Toggle navigation
Toggle navigation
此项目
正在载入...
Sign in
xuning
/
sherpaonnx
转到一个项目
Toggle navigation
项目
群组
代码片段
帮助
Toggle navigation pinning
Project
Activity
Repository
Pipelines
Graphs
Issues
0
Merge Requests
0
Wiki
Network
Create a new issue
Builds
Commits
Authored by
Fangjun Kuang
2024-01-15 21:40:30 +0800
Browse Files
Options
Browse Files
Download
Email Patches
Plain Diff
Committed by
GitHub
2024-01-15 21:40:30 +0800
Commit
59e28518b45323dab2da0acd570b088c19a9e329
59e28518
1 parent
7e0ae677
Add Python API examples for speaker recognition with VAD and ASR. (#532)
隐藏空白字符变更
内嵌
并排对比
正在显示
3 个修改的文件
包含
750 行增加
和
4 行删除
python-api-examples/speaker-identification-with-vad-non-streaming-asr.py
python-api-examples/speaker-identification-with-vad.py
python-api-examples/speaker-identification.py
python-api-examples/speaker-identification-with-vad-non-streaming-asr.py
0 → 100755
查看文件 @
59e2851
#!/usr/bin/env python3
"""
This script shows how to use Python APIs for speaker identification with
a microphone, a VAD model, and a non-streaming ASR model.
Please see also ./generate-subtitles.py
Usage:
(1) Prepare a text file containing speaker related files.
Each line in the text file contains two columns. The first column is the
speaker name, while the second column contains the wave file of the speaker.
If the text file contains multiple wave files for the same speaker, then the
embeddings of these files are averaged.
An example text file is given below:
foo /path/to/a.wav
bar /path/to/b.wav
foo /path/to/c.wav
foobar /path/to/d.wav
Each wave file should contain only a single channel; the sample format
should be int16_t; the sample rate can be arbitrary.
(2) Download a model for computing speaker embeddings
Please visit
https://github.com/k2-fsa/sherpa-onnx/releases/tag/speaker-recongition-models
to download a model. An example is given below:
wget https://github.com/k2-fsa/sherpa-onnx/releases/download/speaker-recongition-models/wespeaker_zh_cnceleb_resnet34.onnx
Note that `zh` means Chinese, while `en` means English.
(3) Download the VAD model
Please visit
https://github.com/snakers4/silero-vad/blob/master/files/silero_vad.onnx
to download silero_vad.onnx
For instance,
wget https://github.com/snakers4/silero-vad/raw/master/files/silero_vad.onnx
(4) Please refer to ./generate-subtitles.py
to download a non-streaming ASR model.
(5) Run this script
Assume the filename of the text file is speaker.txt.
python3 ./python-api-examples/speaker-identification-with-vad.py
\
--silero-vad-model=/path/to/silero_vad.onnx
\
--speaker-file ./speaker.txt
\
--model ./wespeaker_zh_cnceleb_resnet34.onnx
"""
import
argparse
import
sys
from
collections
import
defaultdict
from
pathlib
import
Path
from
typing
import
Dict
,
List
,
Tuple
import
numpy
as
np
import
sherpa_onnx
import
torchaudio
try
:
import
sounddevice
as
sd
except
ImportError
:
print
(
"Please install sounddevice first. You can use"
)
print
()
print
(
" pip install sounddevice"
)
print
()
print
(
"to install it"
)
sys
.
exit
(
-
1
)
g_sample_rate
=
16000
def
register_non_streaming_asr_model_args
(
parser
):
parser
.
add_argument
(
"--tokens"
,
type
=
str
,
help
=
"Path to tokens.txt"
,
)
parser
.
add_argument
(
"--encoder"
,
default
=
""
,
type
=
str
,
help
=
"Path to the transducer encoder model"
,
)
parser
.
add_argument
(
"--decoder"
,
default
=
""
,
type
=
str
,
help
=
"Path to the transducer decoder model"
,
)
parser
.
add_argument
(
"--joiner"
,
default
=
""
,
type
=
str
,
help
=
"Path to the transducer joiner model"
,
)
parser
.
add_argument
(
"--paraformer"
,
default
=
""
,
type
=
str
,
help
=
"Path to the model.onnx from Paraformer"
,
)
parser
.
add_argument
(
"--wenet-ctc"
,
default
=
""
,
type
=
str
,
help
=
"Path to the CTC model.onnx from WeNet"
,
)
parser
.
add_argument
(
"--whisper-encoder"
,
default
=
""
,
type
=
str
,
help
=
"Path to whisper encoder model"
,
)
parser
.
add_argument
(
"--whisper-decoder"
,
default
=
""
,
type
=
str
,
help
=
"Path to whisper decoder model"
,
)
parser
.
add_argument
(
"--whisper-language"
,
default
=
""
,
type
=
str
,
help
=
"""It specifies the spoken language in the input file.
Example values: en, fr, de, zh, jp.
Available languages for multilingual models can be found at
https://github.com/openai/whisper/blob/main/whisper/tokenizer.py#L10
If not specified, we infer the language from the input audio file.
"""
,
)
parser
.
add_argument
(
"--whisper-task"
,
default
=
"transcribe"
,
choices
=
[
"transcribe"
,
"translate"
],
type
=
str
,
help
=
"""For multilingual models, if you specify translate, the output
will be in English.
"""
,
)
parser
.
add_argument
(
"--whisper-tail-paddings"
,
default
=-
1
,
type
=
int
,
help
=
"""Number of tail padding frames.
We have removed the 30-second constraint from whisper, so you need to
choose the amount of tail padding frames by yourself.
Use -1 to use a default value for tail padding.
"""
,
)
parser
.
add_argument
(
"--decoding-method"
,
type
=
str
,
default
=
"greedy_search"
,
help
=
"""Valid values are greedy_search and modified_beam_search.
modified_beam_search is valid only for transducer models.
"""
,
)
parser
.
add_argument
(
"--feature-dim"
,
type
=
int
,
default
=
80
,
help
=
"Feature dimension. Must match the one expected by the model"
,
)
def
get_args
():
parser
=
argparse
.
ArgumentParser
(
formatter_class
=
argparse
.
ArgumentDefaultsHelpFormatter
)
register_non_streaming_asr_model_args
(
parser
)
parser
.
add_argument
(
"--speaker-file"
,
type
=
str
,
required
=
True
,
help
=
"""Path to the speaker file. Read the help doc at the beginning of this
file for the format."""
,
)
parser
.
add_argument
(
"--model"
,
type
=
str
,
required
=
True
,
help
=
"Path to the speaker embedding model file."
,
)
parser
.
add_argument
(
"--silero-vad-model"
,
type
=
str
,
required
=
True
,
help
=
"Path to silero_vad.onnx"
,
)
parser
.
add_argument
(
"--threshold"
,
type
=
float
,
default
=
0.6
)
parser
.
add_argument
(
"--num-threads"
,
type
=
int
,
default
=
1
,
help
=
"Number of threads for neural network computation"
,
)
parser
.
add_argument
(
"--debug"
,
type
=
bool
,
default
=
False
,
help
=
"True to show debug messages"
,
)
parser
.
add_argument
(
"--provider"
,
type
=
str
,
default
=
"cpu"
,
help
=
"Valid values: cpu, cuda, coreml"
,
)
return
parser
.
parse_args
()
def
assert_file_exists
(
filename
:
str
):
assert
Path
(
filename
)
.
is_file
(),
(
f
"{filename} does not exist!
\n
"
"Please refer to "
"https://k2-fsa.github.io/sherpa/onnx/pretrained_models/index.html to download it"
)
def
create_recognizer
(
args
)
->
sherpa_onnx
.
OfflineRecognizer
:
if
args
.
encoder
:
assert
len
(
args
.
paraformer
)
==
0
,
args
.
paraformer
assert
len
(
args
.
wenet_ctc
)
==
0
,
args
.
wenet_ctc
assert
len
(
args
.
whisper_encoder
)
==
0
,
args
.
whisper_encoder
assert
len
(
args
.
whisper_decoder
)
==
0
,
args
.
whisper_decoder
assert_file_exists
(
args
.
encoder
)
assert_file_exists
(
args
.
decoder
)
assert_file_exists
(
args
.
joiner
)
recognizer
=
sherpa_onnx
.
OfflineRecognizer
.
from_transducer
(
encoder
=
args
.
encoder
,
decoder
=
args
.
decoder
,
joiner
=
args
.
joiner
,
tokens
=
args
.
tokens
,
num_threads
=
args
.
num_threads
,
sample_rate
=
args
.
sample_rate
,
feature_dim
=
args
.
feature_dim
,
decoding_method
=
args
.
decoding_method
,
debug
=
args
.
debug
,
)
elif
args
.
paraformer
:
assert
len
(
args
.
wenet_ctc
)
==
0
,
args
.
wenet_ctc
assert
len
(
args
.
whisper_encoder
)
==
0
,
args
.
whisper_encoder
assert
len
(
args
.
whisper_decoder
)
==
0
,
args
.
whisper_decoder
assert_file_exists
(
args
.
paraformer
)
recognizer
=
sherpa_onnx
.
OfflineRecognizer
.
from_paraformer
(
paraformer
=
args
.
paraformer
,
tokens
=
args
.
tokens
,
num_threads
=
args
.
num_threads
,
sample_rate
=
g_sample_rate
,
feature_dim
=
args
.
feature_dim
,
decoding_method
=
args
.
decoding_method
,
debug
=
args
.
debug
,
)
elif
args
.
wenet_ctc
:
assert
len
(
args
.
whisper_encoder
)
==
0
,
args
.
whisper_encoder
assert
len
(
args
.
whisper_decoder
)
==
0
,
args
.
whisper_decoder
assert_file_exists
(
args
.
wenet_ctc
)
recognizer
=
sherpa_onnx
.
OfflineRecognizer
.
from_wenet_ctc
(
model
=
args
.
wenet_ctc
,
tokens
=
args
.
tokens
,
num_threads
=
args
.
num_threads
,
sample_rate
=
args
.
sample_rate
,
feature_dim
=
args
.
feature_dim
,
decoding_method
=
args
.
decoding_method
,
debug
=
args
.
debug
,
)
elif
args
.
whisper_encoder
:
assert_file_exists
(
args
.
whisper_encoder
)
assert_file_exists
(
args
.
whisper_decoder
)
recognizer
=
sherpa_onnx
.
OfflineRecognizer
.
from_whisper
(
encoder
=
args
.
whisper_encoder
,
decoder
=
args
.
whisper_decoder
,
tokens
=
args
.
tokens
,
num_threads
=
args
.
num_threads
,
decoding_method
=
args
.
decoding_method
,
debug
=
args
.
debug
,
language
=
args
.
whisper_language
,
task
=
args
.
whisper_task
,
tail_paddings
=
args
.
whisper_tail_paddings
,
)
else
:
raise
ValueError
(
"Please specify at least one model"
)
return
recognizer
def
load_speaker_embedding_model
(
args
):
config
=
sherpa_onnx
.
SpeakerEmbeddingExtractorConfig
(
model
=
args
.
model
,
num_threads
=
args
.
num_threads
,
debug
=
args
.
debug
,
provider
=
args
.
provider
,
)
if
not
config
.
validate
():
raise
ValueError
(
f
"Invalid config. {config}"
)
extractor
=
sherpa_onnx
.
SpeakerEmbeddingExtractor
(
config
)
return
extractor
def
load_speaker_file
(
args
)
->
Dict
[
str
,
List
[
str
]]:
if
not
Path
(
args
.
speaker_file
)
.
is_file
():
raise
ValueError
(
f
"--speaker-file {args.speaker_file} does not exist"
)
ans
=
defaultdict
(
list
)
with
open
(
args
.
speaker_file
)
as
f
:
for
line
in
f
:
line
=
line
.
strip
()
if
not
line
:
continue
fields
=
line
.
split
()
if
len
(
fields
)
!=
2
:
raise
ValueError
(
f
"Invalid line: {line}. Fields: {fields}"
)
speaker_name
,
filename
=
fields
ans
[
speaker_name
]
.
append
(
filename
)
return
ans
def
load_audio
(
filename
:
str
)
->
Tuple
[
np
.
ndarray
,
int
]:
samples
,
sample_rate
=
torchaudio
.
load
(
filename
)
return
samples
[
0
]
.
contiguous
()
.
numpy
(),
sample_rate
def
compute_speaker_embedding
(
filenames
:
List
[
str
],
extractor
:
sherpa_onnx
.
SpeakerEmbeddingExtractor
,
)
->
np
.
ndarray
:
assert
len
(
filenames
)
>
0
,
"filenames is empty"
ans
=
None
for
filename
in
filenames
:
print
(
f
"processing {filename}"
)
samples
,
sample_rate
=
load_audio
(
filename
)
stream
=
extractor
.
create_stream
()
stream
.
accept_waveform
(
sample_rate
=
sample_rate
,
waveform
=
samples
)
stream
.
input_finished
()
assert
extractor
.
is_ready
(
stream
)
embedding
=
extractor
.
compute
(
stream
)
embedding
=
np
.
array
(
embedding
)
if
ans
is
None
:
ans
=
embedding
else
:
ans
+=
embedding
return
ans
/
len
(
filenames
)
def
main
():
args
=
get_args
()
print
(
args
)
recognizer
=
create_recognizer
(
args
)
extractor
=
load_speaker_embedding_model
(
args
)
speaker_file
=
load_speaker_file
(
args
)
manager
=
sherpa_onnx
.
SpeakerEmbeddingManager
(
extractor
.
dim
)
for
name
,
filename_list
in
speaker_file
.
items
():
embedding
=
compute_speaker_embedding
(
filenames
=
filename_list
,
extractor
=
extractor
,
)
status
=
manager
.
add
(
name
,
embedding
)
if
not
status
:
raise
RuntimeError
(
f
"Failed to register speaker {name}"
)
vad_config
=
sherpa_onnx
.
VadModelConfig
()
vad_config
.
silero_vad
.
model
=
args
.
silero_vad_model
vad_config
.
silero_vad
.
min_silence_duration
=
0.25
vad_config
.
silero_vad
.
min_speech_duration
=
0.25
vad_config
.
sample_rate
=
g_sample_rate
window_size
=
vad_config
.
silero_vad
.
window_size
vad
=
sherpa_onnx
.
VoiceActivityDetector
(
vad_config
,
buffer_size_in_seconds
=
100
)
samples_per_read
=
int
(
0.1
*
g_sample_rate
)
# 0.1 second = 100 ms
devices
=
sd
.
query_devices
()
if
len
(
devices
)
==
0
:
print
(
"No microphone devices found"
)
sys
.
exit
(
0
)
print
(
devices
)
default_input_device_idx
=
sd
.
default
.
device
[
0
]
print
(
f
'Use default device: {devices[default_input_device_idx]["name"]}'
)
print
(
"Started! Please speak"
)
idx
=
0
buffer
=
[]
with
sd
.
InputStream
(
channels
=
1
,
dtype
=
"float32"
,
samplerate
=
g_sample_rate
)
as
s
:
while
True
:
samples
,
_
=
s
.
read
(
samples_per_read
)
# a blocking read
samples
=
samples
.
reshape
(
-
1
)
buffer
=
np
.
concatenate
([
buffer
,
samples
])
while
len
(
buffer
)
>
window_size
:
vad
.
accept_waveform
(
buffer
[:
window_size
])
buffer
=
buffer
[
window_size
:]
while
not
vad
.
empty
():
if
len
(
vad
.
front
.
samples
)
<
0.5
*
g_sample_rate
:
# this segment is too short, skip it
vad
.
pop
()
continue
stream
=
extractor
.
create_stream
()
stream
.
accept_waveform
(
sample_rate
=
g_sample_rate
,
waveform
=
vad
.
front
.
samples
)
stream
.
input_finished
()
embedding
=
extractor
.
compute
(
stream
)
embedding
=
np
.
array
(
embedding
)
name
=
manager
.
search
(
embedding
,
threshold
=
args
.
threshold
)
if
not
name
:
name
=
"unknown"
# Now for non-streaming ASR
asr_stream
=
recognizer
.
create_stream
()
asr_stream
.
accept_waveform
(
sample_rate
=
g_sample_rate
,
waveform
=
vad
.
front
.
samples
)
recognizer
.
decode_stream
(
asr_stream
)
text
=
asr_stream
.
result
.
text
vad
.
pop
()
print
(
f
"
\r
{idx}-{name}: {text}"
)
idx
+=
1
if
__name__
==
"__main__"
:
try
:
main
()
except
KeyboardInterrupt
:
print
(
"
\n
Caught Ctrl + C. Exiting"
)
...
...
python-api-examples/speaker-identification-with-vad.py
0 → 100755
查看文件 @
59e2851
#!/usr/bin/env python3
"""
This script shows how to use Python APIs for speaker identification with
a microphone and a VAD model
Usage:
(1) Prepare a text file containing speaker related files.
Each line in the text file contains two columns. The first column is the
speaker name, while the second column contains the wave file of the speaker.
If the text file contains multiple wave files for the same speaker, then the
embeddings of these files are averaged.
An example text file is given below:
foo /path/to/a.wav
bar /path/to/b.wav
foo /path/to/c.wav
foobar /path/to/d.wav
Each wave file should contain only a single channel; the sample format
should be int16_t; the sample rate can be arbitrary.
(2) Download a model for computing speaker embeddings
Please visit
https://github.com/k2-fsa/sherpa-onnx/releases/tag/speaker-recongition-models
to download a model. An example is given below:
wget https://github.com/k2-fsa/sherpa-onnx/releases/download/speaker-recongition-models/wespeaker_zh_cnceleb_resnet34.onnx
Note that `zh` means Chinese, while `en` means English.
(3) Download the VAD model
Please visit
https://github.com/snakers4/silero-vad/blob/master/files/silero_vad.onnx
to download silero_vad.onnx
For instance,
wget https://github.com/snakers4/silero-vad/raw/master/files/silero_vad.onnx
(4) Run this script
Assume the filename of the text file is speaker.txt.
python3 ./python-api-examples/speaker-identification-with-vad.py
\
--silero-vad-model=/path/to/silero_vad.onnx
\
--speaker-file ./speaker.txt
\
--model ./wespeaker_zh_cnceleb_resnet34.onnx
"""
import
argparse
import
sys
from
collections
import
defaultdict
from
pathlib
import
Path
from
typing
import
Dict
,
List
,
Tuple
import
numpy
as
np
import
sherpa_onnx
import
torchaudio
try
:
import
sounddevice
as
sd
except
ImportError
:
print
(
"Please install sounddevice first. You can use"
)
print
()
print
(
" pip install sounddevice"
)
print
()
print
(
"to install it"
)
sys
.
exit
(
-
1
)
def
get_args
():
parser
=
argparse
.
ArgumentParser
(
formatter_class
=
argparse
.
ArgumentDefaultsHelpFormatter
)
parser
.
add_argument
(
"--speaker-file"
,
type
=
str
,
required
=
True
,
help
=
"""Path to the speaker file. Read the help doc at the beginning of this
file for the format."""
,
)
parser
.
add_argument
(
"--model"
,
type
=
str
,
required
=
True
,
help
=
"Path to the speaker embedding model file."
,
)
parser
.
add_argument
(
"--silero-vad-model"
,
type
=
str
,
required
=
True
,
help
=
"Path to silero_vad.onnx"
,
)
parser
.
add_argument
(
"--threshold"
,
type
=
float
,
default
=
0.6
)
parser
.
add_argument
(
"--num-threads"
,
type
=
int
,
default
=
1
,
help
=
"Number of threads for neural network computation"
,
)
parser
.
add_argument
(
"--debug"
,
type
=
bool
,
default
=
False
,
help
=
"True to show debug messages"
,
)
parser
.
add_argument
(
"--provider"
,
type
=
str
,
default
=
"cpu"
,
help
=
"Valid values: cpu, cuda, coreml"
,
)
return
parser
.
parse_args
()
def
load_speaker_embedding_model
(
args
):
config
=
sherpa_onnx
.
SpeakerEmbeddingExtractorConfig
(
model
=
args
.
model
,
num_threads
=
args
.
num_threads
,
debug
=
args
.
debug
,
provider
=
args
.
provider
,
)
if
not
config
.
validate
():
raise
ValueError
(
f
"Invalid config. {config}"
)
extractor
=
sherpa_onnx
.
SpeakerEmbeddingExtractor
(
config
)
return
extractor
def
load_speaker_file
(
args
)
->
Dict
[
str
,
List
[
str
]]:
if
not
Path
(
args
.
speaker_file
)
.
is_file
():
raise
ValueError
(
f
"--speaker-file {args.speaker_file} does not exist"
)
ans
=
defaultdict
(
list
)
with
open
(
args
.
speaker_file
)
as
f
:
for
line
in
f
:
line
=
line
.
strip
()
if
not
line
:
continue
fields
=
line
.
split
()
if
len
(
fields
)
!=
2
:
raise
ValueError
(
f
"Invalid line: {line}. Fields: {fields}"
)
speaker_name
,
filename
=
fields
ans
[
speaker_name
]
.
append
(
filename
)
return
ans
def
load_audio
(
filename
:
str
)
->
Tuple
[
np
.
ndarray
,
int
]:
samples
,
sample_rate
=
torchaudio
.
load
(
filename
)
return
samples
[
0
]
.
contiguous
()
.
numpy
(),
sample_rate
def
compute_speaker_embedding
(
filenames
:
List
[
str
],
extractor
:
sherpa_onnx
.
SpeakerEmbeddingExtractor
,
)
->
np
.
ndarray
:
assert
len
(
filenames
)
>
0
,
"filenames is empty"
ans
=
None
for
filename
in
filenames
:
print
(
f
"processing {filename}"
)
samples
,
sample_rate
=
load_audio
(
filename
)
stream
=
extractor
.
create_stream
()
stream
.
accept_waveform
(
sample_rate
=
sample_rate
,
waveform
=
samples
)
stream
.
input_finished
()
assert
extractor
.
is_ready
(
stream
)
embedding
=
extractor
.
compute
(
stream
)
embedding
=
np
.
array
(
embedding
)
if
ans
is
None
:
ans
=
embedding
else
:
ans
+=
embedding
return
ans
/
len
(
filenames
)
g_sample_rate
=
16000
def
main
():
args
=
get_args
()
print
(
args
)
extractor
=
load_speaker_embedding_model
(
args
)
speaker_file
=
load_speaker_file
(
args
)
manager
=
sherpa_onnx
.
SpeakerEmbeddingManager
(
extractor
.
dim
)
for
name
,
filename_list
in
speaker_file
.
items
():
embedding
=
compute_speaker_embedding
(
filenames
=
filename_list
,
extractor
=
extractor
,
)
status
=
manager
.
add
(
name
,
embedding
)
if
not
status
:
raise
RuntimeError
(
f
"Failed to register speaker {name}"
)
vad_config
=
sherpa_onnx
.
VadModelConfig
()
vad_config
.
silero_vad
.
model
=
args
.
silero_vad_model
vad_config
.
silero_vad
.
min_silence_duration
=
0.25
vad_config
.
silero_vad
.
min_speech_duration
=
0.25
vad_config
.
sample_rate
=
g_sample_rate
window_size
=
vad_config
.
silero_vad
.
window_size
vad
=
sherpa_onnx
.
VoiceActivityDetector
(
vad_config
,
buffer_size_in_seconds
=
100
)
samples_per_read
=
int
(
0.1
*
g_sample_rate
)
# 0.1 second = 100 ms
devices
=
sd
.
query_devices
()
if
len
(
devices
)
==
0
:
print
(
"No microphone devices found"
)
sys
.
exit
(
0
)
print
(
devices
)
default_input_device_idx
=
sd
.
default
.
device
[
0
]
print
(
f
'Use default device: {devices[default_input_device_idx]["name"]}'
)
print
(
"Started! Please speak"
)
idx
=
0
buffer
=
[]
with
sd
.
InputStream
(
channels
=
1
,
dtype
=
"float32"
,
samplerate
=
g_sample_rate
)
as
s
:
while
True
:
samples
,
_
=
s
.
read
(
samples_per_read
)
# a blocking read
samples
=
samples
.
reshape
(
-
1
)
buffer
=
np
.
concatenate
([
buffer
,
samples
])
while
len
(
buffer
)
>
window_size
:
vad
.
accept_waveform
(
buffer
[:
window_size
])
buffer
=
buffer
[
window_size
:]
while
not
vad
.
empty
():
if
len
(
vad
.
front
.
samples
)
<
0.5
*
g_sample_rate
:
# this segment is too short, skip it
vad
.
pop
()
continue
stream
=
extractor
.
create_stream
()
stream
.
accept_waveform
(
sample_rate
=
g_sample_rate
,
waveform
=
vad
.
front
.
samples
)
vad
.
pop
()
stream
.
input_finished
()
print
(
"Computing"
,
end
=
""
)
embedding
=
extractor
.
compute
(
stream
)
embedding
=
np
.
array
(
embedding
)
name
=
manager
.
search
(
embedding
,
threshold
=
args
.
threshold
)
if
not
name
:
name
=
"unknown"
print
(
f
"
\r
{idx}: Predicted name: {name}"
)
idx
+=
1
if
__name__
==
"__main__"
:
try
:
main
()
except
KeyboardInterrupt
:
print
(
"
\n
Caught Ctrl + C. Exiting"
)
...
...
python-api-examples/speaker-identification.py
查看文件 @
59e2851
#!/usr/bin/env python3
"""
This script shows how to use Python APIs for speaker identification.
This script shows how to use Python APIs for speaker identification with
a microphone.
Usage:
...
...
@@ -43,6 +44,7 @@ python3 ./python-api-examples/speaker-identification.py \
"""
import
argparse
import
queue
import
sys
import
threading
from
collections
import
defaultdict
from
pathlib
import
Path
...
...
@@ -151,7 +153,7 @@ def compute_speaker_embedding(
filenames
:
List
[
str
],
extractor
:
sherpa_onnx
.
SpeakerEmbeddingExtractor
,
)
->
np
.
ndarray
:
assert
len
(
filenames
)
>
0
,
f
"filenames is empty"
assert
len
(
filenames
)
>
0
,
"filenames is empty"
ans
=
None
for
filename
in
filenames
:
...
...
@@ -215,7 +217,7 @@ def main():
global
g_stop
global
g_read_mic_thread
while
True
:
key
=
input
(
"Press
e
nter to start recording"
)
key
=
input
(
"Press
E
nter to start recording"
)
if
key
.
lower
()
in
(
"q"
,
"quit"
):
g_stop
=
True
break
...
...
@@ -224,7 +226,7 @@ def main():
g_buffer
.
queue
.
clear
()
g_read_mic_thread
=
threading
.
Thread
(
target
=
read_mic
)
g_read_mic_thread
.
start
()
input
(
"Press
e
nter to stop recording"
)
input
(
"Press
E
nter to stop recording"
)
g_stop
=
True
g_read_mic_thread
.
join
()
print
(
"Compute embedding"
)
...
...
请
注册
或
登录
后发表评论