Spectrogram Generator

Spectrogram Generator models take in text input and generate a Mel spectrogram. There are several types of Spectrogram Generator architecture; TAO Toolkit supports the FastPitch architecture.

The FastPitch model generates Mel spectrograms and predicts a pitch contour from raw input text. It allows additional control over synthesized utterances through the following options:

  • Modify the pitch contour to control the prosody.

  • Increase or decrease the fundamental frequency in a natural way, which preserves the perceived identity of the speaker.

  • Alter the rate of speech.

  • Specify input as graphemes or phonemes.

  • Switch speakers (if the model has been trained with data from multiple speakers).

The following tasks have been implemented for FastPitch in the TAO Toolkit

  • download_specs

  • dataset_convert

  • train

  • infer

  • export

  • finetune

  • pitch_stats

Example specification files for all the tasks associated with the spectrogram generator component of TTS can be downloaded using the following command:

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tao spectro_gen download_specs \ -o <target_path> \ -r <results_path>

Required Arguments

  • -o: The target path where the spec files will be stored

  • -r: The results and output log directory

The spectrogram generator for TAO Toolkit implements the dataset_convert task to convert and prepare datasets that follow the LJSpeech dataset format.

The dataset_convert task generates manifest files and .txt files with normalized transcripts.

The dataset for TTS consists of a set of utterances in individual audio files (.wav) and a manifest that describes the dataset, with information about one utterance per line (.json).

Each line of the manifest should be in the following format:

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{"audio_filepath": "/path/to/audio.wav", "text": "the transcription of the utterance", "duration": 23.147}

The audio_filepath field should provide an absolute path to the .wav file corresponding to the utterance. The text field should contain the full transcript for the utterance, and the duration field should reflect the duration of the utterance in seconds.

Each entry in the manifest (describing one audio file) should be bordered by { and } and must be contained on one line. The fields that describe the file should be separated by commas and have the form "field_name": value, as shown above.

Since the manifest specifies the path for each utterance, the audio files do not have to be located in the same directory as the manifest, or even in any specific directory structure.

The spec file for TTS using FastPitch includes the trainer, model, training_dataset, validation_dataset, and prior_folder.

The following is a shortened example of a spec file for training on the LJSpeech dataset.

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sample_rate: 22050 train_dataset: ??? validation_dataset: ??? prior_folder: ??? model: learn_alignment: true n_speakers: 1 symbols_embedding_dim: 384 max_token_duration: 75 n_mel_channels: 80 pitch_embedding_kernel_size: 3 n_window_size: 1024 n_window_stride: 256 pitch_fmin: 80 pitch_fmax: 640 pitch_avg: 211.27540199742586 pitch_std: 52.1851002822779 train_ds: dataset: _target_: "nemo.collections.asr.data.audio_to_text.AudioToCharWithPriorAndPitchDataset" manifest_filepath: ${train_dataset} max_duration: null min_duration: 0.1 int_values: false normalize: true sample_rate: ${sample_rate} trim: false sup_data_path: ${prior_folder} n_window_stride: ${model.n_window_stride} n_window_size: ${model.n_window_size} pitch_fmin: ${model.pitch_fmin} pitch_fmax: ${model.pitch_fmax} pitch_avg: ${model.pitch_avg} pitch_std: ${model.pitch_std} vocab: notation: phonemes punct: true spaces: true stresses: true add_blank_at: None pad_with_space: True chars: true improved_version_g2p: true dataloader_params: drop_last: false shuffle: true batch_size: 32 num_workers: 12 validation_ds: dataset: _target_: "nemo.collections.asr.data.audio_to_text.AudioToCharWithPriorAndPitchDataset" manifest_filepath: ${validation_dataset} max_duration: null min_duration: null int_values: false normalize: true sample_rate: ${sample_rate} trim: false sup_data_path: ${prior_folder} n_window_stride: ${model.n_window_stride} n_window_size: ${model.n_window_size} pitch_fmin: ${model.pitch_fmin} pitch_fmax: ${model.pitch_fmax} pitch_avg: ${model.pitch_avg} pitch_std: ${model.pitch_std} vocab: notation: phonemes punct: true spaces: true stresses: true add_blank_at: None pad_with_space: True chars: true improved_version_g2p: true dataloader_params: drop_last: false shuffle: false batch_size: 32 num_workers: 8 preprocessor: _target_: nemo.collections.asr.modules.AudioToMelSpectrogramPreprocessor dither: 0.0 features: ${model.n_mel_channels} frame_splicing: 1 highfreq: 8000 log: true log_zero_guard_type: add log_zero_guard_value: 1e-05 lowfreq: 0 mag_power: 1.0 n_fft: ${model.n_window_size} n_window_size: ${model.n_window_size} n_window_stride: ${model.n_window_stride} normalize: null pad_to: 1 pad_value: 0 preemph: null sample_rate: ${sample_rate} window: hann window_size: null window_stride: null input_fft: #n_embed and padding_idx are added by the model _target_: nemo.collections.tts.modules.transformer.FFTransformerEncoder n_layer: 6 n_head: 1 d_model: ${model.symbols_embedding_dim} d_head: 64 d_inner: 1536 kernel_size: 3 dropout: 0.1 dropatt: 0.1 dropemb: 0.0 d_embed: ${model.symbols_embedding_dim} output_fft: _target_: nemo.collections.tts.modules.transformer.FFTransformerDecoder n_layer: 6 n_head: 1 d_model: ${model.symbols_embedding_dim} d_head: 64 d_inner: 1536 kernel_size: 3 dropout: 0.1 dropatt: 0.1 dropemb: 0.0 alignment_module: _target_: nemo.collections.tts.modules.aligner.AlignmentEncoder n_text_channels: ${model.symbols_embedding_dim} duration_predictor: _target_: nemo.collections.tts.modules.fastpitch.TemporalPredictor input_size: ${model.symbols_embedding_dim} kernel_size: 3 filter_size: 256 dropout: 0.1 n_layers: 2 pitch_predictor: _target_: nemo.collections.tts.modules.fastpitch.TemporalPredictor input_size: ${model.symbols_embedding_dim} kernel_size: 3 filter_size: 256 dropout: 0.1 n_layers: 2 optim: name: lamb lr: 1e-1 betas: [0.9, 0.98] weight_decay: 1e-6 sched: name: NoamAnnealing warmup_steps: 1000 last_epoch: -1 d_model: 1 # Disable scaling based on model dim trainer: max_epochs: 100

The specification can be roughly grouped into three categories:

  • Parameters to configure the trainer

  • Parameters that describe the model

  • Parameters to configure the experiment

This specification can be used with the tao spectro_gen train command.

If you would like to change a parameter for your run without changing the specification file itself, you can specify it on the command line directly. For example, if you would like to change the validation batch size, you can add model.validation_ds.batch_size=1 to your command, which would override the batch size of 32 in the configuration shown above. An example of this is shown in the training instructions below.

Configuring the Trainer

The following parameter is used to configure the trainer element of the Spectrogram Generator.

Parameter

Datatype

Description

Supported Values

max_epochs

int

Specifies the maximum number of epochs to train the model. A field for the trainer parameter.

>0

Configuring the model

The parameters to help configure the FastPitch model are included in the model element. This includes parameters for configuring the following elements:

  1. dataset_config

  2. preprocessor

  3. input_fft

  4. output_fft

  5. alignment_module

  6. duration_predictor

  7. pitch_predictor

  8. optimizer

There are also some global parameters:

Parameter

Datatype

Description

Supported Values

learn_alignment

bool

Enable learning alignment

Valid filepaths

n_speakers

int

The number of speakers in the dataset

symbols_embedding_dim

int

The dimension of the symbols embedding

max_token_duration

int

The maximum duration to clamp the tokens to

pitch_embedding_kernel_size

int

The kernel size of the Conv1d layer generating pitch embeddings

pitch_fmin

float

The fmin input to librosa.pyin. The default value is librosa.note_to_hz(‘C2’)

pitch_fmax

float

The fmax input to librosa.pyin. The default value is librosa.note_to_hz(‘C7’)

pitch_avg

float

The average used to normalize the pitch

pitch_std

float

The std deviation used to normalize the pitch

n_window_stride

int

The stride of the window for fft in samples.

n_window_size

int

The size of the window for fft in samples.

n_mel_channels

int

The number of Mel channels to output

Dataset Configs

The datasets that you use should be specified by <xyz>_ds parameters, depending on the use case:

  • For training using tao spectro_gen train, you should have training_ds to describe your training dataset, and validation_ds to describe your validation dataset.

Each <xyz>_ds config contains two main groups of configuration

  • dataset: The configuration component describing the dataset

  • dataloader: The configuration componenet describing the dataloader

The configurable fields for the dataset field are described in the following table:

Parameter

Datatype

Description

Supported Values

manifest_filepath

string

The filepath to the manifest (.json file) that describes the audio data

Valid filepaths.

min_duration

float

All files with a duration less than the given value (in seconds) will be dropped. The default value is 0.1.

max_duration

float

All files with a duration greater than the given value (in seconds) will be dropped.

sample_rate

int

The target sample rate to load the audio, in Hz.

trim

bool

Whether to trim silence from beginning and end of audio signal using librosa.effects.trim(). The default value is False.

True/False

int_values

bool

If true, load samples as 32-bit integers. The default value is False.

True/False

n_window_stride

int

The stride of window for fft in samples.

n_window_size

int

The size of window for fft in samples.

normalize

bool

The flag to determine whether to normalize the transcript text

True/False

pitch_fmin

float

The fmin input to librosa.pyin. The default value is librosa.note_to_hz(‘C2’)

pitch_fmax

float

The fmax input to librosa.pyin. The default value is librosa.note_to_hz(‘C7’)

pitch_avg

float

The average used to normalize the pitch

pitch_std

float

The std deviation used to normalize the pitch

Note

The pitch_avg and pitch_std parameters provided by default are calculated for the LJSpeech dataset. These values must be re-calculated per speaker.

Similarly, the pitch_fmin and pitch_fmax need to adjusted based on the dataset. The default values may result in poor behaviour.

Vocabulary

This subsection under the dataset component of the <xyz>_ds config defines the configurable fields to generate a vocabulary.

Parameter

Datatype

Description

Supported Values

notation

str

Either ‘chars` or phonemes as general notation

phonemes

punct

bool

Whether to reserve graphemes for basic punctuation

True/False

spaces

bool

Whether to prepend spaces to every punctuation symbol.

True/False

chars

bool

Whether to additionally use chars together with phonemes

True/False

add_blank_at

str

Add blank to labels in the specified order. If this string is empty, then there will be no blank in the labels.

last/last_but_one/None

pad_with_space

bool

Whether to pad text with spaces at the beginning and at the end

True/False

improved_version_g2p

bool

Whether to use the new version of g2p

True/False

Dataloader

Parameter

Datatype

Description

Supported Values

num_workers

integer

The number of worker threads for loading
the dataset

2

shuffle

bool

Whether to shuffle the data. We recommend True for training data and False for validation.

True/False

batch_size

integer

The training data batch size

Preprocessor Config

Parameter

Datatype

Description

Supported Values

dither

float

Amount of white-noise dithering.

>= 0

features

int

Number of mel spectrogram freq bins to output.

derived from model.n_mel_channels

frame_splicing

int

Number of spectrogram frames per model step

highfreq

int

Upper bound on mel basis in Hz.

log

bool

Whether to log the spectrogram

log_zero_guard_type

str

Need to avoid taking the log of zero. There are
two options: “add” or “clamp”.

low_zero_guard_value

float, str

Add or clamp requires the number to add with or
clamp to. log_zero_guard_value can either be a
float or “tiny” or “eps”. torch.finfo is used if
“tiny” or “eps” is passed.

lowfreq

int

Lower bound on mel basis in Hz.

mag_power

int

prior to multiplication with mel basis.

n_fft

int

The size of window for fft in samples. Use one of
window_size or n_window_size.

Derived from model.n_window_size

n_window_size

int

The size of window for fft in samples. Use one of
window_stride or n_window_stride.

Derived from model.n_window_size

n_window_stride

int

The stride of the window for fft.

Derived from model.n_window_stride

normalize

str

other options disable feature normalization.
‘all_features’ normalizes the entire spectrogram

per channel / freq instead.

pad_to

int

a multiple of pad_to.

pad_value

float

The value that shorter mels are padded with.

preemph

float

Amount of pre emphasis to add to audio. Can be
disabled by passing None.

sample_rate

int

The target sample rate to load the audio, in Hz.

Derived from sample_rate

window

string

‘hamming’, ‘blackman’, ‘bartlett’]

window_size

int

Size of window for fft in seconds

window_stride

int

Stride of window for fft in seconds

INPUT / OUTPUT FFT

Parameter

Datatype

Description

Supported Values

n_layer

int

Number of Transformer layers

n_head

int

Number of heads in the MultiHeadAttn module

d_model

int

Hidden size of input and output

Derived from model.symbols_embedding_dim

d_head

int

Hidden size of attention module

d_inner

int

Hidden size of conv layer

kernel_size

int

Kernel size of conv layer

dropout

float

Dropout parameter

dropatt

float

Dropout parameter for attention

d_embed

int

Hidden size of embeddings (input fft only)

Derived from model.symbols_embedding_dim

Alignment Module

Parameter

Datatype

Description

Supported Values

n_text_channels

int

Should match d_model

Duration Predictor / Pitch Predictor

A simple stack of conv, relu, layernorm, dropout layers.

Parameter

Datatype

Description

Supported Values

input_size

int

Should match d_model

Derived from model.symbols_embedding_dim

kernel_size

int

Kernel size for conv layers

filter_size

int

Filter size for conv layers

dropout

float

Dropout parameter

n_layers

int

Number of layers

To train a model from scratch, use the following command:

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tao spectro_gen train \ -e <experiment_spec> \ -g <num_gpus> \ -r /path/to/the/results/directory \ -k <encryption_key>

As mentioned above, you can add additional arguments to override configurations from your experiment specification file. This allows you to create valid spec files that leave these fields blank, to be specified as command line arguments at runtime.

For example, the following command can be used to override the training manifest and validation manifest, the number of epochs to train, and the place to save the model checkpoint:

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tao spectro_gen train \ -e $SPECS_DIR/spectro_gen/train.yaml \ -g 1 \ -k $KEY \ -r $RESULTS_DIR/spectro_gen/train \ train_dataset=$DATA_DIR/ljspeech/ljspeech_train.json \ validation_dataset=$DATA_DIR/ljspeech/ljspeech_val.json \ prior_folder=$RESULTS_DIR/spectro_gen/train/prior_folder \ trainer.max_epochs=5

Required Arguments

  • -e: The experiment specification file to set up training, as in the example given above

  • -r: The path to the results and log directory. Log files, checkpoints, etc., will be stored here

  • -k: The key to encrypt the model

  • Other arguments to override fields in the specification file.

Optional Arguments

  • -g: The number of GPUs to be used in the training in a multi-GPU scenario. The default value is 1.

Training Procedure

At the start of each training experiment, TAO Toolkit will print out a log of the experiment specification, including any parameters added or overridden via the command line. It will also show additional information, such as which GPUs are available, where logs will be saved, how many hours are in each loaded dataset, and how much of each dataset has been filtered.

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GPU available: True, used: True TPU available: False, using: 0 TPU cores IPU available: False, using: 0 IPUs [NeMo W 2021-10-29 21:29:06 exp_manager:414] Exp_manager is logging to /results/spectro_gen/train, but it already exists. [NeMo W 2021-10-29 21:29:06 exp_manager:332] There was no checkpoint folder at checkpoint_dir :/results/spectro_gen/train/checkpoints. Training from scratch. [NeMo I 2021-10-29 21:29:06 exp_manager:220] Experiments will be logged at /results/spectro_gen/train [NeMo I 2021-10-29 21:29:06 exp_manager:569] TensorboardLogger has been set up [NeMo W 2021-10-29 21:29:06 nemo_logging:349] /opt/conda/lib/python3.8/site-packages/pytorch_lightning/callbacks/model_checkpoint.py:240: LightningDeprecationWarning: `ModelCheckpoint(every_n_val_epochs)` is deprecated in v1.4 and will be removed in v1.6. Please use `every_n_epochs` instead. rank_zero_deprecation( [NeMo I 2021-10-29 21:29:12 collections:173] Dataset loaded with 12500 files totalling 22.84 hours [NeMo I 2021-10-29 21:29:12 collections:174] 0 files were filtered totalling 0.00 hours [NeMo I 2021-10-29 21:29:40 collections:173] Dataset loaded with 100 files totalling 0.18 hours [NeMo I 2021-10-29 21:29:40 collections:174] 0 files were filtered totalling 0.00 hours [NeMo I 2021-10-29 21:29:42 features:252] PADDING: 1 [NeMo I 2021-10-29 21:29:42 features:269] STFT using torch initializing ddp: GLOBAL_RANK: 0, MEMBER: 1/1 Added key: store_based_barrier_key:1 to store for rank: 0 Rank 0: Completed store-based barrier for 1 nodes. ---------------------------------------------------------------------------------------------------- distributed_backend=nccl All DDP processes registered. Starting ddp with 1 processes -------------------------------------------------------------------------------------------------

You should next see a full printout of the number of parameters in each module and submodule, as well as the total number of trainable and non-trainable parameters in the model.

In the following table, the fastpitch module contains 45.8 million parameters and its submodule fastpitch.encoder container 21.9 million parameters. The ReLU, PositionalEmbedding, and Dropout modules are listed with no parameters.

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| Name | Type | Params ------------------------------------------------------------------------------------------------------- 0 | mel_loss | MelLoss | 0 1 | pitch_loss | PitchLoss | 0 2 | duration_loss | DurationLoss | 0 3 | aligner | AlignmentEncoder | 1.0 M 4 | aligner.softmax | Softmax | 0 5 | aligner.log_softmax | LogSoftmax | 0 6 | aligner.key_proj | Sequential | 947 K 7 | aligner.key_proj.0 | ConvNorm | 885 K 8 | aligner.key_proj.0.conv | Conv1d | 885 K 9 | aligner.key_proj.1 | ReLU | 0 10 | aligner.key_proj.2 | ConvNorm | 61.5 K 11 | aligner.key_proj.2.conv | Conv1d | 61.5 K 12 | aligner.query_proj | Sequential | 57.9 K 13 | aligner.query_proj.0 | ConvNorm | 38.6 K 14 | aligner.query_proj.0.conv | Conv1d | 38.6 K 15 | aligner.query_proj.1 | ReLU | 0 16 | aligner.query_proj.2 | ConvNorm | 12.9 K 17 | aligner.query_proj.2.conv | Conv1d | 12.9 K 18 | aligner.query_proj.3 | ReLU | 0 19 | aligner.query_proj.4 | ConvNorm | 6.5 K 20 | aligner.query_proj.4.conv | Conv1d | 6.5 K 21 | forward_sum_loss | ForwardSumLoss | 0 22 | forward_sum_loss.log_softmax | LogSoftmax | 0 23 | forward_sum_loss.ctc_loss | CTCLoss | 0 24 | bin_loss | BinLoss | 0 25 | preprocessor | AudioToMelSpectrogramPreprocessor | 0 26 | preprocessor.featurizer | FilterbankFeatures | 0 27 | fastpitch | FastPitchModule | 45.8 M 28 | fastpitch.encoder | FFTransformerEncoder | 21.9 M 29 | fastpitch.encoder.pos_emb | PositionalEmbedding | 0 30 | fastpitch.encoder.drop | Dropout | 0 31 | fastpitch.encoder.layers | ModuleList | 21.8 M 32 | fastpitch.encoder.layers.0 | TransformerLayer | 3.6 M 33 | fastpitch.encoder.layers.0.dec_attn | MultiHeadAttn | 99.3 K 34 | fastpitch.encoder.layers.0.dec_attn.qkv_net | Linear | 73.9 K 35 | fastpitch.encoder.layers.0.dec_attn.drop | Dropout | 0 36 | fastpitch.encoder.layers.0.dec_attn.dropatt | Dropout | 0 37 | fastpitch.encoder.layers.0.dec_attn.o_net | Linear | 24.6 K .. .. 213 | fastpitch.duration_predictor.layers.1.norm | LayerNorm | 512 214 | fastpitch.duration_predictor.layers.1.dropout | Dropout | 0 215 | fastpitch.duration_predictor.fc | Linear | 257 216 | fastpitch.pitch_predictor | TemporalPredictor | 493 K 217 | fastpitch.pitch_predictor.layers | Sequential | 493 K 218 | fastpitch.pitch_predictor.layers.0 | ConvReLUNorm | 295 K 219 | fastpitch.pitch_predictor.layers.0.conv | Conv1d | 295 K 220 | fastpitch.pitch_predictor.layers.0.norm | LayerNorm | 512 221 | fastpitch.pitch_predictor.layers.0.dropout | Dropout | 0 222 | fastpitch.pitch_predictor.layers.1 | ConvReLUNorm | 197 K 223 | fastpitch.pitch_predictor.layers.1.conv | Conv1d | 196 K 224 | fastpitch.pitch_predictor.layers.1.norm | LayerNorm | 512 225 | fastpitch.pitch_predictor.layers.1.dropout | Dropout | 0 226 | fastpitch.pitch_predictor.fc | Linear | 257 227 | fastpitch.pitch_emb | Conv1d | 1.5 K 228 | fastpitch.proj | Linear | 30.8 K ------------------------------------------------------------------------------------------------------- 45.8 M Trainable params 0 Non-trainable params 45.8 M Total params 183.035 Total estimated model params size (MB)

As the model starts training, you should see a progress bar per epoch.

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Epoch 0: 0%| | 0/395 [00:00<00:00, 5504.34it/s][W reducer.cpp:1151] Warning: find_unused_parameters=True was specified in DDP constructor, but did not find any unused parameters in the forward pass. This flag results in an extra traversal of the autograd graph every iteration, which can adversely affect performance. If your model indeed never has any unused parameters in the forward pass, consider turning this flag off. Note that this warning may be a false positive if your model has flow control causing later iterations to have unused parameters. (function operator()) Epoch 0: 6%|▊ | 23/395 [05:06<1:19:07, 12.76s/it, loss=38, v_num=] ...

At the end of training, TAO Toolkit will save the last checkpoint at the path specified by the experiment spec file before finishing.

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[NeMo I 2021-01-20 22:38:48 train:120] Experiment logs saved to '$RESULTS_DIR/spectro_gen/train' [NeMo I 2021-01-20 22:38:48 train:123] Trained model saved to '$RESULTS_DIR/spectro_gen/train/checkpoints/trained-model.tlt' INFO: Internal process exited


Current Limitations

  • Currently, only .wav audio files are supported.

  • Training only supports single speaker dataset.

  • The spectrogram generator can only be trained from scratch.

To perform inference on individual text lines, use the following command:

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tao spectro_gen infer -e <experiment_spec> \ -m <model_checkpoint> \ -g <num_gpus> \ -k $KEY \ -r </path/to/results/directory/for/logs> \ output_path=</path/to/result/directory/for/spectrogram>

Required Arguments

  • -e: The experiment specification file to set up inference. This spec file only needs a file_paths parameter that contains a list of individual file paths.

  • -m: The path to the model checkpoint, which should be a .tlt file.

  • -k: The key to encrypt the model

Optional Arguments

  • -g: The number of GPUs to use for inference in a multi-GPU scenario. The default value is 1.

  • -r: The path to the results and log directory. Log files, checkpoints, etc. will be stored here.

  • Other arguments to override fields in the specification file.

Inference Procedure

At the start of inference, TAO Toolkit will print out the experiment specification, including the audio filepaths on which inference will be performed.

When restoring from the checkpoint, it will then log the original datasets that the checkpoint model was trained and evaluated on. This will show the vocabulary that the model was trained on.

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[NeMo W 2021-10-29 23:08:27 exp_manager:26] Exp_manager is logging to `/results/spectro_gen/infer``, but it already exists. [NeMo W 2021-10-29 23:08:33 modelPT:130] If you intend to do training or fine-tuning, please call the ModelPT.setup_training_data() method and provide a valid configuration file to setup the train data loader. Train config : dataset: _target_: nemo.collections.asr.data.audio_to_text.AudioToCharWithPriorAndPitchDataset manifest_filepath: /data/ljspeech/ljspeech_train.json ... ... dataloader_params: drop_last: false shuffle: true batch_size: 32 num_workers: 12 [NeMo W 2021-10-29 23:08:33 modelPT:137] If you intend to do validation, please call the ModelPT.setup_validation_data() or ModelPT.setup_multiple_validation_data() method and provide a valid configuration file to setup the validation data loader(s). Validation config : dataset: _target_: nemo.collections.asr.data.audio_to_text.AudioToCharWithPriorAndPitchDataset ... ... dataloader_params: drop_last: false shuffle: false batch_size: 32 num_workers: 8 [NeMo I 2021-10-29 23:08:43 features:252] PADDING: 1 [NeMo I 2021-10-29 23:08:43 features:269] STFT using torch Results for by the... saved to /results/spectro_gen/infer/spectro/0.npy Results for direct... saved to /results/spectro_gen/infer/spectro/1.npy Results for uneasy... saved to /results/spectro_gen/infer/spectro/2.npy [NeMo I 2021-10-29 23:08:51 infer:79] Experiment logs saved to '/results/spectro_gen/infer'

The path to the Mel spectrograms generated by the infer task are shown in the last lines of the log.

Current Limitations

  • Currently, only .wav audio files are supported.

To fine-tune a model from a checkpoint, use the following command:

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!tao spectro_gen finetune -e <experiment_spec> \ -m <model_checkpoint> \ -g <num_gpus> \ train_dataset=<train.json> \ validation_dataset=<val.json> \ prior_folder=<prior_dir, could be an empty dir> \ n_speakers=2 \ pitch_fmin=<pitch statistic, see pitch section> \ pitch_fmax=<pitch statistic, see pitch section> \ pitch_avg=<pitch statistic, see pitch section> \ pitch_std=<pitch statistic, see pitch section> \ trainer.max_steps=<num_steps>

Required Arguments

  • -e: The experiment specification file to set up fine-tuning.

  • -m: The path to the model checkpoint from which to fine-tune. The model checkpoint should be a .tlt file.

  • train_dataset: The path to the training manifest, which should be created using dataset_convert dataset_name=merge. See the section below for more details.

  • validation_dataset: The path to the validation manifest.

  • prior_folder: A folder used to store dataset files. If the folder is empty, these files will be computed on the first run and saved to this directory. Future runs will load these files from the directory if they exist.

  • n_speakers: This value should be 2: One for the original speaker, one for the new finetuning speaker.

  • pitch_fmin: The Fmin to be used for pitch extraction. See the section below on how to set this value.

  • pitch_fmax: The Fmax to be used for pitch extraction. See the section below on how to set this value.

  • pitch_avg: The pitch average to be used for pitch extraction. See the section below on how to set this value.

  • pitch_std: The pitch standard deviation to be used for pitch extraction. See the section below on how to set this value.

  • trainer.max_steps: The number of steps used to finetune the model. We recommend adding 1000 for each minute in the finetuning data.

Optional Arguments

  • -g: The number of GPUs to be used for fine-tuning in a multi-GPU scenario (default: 1).

  • -r: The path to the results and log directory. Log files, checkpoints, etc., will be stored here.

  • Other arguments to override fields in the specification file.

Warning

In order to prevent unauthorized use of someone’s voice, TAO will only run finetuning if the text transcripts used in the finetuning data comes from the NVIDIA Custom Voice Recorder tool. Users do not have to use the tool to record their own voice, but the transcripts used must be the same.

Warning

The data from the NVIDIA Custom Voice Recorder tool cannot be used to train a FastPitch model from scratch. Instead, use the data with the finetune endpoint of TAO Text-To-Speech with a pretrained FastPitch model.


Pitch Statistics

To fine tune FastPitch, you need to find and set 4 pitch hyperparameters:

  • Fmin

  • Fmax

  • Mean

  • Std

TAO features the pitch_stats task to help with this process. You must set Fmin and Fmax first. You can then iterate over the finetuning dataset to extract the pitch mean and standard deviation.

Obtaining the fmin and fmax

To get the fmin and fmax values, you will need to start with some defaults and iterate through random samples of the dataset to ensure that the pyin function from librosa extracts the pitch correctly. Then, look at the plotted spectrograms, as well as the predicted f0 (the cyan line), which should match the lowest energy band in the spectrogram. Here is an example of a good match between the predicted f0 and the spectrogram.

good_pitch.png


The following is an example of a bad match between the f0 and the spectrogram. The fmin was likely set too high. The fo algorithm is missing the first two vocalizations and is correctly matching the last half of speech. To fix this, set the fmin value lower.

bad_pitch.png


The following is an example of samples that have low frequency noise. To eliminate the effects of noise, set the fmin value above the noise frequency. Unfortunately, this will result in degraded TTS quality. It would be best to re-record the data in an environment with less noise.

noise.png


To generate these plots, runn the pitch_stats entrypoint with the following options:

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tao spectro_gen pitch_stats num_files=10 \ pitch_fmin=64 \ pitch_fmax=512 \ output_path=/results/spectro_gen/pitch_stats \ compute_stats=false \ render_plots=true \ manifest_filepath=$DATA_DIR/6097_5_mins/6097_manifest_train.json \ --results_dir $RESULTS_DIR/spectro_gen/pitch_stats

Required Arguments:

  • pitch_fmin: The minimum frequency value set by the user as input to extract the pitch

  • pitch_fmax: The maximum frequence value set by the user as input to extract the pitch

  • output_path: The path to the directory where the pitch plots are generated

  • compute_stats: A boolean flag that specifies whether to compute the pitch_mean and pitch_std

  • render_plots: A boolean flag that specifies whether to generate the pitch plots at the output_path

  • manifest_filepath: The path to the dataset

  • num_files: Number of files in the input dataset to visualize the f0 plot.

  • results_dir: The path to the directory where the logs are generated

Note

We recommend setting the compute_stats option to false so you don’t spend time iterating over the entire dataset to compute pitch_mean and pitch_std until you are satisfied with the fmin and fmax values.

Computing the pitch_mean and pitch_std

After you set the pitch_fmin and pitch_fmax, you need to extract the pitch over all training files. After filtering out all 0.0 and nan values from the pitch, you will compute the mean and standard deviation. You can then use these values to fine tune FastPitch. To generate the mean and standard deviation, run the pitch_stats task with the following options:

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tao spectro_gen pitch_stats num_files=10 \ pitch_fmin=64 \ pitch_fmax=512 \ output_path=/results/spectro_gen/pitch_stats \ compute_stats=true \ render_plots=false \ manifest_filepath=$DATA_DIR/6097_5_mins/6097_manifest_train.json \ --results_dir $RESULTS_DIR/spectro_gen/pitch_stats

Note

In the above example, the compute_stats option is set to true while the render_plots option is set to false so that the spectrograms aren’t rendered and predicted f0 again, but we do compute the mean and standard deviation values.

Manifest Creation

For best results, you should fine tune FastPitch by adding the original data as well as data from the new speaker. To create a training manifest file that combines the data, you can use spectro_gen dataset_convert dataset_name=merge with the following parameters:

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!tao spectro_gen dataset_convert dataset_name=merge \ original_json=<original_data.json> \ finetune_json=<finetuning_data.json> \ save_path=<path_to_save_new_json> \ -r <results_dir> \ -e <experiment_spec>

The important arguments are as follows:

  • original_json: The .json file that contains the original data

  • finetune_json: The .json file that contains the finetuning data

A merged .json file will be saved at save_path.

Note

The above code assumes that the original and fine-tuned dataset have gone through dataset_convert to generate the manifest.json files, as mentioned in the preparing the dataset section.

Warning

When merging manifest files, ensure that the audio clips from the original data and the new speaker data share the same sampling rate. If the sampling rates don’t match, you can either resample the data using the command line (method 1) or as part of the code (method 2):

  1. Use the the sox package CLI tool.

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    sox input.wav output.wav rate $RATE

    Where, $RATE is the target sample frequency in Hz.

  2. Use the librosa load function.

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    import librosa audio, sampling_rate = librosa.load( "/path/to/audio.wav", sr=<target_sampling_rate> ) librosa.output.write_wav( "/path/to/target/audio.wav", audio, sr=sampling_rate )


You can export a trained FastPitch model to Riva format, which contains all the model artifacts necessary for deployment to Riva Services. For more details about Riva, see this page.

To export a FastPitch model to the Riva format, use the following command:

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tao spectro_gen export -e <experiment_spec> \ -m <model_checkpoint> \ -r <results_dir> \ -k <encryption_key> \ export_format=RIVA \ export_to=<filename.riva>

Required Arguments

  • -e: The experiment specification file for export. See the Export Spec File section below for more details.

  • -m: The path to the model checkpoint to be exported, which should be a .tlt file

  • -k: The encryption key

Optional Arguments

  • -r: The path to the directory where results will be stored.

Export Spec File

The following is an example spec file for model export:

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# Name of the .tlt EFF archive to be loaded/model to be exported. restore_from: trained-model.tlt # Set export format: RIVA export_format: RIVA # Output EFF archive containing model checkpoint and artifacts required for Riva Services export_to: exported-model.riva

Parameter

Datatype

Description

Default

restore_from

string

The path to the pre-trained model to be exported

trained_model.tlt

export_format

string

The export format

N/A

export_to

string

The target path for the export model

exported-model.riva

A successful run of the model export generates the following log:

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[NeMo W 2021-10-29 23:14:22 exp_manager:26] Exp_manager is logging to `/results/spectro_gen/export``, but it already exists. [NeMo W 2021-10-29 23:14:28 modelPT:130] If you intend to do training or fine-tuning, please call the ModelPT.setup_training_data() method and provide a valid configuration file to setup the train data loader. Train config : dataset: _target_: nemo.collections.asr.data.audio_to_text.AudioToCharWithPriorAndPitchDataset manifest_filepath: /data/ljspeech/ljspeech_train.json max_duration: null min_duration: 0.1 int_values: false normalize: true sample_rate: 22050 ... ... ... [NeMo I 2021-10-29 23:14:35 export:57] Model restored from '/results/spectro_gen/train/checkpoints/trained-model.tlt' [NeMo W 2021-10-29 23:14:38 export_utils:198] Swapped 0 modules Warning: Constant folding - Only steps=1 can be constant folded for opset >= 10 onnx::Slice op. Constant folding not applied. Warning: Constant folding - Only steps=1 can be constant folded for opset >= 10 onnx::Slice op. Constant folding not applied. Warning: Constant folding - Only steps=1 can be constant folded for opset >= 10 onnx::Slice op. Constant folding not applied. [NeMo I 2021-10-29 23:14:58 export:72] Experiment logs saved to '/results/spectro_gen/export' [NeMo I 2021-10-29 23:14:58 export:73] Exported model to '/results/spectro_gen/export/spectro_gen.riva' [NeMo I 2021-10-29 23:15:03 export:80] Exported model is compliant with Riva


© Copyright 2022, NVIDIA.. Last updated on Dec 13, 2022.