REBASED: Transform Speaker Encoder in a Generic Encoder and Implement Emotion Encoder training support (#1349)

* Rename Speaker encoder module to encoder

* Add a generic emotion dataset formatter

* Transform the Speaker Encoder dataset to a generic dataset and create emotion encoder config

* Add class map in emotion config

* Add Base encoder config

* Add evaluation encoder script

* Fix the bug in plot_embeddings

* Enable Weight decay for encoder training

* Add argumnet to disable storage

* Add Perfect Sampler and remove storage

* Add evaluation during encoder training

* Fix lint checks

* Remove useless config parameter

* Active evaluation in speaker encoder test and use multispeaker dataset for this test

* Unit tests fixs

* Remove useless tests for speedup the aux_tests

* Use get_optimizer in Encoder

* Add BaseEncoder Class

* Fix the unitests

* Add Perfect Batch Sampler unit test

* Add compute encoder accuracy in a function
This commit is contained in:
Edresson Casanova
2022-03-11 14:43:40 +01:00
committed by GitHub
parent 36e9ea2f97
commit f81892483d
40 changed files with 962 additions and 2791 deletions
+3 -3
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@@ -3,9 +3,9 @@ import unittest
import torch as T
from tests import get_tests_input_path
from TTS.speaker_encoder.losses import AngleProtoLoss, GE2ELoss, SoftmaxAngleProtoLoss
from TTS.speaker_encoder.models.lstm import LSTMSpeakerEncoder
from TTS.speaker_encoder.models.resnet import ResNetSpeakerEncoder
from TTS.encoder.losses import AngleProtoLoss, GE2ELoss, SoftmaxAngleProtoLoss
from TTS.encoder.models.lstm import LSTMSpeakerEncoder
from TTS.encoder.models.resnet import ResNetSpeakerEncoder
file_path = get_tests_input_path()
+17 -13
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@@ -4,14 +4,14 @@ import shutil
from tests import get_device_id, get_tests_output_path, run_cli
from TTS.config.shared_configs import BaseAudioConfig
from TTS.speaker_encoder.speaker_encoder_config import SpeakerEncoderConfig
from TTS.encoder.configs.speaker_encoder_config import SpeakerEncoderConfig
def run_test_train():
command = (
f"CUDA_VISIBLE_DEVICES='{get_device_id()}' python TTS/bin/train_encoder.py --config_path {config_path} "
f"--coqpit.output_path {output_path} "
"--coqpit.datasets.0.name ljspeech "
"--coqpit.datasets.0.name ljspeech_test "
"--coqpit.datasets.0.meta_file_train metadata.csv "
"--coqpit.datasets.0.meta_file_val metadata.csv "
"--coqpit.datasets.0.path tests/data/ljspeech "
@@ -24,17 +24,21 @@ output_path = os.path.join(get_tests_output_path(), "train_outputs")
config = SpeakerEncoderConfig(
batch_size=4,
num_speakers_in_batch=1,
num_utters_per_speaker=10,
num_loader_workers=0,
max_train_step=2,
num_classes_in_batch=4,
num_utter_per_class=2,
eval_num_classes_in_batch=4,
eval_num_utter_per_class=2,
num_loader_workers=1,
epochs=1,
print_step=1,
save_step=1,
save_step=2,
print_eval=True,
run_eval=True,
audio=BaseAudioConfig(num_mels=80),
)
config.audio.do_trim_silence = True
config.audio.trim_db = 60
config.loss = "ge2e"
config.save_json(config_path)
print(config)
@@ -69,14 +73,14 @@ run_cli(command_train)
shutil.rmtree(continue_path)
# test model with ge2e loss function
config.loss = "ge2e"
config.save_json(config_path)
run_test_train()
# config.loss = "ge2e"
# config.save_json(config_path)
# run_test_train()
# test model with angleproto loss function
config.loss = "angleproto"
config.save_json(config_path)
run_test_train()
# config.loss = "angleproto"
# config.save_json(config_path)
# run_test_train()
# test model with softmaxproto loss function
config.loss = "softmaxproto"
+2 -2
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@@ -6,8 +6,8 @@ import torch
from tests import get_tests_input_path
from TTS.config import load_config
from TTS.speaker_encoder.utils.generic_utils import setup_speaker_encoder_model
from TTS.speaker_encoder.utils.io import save_checkpoint
from TTS.encoder.utils.generic_utils import setup_speaker_encoder_model
from TTS.encoder.utils.io import save_checkpoint
from TTS.tts.utils.speakers import SpeakerManager
from TTS.utils.audio import AudioProcessor
+49
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@@ -8,6 +8,7 @@ from TTS.config.shared_configs import BaseDatasetConfig
from TTS.tts.datasets import load_tts_samples
from TTS.tts.utils.languages import get_language_balancer_weights
from TTS.tts.utils.speakers import get_speaker_balancer_weights
from TTS.encoder.utils.samplers import PerfectBatchSampler
# Fixing random state to avoid random fails
torch.manual_seed(0)
@@ -82,3 +83,51 @@ class TestSamplers(unittest.TestCase):
spk2 += 1
assert is_balanced(spk1, spk2), "Speaker Weighted sampler is supposed to be balanced"
def test_perfect_sampler(self): # pylint: disable=no-self-use
classes = set()
for item in train_samples:
classes.add(item["speaker_name"])
sampler = PerfectBatchSampler(
train_samples,
classes,
batch_size=2 * 3, # total batch size
num_classes_in_batch=2,
label_key="speaker_name",
shuffle=False,
drop_last=True)
batchs = functools.reduce(lambda a, b: a + b, [list(sampler) for i in range(100)])
for batch in batchs:
spk1, spk2 = 0, 0
# for in each batch
for index in batch:
if train_samples[index]["speaker_name"] == "ljspeech-0":
spk1 += 1
else:
spk2 += 1
assert spk1 == spk2, "PerfectBatchSampler is supposed to be perfectly balanced"
def test_perfect_sampler_shuffle(self): # pylint: disable=no-self-use
classes = set()
for item in train_samples:
classes.add(item["speaker_name"])
sampler = PerfectBatchSampler(
train_samples,
classes,
batch_size=2 * 3, # total batch size
num_classes_in_batch=2,
label_key="speaker_name",
shuffle=True,
drop_last=False)
batchs = functools.reduce(lambda a, b: a + b, [list(sampler) for i in range(100)])
for batch in batchs:
spk1, spk2 = 0, 0
# for in each batch
for index in batch:
if train_samples[index]["speaker_name"] == "ljspeech-0":
spk1 += 1
else:
spk2 += 1
assert spk1 == spk2, "PerfectBatchSampler is supposed to be perfectly balanced"
+2 -2
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@@ -66,8 +66,8 @@
"use_mas": false, // use Monotonic Alignment Search if true. Otherwise use pre-computed attention alignments.
// TRAINING
"batch_size": 2, // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
"eval_batch_size":1,
"batch_size": 8, // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
"eval_batch_size": 8,
"r": 1, // Number of decoder frames to predict per iteration. Set the initial values if gradual training is enabled.
"loss_masking": true, // enable / disable loss masking against the sequence padding.
"data_dep_init_iter": 1,
@@ -36,8 +36,8 @@
"warmup_steps": 4000, // Noam decay steps to increase the learning rate from 0 to "lr"
"tb_model_param_stats": false, // true, plots param stats per layer on tensorboard. Might be memory consuming, but good for debugging.
"steps_plot_stats": 10, // number of steps to plot embeddings.
"num_speakers_in_batch": 64, // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
"num_utters_per_speaker": 10, //
"num_classes_in_batch": 64, // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
"num_utter_per_class": 10, //
"num_loader_workers": 8, // number of training data loader processes. Don't set it too big. 4-8 are good values.
"wd": 0.000001, // Weight decay weight.
"checkpoint": true, // If true, it saves checkpoints per "save_step"
+2 -2
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@@ -61,8 +61,8 @@
"reinit_layers": [], // give a list of layer names to restore from the given checkpoint. If not defined, it reloads all heuristically matching layers.
// TRAINING
"batch_size": 1, // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
"eval_batch_size":1,
"batch_size": 8, // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
"eval_batch_size": 8,
"r": 7, // Number of decoder frames to predict per iteration. Set the initial values if gradual training is enabled.
"gradual_training": [[0, 7, 4], [1, 5, 2]], //set gradual training steps [first_step, r, batch_size]. If it is null, gradual training is disabled. For Tacotron, you might need to reduce the 'batch_size' as you proceeed.
"loss_masking": true, // enable / disable loss masking against the sequence padding.
+2 -2
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@@ -61,8 +61,8 @@
"reinit_layers": [], // give a list of layer names to restore from the given checkpoint. If not defined, it reloads all heuristically matching layers.
// TRAINING
"batch_size": 1, // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
"eval_batch_size":1,
"batch_size": 8, // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
"eval_batch_size": 8,
"r": 7, // Number of decoder frames to predict per iteration. Set the initial values if gradual training is enabled.
"gradual_training": [[0, 7, 4], [1, 5, 2]], //set gradual training steps [first_step, r, batch_size]. If it is null, gradual training is disabled. For Tacotron, you might need to reduce the 'batch_size' as you proceeed.
"loss_masking": true, // enable / disable loss masking against the sequence padding.
+1 -1
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@@ -7,7 +7,7 @@ from trainer.logging.tensorboard_logger import TensorboardLogger
from tests import assertHasAttr, assertHasNotAttr, get_tests_data_path, get_tests_input_path, get_tests_output_path
from TTS.config import load_config
from TTS.speaker_encoder.utils.generic_utils import setup_speaker_encoder_model
from TTS.encoder.utils.generic_utils import setup_speaker_encoder_model
from TTS.tts.configs.vits_config import VitsConfig
from TTS.tts.models.vits import Vits, VitsArgs, amp_to_db, db_to_amp, load_audio, spec_to_mel, wav_to_mel, wav_to_spec
from TTS.tts.utils.speakers import SpeakerManager