cherrypick change optimizer and fix import error. (#3040)

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zxcd 1 year ago committed by GitHub
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@ -180,6 +180,7 @@ Via the easy-to-use, efficient, flexible and scalable implementation, our vision
### Recent Update
- 👑 2023.03.09: Add [Wav2vec2ASR-zh](./examples/aishell/asr3).
- 🎉 2023.03.07: Add [TTS ARM Linux C++ Demo](./demos/TTSArmLinux).
- 🔥 2023.03.03 Add Voice Conversion [StarGANv2-VC synthesize pipeline](./examples/vctk/vc3).
- 🎉 2023.02.16: Add [Cantonese TTS](./examples/canton/tts3).
- 🔥 2023.01.10: Add [code-switch asr CLI and Demos](./demos/speech_recognition).
- 👑 2023.01.06: Add [code-switch asr tal_cs recipe](./examples/tal_cs/asr1/).

@ -183,8 +183,9 @@
- 🧩 级联模型应用: 作为传统语音任务的扩展,我们结合了自然语言处理、计算机视觉等任务,实现更接近实际需求的产业级应用。
### 近期更新
- 👑 2023.03.09: 新增 [Wav2vec2ASR-zh](./examples/aishell/asr3).
- 👑 2023.03.09: 新增 [Wav2vec2ASR-zh](./examples/aishell/asr3)
- 🎉 2023.03.07: 新增 [TTS ARM Linux C++ 部署示例](./demos/TTSArmLinux)。
- 🔥 2023.03.03: 新增声音转换模型 [StarGANv2-VC 合成流程](./examples/vctk/vc3)。
- 🎉 2023.02.16: 新增[粤语语音合成](./examples/canton/tts3)。
- 🔥 2023.01.10: 新增[中英混合 ASR CLI 和 Demos](./demos/speech_recognition)。
- 👑 2023.01.06: 新增 [ASR 中英混合 tal_cs 训练推理流程](./examples/tal_cs/asr1/)。

@ -190,9 +190,9 @@ tar xzvf wav2vec2ASR-large-aishell1_ckpt_1.4.0.model.tar.gz
```
You can download the audio demo:
```bash
wget -nc https://paddlespeech.bj.bcebos.com/datasets/single_wav/en/demo_002_en.wav -P data/
wget -nc https://paddlespeech.bj.bcebos.com/datasets/single_wav/zh/demo_01_03.wav -P data/
```
You need to prepare an audio file or use the audio demo above, please confirm the sample rate of the audio is 16K. You can get the result of the audio demo by running the script below.
```bash
CUDA_VISIBLE_DEVICES= ./local/test_wav.sh conf/wav2vec2ASR.yaml conf/tuning/decode.yaml exp/wav2vec2ASR/checkpoints/avg_1 data/demo_002_en.wav
CUDA_VISIBLE_DEVICES= ./local/test_wav.sh conf/wav2vec2ASR.yaml conf/tuning/decode.yaml exp/wav2vec2ASR/checkpoints/avg_1 data/demo_01_03.wav
```

@ -107,6 +107,7 @@ vocab_filepath: data/lang_char/vocab.txt
###########################################
unit_type: 'char'
tokenizer: bert-base-chinese
mean_std_filepath:
preprocess_config: conf/preprocess.yaml
sortagrad: -1 # Feed samples from shortest to longest ; -1: enabled for all epochs, 0: disabled, other: enabled for 'other' epochs
@ -139,12 +140,10 @@ n_epoch: 80
accum_grad: 1
global_grad_clip: 5.0
model_optim: adadelta
model_optim: sgd
model_optim_conf:
lr: 1.0
weight_decay: 0.0
rho: 0.95
epsilon: 1.0e-8
wav2vec2_optim: adam
wav2vec2_optim_conf:
@ -165,3 +164,4 @@ log_interval: 1
checkpoint:
kbest_n: 50
latest_n: 5

@ -0,0 +1,168 @@
############################################
# Network Architecture #
############################################
freeze_wav2vec2: False
normalize_wav: True
output_norm: True
init_type: 'kaiming_uniform' # !Warning: need to convergence
enc:
input_shape: 1024
dnn_blocks: 3
dnn_neurons: 1024
activation: True
normalization: True
dropout_rate: [0.15, 0.15, 0.0]
ctc:
enc_n_units: 1024
blank_id: 0
dropout_rate: 0.0
audio_augment:
speeds: [90, 100, 110]
spec_augment:
time_warp: True
time_warp_window: 5
time_warp_mode: bicubic
freq_mask: True
n_freq_mask: 2
time_mask: True
n_time_mask: 2
replace_with_zero: False
freq_mask_width: 30
time_mask_width: 40
wav2vec2_params_path: exp/wav2vec2/chinese-wav2vec2-large.pdparams
############################################
# Wav2Vec2.0 #
############################################
# vocab_size: 1000000
hidden_size: 1024
num_hidden_layers: 24
num_attention_heads: 16
intermediate_size: 4096
hidden_act: gelu
hidden_dropout: 0.1
activation_dropout: 0.0
attention_dropout: 0.1
feat_proj_dropout: 0.1
feat_quantizer_dropout: 0.0
final_dropout: 0.0
layerdrop: 0.1
initializer_range: 0.02
layer_norm_eps: 1e-5
feat_extract_norm: layer
feat_extract_activation: gelu
conv_dim: [512, 512, 512, 512, 512, 512, 512]
conv_stride: [5, 2, 2, 2, 2, 2, 2]
conv_kernel: [10, 3, 3, 3, 3, 2, 2]
conv_bias: True
num_conv_pos_embeddings: 128
num_conv_pos_embedding_groups: 16
do_stable_layer_norm: True
apply_spec_augment: False
mask_channel_length: 10
mask_channel_min_space: 1
mask_channel_other: 0.0
mask_channel_prob: 0.0
mask_channel_selection: static
mask_feature_length: 10
mask_feature_min_masks: 0
mask_feature_prob: 0.0
mask_time_length: 10
mask_time_min_masks: 2
mask_time_min_space: 1
mask_time_other: 0.0
mask_time_prob: 0.075
mask_time_selection: static
num_codevectors_per_group: 320
num_codevector_groups: 2
contrastive_logits_temperature: 0.1
num_negatives: 100
codevector_dim: 256
proj_codevector_dim: 256
diversity_loss_weight: 0.1
use_weighted_layer_sum: False
# pad_token_id: 0
# bos_token_id: 1
# eos_token_id: 2
add_adapter: False
adapter_kernel_size: 3
adapter_stride: 2
num_adapter_layers: 3
output_hidden_size: None
###########################################
# Data #
###########################################
train_manifest: data/manifest.train
dev_manifest: data/manifest.dev
test_manifest: data/manifest.test
vocab_filepath: data/lang_char/vocab.txt
###########################################
# Dataloader #
###########################################
unit_type: 'char'
tokenizer: bert-base-chinese
mean_std_filepath:
preprocess_config: conf/preprocess.yaml
sortagrad: -1 # Feed samples from shortest to longest ; -1: enabled for all epochs, 0: disabled, other: enabled for 'other' epochs
batch_size: 5 # Different batch_size may cause large differences in results
maxlen_in: 51200000000 # if input length > maxlen-in batchsize is automatically reduced
maxlen_out: 1500000 # if output length > maxlen-out batchsize is automatically reduced
minibatches: 0 # for debug
batch_count: auto
batch_bins: 0
batch_frames_in: 0
batch_frames_out: 0
batch_frames_inout: 0
num_workers: 6
subsampling_factor: 1
num_encs: 1
dist_sampler: True
shortest_first: True
return_lens_rate: True
###########################################
# use speechbrain dataloader #
###########################################
use_sb_pipeline: True # whether use speechbrain pipeline. Default is True.
sb_pipeline_conf: conf/train_with_wav2vec.yaml
###########################################
# Training #
###########################################
n_epoch: 80
accum_grad: 1
global_grad_clip: 5.0
model_optim: adadelta
model_optim_conf:
lr: 1.0
weight_decay: 0.0
rho: 0.95
epsilon: 1.0e-8
wav2vec2_optim: adam
wav2vec2_optim_conf:
lr: 0.0001
weight_decay: 0.0
model_scheduler: newbobscheduler
model_scheduler_conf:
improvement_threshold: 0.0025
annealing_factor: 0.8
patient: 0
wav2vec2_scheduler: newbobscheduler
wav2vec2_scheduler_conf:
improvement_threshold: 0.0025
annealing_factor: 0.9
patient: 0
log_interval: 1
checkpoint:
kbest_n: 50
latest_n: 5

@ -8,9 +8,7 @@ echo "using $ngpu gpus..."
expdir=exp
datadir=data
train_set=train_960
recog_set="test-clean test-other dev-clean dev-other"
recog_set="test-clean"
train_set=train
config_path=$1
decode_config_path=$2
@ -75,7 +73,7 @@ for type in ctc_prefix_beam_search; do
--trans_hyp ${ckpt_prefix}.${type}.rsl.text
python3 utils/compute-wer.py --char=1 --v=1 \
data/manifest.test-clean.text ${ckpt_prefix}.${type}.rsl.text > ${ckpt_prefix}.${type}.error
data/manifest.test.text ${ckpt_prefix}.${type}.rsl.text > ${ckpt_prefix}.${type}.error
echo "decoding ${type} done."
done

@ -14,7 +14,7 @@ ckpt_prefix=$3
audio_file=$4
mkdir -p data
wget -nc https://paddlespeech.bj.bcebos.com/datasets/single_wav/en/demo_002_en.wav -P data/
wget -nc https://paddlespeech.bj.bcebos.com/datasets/single_wav/zh/demo_01_03.wav -P data/
if [ $? -ne 0 ]; then
exit 1
fi

@ -15,11 +15,11 @@ resume= # xx e.g. 30
export FLAGS_cudnn_deterministic=1
. ${MAIN_ROOT}/utils/parse_options.sh || exit 1;
audio_file=data/demo_002_en.wav
audio_file=data/demo_01_03.wav
avg_ckpt=avg_${avg_num}
ckpt=$(basename ${conf_path} | awk -F'.' '{print $1}')
echo "checkpoint name ${ckpt}"git revert -v
echo "checkpoint name ${ckpt}"
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
# prepare data

@ -102,13 +102,11 @@ ssl_dynamic_pretrained_models = {
'params':
'exp/wav2vec2ASR/checkpoints/avg_1.pdparams',
},
},
"wav2vec2ASR_aishell1-zh-16k": {
'1.4': {
'url':
'https://paddlespeech.bj.bcebos.com/s2t/aishell/asr3/wav2vec2ASR-large-aishell1_ckpt_1.4.0.model.tar.gz',
'md5':
'9f0bc943adb822789bf61e674b229d17',
'150e51b8ea5d255ccce6b395de8d916a',
'cfg_path':
'model.yaml',
'ckpt_path':

@ -18,13 +18,13 @@ from pathlib import Path
import paddle
import soundfile
from yacs.config import CfgNode
from paddlenlp.transformers import AutoTokenizer
from paddlespeech.s2t.frontend.featurizer.text_featurizer import TextFeaturizer
from paddlespeech.s2t.models.wav2vec2.wav2vec2_ASR import Wav2vec2ASR
from paddlespeech.s2t.training.cli import default_argument_parser
from paddlespeech.s2t.utils.log import Log
from paddlespeech.s2t.utils.utility import UpdateConfig
from yacs.config import CfgNode
logger = Log(__name__).getlog()
@ -34,8 +34,13 @@ class Wav2vec2Infer():
self.config = config
self.audio_file = args.audio_file
self.text_feature = TextFeaturizer(
unit_type=config.unit_type, vocab=config.vocab_filepath)
if self.config.tokenizer:
self.text_feature = AutoTokenizer.from_pretrained(
self.config.tokenizer)
else:
self.text_feature = TextFeaturizer(
unit_type=config.unit_type, vocab=config.vocab_filepath)
paddle.set_device('gpu' if self.args.ngpu > 0 else 'cpu')
# model
@ -59,14 +64,14 @@ class Wav2vec2Infer():
audio, _ = soundfile.read(
self.audio_file, dtype="int16", always_2d=True)
logger.info(f"audio shape: {audio.shape}")
xs = paddle.to_tensor(audio, dtype='float32').unsqueeze(axis=0)
decode_config = self.config.decode
result_transcripts, result_tokenids = self.model.decode(
xs,
text_feature=self.text_feature,
decoding_method=decode_config.decoding_method,
beam_size=decode_config.beam_size)
beam_size=decode_config.beam_size,
tokenizer=self.config.tokenizer, )
rsl = result_transcripts[0]
utt = Path(self.audio_file).name
logger.info(f"hyp: {utt} {rsl}")

@ -27,8 +27,6 @@ from paddle import inference
from paddle import jit
from paddle.io import DataLoader
from paddle.static import InputSpec
from yacs.config import CfgNode
from paddlespeech.t2s.datasets.am_batch_fn import *
from paddlespeech.t2s.datasets.data_table import DataTable
from paddlespeech.t2s.datasets.vocoder_batch_fn import Clip_static
@ -38,6 +36,7 @@ from paddlespeech.t2s.frontend.mix_frontend import MixFrontend
from paddlespeech.t2s.frontend.zh_frontend import Frontend
from paddlespeech.t2s.modules.normalizer import ZScore
from paddlespeech.utils.dynamic_import import dynamic_import
from yacs.config import CfgNode
# remove [W:onnxruntime: xxx] from ort
ort.set_default_logger_severity(3)
@ -490,6 +489,7 @@ def get_predictor(
device: str='cpu',
# for gpu
use_trt: bool=False,
device_id: int=0,
# for trt
use_dynamic_shape: bool=True,
min_subgraph_size: int=5,
@ -505,6 +505,7 @@ def get_predictor(
params_file (os.PathLike): name of params_file.
device (str): Choose the device you want to run, it can be: cpu/gpu, default is cpu.
use_trt (bool): whether to use TensorRT or not in GPU.
device_id (int): Choose your device id, only valid when the device is gpu, default 0.
use_dynamic_shape (bool): use dynamic shape or not in TensorRT.
use_mkldnn (bool): whether to use MKLDNN or not in CPU.
cpu_threads (int): num of thread when use CPU.
@ -521,7 +522,7 @@ def get_predictor(
config.enable_memory_optim()
config.switch_ir_optim(True)
if device == "gpu":
config.enable_use_gpu(100, 0)
config.enable_use_gpu(100, device_id)
else:
config.disable_gpu()
config.set_cpu_math_library_num_threads(cpu_threads)

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