147 lines
5.4 KiB
Python
147 lines
5.4 KiB
Python
from functools import lru_cache
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from typing import Tuple
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import numpy
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import scipy
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from facefusion import inference_manager
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from facefusion.download import conditional_download_hashes, conditional_download_sources, resolve_download_url
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from facefusion.filesystem import resolve_relative_path
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from facefusion.thread_helper import thread_semaphore
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from facefusion.typing import Audio, AudioChunk, DownloadScope, InferencePool, ModelOptions, ModelSet
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@lru_cache(maxsize = None)
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def create_static_model_set(download_scope : DownloadScope) -> ModelSet:
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return\
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{
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'kim_vocal_2':
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{
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'hashes':
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{
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'voice_extractor':
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{
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'url': resolve_download_url('models-3.0.0', 'kim_vocal_2.hash'),
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'path': resolve_relative_path('../.assets/models/kim_vocal_2.hash')
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}
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},
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'sources':
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{
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'voice_extractor':
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{
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'url': resolve_download_url('models-3.0.0', 'kim_vocal_2.onnx'),
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'path': resolve_relative_path('../.assets/models/kim_vocal_2.onnx')
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}
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}
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}
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}
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def get_inference_pool() -> InferencePool:
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model_sources = get_model_options().get('sources')
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return inference_manager.get_inference_pool(__name__, model_sources)
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def clear_inference_pool() -> None:
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inference_manager.clear_inference_pool(__name__)
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def get_model_options() -> ModelOptions:
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return create_static_model_set('full').get('kim_vocal_2')
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def pre_check() -> bool:
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model_hashes = get_model_options().get('hashes')
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model_sources = get_model_options().get('sources')
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return conditional_download_hashes(model_hashes) and conditional_download_sources(model_sources)
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def batch_extract_voice(audio : Audio, chunk_size : int, step_size : int) -> Audio:
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temp_audio = numpy.zeros((audio.shape[0], 2)).astype(numpy.float32)
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temp_chunk = numpy.zeros((audio.shape[0], 2)).astype(numpy.float32)
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for start in range(0, audio.shape[0], step_size):
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end = min(start + chunk_size, audio.shape[0])
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temp_audio[start:end, ...] += extract_voice(audio[start:end, ...])
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temp_chunk[start:end, ...] += 1
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audio = temp_audio / temp_chunk
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return audio
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def extract_voice(temp_audio_chunk : AudioChunk) -> AudioChunk:
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voice_extractor = get_inference_pool().get('voice_extractor')
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chunk_size = (voice_extractor.get_inputs()[0].shape[3] - 1) * 1024
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trim_size = 3840
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temp_audio_chunk, pad_size = prepare_audio_chunk(temp_audio_chunk.T, chunk_size, trim_size)
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temp_audio_chunk = decompose_audio_chunk(temp_audio_chunk, trim_size)
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temp_audio_chunk = forward(temp_audio_chunk)
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temp_audio_chunk = compose_audio_chunk(temp_audio_chunk, trim_size)
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temp_audio_chunk = normalize_audio_chunk(temp_audio_chunk, chunk_size, trim_size, pad_size)
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return temp_audio_chunk
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def forward(temp_audio_chunk : AudioChunk) -> AudioChunk:
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voice_extractor = get_inference_pool().get('voice_extractor')
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with thread_semaphore():
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temp_audio_chunk = voice_extractor.run(None,
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{
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'input': temp_audio_chunk
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})[0]
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return temp_audio_chunk
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def prepare_audio_chunk(temp_audio_chunk : AudioChunk, chunk_size : int, trim_size : int) -> Tuple[AudioChunk, int]:
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step_size = chunk_size - 2 * trim_size
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pad_size = step_size - temp_audio_chunk.shape[1] % step_size
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audio_chunk_size = temp_audio_chunk.shape[1] + pad_size
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temp_audio_chunk = temp_audio_chunk.astype(numpy.float32) / numpy.iinfo(numpy.int16).max
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temp_audio_chunk = numpy.pad(temp_audio_chunk, ((0, 0), (trim_size, trim_size + pad_size)))
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temp_audio_chunks = []
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for index in range(0, audio_chunk_size, step_size):
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temp_audio_chunks.append(temp_audio_chunk[:, index:index + chunk_size])
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temp_audio_chunk = numpy.concatenate(temp_audio_chunks, axis = 0)
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temp_audio_chunk = temp_audio_chunk.reshape((-1, chunk_size))
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return temp_audio_chunk, pad_size
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def decompose_audio_chunk(temp_audio_chunk : AudioChunk, trim_size : int) -> AudioChunk:
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frame_size = 7680
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frame_overlap = 6656
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frame_total = 3072
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bin_total = 256
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channel_total = 4
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window = scipy.signal.windows.hann(frame_size)
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temp_audio_chunk = scipy.signal.stft(temp_audio_chunk, nperseg = frame_size, noverlap = frame_overlap, window = window)[2]
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temp_audio_chunk = numpy.stack((numpy.real(temp_audio_chunk), numpy.imag(temp_audio_chunk)), axis = -1).transpose((0, 3, 1, 2))
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temp_audio_chunk = temp_audio_chunk.reshape(-1, 2, 2, trim_size + 1, bin_total).reshape(-1, channel_total, trim_size + 1, bin_total)
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temp_audio_chunk = temp_audio_chunk[:, :, :frame_total]
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temp_audio_chunk /= numpy.sqrt(1.0 / window.sum() ** 2)
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return temp_audio_chunk
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def compose_audio_chunk(temp_audio_chunk : AudioChunk, trim_size : int) -> AudioChunk:
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frame_size = 7680
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frame_overlap = 6656
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frame_total = 3072
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bin_total = 256
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window = scipy.signal.windows.hann(frame_size)
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temp_audio_chunk = numpy.pad(temp_audio_chunk, ((0, 0), (0, 0), (0, trim_size + 1 - frame_total), (0, 0)))
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temp_audio_chunk = temp_audio_chunk.reshape(-1, 2, trim_size + 1, bin_total).transpose((0, 2, 3, 1))
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temp_audio_chunk = temp_audio_chunk[:, :, :, 0] + 1j * temp_audio_chunk[:, :, :, 1]
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temp_audio_chunk = scipy.signal.istft(temp_audio_chunk, nperseg = frame_size, noverlap = frame_overlap, window = window)[1]
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temp_audio_chunk *= numpy.sqrt(1.0 / window.sum() ** 2)
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return temp_audio_chunk
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def normalize_audio_chunk(temp_audio_chunk : AudioChunk, chunk_size : int, trim_size : int, pad_size : int) -> AudioChunk:
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temp_audio_chunk = temp_audio_chunk.reshape((-1, 2, chunk_size))
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temp_audio_chunk = temp_audio_chunk[:, :, trim_size:-trim_size].transpose(1, 0, 2)
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temp_audio_chunk = temp_audio_chunk.reshape(2, -1)[:, :-pad_size].T
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return temp_audio_chunk
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