Files
facefusion/facefusion/voice_extractor.py
T
Henry Ruhs 8bf9170577 3.5.0 (#977)
* Mark as NEXT

* Reduce caching to avoid RAM explosion

* Reduce caching to avoid RAM explosion

* Update dependencies

* add face-detector-pad-factor

* update facefusion.ini

* fix test

* change pad to margin

* fix order

* add prepare margin

* use 50% max margin

* Minor fixes part2

* Minor fixes part3

* Minor fixes part4

* Minor fixes part1

* Downgrade onnxruntime as of BiRefNet broken on CPU

add test

update

update facefusion.ini

add birefnet

* rename models

add more models

* Fix versions

* Add .claude to gitignore

* add normalize color

add 4 channel

add colors

* worflows

* cleanup

* cleanup

* cleanup

* cleanup

* add more models (#961)

* Fix naming

* changes

* Fix style and mock Gradio

* Fix style and mock Gradio

* Fix style and mock Gradio

* apply clamp

* remove clamp

* Add normalizer test

* Introduce sanitizer for the rescue (#963)

* Introduce sanitizer for the rescue

* Introduce sanitizer for the rescue

* Introduce sanitizer for the rescue

* prepare ffmpeg for alpha support

* Some cleanup

* Some cleanup

* Fix CI

* List as TypeAlias is not allowed (#967)

* List as TypeAlias is not allowed

* List as TypeAlias is not allowed

* List as TypeAlias is not allowed

* List as TypeAlias is not allowed

* Add mpeg and mxf support (#968)

* Add mpeg support

* Add mxf support

* Adjust fix_xxx_encoder for the new formats

* Extend output pattern for batch-run (#969)

* Extend output pattern for batch-run

* Add {target_extension} to allowed mixed files

* Catch invalid output pattern keys

* alpha support

* cleanup

* cleanup

* add ProcessorOutputs type

* fix preview and streamer, support alpha for background_remover

* Refactor/open close processors (#972)

* Introduce open/close processors

* Add locales for translator

* Introduce __autoload__ for translator

* More cleanup

* Fix import issues

* Resolve the scope situation for locals

* Fix installer by not using translator

* Fixes after merge

* Fixes after merge

* Fix translator keys in ui

* Use LOCALS in installer

* Update and partial fix DirectML

* Use latest onnxruntime

* Fix performance

* Fix lint issues

* fix mask

* fix lint

* fix lint

* Remove default from translator.get()

* remove 'framerate='

* fix test

* Rename and reorder models

* Align naming

* add alpha preview

* fix frame-by-frame

* Add alpha effect via css

* preview support alpha channel

* fix preview modes

* Use official assets repositories

* Add support for u2net_cloth

* fix naming

* Add more models

* Add vendor, license and year direct to the models

* Add vendor, license and year direct to the models

* Update dependencies, Minor CSS adjustment

* Ready for 3.5.0

* Fix naming

* Update about messages

* Fix return

* Use groups to show/hide

* Update preview

* Conditional merge mask

* Conditional merge mask

* Fix import order

---------

Co-authored-by: harisreedhar <h4harisreedhar.s.s@gmail.com>
Co-authored-by: Harisreedhar <46858047+harisreedhar@users.noreply.github.com>
2025-11-03 14:05:15 +01:00

214 lines
7.6 KiB
Python

from functools import lru_cache
from typing import Tuple
import numpy
import scipy
from facefusion import inference_manager, state_manager
from facefusion.download import conditional_download_hashes, conditional_download_sources, resolve_download_url
from facefusion.filesystem import resolve_relative_path
from facefusion.thread_helper import thread_semaphore
from facefusion.types import Audio, AudioChunk, DownloadScope, DownloadSet, InferencePool, ModelSet, Voice, VoiceChunk
@lru_cache()
def create_static_model_set(download_scope : DownloadScope) -> ModelSet:
return\
{
'kim_vocal_1':
{
'__metadata__':
{
'vendor': 'KimberleyJensen',
'license': 'Non-Commercial',
'year': 2023
},
'hashes':
{
'voice_extractor':
{
'url': resolve_download_url('models-3.4.0', 'kim_vocal_1.hash'),
'path': resolve_relative_path('../.assets/models/kim_vocal_1.hash')
}
},
'sources':
{
'voice_extractor':
{
'url': resolve_download_url('models-3.4.0', 'kim_vocal_1.onnx'),
'path': resolve_relative_path('../.assets/models/kim_vocal_1.onnx')
}
}
},
'kim_vocal_2':
{
'__metadata__':
{
'vendor': 'KimberleyJensen',
'license': 'Non-Commercial',
'year': 2023
},
'hashes':
{
'voice_extractor':
{
'url': resolve_download_url('models-3.0.0', 'kim_vocal_2.hash'),
'path': resolve_relative_path('../.assets/models/kim_vocal_2.hash')
}
},
'sources':
{
'voice_extractor':
{
'url': resolve_download_url('models-3.0.0', 'kim_vocal_2.onnx'),
'path': resolve_relative_path('../.assets/models/kim_vocal_2.onnx')
}
}
},
'uvr_mdxnet':
{
'__metadata__':
{
'vendor': 'Unknown',
'license': 'Non-Commercial',
'year': 2023
},
'hashes':
{
'voice_extractor':
{
'url': resolve_download_url('models-3.4.0', 'uvr_mdxnet.hash'),
'path': resolve_relative_path('../.assets/models/uvr_mdxnet.hash')
}
},
'sources':
{
'voice_extractor':
{
'url': resolve_download_url('models-3.4.0', 'uvr_mdxnet.onnx'),
'path': resolve_relative_path('../.assets/models/uvr_mdxnet.onnx')
}
}
}
}
def get_inference_pool() -> InferencePool:
model_names = [ state_manager.get_item('voice_extractor_model') ]
_, model_source_set = collect_model_downloads()
return inference_manager.get_inference_pool(__name__, model_names, model_source_set)
def clear_inference_pool() -> None:
model_names = [ state_manager.get_item('voice_extractor_model') ]
inference_manager.clear_inference_pool(__name__, model_names)
def collect_model_downloads() -> Tuple[DownloadSet, DownloadSet]:
model_set = create_static_model_set('full')
model_hash_set = {}
model_source_set = {}
for voice_extractor_model in [ 'kim_vocal_1', 'kim_vocal_2', 'uvr_mdxnet' ]:
if state_manager.get_item('voice_extractor_model') == voice_extractor_model:
model_hash_set[voice_extractor_model] = model_set.get(voice_extractor_model).get('hashes').get('voice_extractor')
model_source_set[voice_extractor_model] = model_set.get(voice_extractor_model).get('sources').get('voice_extractor')
return model_hash_set, model_source_set
def pre_check() -> bool:
model_hash_set, model_source_set = collect_model_downloads()
return conditional_download_hashes(model_hash_set) and conditional_download_sources(model_source_set)
def batch_extract_voice(audio : Audio, chunk_size : int, step_size : int) -> Voice:
temp_voice = numpy.zeros((audio.shape[0], 2)).astype(numpy.float32)
temp_voice_chunk = numpy.zeros((audio.shape[0], 2)).astype(numpy.float32)
for start in range(0, audio.shape[0], step_size):
end = min(start + chunk_size, audio.shape[0])
temp_voice[start:end, ...] += extract_voice(audio[start:end, ...])
temp_voice_chunk[start:end, ...] += 1
voice = temp_voice / temp_voice_chunk
return voice
def extract_voice(temp_audio_chunk : AudioChunk) -> VoiceChunk:
voice_extractor = get_inference_pool().get(state_manager.get_item('voice_extractor_model'))
voice_trim_size = 3840
voice_chunk_size = (voice_extractor.get_inputs()[0].shape[3] - 1) * 1024
temp_audio_chunk, audio_pad_size = prepare_audio_chunk(temp_audio_chunk.T, voice_chunk_size, voice_trim_size)
temp_audio_chunk = decompose_audio_chunk(temp_audio_chunk, voice_trim_size)
temp_audio_chunk = forward(temp_audio_chunk)
temp_audio_chunk = compose_audio_chunk(temp_audio_chunk, voice_trim_size)
temp_audio_chunk = normalize_audio_chunk(temp_audio_chunk, voice_chunk_size, voice_trim_size, audio_pad_size)
return temp_audio_chunk
def forward(temp_audio_chunk : AudioChunk) -> AudioChunk:
voice_extractor = get_inference_pool().get(state_manager.get_item('voice_extractor_model'))
with thread_semaphore():
temp_audio_chunk = voice_extractor.run(None,
{
'input': temp_audio_chunk
})[0]
return temp_audio_chunk
def prepare_audio_chunk(temp_audio_chunk : AudioChunk, chunk_size : int, audio_trim_size : int) -> Tuple[AudioChunk, int]:
audio_step_size = chunk_size - 2 * audio_trim_size
audio_pad_size = audio_step_size - temp_audio_chunk.shape[1] % audio_step_size
audio_chunk_size = temp_audio_chunk.shape[1] + audio_pad_size
temp_audio_chunk = temp_audio_chunk.astype(numpy.float32) / numpy.iinfo(numpy.int16).max
temp_audio_chunk = numpy.pad(temp_audio_chunk, ((0, 0), (audio_trim_size, audio_trim_size + audio_pad_size)))
temp_audio_chunks = []
for index in range(0, audio_chunk_size, audio_step_size):
temp_audio_chunks.append(temp_audio_chunk[:, index:index + chunk_size])
temp_audio_chunk = numpy.concatenate(temp_audio_chunks, axis = 0)
temp_audio_chunk = temp_audio_chunk.reshape((-1, chunk_size))
return temp_audio_chunk, audio_pad_size
def decompose_audio_chunk(temp_audio_chunk : AudioChunk, audio_trim_size : int) -> AudioChunk:
audio_frame_size = 7680
audio_frame_overlap = 6656
audio_frame_total = 3072
audio_bin_total = 256
audio_channel_total = 4
window = scipy.signal.windows.hann(audio_frame_size)
temp_audio_chunk = scipy.signal.stft(temp_audio_chunk, nperseg = audio_frame_size, noverlap = audio_frame_overlap, window = window)[2]
temp_audio_chunk = numpy.stack((numpy.real(temp_audio_chunk), numpy.imag(temp_audio_chunk)), axis = -1).transpose((0, 3, 1, 2))
temp_audio_chunk = temp_audio_chunk.reshape(-1, 2, 2, audio_trim_size + 1, audio_bin_total).reshape(-1, audio_channel_total, audio_trim_size + 1, audio_bin_total)
temp_audio_chunk = temp_audio_chunk[:, :, :audio_frame_total]
temp_audio_chunk /= numpy.sqrt(1.0 / window.sum() ** 2)
return temp_audio_chunk
def compose_audio_chunk(temp_audio_chunk : AudioChunk, audio_trim_size : int) -> AudioChunk:
audio_frame_size = 7680
audio_frame_overlap = 6656
audio_frame_total = 3072
audio_bin_total = 256
window = scipy.signal.windows.hann(audio_frame_size)
temp_audio_chunk = numpy.pad(temp_audio_chunk, ((0, 0), (0, 0), (0, audio_trim_size + 1 - audio_frame_total), (0, 0)))
temp_audio_chunk = temp_audio_chunk.reshape(-1, 2, audio_trim_size + 1, audio_bin_total).transpose((0, 2, 3, 1))
temp_audio_chunk = temp_audio_chunk[:, :, :, 0] + 1j * temp_audio_chunk[:, :, :, 1]
temp_audio_chunk = scipy.signal.istft(temp_audio_chunk, nperseg = audio_frame_size, noverlap = audio_frame_overlap, window = window)[1]
temp_audio_chunk *= numpy.sqrt(1.0 / window.sum() ** 2)
return temp_audio_chunk
def normalize_audio_chunk(temp_audio_chunk : AudioChunk, chunk_size : int, audio_trim_size : int, audio_pad_size : int) -> AudioChunk:
temp_audio_chunk = temp_audio_chunk.reshape((-1, 2, chunk_size))
temp_audio_chunk = temp_audio_chunk[:, :, audio_trim_size:-audio_trim_size].transpose(1, 0, 2)
temp_audio_chunk = temp_audio_chunk.reshape(2, -1)[:, :-audio_pad_size].T
return temp_audio_chunk