daihui.zhang
commited on
Commit
·
4d568e6
1
Parent(s):
9ce0bfb
update add fix
Browse files- .gitignore +1 -1
- main.py +0 -0
- moyoyo_asr_models/Qwen2.5-3B.Q5_0.gguf +3 -0
- moyoyo_asr_models/qwen2.5-7b-instruct-q5_0-00001-of-00002.gguf +3 -0
- moyoyo_asr_models/qwen2.5-7b-instruct-q5_0-00002-of-00002.gguf +3 -0
- pyproject.toml +3 -0
- requirements.txt +31 -23
- transcribe/helpers/__init__.py +0 -0
- transcribe/{translator.py → helpers/translator.py} +0 -0
- transcribe/helpers/vadprocessor.py +251 -0
- transcribe/{whisper.py → helpers/whisper.py} +0 -0
- transcribe/pipelines/__init__.py +1 -0
- transcribe/pipelines/base.py +1 -0
- transcribe/pipelines/pipe_translate.py +1 -1
- transcribe/pipelines/pipe_vad.py +68 -1
- transcribe/pipelines/pipe_whisper.py +1 -1
- transcribe/translatepipes.py +19 -10
- transcribe/utils.py +6 -0
- transcribe/whisper_llm_serve.py +50 -22
- uv.lock +100 -0
.gitignore
CHANGED
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@@ -68,7 +68,7 @@ instance/
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# Sphinx documentation
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docs/_build/
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-
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# PyBuilder
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.pybuilder/
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target/
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# Sphinx documentation
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docs/_build/
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+
test/
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# PyBuilder
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.pybuilder/
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target/
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main.py
ADDED
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File without changes
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moyoyo_asr_models/Qwen2.5-3B.Q5_0.gguf
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@@ -0,0 +1,3 @@
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version https://git-lfs.github.com/spec/v1
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+
oid sha256:35a1eec88348fd963b06bb2e3ba8d4d27d4f7a79d9ed4bab645f931d7b2b7ac7
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+
size 2169666016
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moyoyo_asr_models/qwen2.5-7b-instruct-q5_0-00001-of-00002.gguf
ADDED
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@@ -0,0 +1,3 @@
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+
version https://git-lfs.github.com/spec/v1
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+
oid sha256:715a0c76b1b074ed65fe0fabf88b1df6f5f638318f15ba9f62a04d493e236d5f
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+
size 4001112160
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moyoyo_asr_models/qwen2.5-7b-instruct-q5_0-00002-of-00002.gguf
ADDED
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@@ -0,0 +1,3 @@
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+
version https://git-lfs.github.com/spec/v1
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+
oid sha256:6a254c888ebffcddf98b0c277433a339d9a8cbf970f246b0b6e6a9195e5d6cfb
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+
size 1314064416
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pyproject.toml
CHANGED
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@@ -6,12 +6,15 @@ readme = "README.md"
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requires-python = ">=3.11"
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dependencies = [
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"av>=14.2.0",
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"librosa>=0.11.0",
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"numpy>=2.1.3",
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"onnxruntime>=1.21.0",
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"pyaudio>=0.2.14",
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"pydantic>=2.11.2",
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"setuptools>=78.1.0",
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"soundfile>=0.13.1",
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"torch>=2.6.0",
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"tqdm>=4.67.1",
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requires-python = ">=3.11"
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dependencies = [
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"av>=14.2.0",
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+
"fastapi>=0.115.12",
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"librosa>=0.11.0",
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"numpy>=2.1.3",
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"onnxruntime>=1.21.0",
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"pyaudio>=0.2.14",
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"pydantic>=2.11.2",
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+
"pydub>=0.25.1",
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"setuptools>=78.1.0",
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+
"silero-vad>=5.1.2",
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"soundfile>=0.13.1",
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"torch>=2.6.0",
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"tqdm>=4.67.1",
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requirements.txt
CHANGED
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@@ -2,6 +2,8 @@
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# uv pip compile pyproject.toml -o requirements.txt
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annotated-types==0.7.0
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# via pydantic
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audioread==3.0.1
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# via librosa
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av==14.3.0
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@@ -16,8 +18,8 @@ coloredlogs==15.0.1
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# via onnxruntime
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decorator==5.2.1
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# via librosa
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-
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-
# via
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filelock==3.18.0
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# via torch
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flatbuffers==25.2.10
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@@ -27,11 +29,11 @@ fsspec==2025.3.2
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humanfriendly==10.0
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# via coloredlogs
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idna==3.10
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-
# via requests
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-
jinja2==3.1.6
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# via
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-
#
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-
#
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joblib==1.4.2
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# via
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# librosa
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@@ -40,8 +42,6 @@ lazy-loader==0.4
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# via librosa
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librosa==0.11.0
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# via trans (pyproject.toml)
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-
llama-cpp-python==0.3.8
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-
# via trans (pyproject.toml)
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llvmlite==0.44.0
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# via numba
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markupsafe==3.0.2
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@@ -58,25 +58,23 @@ numpy==2.1.3
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# via
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# trans (pyproject.toml)
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# librosa
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-
# llama-cpp-python
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# numba
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# onnxruntime
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-
# pywhispercpp
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# scikit-learn
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# scipy
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# soundfile
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# soxr
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onnxruntime==1.21.0
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-
# via
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packaging==24.2
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# via
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# lazy-loader
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# onnxruntime
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# pooch
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platformdirs==4.3.7
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-
# via
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-
# pooch
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-
# pywhispercpp
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pooch==1.8.2
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# via librosa
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protobuf==6.30.2
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@@ -86,15 +84,15 @@ pyaudio==0.2.14
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pycparser==2.22
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# via cffi
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pydantic==2.11.2
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-
# via
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pydantic-core==2.33.1
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# via pydantic
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-
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# via trans (pyproject.toml)
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requests==2.32.3
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-
# via
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# pooch
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# pywhispercpp
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scikit-learn==1.6.1
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# via librosa
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scipy==1.15.2
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@@ -103,12 +101,18 @@ scipy==1.15.2
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# scikit-learn
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setuptools==78.1.0
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# via trans (pyproject.toml)
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soundfile==0.13.1
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# via
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# trans (pyproject.toml)
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# librosa
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soxr==0.5.0.post1
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# via librosa
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sympy==1.13.1
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# via
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# onnxruntime
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threadpoolctl==3.6.0
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# via scikit-learn
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torch==2.6.0
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-
# via trans (pyproject.toml)
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-
tqdm==4.67.1
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# via
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# trans (pyproject.toml)
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#
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typing-extensions==4.13.1
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# via
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# librosa
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-
# llama-cpp-python
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# pydantic
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# pydantic-core
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# torch
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# uv pip compile pyproject.toml -o requirements.txt
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annotated-types==0.7.0
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# via pydantic
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+
anyio==4.9.0
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+
# via starlette
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audioread==3.0.1
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# via librosa
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av==14.3.0
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# via onnxruntime
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decorator==5.2.1
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# via librosa
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+
fastapi==0.115.12
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| 22 |
+
# via trans (pyproject.toml)
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| 23 |
filelock==3.18.0
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| 24 |
# via torch
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flatbuffers==25.2.10
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humanfriendly==10.0
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# via coloredlogs
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idna==3.10
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# via
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+
# anyio
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+
# requests
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+
jinja2==3.1.6
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+
# via torch
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joblib==1.4.2
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| 38 |
# via
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# librosa
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# via librosa
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| 43 |
librosa==0.11.0
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# via trans (pyproject.toml)
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llvmlite==0.44.0
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| 46 |
# via numba
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markupsafe==3.0.2
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# via
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# trans (pyproject.toml)
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# librosa
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# numba
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| 62 |
# onnxruntime
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# scikit-learn
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# scipy
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| 65 |
# soundfile
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| 66 |
# soxr
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onnxruntime==1.21.0
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+
# via
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+
# trans (pyproject.toml)
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| 70 |
+
# silero-vad
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| 71 |
packaging==24.2
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| 72 |
# via
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# lazy-loader
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| 74 |
# onnxruntime
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| 75 |
# pooch
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| 76 |
platformdirs==4.3.7
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+
# via pooch
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| 78 |
pooch==1.8.2
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# via librosa
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protobuf==6.30.2
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| 84 |
pycparser==2.22
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| 85 |
# via cffi
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| 86 |
pydantic==2.11.2
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+
# via
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| 88 |
+
# trans (pyproject.toml)
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| 89 |
+
# fastapi
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| 90 |
pydantic-core==2.33.1
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| 91 |
# via pydantic
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| 92 |
+
pydub==0.25.1
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# via trans (pyproject.toml)
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requests==2.32.3
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+
# via pooch
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scikit-learn==1.6.1
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# via librosa
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scipy==1.15.2
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| 101 |
# scikit-learn
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setuptools==78.1.0
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| 103 |
# via trans (pyproject.toml)
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| 104 |
+
silero-vad==5.1.2
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| 105 |
+
# via trans (pyproject.toml)
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| 106 |
+
sniffio==1.3.1
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| 107 |
+
# via anyio
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| 108 |
soundfile==0.13.1
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| 109 |
# via
|
| 110 |
# trans (pyproject.toml)
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| 111 |
# librosa
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| 112 |
soxr==0.5.0.post1
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| 113 |
# via librosa
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| 114 |
+
starlette==0.46.1
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| 115 |
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# via fastapi
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| 116 |
sympy==1.13.1
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| 117 |
# via
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| 118 |
# onnxruntime
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threadpoolctl==3.6.0
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# via scikit-learn
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torch==2.6.0
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| 123 |
# via
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| 124 |
# trans (pyproject.toml)
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+
# silero-vad
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| 126 |
+
# torchaudio
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| 127 |
+
torchaudio==2.6.0
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| 128 |
+
# via silero-vad
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| 129 |
+
tqdm==4.67.1
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| 130 |
+
# via trans (pyproject.toml)
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| 131 |
typing-extensions==4.13.1
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| 132 |
# via
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+
# anyio
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| 134 |
+
# fastapi
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| 135 |
# librosa
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# pydantic
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# pydantic-core
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# torch
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transcribe/helpers/__init__.py
ADDED
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File without changes
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transcribe/{translator.py → helpers/translator.py}
RENAMED
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File without changes
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transcribe/helpers/vadprocessor.py
ADDED
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@@ -0,0 +1,251 @@
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|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 1 |
+
import gc
|
| 2 |
+
import time
|
| 3 |
+
import numpy as np
|
| 4 |
+
import onnxruntime
|
| 5 |
+
from datetime import timedelta
|
| 6 |
+
from pydub import AudioSegment
|
| 7 |
+
from silero_vad import load_silero_vad, get_speech_timestamps
|
| 8 |
+
import os
|
| 9 |
+
import logging
|
| 10 |
+
|
| 11 |
+
|
| 12 |
+
class SileroVADProcessor:
|
| 13 |
+
"""
|
| 14 |
+
A class for processing audio files using Silero VAD to detect voice activity
|
| 15 |
+
and extract voice segments from audio files.
|
| 16 |
+
"""
|
| 17 |
+
|
| 18 |
+
def __init__(self,
|
| 19 |
+
activate_threshold=0.5,
|
| 20 |
+
fusion_threshold=0.3,
|
| 21 |
+
min_speech_duration=0.25,
|
| 22 |
+
max_speech_duration=20,
|
| 23 |
+
min_silence_duration=250,
|
| 24 |
+
sample_rate=16000,
|
| 25 |
+
ort_providers=None):
|
| 26 |
+
"""
|
| 27 |
+
Initialize the SileroVADProcessor.
|
| 28 |
+
|
| 29 |
+
Args:
|
| 30 |
+
activate_threshold (float): Threshold for voice activity detection
|
| 31 |
+
fusion_threshold (float): Threshold for merging close speech segments (seconds)
|
| 32 |
+
min_speech_duration (float): Minimum duration of speech to be considered valid (seconds)
|
| 33 |
+
max_speech_duration (float): Maximum duration of speech (seconds)
|
| 34 |
+
min_silence_duration (int): Minimum silence duration (ms)
|
| 35 |
+
sample_rate (int): Sample rate of the audio (8000 or 16000 Hz)
|
| 36 |
+
ort_providers (list): ONNX Runtime providers for acceleration
|
| 37 |
+
"""
|
| 38 |
+
# VAD parameters
|
| 39 |
+
self.activate_threshold = activate_threshold
|
| 40 |
+
self.fusion_threshold = fusion_threshold
|
| 41 |
+
self.min_speech_duration = min_speech_duration
|
| 42 |
+
self.max_speech_duration = max_speech_duration
|
| 43 |
+
self.min_silence_duration = min_silence_duration
|
| 44 |
+
self.sample_rate = sample_rate
|
| 45 |
+
self.ort_providers = ort_providers if ort_providers else []
|
| 46 |
+
|
| 47 |
+
# Initialize logger
|
| 48 |
+
self.logger = logging.getLogger(__name__)
|
| 49 |
+
|
| 50 |
+
# Load Silero VAD model
|
| 51 |
+
self._init_onnx_session()
|
| 52 |
+
self.silero_vad = load_silero_vad(onnx=True)
|
| 53 |
+
|
| 54 |
+
def _init_onnx_session(self):
|
| 55 |
+
"""Initialize ONNX Runtime session with appropriate settings."""
|
| 56 |
+
session_opts = onnxruntime.SessionOptions()
|
| 57 |
+
session_opts.log_severity_level = 3
|
| 58 |
+
session_opts.inter_op_num_threads = 0
|
| 59 |
+
session_opts.intra_op_num_threads = 0
|
| 60 |
+
session_opts.enable_cpu_mem_arena = True
|
| 61 |
+
session_opts.execution_mode = onnxruntime.ExecutionMode.ORT_SEQUENTIAL
|
| 62 |
+
session_opts.graph_optimization_level = onnxruntime.GraphOptimizationLevel.ORT_ENABLE_ALL
|
| 63 |
+
|
| 64 |
+
session_opts.add_session_config_entry("session.intra_op.allow_spinning", "1")
|
| 65 |
+
session_opts.add_session_config_entry("session.inter_op.allow_spinning", "1")
|
| 66 |
+
session_opts.add_session_config_entry("session.set_denormal_as_zero", "1")
|
| 67 |
+
|
| 68 |
+
# Set the session_opts to be used by silero_vad
|
| 69 |
+
# onnxruntime.capi._pybind_state.get_default_session_options(session_opts)
|
| 70 |
+
|
| 71 |
+
def load_audio(self, audio_path):
|
| 72 |
+
"""
|
| 73 |
+
Load audio file and prepare it for VAD processing.
|
| 74 |
+
|
| 75 |
+
Args:
|
| 76 |
+
audio_path (str): Path to the audio file
|
| 77 |
+
|
| 78 |
+
Returns:
|
| 79 |
+
numpy.ndarray: Audio data as numpy array
|
| 80 |
+
"""
|
| 81 |
+
self.logger.info(f"Loading audio from {audio_path}")
|
| 82 |
+
audio_segment = AudioSegment.from_file(audio_path)
|
| 83 |
+
audio_segment = audio_segment.set_channels(1).set_frame_rate(self.sample_rate)
|
| 84 |
+
|
| 85 |
+
# Convert to numpy array and normalize
|
| 86 |
+
dtype = np.float16 if self.use_gpu_fp16 else np.float32
|
| 87 |
+
audio_array = np.array(audio_segment.get_array_of_samples(), dtype=dtype) * 0.000030517578 # 1/32768
|
| 88 |
+
|
| 89 |
+
self.audio_segment = audio_segment # Store for later use
|
| 90 |
+
return audio_array
|
| 91 |
+
|
| 92 |
+
@property
|
| 93 |
+
def model(self):
|
| 94 |
+
return self.silero_vad
|
| 95 |
+
|
| 96 |
+
def process_timestamps(self, timestamps):
|
| 97 |
+
"""
|
| 98 |
+
Process VAD timestamps: filter short segments and merge close segments.
|
| 99 |
+
|
| 100 |
+
Args:
|
| 101 |
+
timestamps (list): List of (start, end) tuples
|
| 102 |
+
|
| 103 |
+
Returns:
|
| 104 |
+
list: Processed list of (start, end) tuples
|
| 105 |
+
"""
|
| 106 |
+
# Filter out short durations
|
| 107 |
+
filtered_timestamps = [(start, end) for start, end in timestamps
|
| 108 |
+
if (end - start) >= self.min_speech_duration]
|
| 109 |
+
|
| 110 |
+
# Fuse timestamps in two passes for better merging
|
| 111 |
+
fused_timestamps_1st = []
|
| 112 |
+
for start, end in filtered_timestamps:
|
| 113 |
+
if fused_timestamps_1st and (start - fused_timestamps_1st[-1][1] <= self.fusion_threshold):
|
| 114 |
+
fused_timestamps_1st[-1] = (fused_timestamps_1st[-1][0], end)
|
| 115 |
+
else:
|
| 116 |
+
fused_timestamps_1st.append((start, end))
|
| 117 |
+
|
| 118 |
+
fused_timestamps_2nd = []
|
| 119 |
+
for start, end in fused_timestamps_1st:
|
| 120 |
+
if fused_timestamps_2nd and (start - fused_timestamps_2nd[-1][1] <= self.fusion_threshold):
|
| 121 |
+
fused_timestamps_2nd[-1] = (fused_timestamps_2nd[-1][0], end)
|
| 122 |
+
else:
|
| 123 |
+
fused_timestamps_2nd.append((start, end))
|
| 124 |
+
|
| 125 |
+
return fused_timestamps_2nd
|
| 126 |
+
|
| 127 |
+
def format_time(self, seconds):
|
| 128 |
+
"""
|
| 129 |
+
Convert seconds to VTT time format 'hh:mm:ss.mmm'.
|
| 130 |
+
|
| 131 |
+
Args:
|
| 132 |
+
seconds (float): Time in seconds
|
| 133 |
+
|
| 134 |
+
Returns:
|
| 135 |
+
str: Formatted time string
|
| 136 |
+
"""
|
| 137 |
+
td = timedelta(seconds=seconds)
|
| 138 |
+
td_sec = td.total_seconds()
|
| 139 |
+
total_seconds = int(td_sec)
|
| 140 |
+
milliseconds = int((td_sec - total_seconds) * 1000)
|
| 141 |
+
hours = total_seconds // 3600
|
| 142 |
+
minutes = (total_seconds % 3600) // 60
|
| 143 |
+
seconds = total_seconds % 60
|
| 144 |
+
return f"{hours:02}:{minutes:02}:{seconds:02}.{milliseconds:03}"
|
| 145 |
+
|
| 146 |
+
def detect_speech(self, audio:np.array):
|
| 147 |
+
"""
|
| 148 |
+
Run VAD on the audio file to detect speech segments.
|
| 149 |
+
|
| 150 |
+
Args:
|
| 151 |
+
audio_path (str): Path to the audio file
|
| 152 |
+
|
| 153 |
+
Returns:
|
| 154 |
+
list: List of processed timestamps as (start, end) tuples
|
| 155 |
+
"""
|
| 156 |
+
self.logger.info("Starting VAD process")
|
| 157 |
+
start_time = time.time()
|
| 158 |
+
# Get speech timestamps
|
| 159 |
+
raw_timestamps = get_speech_timestamps(
|
| 160 |
+
audio,
|
| 161 |
+
model=self.silero_vad,
|
| 162 |
+
threshold=self.activate_threshold,
|
| 163 |
+
max_speech_duration_s=self.max_speech_duration,
|
| 164 |
+
min_speech_duration_ms=int(self.min_speech_duration * 1000),
|
| 165 |
+
min_silence_duration_ms=self.min_silence_duration,
|
| 166 |
+
return_seconds=True
|
| 167 |
+
)
|
| 168 |
+
|
| 169 |
+
# Convert to simple format and process
|
| 170 |
+
timestamps = [(item['start'], item['end']) for item in raw_timestamps]
|
| 171 |
+
processed_timestamps = self.process_timestamps(timestamps)
|
| 172 |
+
|
| 173 |
+
# Clean up
|
| 174 |
+
del audio
|
| 175 |
+
gc.collect()
|
| 176 |
+
|
| 177 |
+
self.logger.info(f"VAD completed in {time.time() - start_time:.3f} seconds")
|
| 178 |
+
return processed_timestamps
|
| 179 |
+
|
| 180 |
+
"""
|
| 181 |
+
Save timestamps in both second and sample indices formats.
|
| 182 |
+
|
| 183 |
+
Args:
|
| 184 |
+
timestamps (list): List of (start, end) tuples
|
| 185 |
+
output_prefix (str): Prefix for output files
|
| 186 |
+
"""
|
| 187 |
+
# Save timestamps in seconds (VTT format)
|
| 188 |
+
seconds_path = f"{output_prefix}_timestamps_second.txt"
|
| 189 |
+
with open(seconds_path, "w", encoding='UTF-8') as file:
|
| 190 |
+
self.logger.info("Saving timestamps in seconds format")
|
| 191 |
+
for start, end in timestamps:
|
| 192 |
+
s_time = self.format_time(start)
|
| 193 |
+
e_time = self.format_time(end)
|
| 194 |
+
line = f"{s_time} --> {e_time}\n"
|
| 195 |
+
file.write(line)
|
| 196 |
+
|
| 197 |
+
# Save timestamps in sample indices
|
| 198 |
+
indices_path = f"{output_prefix}_timestamps_indices.txt"
|
| 199 |
+
with open(indices_path, "w", encoding='UTF-8') as file:
|
| 200 |
+
self.logger.info("Saving timestamps in indices format")
|
| 201 |
+
for start, end in timestamps:
|
| 202 |
+
line = f"{int(start * self.sample_rate)} --> {int(end * self.sample_rate)}\n"
|
| 203 |
+
file.write(line)
|
| 204 |
+
|
| 205 |
+
self.logger.info(f"Timestamps saved to {seconds_path} and {indices_path}")
|
| 206 |
+
|
| 207 |
+
def extract_speech_segments(self, audio_segment, timestamps):
|
| 208 |
+
"""
|
| 209 |
+
Extract speech segments from the audio and combine them into a single audio file.
|
| 210 |
+
|
| 211 |
+
Args:
|
| 212 |
+
timestamps (list): List of (start, end) tuples indicating speech segments
|
| 213 |
+
|
| 214 |
+
Returns:
|
| 215 |
+
AudioSegment: The combined speech segments
|
| 216 |
+
"""
|
| 217 |
+
audio_segment = audio_segment.numpy()
|
| 218 |
+
combined_speech = np.array([], dtype=np.float32)
|
| 219 |
+
|
| 220 |
+
# Extract and combine each speech segment
|
| 221 |
+
for i, (start, end) in enumerate(timestamps):
|
| 222 |
+
# Convert seconds to milliseconds for pydub
|
| 223 |
+
start_ms = int(start * 1000)
|
| 224 |
+
end_ms = int(end * 1000)
|
| 225 |
+
|
| 226 |
+
# Ensure the end time does not exceed the length of the audio segment
|
| 227 |
+
if end_ms > len(audio_segment):
|
| 228 |
+
end_ms = len(audio_segment)
|
| 229 |
+
|
| 230 |
+
# Extract the segment
|
| 231 |
+
segment = audio_segment[start_ms:end_ms]
|
| 232 |
+
|
| 233 |
+
# Add to combined audio
|
| 234 |
+
combined_speech = np.append(combined_speech, segment)
|
| 235 |
+
|
| 236 |
+
return combined_speech
|
| 237 |
+
|
| 238 |
+
def process_audio(self, audio_array:np.array):
|
| 239 |
+
"""
|
| 240 |
+
Complete processing pipeline: detect speech, save timestamps, and optionally extract speech.
|
| 241 |
+
|
| 242 |
+
Returns:
|
| 243 |
+
tuple: (timestamps, output_speech_path if extract_speech else None)
|
| 244 |
+
"""
|
| 245 |
+
|
| 246 |
+
# Run VAD to detect speech
|
| 247 |
+
timestamps = self.detect_speech(audio_array)
|
| 248 |
+
|
| 249 |
+
combined_speech = self.extract_speech_segments(audio_array, timestamps)
|
| 250 |
+
|
| 251 |
+
return timestamps, combined_speech
|
transcribe/{whisper.py → helpers/whisper.py}
RENAMED
|
File without changes
|
transcribe/pipelines/__init__.py
CHANGED
|
@@ -1,4 +1,5 @@
|
|
| 1 |
|
| 2 |
from .pipe_translate import TranslatePipe
|
| 3 |
from .pipe_whisper import WhisperPipe
|
|
|
|
| 4 |
from .base import MetaItem
|
|
|
|
| 1 |
|
| 2 |
from .pipe_translate import TranslatePipe
|
| 3 |
from .pipe_whisper import WhisperPipe
|
| 4 |
+
from .pipe_vad import VadPipe
|
| 5 |
from .base import MetaItem
|
transcribe/pipelines/base.py
CHANGED
|
@@ -12,6 +12,7 @@ class Segment:
|
|
| 12 |
@dataclass
|
| 13 |
class MetaItem:
|
| 14 |
segments: list[Segment] = field(default_factory=list)
|
|
|
|
| 15 |
audio: bytes = b''
|
| 16 |
transcribe_content: str = ''
|
| 17 |
translate_content: str = ''
|
|
|
|
| 12 |
@dataclass
|
| 13 |
class MetaItem:
|
| 14 |
segments: list[Segment] = field(default_factory=list)
|
| 15 |
+
source_audio: bytes = b""
|
| 16 |
audio: bytes = b''
|
| 17 |
transcribe_content: str = ''
|
| 18 |
translate_content: str = ''
|
transcribe/pipelines/pipe_translate.py
CHANGED
|
@@ -1,7 +1,7 @@
|
|
| 1 |
|
| 2 |
from .base import MetaItem, BasePipe, Segment
|
| 3 |
from llama_cpp import Llama
|
| 4 |
-
from ..translator import QwenTranslator
|
| 5 |
from config import LLM_MODEL_PATH, LLM_SYS_PROMPT
|
| 6 |
|
| 7 |
|
|
|
|
| 1 |
|
| 2 |
from .base import MetaItem, BasePipe, Segment
|
| 3 |
from llama_cpp import Llama
|
| 4 |
+
from ..helpers.translator import QwenTranslator
|
| 5 |
from config import LLM_MODEL_PATH, LLM_SYS_PROMPT
|
| 6 |
|
| 7 |
|
transcribe/pipelines/pipe_vad.py
CHANGED
|
@@ -1,3 +1,70 @@
|
|
| 1 |
|
| 2 |
-
|
| 3 |
from .base import MetaItem, BasePipe
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 1 |
|
|
|
|
| 2 |
from .base import MetaItem, BasePipe
|
| 3 |
+
from ..helpers.vadprocessor import SileroVADProcessor
|
| 4 |
+
import numpy as np
|
| 5 |
+
from silero_vad import read_audio, get_speech_timestamps,collect_chunks, VADIterator
|
| 6 |
+
import torch
|
| 7 |
+
|
| 8 |
+
class VadPipe(BasePipe):
|
| 9 |
+
model = None
|
| 10 |
+
sample_rate=16000
|
| 11 |
+
window_size_samples = 512
|
| 12 |
+
|
| 13 |
+
@classmethod
|
| 14 |
+
def init(cls):
|
| 15 |
+
if cls.model is None:
|
| 16 |
+
cls.model = SileroVADProcessor(
|
| 17 |
+
activate_threshold=0.5,
|
| 18 |
+
fusion_threshold=0.3,
|
| 19 |
+
min_speech_duration=0.25,
|
| 20 |
+
max_speech_duration=20,
|
| 21 |
+
min_silence_duration=250,
|
| 22 |
+
sample_rate=cls.sample_rate
|
| 23 |
+
)
|
| 24 |
+
|
| 25 |
+
@property
|
| 26 |
+
def vad_iterator(self):
|
| 27 |
+
return VADIterator(self.model.silero_vad, sampling_rate=self.sample_rate,)
|
| 28 |
+
|
| 29 |
+
def process(self, in_data: MetaItem) -> MetaItem:
|
| 30 |
+
source_audio = in_data.source_audio
|
| 31 |
+
source_audio = np.frombuffer(source_audio, dtype=np.float32)
|
| 32 |
+
speech_segments = []
|
| 33 |
+
is_speech_active = False
|
| 34 |
+
# current_segment_end = len(source_audio)
|
| 35 |
+
|
| 36 |
+
for i in range(0, len(source_audio), self.window_size_samples):
|
| 37 |
+
|
| 38 |
+
window = source_audio[i:i+self.window_size_samples]
|
| 39 |
+
|
| 40 |
+
if len(window) < self.window_size_samples:
|
| 41 |
+
padded_window = np.zeros(self.window_size_samples, dtype=np.float32)
|
| 42 |
+
padded_window[:len(window)] = window
|
| 43 |
+
window = padded_window
|
| 44 |
+
|
| 45 |
+
speech_dict = self.vad_iterator(window, return_seconds=False)
|
| 46 |
+
if not speech_dict:
|
| 47 |
+
continue
|
| 48 |
+
|
| 49 |
+
# 计算当前偏移量
|
| 50 |
+
|
| 51 |
+
if speech_dict and 'start' in speech_dict and not is_speech_active:
|
| 52 |
+
is_speech_active = True
|
| 53 |
+
# current_segment_start = speech_dict['start'] + i
|
| 54 |
+
|
| 55 |
+
if is_speech_active:
|
| 56 |
+
speech_segments.append(window)
|
| 57 |
+
# # 如果检测到语音结束
|
| 58 |
+
# if speech_dict and 'end' in speech_dict and is_speech_active:
|
| 59 |
+
# # 调整语音结束时间,加上窗口偏移
|
| 60 |
+
# current_segment_end = min(speech_dict['end'] + i, current_segment_end)
|
| 61 |
+
# is_speech_active = False
|
| 62 |
+
# speech_audio = source_audio[current_segment_start: current_segment_end]
|
| 63 |
+
# speech_segments.append(speech_audio)
|
| 64 |
+
|
| 65 |
+
self.vad_iterator.reset_states()
|
| 66 |
+
combied_audio = np.concatenate(speech_segments, axis=0).tobytes() if len(speech_segments) else b""
|
| 67 |
+
in_data.audio = combied_audio
|
| 68 |
+
in_data.source_audio = b""
|
| 69 |
+
|
| 70 |
+
return in_data
|
transcribe/pipelines/pipe_whisper.py
CHANGED
|
@@ -1,7 +1,7 @@
|
|
| 1 |
|
| 2 |
|
| 3 |
from .base import MetaItem, BasePipe, Segment
|
| 4 |
-
from ..whisper import WhisperCPP
|
| 5 |
|
| 6 |
|
| 7 |
class WhisperPipe(BasePipe):
|
|
|
|
| 1 |
|
| 2 |
|
| 3 |
from .base import MetaItem, BasePipe, Segment
|
| 4 |
+
from ..helpers.whisper import WhisperCPP
|
| 5 |
|
| 6 |
|
| 7 |
class WhisperPipe(BasePipe):
|
transcribe/translatepipes.py
CHANGED
|
@@ -1,4 +1,4 @@
|
|
| 1 |
-
from transcribe.pipelines import WhisperPipe, TranslatePipe, MetaItem
|
| 2 |
import multiprocessing as mp
|
| 3 |
import config
|
| 4 |
|
|
@@ -10,21 +10,23 @@ class TranslatePipes:
|
|
| 10 |
# self.result_queue = mp.Queue()
|
| 11 |
|
| 12 |
# whisper 转录
|
| 13 |
-
self._whisper_pipe = WhisperPipe()
|
| 14 |
|
| 15 |
# llm 翻译
|
| 16 |
-
self._translate_pipe = TranslatePipe()
|
| 17 |
|
| 18 |
-
|
| 19 |
-
self.
|
| 20 |
-
|
| 21 |
-
|
| 22 |
-
|
| 23 |
-
|
|
|
|
| 24 |
|
| 25 |
def wait_ready(self):
|
| 26 |
self._whisper_pipe.wait()
|
| 27 |
self._translate_pipe.wait()
|
|
|
|
| 28 |
|
| 29 |
def translate(self, text, src_lang, dst_lang) -> MetaItem:
|
| 30 |
item = MetaItem(
|
|
@@ -40,13 +42,20 @@ class TranslatePipes:
|
|
| 40 |
self._whisper_pipe.input_queue.put(item)
|
| 41 |
return self._whisper_pipe.output_queue.get()
|
| 42 |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 43 |
|
| 44 |
if __name__ == "__main__":
|
| 45 |
import soundfile
|
| 46 |
tp = TranslatePipes()
|
| 47 |
# result = tp.translate("你好,今天天气怎么样?", src_lang="zh", dst_lang="en")
|
| 48 |
mel, _, = soundfile.read("assets/jfk.flac")
|
| 49 |
-
result = tp.transcrible(mel, 'en')
|
|
|
|
| 50 |
print(result)
|
| 51 |
|
| 52 |
|
|
|
|
| 1 |
+
from transcribe.pipelines import WhisperPipe, TranslatePipe, MetaItem, VadPipe
|
| 2 |
import multiprocessing as mp
|
| 3 |
import config
|
| 4 |
|
|
|
|
| 10 |
# self.result_queue = mp.Queue()
|
| 11 |
|
| 12 |
# whisper 转录
|
| 13 |
+
self._whisper_pipe = self._launch_process(WhisperPipe())
|
| 14 |
|
| 15 |
# llm 翻译
|
| 16 |
+
self._translate_pipe = self._launch_process(TranslatePipe())
|
| 17 |
|
| 18 |
+
# vad
|
| 19 |
+
self._vad_pipe = self._launch_process(VadPipe())
|
| 20 |
+
|
| 21 |
+
def _launch_process(self, process_obj):
|
| 22 |
+
process_obj.daemon = True
|
| 23 |
+
process_obj.start()
|
| 24 |
+
return process_obj
|
| 25 |
|
| 26 |
def wait_ready(self):
|
| 27 |
self._whisper_pipe.wait()
|
| 28 |
self._translate_pipe.wait()
|
| 29 |
+
self._vad_pipe.wait()
|
| 30 |
|
| 31 |
def translate(self, text, src_lang, dst_lang) -> MetaItem:
|
| 32 |
item = MetaItem(
|
|
|
|
| 42 |
self._whisper_pipe.input_queue.put(item)
|
| 43 |
return self._whisper_pipe.output_queue.get()
|
| 44 |
|
| 45 |
+
def voice_detect(self, audio_buffer:bytes) -> MetaItem:
|
| 46 |
+
item = MetaItem(source_audio=audio_buffer)
|
| 47 |
+
self._vad_pipe.input_queue.put(item)
|
| 48 |
+
return self._vad_pipe.output_queue.get()
|
| 49 |
+
|
| 50 |
+
|
| 51 |
|
| 52 |
if __name__ == "__main__":
|
| 53 |
import soundfile
|
| 54 |
tp = TranslatePipes()
|
| 55 |
# result = tp.translate("你好,今天天气怎么样?", src_lang="zh", dst_lang="en")
|
| 56 |
mel, _, = soundfile.read("assets/jfk.flac")
|
| 57 |
+
# result = tp.transcrible(mel, 'en')
|
| 58 |
+
result = tp.voice_detect(mel)
|
| 59 |
print(result)
|
| 60 |
|
| 61 |
|
transcribe/utils.py
CHANGED
|
@@ -2,6 +2,8 @@ import os
|
|
| 2 |
import textwrap
|
| 3 |
from pathlib import Path
|
| 4 |
import logging
|
|
|
|
|
|
|
| 5 |
|
| 6 |
|
| 7 |
import av
|
|
@@ -86,3 +88,7 @@ def resample(file: str, sr: int = 16000):
|
|
| 86 |
|
| 87 |
output_container.close()
|
| 88 |
return resampled_file
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 2 |
import textwrap
|
| 3 |
from pathlib import Path
|
| 4 |
import logging
|
| 5 |
+
import numpy as np
|
| 6 |
+
from scipy.io.wavfile import write
|
| 7 |
|
| 8 |
|
| 9 |
import av
|
|
|
|
| 88 |
|
| 89 |
output_container.close()
|
| 90 |
return resampled_file
|
| 91 |
+
|
| 92 |
+
|
| 93 |
+
def save_to_wave(filename, data:np.ndarray, sample_rate=16000):
|
| 94 |
+
write(filename, sample_rate, data)
|
transcribe/whisper_llm_serve.py
CHANGED
|
@@ -5,13 +5,13 @@ import multiprocessing as mp
|
|
| 5 |
import numpy as np
|
| 6 |
from logging import getLogger
|
| 7 |
|
| 8 |
-
import
|
| 9 |
import time
|
| 10 |
import json
|
| 11 |
import threading
|
| 12 |
from .server import ServeClientBase
|
| 13 |
-
|
| 14 |
-
|
| 15 |
from api_model import TransResult, Message
|
| 16 |
from .utils import log_block
|
| 17 |
from .translatepipes import TranslatePipes
|
|
@@ -23,14 +23,11 @@ translate_pipes = TranslatePipes()
|
|
| 23 |
translate_pipes.wait_ready()
|
| 24 |
logger.info("Pipeline is ready.")
|
| 25 |
|
| 26 |
-
def save_to_wave(filename, data:np.ndarray, sample_rate=16000):
|
| 27 |
-
write(filename, sample_rate, data)
|
| 28 |
|
| 29 |
|
| 30 |
|
| 31 |
class PyWhiperCppServe(ServeClientBase):
|
| 32 |
|
| 33 |
-
|
| 34 |
def __init__(self, websocket, language=None, dst_lang=None, client_uid=None,):
|
| 35 |
super().__init__(client_uid, websocket)
|
| 36 |
self.language = language
|
|
@@ -42,16 +39,23 @@ class PyWhiperCppServe(ServeClientBase):
|
|
| 42 |
|
| 43 |
self.lock = threading.Lock()
|
| 44 |
self.frames_np = None
|
|
|
|
|
|
|
| 45 |
self.sample_rate = 16000
|
| 46 |
|
| 47 |
self.send_ready_state()
|
| 48 |
|
| 49 |
-
self.
|
| 50 |
-
self.
|
| 51 |
-
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 52 |
|
| 53 |
def send_ready_state(self):
|
| 54 |
-
|
| 55 |
self.websocket.send(json.dumps({
|
| 56 |
"uid": self.client_uid,
|
| 57 |
"message": self.SERVER_READY,
|
|
@@ -63,12 +67,34 @@ class PyWhiperCppServe(ServeClientBase):
|
|
| 63 |
self.dst_lang = dst_lang
|
| 64 |
|
| 65 |
def add_frames(self, frame_np):
|
| 66 |
-
|
| 67 |
-
|
| 68 |
-
|
| 69 |
-
|
| 70 |
-
|
| 71 |
-
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 72 |
|
| 73 |
|
| 74 |
def update_audio_buffer(self, last_offset):
|
|
@@ -99,8 +125,10 @@ class PyWhiperCppServe(ServeClientBase):
|
|
| 99 |
log_block("LLM translate input", f"{text}")
|
| 100 |
start_time = time.perf_counter()
|
| 101 |
ret = translate_pipes.translate(text, self.language, self.dst_lang)
|
|
|
|
| 102 |
log_block("LLM translate time", f"{(time.perf_counter() - start_time):.3f}", "s")
|
| 103 |
-
|
|
|
|
| 104 |
|
| 105 |
|
| 106 |
|
|
@@ -109,8 +137,8 @@ class PyWhiperCppServe(ServeClientBase):
|
|
| 109 |
# 当左边确认后,右边段才会进入观察
|
| 110 |
# 当左边确认后,会从缓冲区中删除对应的buffer,减少下次输入的数据量
|
| 111 |
left_watch_idx, left_watch_sequences, right_watch_sequences, is_end_sentence = segments_split(segments, audio_buffer)
|
| 112 |
-
left_watch_string =
|
| 113 |
-
right_watch_string =
|
| 114 |
|
| 115 |
if left_watch_idx != 0:
|
| 116 |
# 将观察字符串临时存储
|
|
@@ -121,8 +149,8 @@ class PyWhiperCppServe(ServeClientBase):
|
|
| 121 |
|
| 122 |
# 整句消除 后两句之前的内容
|
| 123 |
left_watch_idx, left_watch_sequences, right_watch_sequences, is_end_sentence = sequences_split(segments, audio_buffer)
|
| 124 |
-
left_watch_string =
|
| 125 |
-
right_watch_string =
|
| 126 |
if left_watch_idx != 0:
|
| 127 |
return left_watch_idx, left_watch_string, right_watch_string, is_end_sentence
|
| 128 |
|
|
@@ -157,7 +185,7 @@ class PyWhiperCppServe(ServeClientBase):
|
|
| 157 |
# logger.error(f"{e}")
|
| 158 |
|
| 159 |
def handle_transcription_output(self, segments, audio_buffer):
|
| 160 |
-
texts =
|
| 161 |
if not len(texts):
|
| 162 |
return
|
| 163 |
self._segment_manager.handle(texts)
|
|
|
|
| 5 |
import numpy as np
|
| 6 |
from logging import getLogger
|
| 7 |
|
| 8 |
+
from .utils import save_to_wave
|
| 9 |
import time
|
| 10 |
import json
|
| 11 |
import threading
|
| 12 |
from .server import ServeClientBase
|
| 13 |
+
import queue
|
| 14 |
+
import collections
|
| 15 |
from api_model import TransResult, Message
|
| 16 |
from .utils import log_block
|
| 17 |
from .translatepipes import TranslatePipes
|
|
|
|
| 23 |
translate_pipes.wait_ready()
|
| 24 |
logger.info("Pipeline is ready.")
|
| 25 |
|
|
|
|
|
|
|
| 26 |
|
| 27 |
|
| 28 |
|
| 29 |
class PyWhiperCppServe(ServeClientBase):
|
| 30 |
|
|
|
|
| 31 |
def __init__(self, websocket, language=None, dst_lang=None, client_uid=None,):
|
| 32 |
super().__init__(client_uid, websocket)
|
| 33 |
self.language = language
|
|
|
|
| 39 |
|
| 40 |
self.lock = threading.Lock()
|
| 41 |
self.frames_np = None
|
| 42 |
+
self._frame_queue = queue.Queue()
|
| 43 |
+
self._previous_frame_queue = collections.deque(maxlen=2)
|
| 44 |
self.sample_rate = 16000
|
| 45 |
|
| 46 |
self.send_ready_state()
|
| 47 |
|
| 48 |
+
self.run_in_thread(self.speech_to_text)
|
| 49 |
+
self.run_in_thread(self.get_frame_from_queue)
|
| 50 |
+
|
| 51 |
+
self.text_sep = "" if self.language == "zh" else " "
|
| 52 |
+
|
| 53 |
+
def run_in_thread(self, func):
|
| 54 |
+
t = threading.Thread(target=func)
|
| 55 |
+
t.daemon = True
|
| 56 |
+
t.start()
|
| 57 |
|
| 58 |
def send_ready_state(self):
|
|
|
|
| 59 |
self.websocket.send(json.dumps({
|
| 60 |
"uid": self.client_uid,
|
| 61 |
"message": self.SERVER_READY,
|
|
|
|
| 67 |
self.dst_lang = dst_lang
|
| 68 |
|
| 69 |
def add_frames(self, frame_np):
|
| 70 |
+
self._frame_queue.put(frame_np)
|
| 71 |
+
|
| 72 |
+
def get_prev_frame(self, ):
|
| 73 |
+
if len(self._previous_frame_queue) == 2:
|
| 74 |
+
return self._previous_frame_queue[-1]
|
| 75 |
+
|
| 76 |
+
|
| 77 |
+
def get_frame_from_queue(self,):
|
| 78 |
+
while True:
|
| 79 |
+
try:
|
| 80 |
+
frame_np = self._frame_queue.get(timeout=0.1)
|
| 81 |
+
# frame_np = item.source_audio
|
| 82 |
+
# self._previous_frame_queue.appendleft(frame_np.copy())
|
| 83 |
+
# prev_frame_np = self.get_prev_frame()
|
| 84 |
+
# if prev_frame_np is not None:
|
| 85 |
+
# frame_np = np.concatenate([prev_frame_np[int(-0.05 * self.sample_rate):],frame_np], axis=0)
|
| 86 |
+
# item = translate_pipes.voice_detect(frame_np.tobytes())
|
| 87 |
+
# if item.audio == b"":
|
| 88 |
+
# continue
|
| 89 |
+
# frame_np = np.frombuffer(item.audio, dtype=np.float32)
|
| 90 |
+
with self.lock:
|
| 91 |
+
if self.frames_np is None:
|
| 92 |
+
self.frames_np = frame_np.copy()
|
| 93 |
+
else:
|
| 94 |
+
self.frames_np = np.append(self.frames_np,frame_np)
|
| 95 |
+
except queue.Empty:
|
| 96 |
+
pass
|
| 97 |
+
|
| 98 |
|
| 99 |
|
| 100 |
def update_audio_buffer(self, last_offset):
|
|
|
|
| 125 |
log_block("LLM translate input", f"{text}")
|
| 126 |
start_time = time.perf_counter()
|
| 127 |
ret = translate_pipes.translate(text, self.language, self.dst_lang)
|
| 128 |
+
translated_text = ret.translate_content
|
| 129 |
log_block("LLM translate time", f"{(time.perf_counter() - start_time):.3f}", "s")
|
| 130 |
+
log_block("LLM translate out", f"{translated_text}")
|
| 131 |
+
return translated_text
|
| 132 |
|
| 133 |
|
| 134 |
|
|
|
|
| 137 |
# 当左边确认后,右边段才会进入观察
|
| 138 |
# 当左边确认后,会从缓冲区中删除对应的buffer,减少下次输入的数据量
|
| 139 |
left_watch_idx, left_watch_sequences, right_watch_sequences, is_end_sentence = segments_split(segments, audio_buffer)
|
| 140 |
+
left_watch_string = self.text_sep.join(i.text for i in left_watch_sequences)
|
| 141 |
+
right_watch_string = self.text_sep.join(i.text for i in right_watch_sequences)
|
| 142 |
|
| 143 |
if left_watch_idx != 0:
|
| 144 |
# 将观察字符串临时存储
|
|
|
|
| 149 |
|
| 150 |
# 整句消除 后两句之前的内容
|
| 151 |
left_watch_idx, left_watch_sequences, right_watch_sequences, is_end_sentence = sequences_split(segments, audio_buffer)
|
| 152 |
+
left_watch_string = self.text_sep.join(i.text for i in left_watch_sequences)
|
| 153 |
+
right_watch_string = self.text_sep.join(i.text for i in right_watch_sequences)
|
| 154 |
if left_watch_idx != 0:
|
| 155 |
return left_watch_idx, left_watch_string, right_watch_string, is_end_sentence
|
| 156 |
|
|
|
|
| 185 |
# logger.error(f"{e}")
|
| 186 |
|
| 187 |
def handle_transcription_output(self, segments, audio_buffer):
|
| 188 |
+
texts = self.text_sep.join(i.text for i in segments)
|
| 189 |
if not len(texts):
|
| 190 |
return
|
| 191 |
self._segment_manager.handle(texts)
|
uv.lock
CHANGED
|
@@ -16,6 +16,20 @@ wheels = [
|
|
| 16 |
{ url = "https://files.pythonhosted.org/packages/78/b6/6307fbef88d9b5ee7421e68d78a9f162e0da4900bc5f5793f6d3d0e34fb8/annotated_types-0.7.0-py3-none-any.whl", hash = "sha256:1f02e8b43a8fbbc3f3e0d4f0f4bfc8131bcb4eebe8849b8e5c773f3a1c582a53", size = 13643 },
|
| 17 |
]
|
| 18 |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 19 |
[[package]]
|
| 20 |
name = "audioop-lts"
|
| 21 |
version = "0.2.1"
|
|
@@ -223,6 +237,20 @@ wheels = [
|
|
| 223 |
{ url = "https://files.pythonhosted.org/packages/4e/8c/f3147f5c4b73e7550fe5f9352eaa956ae838d5c51eb58e7a25b9f3e2643b/decorator-5.2.1-py3-none-any.whl", hash = "sha256:d316bb415a2d9e2d2b3abcc4084c6502fc09240e292cd76a76afc106a1c8e04a", size = 9190 },
|
| 224 |
]
|
| 225 |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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| 226 |
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| 1177 |
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|
| 1178 |
[[package]]
|
| 1179 |
name = "tqdm"
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| 1180 |
version = "4.67.1"
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|
| 1193 |
source = { virtual = "." }
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| 1194 |
dependencies = [
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| 1195 |
{ name = "av" },
|
| 1196 |
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{ name = "fastapi" },
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| 1197 |
{ name = "librosa" },
|
| 1198 |
{ name = "numpy" },
|
| 1199 |
{ name = "onnxruntime" },
|
| 1200 |
{ name = "pyaudio" },
|
| 1201 |
{ name = "pydantic" },
|
| 1202 |
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{ name = "pydub" },
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| 1203 |
{ name = "setuptools" },
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| 1204 |
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{ name = "silero-vad" },
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{ name = "soundfile" },
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| 1206 |
{ name = "torch" },
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| 1212 |
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| 1213 |
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| 1214 |
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| 1220 |
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| 1222 |
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| 1223 |
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| 1224 |
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| 1225 |
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| 1226 |
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