merge master ,加入speaker change locating pipeline

Link: https://code.alibaba-inc.com/Ali-MaaS/MaaS-lib/codereview/12601179
This commit is contained in:
tongmu.wh
2023-05-12 17:53:56 +08:00
committed by xingjun.wxj
parent 1cd67ef060
commit ca0cf2b5cf
5 changed files with 457 additions and 7 deletions

View File

@@ -182,6 +182,7 @@ class Models(object):
generic_sv = 'generic-sv'
ecapa_tdnn_sv = 'ecapa-tdnn-sv'
campplus_sv = 'cam++-sv'
scl_sd = 'scl-sd'
rdino_tdnn_sv = 'rdino_ecapa-tdnn-sv'
generic_lm = 'generic-lm'
@@ -481,6 +482,7 @@ class Pipelines(object):
vad_inference = 'vad-inference'
speaker_verification = 'speaker-verification'
speaker_verification_rdino = 'speaker-verification-rdino'
speaker_change_locating = 'speaker-change-locating'
lm_inference = 'language-score-prediction'
speech_timestamp_inference = 'speech-timestamp-inference'

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@@ -76,11 +76,13 @@ class CAMPPlus(nn.Module):
bn_size=4,
init_channels=128,
config_str='batchnorm-relu',
memory_efficient=True):
memory_efficient=True,
output_level='segment'):
super(CAMPPlus, self).__init__()
self.head = FCM(feat_dim=feat_dim)
channels = self.head.out_channels
self.output_level = output_level
self.xvector = nn.Sequential(
OrderedDict([
@@ -118,10 +120,14 @@ class CAMPPlus(nn.Module):
self.xvector.add_module('out_nonlinear',
get_nonlinear(config_str, channels))
self.xvector.add_module('stats', StatsPool())
self.xvector.add_module(
'dense',
DenseLayer(channels * 2, embedding_size, config_str='batchnorm_'))
if self.output_level == 'segment':
self.xvector.add_module('stats', StatsPool())
self.xvector.add_module(
'dense',
DenseLayer(
channels * 2, embedding_size, config_str='batchnorm_'))
else:
assert self.output_level == 'frame', '`output_level` should be set to \'segment\' or \'frame\'. '
for m in self.modules():
if isinstance(m, (nn.Conv1d, nn.Linear)):
@@ -133,6 +139,8 @@ class CAMPPlus(nn.Module):
x = x.permute(0, 2, 1) # (B,T,F) => (B,F,T)
x = self.head(x)
x = self.xvector(x)
if self.output_level == 'frame':
x = x.transpose(1, 2)
return x

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@@ -0,0 +1,319 @@
# Copyright (c) Alibaba, Inc. and its affiliates.
import os
from collections import OrderedDict
from typing import Any, Dict, Union
import numpy as np
import torch
import torch.nn as nn
import torch.nn.functional as F
import torchaudio.compliance.kaldi as Kaldi
from modelscope.metainfo import Models
from modelscope.models import MODELS, TorchModel
from modelscope.models.audio.sv.DTDNN import CAMPPlus
from modelscope.utils.constant import Tasks
class MultiHeadSelfAttention(nn.Module):
def __init__(self, n_units, h=8, dropout=0.1):
super(MultiHeadSelfAttention, self).__init__()
self.linearQ = nn.Linear(n_units, n_units)
self.linearK = nn.Linear(n_units, n_units)
self.linearV = nn.Linear(n_units, n_units)
self.linearO = nn.Linear(n_units, n_units)
self.d_k = n_units // h
self.h = h
self.dropout = nn.Dropout(p=dropout)
self.att = None
def forward(self, x, batch_size):
# x: (BT, F)
q = self.linearQ(x).reshape(batch_size, -1, self.h, self.d_k)
k = self.linearK(x).reshape(batch_size, -1, self.h, self.d_k)
v = self.linearV(x).reshape(batch_size, -1, self.h, self.d_k)
scores = torch.matmul(q.transpose(1, 2), k.permute(
0, 2, 3, 1)) / np.sqrt(self.d_k)
# scores: (B, h, T, T)
self.att = F.softmax(scores, dim=3)
p_att = self.dropout(self.att)
# v : (B, T, h, d_k)
# p_att : (B, h, T, T)
x = torch.matmul(p_att, v.transpose(1, 2))
# x : (B, h, T, d_k)
x = x.transpose(1, 2).reshape(-1, self.h * self.d_k)
return self.linearO(x)
class PositionwiseFeedForward(nn.Module):
def __init__(self, n_units, d_units, dropout):
super(PositionwiseFeedForward, self).__init__()
self.linear1 = nn.Linear(n_units, d_units)
self.linear2 = nn.Linear(d_units, n_units)
self.dropout = nn.Dropout(p=dropout)
def forward(self, x):
return self.linear2(self.dropout(F.relu(self.linear1(x))))
class PosEncoding(nn.Module):
def __init__(self, max_seq_len, d_word_vec):
super(PosEncoding, self).__init__()
pos_enc = np.array([[
pos / np.power(10000, 2.0 * (j // 2) / d_word_vec)
for j in range(d_word_vec)
] for pos in range(max_seq_len)])
pos_enc[:, 0::2] = np.sin(pos_enc[:, 0::2])
pos_enc[:, 1::2] = np.cos(pos_enc[:, 1::2])
pad_row = np.zeros([1, d_word_vec])
pos_enc = np.concatenate([pad_row, pos_enc]).astype(np.float32)
self.pos_enc = torch.nn.Embedding(max_seq_len + 1, d_word_vec)
self.pos_enc.weight = torch.nn.Parameter(
torch.from_numpy(pos_enc), requires_grad=False)
def forward(self, input_len):
max_len = torch.max(input_len)
input_pos = torch.LongTensor([
list(range(1, len + 1)) + [0] * (max_len - len)
for len in input_len
])
return self.pos_enc(input_pos)
class TransformerEncoder(nn.Module):
def __init__(self,
idim,
n_units=256,
n_layers=2,
e_units=512,
h=4,
dropout=0.1):
super(TransformerEncoder, self).__init__()
self.linear_in = nn.Linear(idim, n_units)
self.lnorm_in = nn.LayerNorm(n_units)
self.n_layers = n_layers
self.dropout = nn.Dropout(p=dropout)
for i in range(n_layers):
setattr(self, '{}{:d}'.format('lnorm1_', i), nn.LayerNorm(n_units))
setattr(self, '{}{:d}'.format('self_att_', i),
MultiHeadSelfAttention(n_units, h))
setattr(self, '{}{:d}'.format('lnorm2_', i), nn.LayerNorm(n_units))
setattr(self, '{}{:d}'.format('ff_', i),
PositionwiseFeedForward(n_units, e_units, dropout))
self.lnorm_out = nn.LayerNorm(n_units)
def forward(self, x):
# x: [B, num_anchors, T, n_in]
bs, num, tframe, dim = x.size()
x = x.reshape(bs * num, tframe, -1) # [B*num_anchors, T, dim]
# x: (B, T, F) ... batch, time, (mel)freq
B_size, T_size, _ = x.shape
# e: (BT, F)
e = self.linear_in(x.reshape(B_size * T_size, -1))
# Encoder stack
for i in range(self.n_layers):
# layer normalization
e = getattr(self, '{}{:d}'.format('lnorm1_', i))(e)
# self-attention
s = getattr(self, '{}{:d}'.format('self_att_', i))(e, x.shape[0])
# residual
e = e + self.dropout(s)
# layer normalization
e = getattr(self, '{}{:d}'.format('lnorm2_', i))(e)
# positionwise feed-forward
s = getattr(self, '{}{:d}'.format('ff_', i))(e)
# residual
e = e + self.dropout(s)
# final layer normalization
# output: (BT, F)
# output: (B, F, T)
output = self.lnorm_out(e).reshape(B_size, T_size, -1)
output = output.reshape(bs, num, tframe,
-1) # [B, num_anchors, T, dim]
return output
class TransformerEncoder_out(nn.Module):
def __init__(self,
idim,
n_units=256,
n_layers=2,
e_units=512,
h=4,
dropout=0.1):
super(TransformerEncoder_out, self).__init__()
self.linear_in = nn.Linear(idim, n_units)
self.lnorm_in = nn.LayerNorm(n_units)
self.n_layers = n_layers
self.dropout = nn.Dropout(p=dropout)
for i in range(n_layers):
setattr(self, '{}{:d}'.format('lnorm1_', i), nn.LayerNorm(n_units))
setattr(self, '{}{:d}'.format('self_att_', i),
MultiHeadSelfAttention(n_units, h))
setattr(self, '{}{:d}'.format('lnorm2_', i), nn.LayerNorm(n_units))
setattr(self, '{}{:d}'.format('ff_', i),
PositionwiseFeedForward(n_units, e_units, dropout))
self.lnorm_out = nn.LayerNorm(n_units)
def forward(self, x):
# x: (B, T, F)
B_size, T_size, _ = x.shape
# e: (BT, F)
e = self.linear_in(x.reshape(B_size * T_size, -1))
# Encoder stack
for i in range(self.n_layers):
# layer normalization
e = getattr(self, '{}{:d}'.format('lnorm1_', i))(e)
# self-attention
s = getattr(self, '{}{:d}'.format('self_att_', i))(e, x.shape[0])
# residual
e = e + self.dropout(s)
# layer normalization
e = getattr(self, '{}{:d}'.format('lnorm2_', i))(e)
# positionwise feed-forward
s = getattr(self, '{}{:d}'.format('ff_', i))(e)
# residual
e = e + self.dropout(s)
# final layer normalization
# output: (BT, F)
# output: (B, T, F)
output = self.lnorm_out(e).reshape(B_size, T_size, -1)
return output
class OutLayer(nn.Module):
def __init__(self, n_units=256, num_anchors=2):
super(OutLayer, self).__init__()
self.combine = TransformerEncoder_out(num_anchors * n_units, n_units)
self.out_linear = nn.Linear(n_units // num_anchors, 1)
def forward(self, input):
# input: [B, num_anchors, T, dim]
bs, num, tframe, dim = input.size()
output = input.permute(0, 2, 1,
3).reshape(bs, tframe,
-1) # [Bs, t, num_anchors*dim]
output = self.combine(output) # [Bs, t, n_units]
output = output.reshape(
bs, tframe, num, -1) # [Bs, t, num_anchors, n_units//num_anchors]
output = self.out_linear(output).squeeze(-1) # [Bs, t, num_anchors]
return output
class TransformerDetector(nn.Module):
def __init__(self,
frame_dim=512,
anchor_dim=192,
hidden_dim=256,
max_seq_len=1000):
super(TransformerDetector, self).__init__()
self.detection = TransformerEncoder(
idim=frame_dim + anchor_dim, n_units=hidden_dim)
self.output = OutLayer(n_units=hidden_dim)
self.pos_enc = PosEncoding(max_seq_len, hidden_dim)
def forward(self, feats, anchors):
# feats: [1, t, fdim]
num_frames = feats.shape[1]
num_anchors = anchors.shape[1]
bs = feats.shape[0]
feats = feats.unsqueeze(1).repeat(
1, num_anchors, 1, 1) # shape: [Bs, num_anchors, t, fdim]
anchors = anchors.unsqueeze(2).repeat(
1, 1, num_frames, 1) # shape: [Bs, num_anchors, t, xdim]
sd_in = torch.cat((feats, anchors),
dim=-1) # shape: [Bs, num_anchors, t, fdim+xdim]
sd_out = self.detection(sd_in) # shape: [Bs, num_anchors, t, sd_dim]
# pos
pos_emb = self.pos_enc(torch.tensor([num_frames] * (bs * num_anchors)))
pos_emb = pos_emb.reshape(bs, num_anchors, num_frames, -1)
sd_out += pos_emb
# output
output = self.output(sd_out) # shape: [Bs, t, num_anchors]
return output
@MODELS.register_module(Tasks.speaker_diarization, module_name=Models.scl_sd)
class SpeakerChangeLocatorTransformer(TorchModel):
r"""A speaekr change locator using the transformer architecture as the backbone.
Args:
model_dir: A model dir.
model_config: The model config.
"""
def __init__(self, model_dir, model_config: Dict[str, Any], *args,
**kwargs):
super().__init__(model_dir, model_config, *args, **kwargs)
self.model_config = model_config
self.feature_dim = self.model_config['fbank_dim']
frame_size = self.model_config['frame_size']
anchor_size = self.model_config['anchor_size']
self.encoder = CAMPPlus(self.feature_dim, output_level='frame')
self.backend = TransformerDetector(
frame_dim=frame_size, anchor_dim=anchor_size)
pretrained_encoder = kwargs['pretrained_encoder']
pretrained_backend = kwargs['pretrained_backend']
self.__load_check_point(pretrained_encoder, pretrained_backend)
self.encoder.eval()
self.backend.eval()
def forward(self, audio, anchors):
assert len(audio.shape) == 2 and audio.shape[
0] == 1, 'modelscope error: the shape of input audio to model needs to be [1, T]'
assert len(
anchors.shape
) == 3 and anchors.shape[0] == 1 and anchors.shape[
1] == 2, 'modelscope error: the shape of input anchors to model needs to be [1, 2, D]'
# audio shape: [1, T]
feature = self.__extract_feature(audio)
frame_state = self.encoder(feature)
output = self.backend(frame_state, anchors)
output = output.squeeze(0).detach().cpu().sigmoid()
time_scale_factor = int(np.ceil(feature.shape[1] / output.shape[0]))
output = output.unsqueeze(1).expand(-1, time_scale_factor,
-1).reshape(-1, output.shape[-1])
return output
def __extract_feature(self, audio):
feature = Kaldi.fbank(audio, num_mel_bins=self.feature_dim)
feature = feature - feature.mean(dim=0, keepdim=True)
feature = feature.unsqueeze(0)
return feature
def __load_check_point(self,
pretrained_encoder,
pretrained_backend,
device=None):
if not device:
device = torch.device('cpu')
self.encoder.load_state_dict(
torch.load(
os.path.join(self.model_dir, pretrained_encoder),
map_location=device))
self.backend.load_state_dict(
torch.load(
os.path.join(self.model_dir, pretrained_backend),
map_location=device))

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@@ -0,0 +1,105 @@
# Copyright (c) Alibaba, Inc. and its affiliates.
import io
from typing import Any, Dict, List, Union
import numpy as np
import soundfile as sf
import torch
from modelscope.fileio import File
from modelscope.metainfo import Pipelines
from modelscope.outputs import OutputKeys
from modelscope.pipelines.base import InputModel, Pipeline
from modelscope.pipelines.builder import PIPELINES
from modelscope.utils.constant import Tasks
from modelscope.utils.logger import get_logger
logger = get_logger()
__all__ = ['SpeakerChangeLocatingPipeline']
@PIPELINES.register_module(
Tasks.speaker_diarization, module_name=Pipelines.speaker_change_locating)
class SpeakerChangeLocatingPipeline(Pipeline):
"""Speaker Change Locating Inference Pipeline
use `model` to create a speaker change Locating pipeline.
Args:
model (SpeakerChangeLocatingPipeline): A model instance, or a model local dir, or a model id in the model hub.
kwargs (dict, `optional`):
Extra kwargs passed into the pipeline's constructor.
Example:
>>> from modelscope.pipelines import pipeline
>>> from modelscope.utils.constant import Tasks
>>> p = pipeline(
>>> task=Tasks.speaker_diarization, model='damo/speech_campplus-transformer_scl_zh-cn_16k-common')
>>> print(p(audio))
"""
def __init__(self, model: InputModel, **kwargs):
"""use `model` to create a speaker change Locating pipeline for prediction
Args:
model (str): a valid offical model id
"""
super().__init__(model=model, **kwargs)
self.model_config = self.model.model_config
self.config = self.model.model_config
self.anchor_size = self.config['anchor_size']
def __call__(self, audio: str, embds: List = None) -> Dict[str, Any]:
if embds is not None:
assert len(embds) == 2
assert isinstance(embds[0], np.ndarray) and isinstance(
embds[1], np.ndarray)
assert embds[0].shape == (
self.anchor_size, ) and embds[1].shape == (self.anchor_size, )
else:
embd1 = np.zeros(self.anchor_size // 2)
embd2 = np.ones(self.anchor_size - self.anchor_size // 2)
embd3 = np.ones(self.anchor_size // 2)
embd4 = np.zeros(self.anchor_size - self.anchor_size // 2)
embds = [
np.stack([embd1, embd2], axis=1).flatten(),
np.stack([embd3, embd4], axis=1).flatten(),
]
anchors = torch.from_numpy(np.stack(embds,
axis=0)).float().unsqueeze(0)
output = self.preprocess(audio)
output = self.forward(output, anchors)
output = self.postprocess(output)
return output
def forward(self, input: torch.Tensor, anchors: torch.Tensor):
output = self.model(input, anchors)
return output
def postprocess(self, input: torch.Tensor) -> Dict[str, Any]:
predict = np.where(np.diff(input.argmax(-1).numpy()))
try:
predict = predict[0][0] * 0.01 + 0.02
predict = round(predict, 2)
return {OutputKeys.TEXT: f'The change point is at {predict}s.'}
except Exception:
return {OutputKeys.TEXT: 'No change point is found.'}
def preprocess(self, input: str) -> torch.Tensor:
if isinstance(input, str):
file_bytes = File.read(input)
data, fs = sf.read(io.BytesIO(file_bytes), dtype='float32')
if len(data.shape) == 2:
data = data[:, 0]
if fs != self.model_config['sample_rate']:
raise ValueError(
'modelscope error: Only support %d sample rate files'
% self.model_cfg['sample_rate'])
data = torch.from_numpy(data).unsqueeze(0)
else:
raise ValueError(
'modelscope error: The input type is restricted to audio file address'
% i)
return data

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@@ -2,7 +2,7 @@
import os.path
import unittest
from typing import Any, Dict, List
from typing import Any, Dict, List, Union
from modelscope.outputs import OutputKeys
from modelscope.pipelines import pipeline
@@ -16,20 +16,25 @@ logger = get_logger()
SPEAKER1_A_EN_16K_WAV = 'data/test/audios/speaker1_a_en_16k.wav'
SPEAKER1_B_EN_16K_WAV = 'data/test/audios/speaker1_b_en_16k.wav'
SPEAKER2_A_EN_16K_WAV = 'data/test/audios/speaker2_a_en_16k.wav'
SCL_EXAMPLE_WAV = 'data/test/audios/scl_example1.wav'
class SpeakerVerificationTest(unittest.TestCase, DemoCompatibilityCheck):
ecapatdnn_voxceleb_16k_model_id = 'damo/speech_ecapa-tdnn_sv_en_voxceleb_16k'
campplus_voxceleb_16k_model_id = 'damo/speech_campplus_sv_en_voxceleb_16k'
rdino_voxceleb_16k_model_id = 'damo/speech_rdino_ecapa_tdnn_sv_en_voxceleb_16k'
speaker_change_locating_cn_model_id = 'damo/speech_campplus-transformer_scl_zh-cn_16k-common'
def setUp(self) -> None:
self.task = Tasks.speaker_verification
def run_pipeline(self,
model_id: str,
audios: List[str],
audios: Union[List[str], str],
task: str = None,
model_revision=None) -> Dict[str, Any]:
if task is not None:
self.task = task
p = pipeline(
task=self.task, model=model_id, model_revision=model_revision)
result = p(audios)
@@ -66,6 +71,17 @@ class SpeakerVerificationTest(unittest.TestCase, DemoCompatibilityCheck):
print(result)
self.assertTrue(OutputKeys.SCORE in result)
@unittest.skipUnless(test_level() >= 0, 'skip test in current test level')
def test_run_with_speaker_change_locating_cn_16k(self):
logger.info(
'Run speaker change locating for campplus-transformer model')
result = self.run_pipeline(
model_id=self.speaker_change_locating_cn_model_id,
task=Tasks.speaker_diarization,
audios=SCL_EXAMPLE_WAV)
print(result)
self.assertTrue(OutputKeys.TEXT in result)
@unittest.skip('demo compatibility test is only enabled on a needed-basis')
def test_demo_compatibility(self):
self.compatibility_check()