mirror of https://github.com/coqui-ai/TTS.git
113 lines
3.7 KiB
Python
113 lines
3.7 KiB
Python
import os
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import unittest
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import torch
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from trainer.generic_utils import count_parameters
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from tests import get_tests_input_path
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from TTS.vc.models.freevc import FreeVC, FreeVCConfig
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# pylint: disable=unused-variable
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# pylint: disable=no-self-use
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torch.manual_seed(1)
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use_cuda = torch.cuda.is_available()
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device = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")
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c = FreeVCConfig()
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WAV_FILE = os.path.join(get_tests_input_path(), "example_1.wav")
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BATCH_SIZE = 3
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class TestFreeVC(unittest.TestCase):
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def _create_inputs(self, config, batch_size=2):
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spec = torch.rand(batch_size, 30, config.audio["filter_length"] // 2 + 1).to(device)
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mel = torch.rand(batch_size, 30, config.audio["n_mel_channels"]).to(device)
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spec_lengths = torch.randint(20, 30, (batch_size,)).long().to(device)
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spec_lengths[-1] = spec.size(2)
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waveform = torch.rand(batch_size, spec.size(2) * config.audio["hop_length"]).to(device)
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return mel, spec, spec_lengths, waveform
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@staticmethod
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def _create_inputs_inference():
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source_wav = torch.rand(16000)
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target_wav = torch.rand(16000)
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return source_wav, target_wav
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def test_methods(self):
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config = FreeVCConfig()
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model = FreeVC(config).to(device)
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model.load_pretrained_speaker_encoder()
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model.init_multispeaker(config)
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wavlm_feats = model.extract_wavlm_features(torch.rand(1, 16000))
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assert wavlm_feats.shape == (1, 1024, 49), wavlm_feats.shape
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def test_load_audio(self):
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config = FreeVCConfig()
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model = FreeVC(config).to(device)
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wav = model.load_audio(WAV_FILE)
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wav2 = model.load_audio(wav)
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assert all(torch.isclose(wav, wav2))
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def _test_forward(self, batch_size):
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# create model
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config = FreeVCConfig()
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model = FreeVC(config).to(device)
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model.train()
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print(" > Num parameters for FreeVC model:%s" % (count_parameters(model)))
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mel, spec, spec_lengths, waveform = self._create_inputs(config, batch_size)
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wavlm_vec = model.extract_wavlm_features(waveform)
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wavlm_vec_lengths = torch.ones(batch_size, dtype=torch.long)
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y = model.forward(wavlm_vec, spec, None, mel, spec_lengths, wavlm_vec_lengths)
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# TODO: assert with training implementation
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def test_forward(self):
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self._test_forward(1)
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self._test_forward(3)
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def _test_inference(self, batch_size):
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config = FreeVCConfig()
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model = FreeVC(config).to(device)
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model.eval()
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mel, _, _, waveform = self._create_inputs(config, batch_size)
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wavlm_vec = model.extract_wavlm_features(waveform)
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wavlm_vec_lengths = torch.ones(batch_size, dtype=torch.long)
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output_wav = model.inference(wavlm_vec, None, mel, wavlm_vec_lengths)
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assert (
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output_wav.shape[-1] // config.audio.hop_length == wavlm_vec.shape[-1]
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), f"{output_wav.shape[-1] // config.audio.hop_length} != {wavlm_vec.shape}"
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def test_inference(self):
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self._test_inference(1)
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self._test_inference(3)
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def test_voice_conversion(self):
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config = FreeVCConfig()
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model = FreeVC(config).to(device)
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model.eval()
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source_wav, target_wav = self._create_inputs_inference()
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output_wav = model.voice_conversion(source_wav, target_wav)
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assert (
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output_wav.shape[0] + config.audio.hop_length == source_wav.shape[0]
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), f"{output_wav.shape} != {source_wav.shape}"
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def test_train_step(self): ...
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def test_train_eval_log(self): ...
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def test_test_run(self): ...
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def test_load_checkpoint(self): ...
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def test_get_criterion(self): ...
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def test_init_from_config(self): ...
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