token = [0] * sub_token_len
fbank_beg_i = [len(source_ids_i)]
source_ids_i += sub_token
fbank_mask_i += [1] * len(sub_token)
source_mask = [-100] * len(source_ids_i)
target_out = f"{target_out}<|im_end|>"
target_ids = tokenizer.encode(target_out)
input_ids += source_ids_i + target_ids
labels += source_mask + target_ids
fbank_mask += fbank_mask_i
fbank_beg.append(fbank_beg_i)
input_ids = torch.tensor(input_ids, dtype=torch.int64) # [: self.max_token_length]
attention_mask = torch.tensor([1] * len(input_ids), dtype=torch.int32)
labels = torch.tensor(labels, dtype=torch.int64) # [: self.max_token_length]
source_ids = torch.tensor(source_ids_i, dtype=torch.int64)
target_ids = torch.tensor(target_ids, dtype=torch.int64)
fbank = speech[0, :, :]
fbank_lens = speech_lengths
fbank_mask = torch.tensor(fbank_mask, dtype=torch.float32)
fbank_beg = torch.tensor(fbank_beg, dtype=torch.int32)
output = {
"speech": fbank[None, :, :],
"speech_lengths": fbank_lens[:, None],
"fbank_mask": fbank_mask[None, :],
"fbank_beg": fbank_beg[None,],
"input_ids": input_ids[None, :],
"attention_mask": attention_mask[None, :],
"labels_ids": labels[None, :],
"source_ids": source_ids[None, :],
"target_ids": target_ids[None, :],
}
return output
def inference(
self,
data_in,
data_lengths=None,
key: list = None,
tokenizer=None,
frontend=None,
**kwargs,
):
meta_data = {}
prompt = kwargs.get("prompt", None)
if kwargs.get("batch_size", 1) > 1:
raise NotImplementedError("batch decoding is not implemented")
contents = self.data_template(data_in[0])
output = self.data_load_speech(contents, tokenizer, frontend, meta_data=meta_data, **kwargs)
batch = to_device(output, kwargs["device"])
# audio encoder
speech = batch["speech"]
speech_lengths = batch["speech_lengths"][:, 0]
# fp16
if kwargs.get("fp16", False):
speech = speech.to(torch.float16)
elif kwargs.get("bf16", False):
speech = speech.to(torch.bfloat16)
# audio encoder
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
# audio_adaptor
encoder_out, encoder_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
input_ids = batch["input_ids"]
source_ids = batch["source_ids"]
if not kwargs.get("tearchforing", False):
input_ids = source_ids
input_ids[input_ids < 0] = 0
inputs_embeds = self.llm.model.get_input_embeddings()(input_ids)
batch_size, token_num, dims = inputs_embeds.shape
fbank_beg = batch["fbank_beg"]
for batch_idx in range(batch_size):
min_len = encoder_out_lens[batch_idx].item()
fbank_beg_idx = fbank_beg[batch_idx]
inputs_embeds[batch_idx, fbank_beg_idx : fbank_beg_idx + min_len, :] = encoder_out[
batch_idx, :min_len, :
]
llm_dtype = kwargs.get("llm_dtype", "fp32")
if llm_dtype == "fp32":
llm_dtype = "fp16" if kwargs.get("fp16", False) else llm_dtype
llm_dtype = "bf16" if kwargs.get("bf16", False) else llm_dtype
with torch.cuda.amp.autocast(
enabled=True if llm_dtype != "fp32" else False, dtype=dtype_map[llm_dtype]
):
label = contents["assistant"][0]
self.llm = self.llm.to(dtype_map[llm_dtype])
inputs_embeds = inputs_embeds.to(dtype_map[llm_dtype])
if not kwargs.get("tearchforing", False):
generated_ids = self.llm.generate(
inputs_embeds=inputs_embeds, max_new_tokens=kwargs.get("max_length", 512)
)
# generated_ids = [
# output_ids[len(input_id) :]
# for input_id, output_ids in zip(input_ids, generated_ids)
# ]
response = tokenizer.batch_decode(
generated_ids, skip_special_tokens=kwargs.get("skip_special_tokens", True)
)[0]
loss = None
else:
labels_ids = batch["labels_ids"]
labels_ids[labels_ids == -1] = -100
attention_mask = batch.get("attention_mask", None)
# attention_mask = attention_mask.to(dtype_map[llm_dtype])
model_outputs = self.llm(
inputs_embeds=inputs_embeds, attention_mask=attention_mask, labels=labels_ids
)
preds = torch.argmax(model_outputs.logits, -1)[:, source_ids.shape[1] :]
response = tokenizer.batch_decode(
preds,
add_special_tokens=False,
skip_special_tokens=kwargs.get("skip_special_tokens", True),
)[0]
loss = model_outputs.loss.item()
ibest_writer = None
if kwargs.get("output_dir") is not None:
if not hasattr(self, "writer"):
self.writer = DatadirWriter(kwargs.get("output_dir"))
ibest_writer = self.writer[f"{0 + 1}best_recog"]
results = []
response_clean = re.sub("[^\w\s\u3000\u4e00-\u9fff]+", "", response)
result_i = {"key": key[0], "text": response, "text_tn": response_clean, "label": label}
if loss is not None:
result_i["loss"] = loss
results.append(result_i)
if ibest_writer is not None:
ibest_writer["text"][key[0]] = response
ibest_writer["label"][key[0]] = label
ibest_writer["text_tn"][key[0]] = response_clean
return results, meta_data
@tables.register("model_classes", "LLMASR3")
class LLMASR3(LLMASR2):
""" """
def __init__(
self,
*args,
**kwargs,
):
super().__init__(*args, **kwargs)
def encode(self, speech, speech_lengths):
# audio encoder
encoder_out, encoder_out_lens = self.audio_encoder(speech, speech_lengths)
return encoder_out, encoder_out_lens
@tables.register("model_classes", "LLMASR4")
class LLMASR4(nn.Module):
""" """
def __init__(
self,
specaug: str = None,
specaug_conf: dict = None,
normalize: str = None,
normalize_conf: dict = None,
audio_encoder: str = None,
audio_encoder_conf: dict = None,
audio_adaptor: str = None,
audio_adaptor_conf: dict = None,
decoder: str = None,
decoder_conf: dict = None,
ctc: str = None,
ctc_conf: dict = None,
ctc_weight: float = 0.5,
llm: str = None,
llm_conf: dict = None,
input_size: int = 80,
vocab_size: int = -1,
ignore_id: int = -1,
blank_id: int = 0,
sos: int = 1,
eos: int = 2,
lsm_weight: float = 0.0,
length_normalized_loss: bool = False,
report_cer: bool = True,
report_wer: bool = True,
sym_space: str = "<space>",
sym_blank: str = "<blank>",
# extract_feats_in_collect_stats: bool = True,
share_embedding: bool = False,
# preencoder: Optional[AbsPreEncoder] = None,
# postencoder: Optional[AbsPostEncoder] = None,
**kwargs,
):
super().__init__()
# audio encoder
hub = audio_encoder_conf.get("hub", None)
if hub == "ms":
from funasr import AutoModel
model = AutoModel(model=audio_encoder, model_revision="master")
# frontend = model.kwargs.get("frontend")
audio_encoder_output_size = model.model.encoder_output_size
audio_encoder = (
model.model.model.encoder if hasattr(model.model, "model") else model.model.encoder
)
# self.frontend = frontend
elif hub == "hf":
pass
else:
encoder_class = tables.encoder_classes.get(audio_encoder)
audio_encoder = encoder_class(input_size=input_size, **audio_encoder_conf)
audio_encoder_output_size = audio_encoder.output_size()
freeze = audio_encoder_conf.get("freeze", True)
freeze_layer_num = int(audio_encoder_conf.get("freeze_layer_num", -1))
# if freeze_layer_num > 0:
# freeze_layer_num = range(freeze_layer_num)
if freeze:
for name, param in audio_encoder.named_parameters():
if freeze_layer_num > 0:
idx = re.search(r"\.\d+\.", name)
if idx is not None:
beg, end = idx.regs[0]
layer_id = int(name[beg + 1 : end - 1])
if layer_id < freeze_layer_num:
param.requires_grad = False
elif "ln_post." not in name:
param.requires_grad = False
else:
param.requires_grad = False
audio_encoder.eval()
self.audio_encoder = audio_encoder
# llm
self.llm = None
from transformers import AutoModelForCausalLM, AutoTokenizer, AutoConfig
init_param_path = llm_conf.get("init_param_path", "vicuna-7b-v1.5")
model = AutoModelForCausalLM.from_pretrained(
init_param_path,
load_in_8bit=None,
device_map=None,
use_cache=None,
)
freeze = llm_conf.get("freeze", True)
if freeze:
for name, param in model.named_parameters():
param.requires_grad = False
model.eval()
self.llm_dtype = llm_conf.get("llm_dtype", "fp32")
self.llm = model.to(dtype_map[self.llm_dtype])
llm_dim = model.get_input_embeddings().weight.shape[-1]
# adaptor
adaptor_class = tables.adaptor_classes.get(audio_adaptor)
audio_adaptor_conf["encoder_dim"] = audio_encoder_output_size
audio_adaptor_conf["llm_dim"] = llm_dim
audio_adaptor = adaptor_class(**audio_adaptor_conf)
init_param_path = audio_adaptor_conf.get("init_param_path", None)
if init_param_path is not None:
src_state = torch.load(init_param_path, map_location="cpu")
flag = audio_adaptor.load_state_dict(src_state, strict=False)
logging.info(f"Loading audio_adaptor ckpt: {init_param_path}, status: {flag}")
self.audio_adaptor = audio_adaptor
self.error_calculator = None
self.length_normalized_loss = length_normalized_loss
self.beam_search = None
def forward(
self,
speech: torch.Tensor,
speech_lengths: torch.Tensor,
input_ids: torch.Tensor,
attention_mask: torch.Tensor,
labels_ids: torch.Tensor,
fbank_beg: torch.Tensor,
fbank_mask: torch.Tensor,
**kwargs,
) -> Tuple[torch.Tensor, Dict[str, torch.Tensor], torch.Tensor]:
"""Encoder + Decoder + Calc loss
Args:
speech: (Batch, Length, ...)
speech_lengths: (Batch, )
text: (Batch, Length)
text_lengths: (Batch,)
"""
import pdb
pdb.set_trace()
if len(speech_lengths.size()) > 1:
speech_lengths = speech_lengths[:, 0]
batch_size_speech, frames, _ = speech.shape
batch_size, token_num = input_ids.shape
with torch.cuda.amp.autocast(enabled=False):
# audio encoder
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
# audio_adaptor
encoder_out, encoder_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
input_ids[input_ids < 0] = 0
inputs_embeds = self.llm.model.get_input_embeddings()(input_ids)
batch_size, token_num, dims = inputs_embeds.shape
fake_token_len = kwargs.get("fake_token_len")
fake_token_len[fake_token_len < 0] = 0
fbank_beg[fbank_beg < 0] = 0
speech_idx = 0
for batch_idx in range(batch_size):
for turn_id in range(fbank_beg.shape[1]):
fbank_beg_idx = fbank_beg[batch_idx, turn_id].item()
if fbank_beg_idx > 0:
speech_token_len = fake_token_len[batch_idx, turn_id]
speech_token = encoder_out[speech_idx, :speech_token_len, :]
try:
inputs_embeds[
batch_idx, fbank_beg_idx : fbank_beg_idx + speech_token_len, :
] = speech_token
except Exception as e:
logging.error(f"{str(e)}, {traceback.format_exc()}")
logging.info(
f"batch_idx: {batch_idx}, inputs_embeds: {inputs_embeds.shape}, fbank_beg_idx: {fbank_beg_idx}, speech_token_len: {speech_token_len}, encoder_out: {encoder_out.shape}, encoder_out_lens: {encoder_out_lens[speech_idx].item()}"
)
speech_token_len = encoder_out_lens[speech_idx].item()
speech_token = encoder_out[speech_idx, turn_id, :speech_token_len, :]
inputs_embeds[
batch_idx, fbank_beg_idx : fbank_beg_idx + speech_token_len, :
] = speech_token
speech_idx += 1
with torch.cuda.amp.autocast(
enabled=True if self.llm_dtype != "fp32" else False, dtype=dtype_map[self.llm_dtype]
):
labels_ids[labels_ids == -1] = -100
attention_mask[attention_mask < 0] = 0
model_outputs = self.llm(
inputs_embeds=inputs_embeds.to(dtype_map[self.llm_dtype]),
attention_mask=attention_mask,
labels=labels_ids,
)
loss = model_outputs.loss
stats = {}
with torch.no_grad():
preds = torch.argmax(model_outputs.logits, -1)
acc_att = compute_accuracy(preds[:, :-1], labels_ids[:, 1:], ignore_label=-100)
stats["acc"] = acc_att
stats["loss"] = torch.clone(loss.detach())
stats["batch_size"] = batch_size
stats["batch_size_speech"] = batch_size_speech
stats["batch_size_x_frames"] = frames * batch_size_speech
stats["batch_size_real_frames"] = speech_lengths.sum().item()
stats["padding_frames"] = stats["batch_size_x_frames"] - stats["batch_size_real_frames"]
stats["batch_size_x_tokens"] = token_num * batch_size
stats["batch_size_real_tokens"] = attention_mask.sum().item()
stats["padding_tokens"] = stats["batch_size_x_tokens"] - stats["batch_size_real_tokens"]
# force_gatherable: to-device and to-tensor if scalar for DataParallel
if self.length_normalized_loss:
batch_size = int((labels_ids > 0 + 1).sum())
loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device)
return loss, stats, weight
def encode(self, speech, speech_lengths):
# audio encoder
encoder_out, encoder_out_lens = self.audio_encoder(speech.permute(0, 2, 1), speech_lengths)
return encoder_out, encoder_out_lens
def data_template(self, data):
system, user, assistant = [], [], []
for i, item in enumerate(data):
@ -685,11 +1230,10 @@ class LLMASR2(nn.Module):
# fp16
if kwargs.get("fp16", False):
speech = speech.to(torch.float16)
encoder_out_lens = encoder_out_lens.to(torch.float16)
elif kwargs.get("bf16", False):
speech = speech.to(torch.bfloat16)
encoder_out_lens = encoder_out_lens.to(torch.bfloat16)
encoder_out, encoder_out_lens = self.audio_encoder(speech.permute(0, 2, 1), speech_lengths)
# audio encoder
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
# audio_adaptor
encoder_out, encoder_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
@ -712,11 +1256,16 @@ class LLMASR2(nn.Module):
]
llm_dtype = kwargs.get("llm_dtype", "fp32")
dtype_map = {"bf16": torch.bfloat16, "fp16": torch.float16, "fp32": torch.float32}
with torch.cuda.amp.autocast(dtype=dtype_map[llm_dtype]):
if llm_dtype == "fp32":
llm_dtype = "fp16" if kwargs.get("fp16", False) else llm_dtype
llm_dtype = "bf16" if kwargs.get("bf16", False) else llm_dtype
with torch.cuda.amp.autocast(
enabled=True if llm_dtype != "fp32" else False, dtype=dtype_map[llm_dtype]
):
label = contents["assistant"][0]
# self.llm = self.llm.to(dtype_map[llm_dtype])
# inputs_embeds = inputs_embeds.to(dtype_map[llm_dtype])
self.llm = self.llm.to(dtype_map[llm_dtype])
inputs_embeds = inputs_embeds.to(dtype_map[llm_dtype])
if not kwargs.get("tearchforing", False):

View File

@ -494,6 +494,8 @@ class CifPredictorV2Export(torch.nn.Module):
token_num_floor = torch.floor(token_num)
return hidden, alphas, token_num_floor
@torch.jit.script
def cif_v1_export(hidden, alphas, threshold: float):
device = hidden.device
@ -504,7 +506,7 @@ def cif_v1_export(hidden, alphas, threshold: float):
frames = torch.zeros(batch_size, len_time, hidden_size, dtype=dtype, device=device)
fires = torch.zeros(batch_size, len_time, dtype=dtype, device=device)
prefix_sum = torch.cumsum(alphas, dim=1, dtype=torch.float64).to(torch.float32) # cumsum precision degradation cause wrong result in extreme
prefix_sum = torch.cumsum(alphas, dim=1)
prefix_sum_floor = torch.floor(prefix_sum)
dislocation_prefix_sum = torch.roll(prefix_sum, 1, dims=1)
dislocation_prefix_sum_floor = torch.floor(dislocation_prefix_sum)
@ -516,9 +518,7 @@ def cif_v1_export(hidden, alphas, threshold: float):
fires[fire_idxs] = 1
fires = fires + prefix_sum - prefix_sum_floor
prefix_sum_hidden = torch.cumsum(
alphas.unsqueeze(-1).tile((1, 1, hidden_size)) * hidden, dim=1
)
prefix_sum_hidden = torch.cumsum(alphas.unsqueeze(-1).tile((1, 1, hidden_size)) * hidden, dim=1)
frames = prefix_sum_hidden[fire_idxs]
shift_frames = torch.roll(frames, 1, dims=0)
@ -530,26 +530,22 @@ def cif_v1_export(hidden, alphas, threshold: float):
shift_frames[shift_batch_idxs] = 0
remains = fires - torch.floor(fires)
remain_frames = (
remains[fire_idxs].unsqueeze(-1).tile((1, hidden_size)) * hidden[fire_idxs]
)
remain_frames = remains[fire_idxs].unsqueeze(-1).tile((1, hidden_size)) * hidden[fire_idxs]
shift_remain_frames = torch.roll(remain_frames, 1, dims=0)
shift_remain_frames[shift_batch_idxs] = 0
frames = frames - shift_frames + shift_remain_frames - remain_frames
max_label_len = alphas.sum(dim=-1)
max_label_len = torch.floor(max_label_len).max().to(dtype=torch.int64)
max_label_len = batch_len.max()
frame_fires = torch.zeros(
batch_size, max_label_len, hidden_size, dtype=dtype, device=device
)
frame_fires = torch.zeros(batch_size, max_label_len, hidden_size, dtype=dtype, device=device)
indices = torch.arange(max_label_len, device=device).expand(batch_size, -1)
frame_fires_idxs = indices < batch_len.unsqueeze(1)
frame_fires[frame_fires_idxs] = frames
return frame_fires, fires
@torch.jit.script
def cif_export(hidden, alphas, threshold: float):
batch_size, len_time, hidden_size = hidden.size()
@ -671,7 +667,7 @@ def cif_wo_hidden_v1(alphas, threshold, return_fire_idxs=False):
fires = torch.zeros(batch_size, len_time, dtype=dtype, device=device)
prefix_sum = torch.cumsum(alphas, dim=1, dtype=torch.float64).to(torch.float32) # cumsum precision degradation cause wrong result in extreme
prefix_sum = torch.cumsum(alphas, dim=1)
prefix_sum_floor = torch.floor(prefix_sum)
dislocation_prefix_sum = torch.roll(prefix_sum, 1, dims=1)
dislocation_prefix_sum_floor = torch.floor(dislocation_prefix_sum)
@ -693,11 +689,8 @@ def cif_v1(hidden, alphas, threshold):
device = hidden.device
dtype = hidden.dtype
batch_size, len_time, hidden_size = hidden.size()
frames = torch.zeros(batch_size, len_time, hidden_size,
dtype=dtype, device=device)
prefix_sum_hidden = torch.cumsum(
alphas.unsqueeze(-1).tile((1, 1, hidden_size)) * hidden, dim=1
)
frames = torch.zeros(batch_size, len_time, hidden_size, dtype=dtype, device=device)
prefix_sum_hidden = torch.cumsum(alphas.unsqueeze(-1).tile((1, 1, hidden_size)) * hidden, dim=1)
frames = prefix_sum_hidden[fire_idxs]
shift_frames = torch.roll(frames, 1, dims=0)
@ -709,21 +702,16 @@ def cif_v1(hidden, alphas, threshold):
shift_frames[shift_batch_idxs] = 0
remains = fires - torch.floor(fires)
remain_frames = (
remains[fire_idxs].unsqueeze(-1).tile((1,
hidden_size)) * hidden[fire_idxs]
)
remain_frames = remains[fire_idxs].unsqueeze(-1).tile((1, hidden_size)) * hidden[fire_idxs]
shift_remain_frames = torch.roll(remain_frames, 1, dims=0)
shift_remain_frames[shift_batch_idxs] = 0
frames = frames - shift_frames + shift_remain_frames - remain_frames
max_label_len = torch.round(alphas.sum(-1)).int().max() # torch.round to calculate the max length
max_label_len = batch_len.max()
frame_fires = torch.zeros(
batch_size, max_label_len, hidden_size, dtype=dtype, device=device
)
frame_fires = torch.zeros(batch_size, max_label_len, hidden_size, dtype=dtype, device=device)
indices = torch.arange(max_label_len, device=device).expand(batch_size, -1)
frame_fires_idxs = indices < batch_len.unsqueeze(1)
frame_fires[frame_fires_idxs] = frames

View File

@ -16,6 +16,7 @@ from funasr.train_utils.device_funcs import force_gatherable
from . import whisper_lib as whisper
from funasr.utils.load_utils import load_audio_text_image_video, extract_fbank
from funasr.utils.datadir_writer import DatadirWriter
from funasr.models.ctc.ctc import CTC
from funasr.register import tables
@ -1035,6 +1036,7 @@ class SenseVoiceSANM(nn.Module):
self.length_normalized_loss = length_normalized_loss
self.beam_search = None
self.activation_checkpoint = kwargs.get("activation_checkpoint", False)
self.encoder_output_size = encoder_output_size
def forward(
self,
@ -1256,7 +1258,7 @@ class SenseVoiceSANM(nn.Module):
if isinstance(task, str):
task = [task]
task = "".join([f"<|{x}|>" for x in task])
sos = kwargs.get("model_conf").get("sos")
if isinstance(sos, str):
initial_prompt = kwargs.get("initial_prompt", f"<|startoftranscript|>{task}")
@ -1270,7 +1272,9 @@ class SenseVoiceSANM(nn.Module):
language = DecodingOptions.get("language", None)
language = None if language == "auto" else language
initial_prompt = kwargs.get("initial_prompt", f"{task}")
initial_prompt_lid = f"{initial_prompt}<|{language}|>" if language is not None else initial_prompt
initial_prompt_lid = (
f"{initial_prompt}<|{language}|>" if language is not None else initial_prompt
)
initial_prompt_lid_int = tokenizer.encode(initial_prompt_lid, allowed_special="all")
sos_int = [sos] + initial_prompt_lid_int
eos = kwargs.get("model_conf").get("eos")
@ -1303,9 +1307,7 @@ class SenseVoiceSANM(nn.Module):
)
self.beam_search.event_score_ga = DecodingOptions.get("gain_tokens_score", [1, 1, 1, 1])
encoder_out, encoder_out_lens = self.encode(
speech[None, :, :], speech_lengths
)
encoder_out, encoder_out_lens = self.encode(speech[None, :, :], speech_lengths)
if text_token_int is not None:
i = 0
@ -1384,3 +1386,279 @@ class SenseVoiceSANM(nn.Module):
ibest_writer["text"][key[i]] = text
return results, meta_data
from funasr.models.paraformer.search import Hypothesis
from funasr.utils import postprocess_utils
@tables.register("model_classes", "SenseVoiceSANMCTC")
class SenseVoiceSANMCTC(nn.Module):
"""CTC-attention hybrid Encoder-Decoder model"""
def __init__(
self,
specaug: str = None,
specaug_conf: dict = None,
normalize: str = None,
normalize_conf: dict = None,
encoder: str = None,
encoder_conf: dict = None,
ctc_conf: dict = None,
input_size: int = 80,
vocab_size: int = -1,
ignore_id: int = -1,
blank_id: int = 0,
sos: int = 1,
eos: int = 2,
length_normalized_loss: bool = False,
**kwargs,
):
super().__init__()
if specaug is not None:
specaug_class = tables.specaug_classes.get(specaug)
specaug = specaug_class(**specaug_conf)
if normalize is not None:
normalize_class = tables.normalize_classes.get(normalize)
normalize = normalize_class(**normalize_conf)
encoder_class = tables.encoder_classes.get(encoder)
encoder = encoder_class(input_size=input_size, **encoder_conf)
encoder_output_size = encoder.output_size()
if ctc_conf is None:
ctc_conf = {}
ctc = CTC(odim=vocab_size, encoder_output_size=encoder_output_size, **ctc_conf)
self.blank_id = blank_id
self.sos = sos if sos is not None else vocab_size - 1
self.eos = eos if eos is not None else vocab_size - 1
self.vocab_size = vocab_size
self.ignore_id = ignore_id
self.specaug = specaug
self.normalize = normalize
self.encoder = encoder
self.error_calculator = None
self.ctc = ctc
self.length_normalized_loss = length_normalized_loss
self.encoder_output_size = encoder_output_size
self.lid_dict = {"zh": 3, "en": 4, "yue": 7, "ja": 11, "ko": 12, "nospeech": 13}
self.textnorm_dict = {"withtextnorm": 14, "wotextnorm": 15}
self.embed = torch.nn.Embedding(8 + len(self.lid_dict) + len(self.textnorm_dict), 560)
def forward(
self,
speech: torch.Tensor,
speech_lengths: torch.Tensor,
text: torch.Tensor,
text_lengths: torch.Tensor,
**kwargs,
):
"""Encoder + Decoder + Calc loss
Args:
speech: (Batch, Length, ...)
speech_lengths: (Batch, )
text: (Batch, Length)
text_lengths: (Batch,)
"""
# import pdb;
# pdb.set_trace()
if len(text_lengths.size()) > 1:
text_lengths = text_lengths[:, 0]
if len(speech_lengths.size()) > 1:
speech_lengths = speech_lengths[:, 0]
batch_size = speech.shape[0]
# 1. Encoder
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
loss_ctc, cer_ctc = None, None
stats = dict()
loss_ctc, cer_ctc = self._calc_ctc_loss(encoder_out, encoder_out_lens, text, text_lengths)
loss = loss_ctc
# Collect total loss stats
stats["loss"] = torch.clone(loss.detach())
# force_gatherable: to-device and to-tensor if scalar for DataParallel
if self.length_normalized_loss:
batch_size = int((text_lengths + 1).sum())
loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device)
return loss, stats, weight
def encode(
self,
speech: torch.Tensor,
speech_lengths: torch.Tensor,
**kwargs,
):
"""Frontend + Encoder. Note that this method is used by asr_inference.py
Args:
speech: (Batch, Length, ...)
speech_lengths: (Batch, )
ind: int
"""
# Data augmentation
if self.specaug is not None and self.training:
speech, speech_lengths = self.specaug(speech, speech_lengths)
# Normalization for feature: e.g. Global-CMVN, Utterance-CMVN
if self.normalize is not None:
speech, speech_lengths = self.normalize(speech, speech_lengths)
# Forward encoder
# feats: (Batch, Length, Dim)
# -> encoder_out: (Batch, Length2, Dim2)
encoder_out, encoder_out_lens = self.encoder(speech, speech_lengths)
return encoder_out, encoder_out_lens
def _calc_ctc_loss(
self,
encoder_out: torch.Tensor,
encoder_out_lens: torch.Tensor,
ys_pad: torch.Tensor,
ys_pad_lens: torch.Tensor,
):
# Calc CTC loss
loss_ctc = self.ctc(encoder_out, encoder_out_lens, ys_pad, ys_pad_lens)
# Calc CER using CTC
cer_ctc = None
if not self.training and self.error_calculator is not None:
ys_hat = self.ctc.argmax(encoder_out).data
cer_ctc = self.error_calculator(ys_hat.cpu(), ys_pad.cpu(), is_ctc=True)
return loss_ctc, cer_ctc
def inference(
self,
data_in,
data_lengths=None,
key: list = None,
tokenizer=None,
frontend=None,
**kwargs,
):
if kwargs.get("batch_size", 1) > 1:
raise NotImplementedError("batch decoding is not implemented")
meta_data = {}
if (
isinstance(data_in, torch.Tensor) and kwargs.get("data_type", "sound") == "fbank"
): # fbank
speech, speech_lengths = data_in, data_lengths
if len(speech.shape) < 3:
speech = speech[None, :, :]
if speech_lengths is None:
speech_lengths = speech.shape[1]
else:
# extract fbank feats
time1 = time.perf_counter()
audio_sample_list = load_audio_text_image_video(
data_in,
fs=frontend.fs,
audio_fs=kwargs.get("fs", 16000),
data_type=kwargs.get("data_type", "sound"),
tokenizer=tokenizer,
)
time2 = time.perf_counter()
meta_data["load_data"] = f"{time2 - time1:0.3f}"
speech, speech_lengths = extract_fbank(
audio_sample_list, data_type=kwargs.get("data_type", "sound"), frontend=frontend
)
time3 = time.perf_counter()
meta_data["extract_feat"] = f"{time3 - time2:0.3f}"
meta_data["batch_data_time"] = (
speech_lengths.sum().item() * frontend.frame_shift * frontend.lfr_n / 1000
)
speech = speech.to(device=kwargs["device"])
speech_lengths = speech_lengths.to(device=kwargs["device"])
language = kwargs.get("language", None)
if language is not None:
language_query = self.embed(
torch.LongTensor(
[[self.lid_dict[language] if language in self.lid_dict else 0]]
).to(speech.device)
).repeat(speech.size(0), 1, 1)
else:
language_query = self.embed(torch.LongTensor([[0]]).to(speech.device)).repeat(
speech.size(0), 1, 1
)
textnorm = kwargs.get("text_norm", "wotextnorm")
textnorm_query = self.embed(
torch.LongTensor([[self.textnorm_dict[textnorm]]]).to(speech.device)
).repeat(speech.size(0), 1, 1)
speech = torch.cat((textnorm_query, speech), dim=1)
speech_lengths += 1
event_emo_query = self.embed(torch.LongTensor([[1, 2]]).to(speech.device)).repeat(
speech.size(0), 1, 1
)
input_query = torch.cat((language_query, event_emo_query), dim=1)
speech = torch.cat((input_query, speech), dim=1)
speech_lengths += 3
# Encoder
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
if isinstance(encoder_out, tuple):
encoder_out = encoder_out[0]
# c. Passed the encoder result and the beam search
ctc_logits = self.ctc.log_softmax(encoder_out)
results = []
b, n, d = encoder_out.size()
if isinstance(key[0], (list, tuple)):
key = key[0]
if len(key) < b:
key = key * b
for i in range(b):
x = ctc_logits[i, : encoder_out_lens[i], :]
yseq = x.argmax(dim=-1)
yseq = torch.unique_consecutive(yseq, dim=-1)
yseq = torch.tensor([self.sos] + yseq.tolist() + [self.eos], device=yseq.device)
nbest_hyps = [Hypothesis(yseq=yseq)]
for nbest_idx, hyp in enumerate(nbest_hyps):
ibest_writer = None
if kwargs.get("output_dir") is not None:
if not hasattr(self, "writer"):
self.writer = DatadirWriter(kwargs.get("output_dir"))
ibest_writer = self.writer[f"{nbest_idx + 1}best_recog"]
# remove sos/eos and get results
last_pos = -1
if isinstance(hyp.yseq, list):
token_int = hyp.yseq[1:last_pos]
else:
token_int = hyp.yseq[1:last_pos].tolist()
# remove blank symbol id, which is assumed to be 0
token_int = list(
filter(
lambda x: x != self.eos and x != self.sos and x != self.blank_id, token_int
)
)
# Change integer-ids to tokens
text = tokenizer.decode(token_int)
result_i = {"key": key[i], "text": text}
results.append(result_i)
if ibest_writer is not None:
ibest_writer["token"][key[i]] = " ".join(token)
ibest_writer["text"][key[i]] = text_postprocessed
return results, meta_data

View File

@ -10,36 +10,6 @@ import torch.optim
import pdb
def filter_state_dict(
dst_state: Dict[str, Union[float, torch.Tensor]],
src_state: Dict[str, Union[float, torch.Tensor]],
):
"""Filter name, size mismatch instances between dicts.
Args:
dst_state: reference state dict for filtering
src_state: target state dict for filtering
"""
match_state = {}
for key, value in src_state.items():
if key in dst_state and (dst_state[key].size() == src_state[key].size()):
match_state[key] = value
else:
if key not in dst_state:
logging.warning(
f"Filter out {key} from pretrained dict"
+ " because of name not found in target dict"
)
else:
logging.warning(
f"Filter out {key} from pretrained dict"
+ " because of size mismatch"
+ f"({dst_state[key].size()}-{src_state[key].size()})"
)
return match_state
def load_pretrained_model(
path: str,
model: torch.nn.Module,
@ -62,7 +32,7 @@ def load_pretrained_model(
obj = model
dst_state = obj.state_dict()
print(f"ckpt: {path}")
logging.info(f"ckpt: {path}")
if oss_bucket is None:
src_state = torch.load(path, map_location=map_location)
@ -77,8 +47,24 @@ def load_pretrained_model(
if isinstance(scope_map, str):
scope_map = scope_map.split(",")
scope_map += ["module.", "None"]
logging.info(f"scope_map: {scope_map}")
if excludes is not None:
if isinstance(excludes, str):
excludes = excludes.split(",")
logging.info(f"excludes: {excludes}")
for k in dst_state.keys():
excludes_flag = False
if excludes is not None:
for k_ex in excludes:
if k.startswith(k_ex):
logging.info(f"key: {k} matching: {k_ex}, excluded")
excludes_flag = True
break
if excludes_flag:
continue
k_src = k
@ -92,25 +78,25 @@ def load_pretrained_model(
if dst_prefix == "" and (src_prefix + k) in src_state.keys():
k_src = src_prefix + k
if not k_src.startswith("module."):
print(f"init param, map: {k} from {k_src} in ckpt")
logging.info(f"init param, map: {k} from {k_src} in ckpt")
elif (
k.startswith(dst_prefix)
and k.replace(dst_prefix, src_prefix, 1) in src_state.keys()
):
k_src = k.replace(dst_prefix, src_prefix, 1)
if not k_src.startswith("module."):
print(f"init param, map: {k} from {k_src} in ckpt")
logging.info(f"init param, map: {k} from {k_src} in ckpt")
if k_src in src_state.keys():
if ignore_init_mismatch and dst_state[k].shape != src_state[k_src].shape:
print(
logging.info(
f"ignore_init_mismatch:{ignore_init_mismatch}, dst: {k, dst_state[k].shape}, src: {k_src, src_state[k_src].shape}"
)
else:
dst_state[k] = src_state[k_src]
else:
print(f"Warning, miss key in ckpt: {k}, mapped: {k_src}")
print(f"Warning, miss key in ckpt: {k}, {path}")
flag = obj.load_state_dict(dst_state, strict=True)
# print(flag)
logging.info(f"Loading ckpt: {path}, status: {flag}")

View File

@ -29,9 +29,10 @@ def maybe_autocast(dtype=None, use_deepspeed=False):
with torch.cuda.amp.autocast(enabled=True, dtype=dtype, cache_enabled=False):
yield
else:
if dtype == torch.float16:
with autocast(enabled=True):
yield
if dtype == torch.float16 or dtype == torch.bfloat16:
yield
# with autocast(enabled=True, dtype=dtype):
# yield
else:
yield
@ -60,6 +61,7 @@ class Trainer:
use_ddp: bool = False,
use_fsdp: bool = False,
use_fp16: bool = False,
use_bf16: bool = False,
use_deepspeed: bool = False,
output_dir: str = "./",
**kwargs,
@ -78,7 +80,7 @@ class Trainer:
output_dir (str): The directory where model checkpoints will be saved. Default is './'.
resume (str, optional): The file path to a checkpoint to resume training from.
"""
self.rank = kwargs.get("rank", 0)
self.rank = rank
self.local_rank = local_rank
self.world_size = world_size
self.use_ddp = use_ddp
@ -98,8 +100,11 @@ class Trainer:
self.batch_total = 0
self.dtype = torch.float32
self.use_fp16 = use_fp16
self.use_bf16 = use_bf16
if self.use_fp16:
self.dtype = torch.float16
if self.use_bf16:
self.dtype = torch.bfloat16
self.save_checkpoint_interval = kwargs.get("save_checkpoint_interval", 5000)
self.validate_interval = kwargs.get("validate_interval", 5000)
self.keep_nbest_models = kwargs.get("keep_nbest_models", 500)
@ -147,6 +152,16 @@ class Trainer:
self.use_deepspeed = use_deepspeed
self.deepspeed_config = kwargs.get("deepspeed_config", "")
excludes = kwargs.get("excludes", None)
if excludes is not None:
if isinstance(excludes, str):
excludes = excludes.split(",")
self.excludes = excludes
effective_save_name_excludes = kwargs.get("effective_save_name_excludes", None)
if effective_save_name_excludes is not None:
if isinstance(effective_save_name_excludes, str):
effective_save_name_excludes = effective_save_name_excludes.split(",")
self.effective_save_name_excludes = effective_save_name_excludes
def save_checkpoint(
self,
@ -277,11 +292,12 @@ class Trainer:
elif self.use_fsdp:
pass
elif self.rank == 0:
logging.info(f"Save checkpoint: {epoch}, rank: {self.local_rank}\n")
logging.info(
f"Save checkpoint: {epoch}, rank: {self.rank}, local_rank: {self.local_rank}\n"
)
# self.step_or_epoch += 1
state = {
"epoch": epoch,
"state_dict": model.state_dict(),
"optimizer": optim.state_dict(),
"scheduler": scheduler.state_dict(),
"saved_ckpts": self.saved_ckpts,
@ -299,7 +315,24 @@ class Trainer:
}
step = step_in_epoch
if hasattr(model, "module"):
state["state_dict"] = model.module.state_dict()
state_dict = model.module.state_dict()
else:
state_dict = model.state_dict()
if self.effective_save_name_excludes is not None:
logging.info(f"effective_save_name_excludes: {self.effective_save_name_excludes}")
dst_state_dict = {}
for k in state_dict.keys():
for k_ex in self.effective_save_name_excludes:
k_tmp = k.replace("module.", "")
if k.startswith(k_ex):
logging.info(f"key: {k} matching: {k_ex}, not save it")
break
else:
dst_state_dict[k] = state_dict[k]
state["state_dict"] = dst_state_dict
else:
state["state_dict"] = state_dict
if scaler:
state["scaler_state"] = scaler.state_dict()
@ -440,6 +473,16 @@ class Trainer:
src_state = checkpoint["state_dict"]
dst_state = model.state_dict()
for k in dst_state.keys():
excludes_flag = False
if self.excludes is not None:
for k_ex in self.excludes:
k_tmp = k.replace("module.", "")
if k_tmp.startswith(k_ex):
logging.info(f"key: {k} matching: {k_ex}, excluded")
excludes_flag = True
break
if excludes_flag:
continue
if not k.startswith("module.") and "module." + k in src_state.keys():
k_ddp = "module." + k
elif k.startswith("module.") and "module." + k not in src_state.keys():
@ -640,7 +683,7 @@ class Trainer:
scaled_loss = model.backward(loss)
else:
loss = loss / self.accum_grad
if self.use_fp16:
if self.use_fp16 or self.use_bf16:
scaler.scale(loss).backward()
else:
loss.backward()
@ -668,7 +711,7 @@ class Trainer:
# Execute an optimization step (update model parameters)
if self.use_ddp or self.use_fsdp:
dist.barrier()
if self.use_fp16:
if self.use_fp16 or self.use_bf16:
scaler.step(optim)
scaler.update()
else: