Files
facefusion/facefusion/voice_extractor.py
Henry Ruhs 4ccf4c24c7 Next (#477)
* Add real_hatgan_x4 model

* Mark it as NEXT

* Force download to be executed and exit

* Fix frame per second interpolation

* 5 to 68 landmark (#456)

* changes

* changes

* Adjust model url

* Cleanup 5 to 68 landmark convertion

* Move everything to face analyser

* Introduce matrix only face helper

* Revert facefusion.ini

* Adjust limit due false positive analysis

* changes (#457)

* Use pixel format yuv422p to merge video

* Fix some code

* Minor cleanup

* Add gpen_bfr_1024 and gpen_bfr_2048

* Revert it back to yuv420p due compatibility issues

* Add debug back to ffmpeg

* Add debug back to ffmpeg

* Migrate to conda (#461)

* Migrate from venv to conda

* Migrate from venv to conda

* Message when conda is not activated

* Use release for every slider (#463)

* Use release event handler for every slider

* Move more sliders to release handler

* Move more sliders to release handler

* Add get_ui_components() to simplify code

* Revert some changes on frame slider

* Add the first iteration of a frame colorizer

* Support for the DDColor model

* Improve model file handling

* Improve model file handling part2

* Remove deoldify

* Remove deoldify

* Voice separator (#468)

* changes

* changes

* changes

* changes

* changes

* changes

* Rename audio extractor to voice extractor

* Cosmetic changes

* Cosmetic changes

* Fix fps lowering and boosting

* Fix fps lowering and boosting

* Fix fps lowering and boosting

* Some refactoring for audio.py and some astype() here and there (#470)

* Some refactoring for audio.py and some astype() here and there

* Fix lint

* Spacing

* Add mp3 to benchmark suite for lip syncer testing

* Improve naming

* Adjust chunk size

* Use higher quality

* Revert "Use higher quality"

This reverts commit d32f28757251ecc0f48214073adf54f3631b1289.

* Improve naming in ffmpeg.py

* Simplify code

* Better fps calculation

* Fix naming here and there

* Add back real esrgan x2

* Remove trailing comma

* Update wording and README

* Use semaphore to prevent frame colorizer memory issues

* Revert "Remove deoldify"

This reverts commit bd8034cbc71fe701f78dddec3057dc98593b2162.

* Remove unused type from frame colorizer

* Adjust naming

* Add missing clear of model initializer

* Change nvenc preset mappping to support old FFMPEG 4

* Update onnxruntime to 1.17.1

* Fix lint

* Prepare 2.5.0

* Fix Gradio overrides

* Add Deoldify Artistic back

* Feat/audio refactoring (#476)

* Improve audio naming and variables

* Improve audio naming and variables

* Refactor voice extractor like crazy

* Refactor voice extractor like crazy

* Remove spaces

* Update the usage

---------

Co-authored-by: Harisreedhar <46858047+harisreedhar@users.noreply.github.com>
2024-04-09 15:40:55 +02:00

133 lines
5.5 KiB
Python

from typing import Any, Tuple
from time import sleep
import threading
import scipy
import numpy
import onnxruntime
import facefusion.globals
from facefusion import process_manager
from facefusion.typing import ModelSet, AudioChunk, Audio
from facefusion.execution import apply_execution_provider_options
from facefusion.filesystem import resolve_relative_path, is_file
from facefusion.download import conditional_download
VOICE_EXTRACTOR = None
THREAD_SEMAPHORE : threading.Semaphore = threading.Semaphore()
THREAD_LOCK : threading.Lock = threading.Lock()
MODELS : ModelSet =\
{
'voice_extractor':
{
'url': 'https://github.com/facefusion/facefusion-assets/releases/download/models/voice_extractor.onnx',
'path': resolve_relative_path('../.assets/models/voice_extractor.onnx')
}
}
def get_voice_extractor() -> Any:
global VOICE_EXTRACTOR
with THREAD_LOCK:
while process_manager.is_checking():
sleep(0.5)
if VOICE_EXTRACTOR is None:
model_path = MODELS.get('voice_extractor').get('path')
VOICE_EXTRACTOR = onnxruntime.InferenceSession(model_path, providers = apply_execution_provider_options(facefusion.globals.execution_providers))
return VOICE_EXTRACTOR
def clear_voice_extractor() -> None:
global VOICE_EXTRACTOR
VOICE_EXTRACTOR = None
def pre_check() -> bool:
download_directory_path = resolve_relative_path('../.assets/models')
model_url = MODELS.get('voice_extractor').get('url')
model_path = MODELS.get('voice_extractor').get('path')
if not facefusion.globals.skip_download:
process_manager.check()
conditional_download(download_directory_path, [ model_url ])
process_manager.end()
return is_file(model_path)
def batch_extract_voice(audio : Audio, chunk_size : int, step_size : int) -> Audio:
temp_audio = numpy.zeros((audio.shape[0], 2)).astype(numpy.float32)
temp_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_audio[start:end, ...] += extract_voice(audio[start:end, ...])
temp_chunk[start:end, ...] += 1
audio = temp_audio / temp_chunk
return audio
def extract_voice(temp_audio_chunk : AudioChunk) -> AudioChunk:
voice_extractor = get_voice_extractor()
chunk_size = 1024 * (voice_extractor.get_inputs()[0].shape[3] - 1)
trim_size = 3840
temp_audio_chunk, pad_size = prepare_audio_chunk(temp_audio_chunk.T, chunk_size, trim_size)
temp_audio_chunk = decompose_audio_chunk(temp_audio_chunk, trim_size)
with THREAD_SEMAPHORE:
temp_audio_chunk = voice_extractor.run(None,
{
voice_extractor.get_inputs()[0].name: temp_audio_chunk
})[0]
temp_audio_chunk = compose_audio_chunk(temp_audio_chunk, trim_size)
temp_audio_chunk = normalize_audio_chunk(temp_audio_chunk, chunk_size, trim_size, pad_size)
return temp_audio_chunk
def prepare_audio_chunk(temp_audio_chunk : AudioChunk, chunk_size : int, trim_size : int) -> Tuple[AudioChunk, int]:
step_size = chunk_size - 2 * trim_size
pad_size = step_size - temp_audio_chunk.shape[1] % step_size
audio_chunk_size = temp_audio_chunk.shape[1] + 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), (trim_size, trim_size + pad_size)))
temp_audio_chunks = []
for index in range(0, audio_chunk_size, step_size):
temp_audio_chunk = temp_audio_chunk[:, index:index + chunk_size]
temp_audio_chunks.append(temp_audio_chunk)
temp_audio_chunk = numpy.concatenate(temp_audio_chunks, axis = 0)
temp_audio_chunk = temp_audio_chunk.reshape((-1, chunk_size))
return temp_audio_chunk, pad_size
def decompose_audio_chunk(temp_audio_chunk : AudioChunk, trim_size : int) -> AudioChunk:
frame_size = 7680
frame_overlap = 6656
voice_extractor_shape = get_voice_extractor().get_inputs()[0].shape
window = scipy.signal.windows.hann(frame_size)
temp_audio_chunk = scipy.signal.stft(temp_audio_chunk, nperseg = frame_size, noverlap = 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, trim_size + 1, voice_extractor_shape[3]).reshape(-1, voice_extractor_shape[1], trim_size + 1, voice_extractor_shape[3])
temp_audio_chunk = temp_audio_chunk[:, :, :voice_extractor_shape[2]]
temp_audio_chunk /= numpy.sqrt(1.0 / window.sum() ** 2)
return temp_audio_chunk
def compose_audio_chunk(temp_audio_chunk : AudioChunk, trim_size : int) -> AudioChunk:
frame_size = 7680
frame_overlap = 6656
voice_extractor_shape = get_voice_extractor().get_inputs()[0].shape
window = scipy.signal.windows.hann(frame_size)
temp_audio_chunk = numpy.pad(temp_audio_chunk, ((0, 0), (0, 0), (0, trim_size + 1 - voice_extractor_shape[2]), (0, 0)))
temp_audio_chunk = temp_audio_chunk.reshape(-1, 2, trim_size + 1, voice_extractor_shape[3]).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 = frame_size, noverlap = 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, trim_size : int, pad_size : int) -> AudioChunk:
temp_audio_chunk = temp_audio_chunk.reshape((-1, 2, chunk_size))
temp_audio_chunk = temp_audio_chunk[:, :, trim_size:-trim_size].transpose(1, 0, 2)
temp_audio_chunk = temp_audio_chunk.reshape(2, -1)[:, :-pad_size].T
return temp_audio_chunk