voice-changer/server/voice_changer/DiffusionSVC/pipeline/Pipeline.py
w-okada b215f3ba84 Modification:
- Timer update
  - Diffusion SVC Performance monitor
2023-12-21 04:11:25 +09:00

194 lines
7.4 KiB
Python

from typing import Any
import torch
import torch.nn.functional as F
from torch.cuda.amp import autocast
from Exceptions import (
DeviceCannotSupportHalfPrecisionException,
DeviceChangingException,
HalfPrecisionChangingException,
NotEnoughDataExtimateF0,
)
from mods.log_control import VoiceChangaerLogger
from voice_changer.DiffusionSVC.inferencer.Inferencer import Inferencer
from voice_changer.DiffusionSVC.pitchExtractor.PitchExtractor import PitchExtractor
from voice_changer.RVC.embedder.Embedder import Embedder
from voice_changer.common.VolumeExtractor import VolumeExtractor
from torchaudio.transforms import Resample
from voice_changer.utils.Timer import Timer2
logger = VoiceChangaerLogger.get_instance().getLogger()
class Pipeline(object):
embedder: Embedder
inferencer: Inferencer
pitchExtractor: PitchExtractor
index: Any | None
big_npy: Any | None
# feature: Any | None
targetSR: int
device: torch.device
isHalf: bool
def __init__(
self,
embedder: Embedder,
inferencer: Inferencer,
pitchExtractor: PitchExtractor,
# index: Any | None,
targetSR,
device,
isHalf,
resamplerIn: Resample,
resamplerOut: Resample,
):
self.inferencer = inferencer
inferencer_block_size, inferencer_sampling_rate = inferencer.getConfig()
self.hop_size = inferencer_block_size * 16000 / inferencer_sampling_rate # 16000はオーディオのサンプルレート。16Kで処理
self.inferencer_block_size = inferencer_block_size
self.inferencer_sampling_rate = inferencer_sampling_rate
self.volumeExtractor = VolumeExtractor(self.hop_size)
self.embedder = embedder
self.pitchExtractor = pitchExtractor
self.resamplerIn = resamplerIn
self.resamplerOut = resamplerOut
logger.info("VOLUME EXTRACTOR" + str(self.volumeExtractor))
logger.info("GENERATE INFERENCER" + str(self.inferencer))
logger.info("GENERATE EMBEDDER" + str(self.embedder))
logger.info("GENERATE PITCH EXTRACTOR" + str(self.pitchExtractor))
self.targetSR = targetSR
self.device = device
self.isHalf = False
def getPipelineInfo(self):
volumeExtractorInfo = self.volumeExtractor.getVolumeExtractorInfo()
inferencerInfo = self.inferencer.getInferencerInfo() if self.inferencer else {}
embedderInfo = self.embedder.getEmbedderInfo()
pitchExtractorInfo = self.pitchExtractor.getPitchExtractorInfo()
return {"volumeExtractor": volumeExtractorInfo, "inferencer": inferencerInfo, "embedder": embedderInfo, "pitchExtractor": pitchExtractorInfo, "isHalf": self.isHalf}
def setPitchExtractor(self, pitchExtractor: PitchExtractor):
self.pitchExtractor = pitchExtractor
@torch.no_grad()
def extract_volume_and_mask(self, audio: torch.Tensor, threshold: float):
volume_t = self.volumeExtractor.extract_t(audio)
mask = self.volumeExtractor.get_mask_from_volume_t(volume_t, self.inferencer_block_size, threshold=threshold)
volume = volume_t.unsqueeze(-1).unsqueeze(0)
return volume, mask
def exec(
self,
sid,
audio, # torch.tensor [n]
sr,
pitchf, # np.array [m]
feature, # np.array [m, feat]
f0_up_key,
k_step,
infer_speedup,
silence_front,
embOutputLayer,
useFinalProj,
protect=0.5,
skip_diffusion=True,
):
use_timer = False
# print("---------- pipe line --------------------")
with Timer2("pre-process", use_timer) as t:
audio_t = torch.from_numpy(audio).float().unsqueeze(0).to(self.device)
audio16k = self.resamplerIn(audio_t)
volume, mask = self.extract_volume_and_mask(audio16k, threshold=-60.0)
sid = torch.tensor(sid, device=self.device).unsqueeze(0).long()
n_frames = int(audio16k.size(-1) // self.hop_size + 1)
# print("[Timer::1: ]", t.secs)
with Timer2("extract pitch", use_timer) as t:
# ピッチ検出
try:
# pitch = self.pitchExtractor.extract(
# audio16k.squeeze(),
# pitchf,
# f0_up_key,
# int(self.hop_size), # 処理のwindowサイズ (44100における512)
# silence_front=silence_front,
# )
pitch = self.pitchExtractor.extract(
audio,
sr,
self.inferencer_block_size,
self.inferencer_sampling_rate,
pitchf,
f0_up_key,
silence_front=silence_front,
)
pitch = torch.tensor(pitch[-n_frames:], device=self.device, dtype=torch.float).unsqueeze(0).long()
except IndexError as e: # NOQA
raise NotEnoughDataExtimateF0()
# tensor型調整
feats = audio16k.squeeze()
if feats.dim() == 2: # double channels
feats = feats.mean(-1)
feats = feats.view(1, -1)
# print("[Timer::2: ]", t.secs)
with Timer2("extract feature", use_timer) as t:
# embedding
with autocast(enabled=self.isHalf):
try:
feats = self.embedder.extractFeatures(feats, embOutputLayer, useFinalProj)
if torch.isnan(feats).all():
raise DeviceCannotSupportHalfPrecisionException()
except RuntimeError as e:
if "HALF" in e.__str__().upper():
raise HalfPrecisionChangingException()
elif "same device" in e.__str__():
raise DeviceChangingException()
else:
raise e
feats = F.interpolate(feats.permute(0, 2, 1), size=int(n_frames), mode="nearest").permute(0, 2, 1)
# print("[Timer::3: ]", t.secs)
with Timer2("infer", use_timer) as t:
# 推論実行
try:
with torch.no_grad():
with autocast(enabled=self.isHalf):
audio1 = (
torch.clip(
self.inferencer.infer(audio16k, feats, pitch.unsqueeze(-1), volume, mask, sid, k_step, infer_speedup, silence_front=silence_front, skip_diffusion=skip_diffusion).to(dtype=torch.float32),
-1.0,
1.0,
)
* 32767.5
).data.to(dtype=torch.int16)
except RuntimeError as e:
if "HALF" in e.__str__().upper():
print("11", e)
raise HalfPrecisionChangingException()
else:
raise e
# print("[Timer::4: ]", t.secs)
with Timer2("post-process", use_timer) as t: # NOQA
feats_buffer = feats.squeeze(0).detach().cpu()
if pitch is not None:
pitch_buffer = pitch.squeeze(0).detach().cpu()
else:
pitch_buffer = None
del pitch, pitchf, feats, sid
audio1 = self.resamplerOut(audio1.float())
# print("[Timer::5: ]", t.secs)
return audio1, pitch_buffer, feats_buffer