--- language: sw license: cc-by-sa-4.0 tags: - tensorflowtts - audio - text-to-speech - mel-to-wav inference: false datasets: - bookbot/OpenBible_Swahili --- # MB-MelGAN HiFi PostNets SW v1 MB-MelGAN HiFi PostNets SW v1 is a mel-to-wav model based on the [MB-MelGAN](https://arxiv.org/abs/2005.05106) architecture with [HiFi-GAN](https://arxiv.org/abs/2010.05646) discriminator. This model was trained from scratch on a synthetic audio dataset. Instead of training on ground truth waveform spectrograms, this model was trained on the generated PostNet spectrograms of [LightSpeech MFA SW v1](https://huggingface.co/bookbot/lightspeech-mfa-sw-v1). The list of real speakers include: - sw-KE-OpenBible This model was trained using the [TensorFlowTTS](https://github.com/TensorSpeech/TensorFlowTTS) framework. All training was done on a Scaleway RENDER-S VM with a Tesla P100 GPU. All necessary scripts used for training could be found in this [Github Fork](https://github.com/bookbot-hive/TensorFlowTTS), as well as the [Training metrics](https://huggingface.co/bookbot/mb-melgan-hifi-postnets-sw-v1/tensorboard) logged via Tensorboard. ## Model | Model | Config | SR (Hz) | Mel range (Hz) | FFT / Hop / Win (pt) | #steps | | ------------------------------- | ----------------------------------------------------------------------------------------- | ------- | -------------- | -------------------- | ------ | | `mb-melgan-hifi-postnets-sw-v1` | [Link](https://huggingface.co/bookbot/mb-melgan-hifi-postnets-sw-v1/blob/main/config.yml) | 44.1K | 20-11025 | 2048 / 512 / None | 1M | ## Training Procedure
Feature Extraction Setting sampling_rate: 44100 hop_size: 512 # Hop size. format: "npy"
Generator Network Architecture Setting model_type: "multiband_melgan_generator" multiband_melgan_generator_params: out_channels: 4 # Number of output channels (number of subbands). kernel_size: 7 # Kernel size of initial and final conv layers. filters: 384 # Initial number of channels for conv layers. upsample_scales: [8, 4, 4] # List of Upsampling scales. stack_kernel_size: 3 # Kernel size of dilated conv layers in residual stack. stacks: 4 # Number of stacks in a single residual stack module. is_weight_norm: false # Use weight-norm or not.
Discriminator Network Architecture Setting multiband_melgan_discriminator_params: out_channels: 1 # Number of output channels. scales: 3 # Number of multi-scales. downsample_pooling: "AveragePooling1D" # Pooling type for the input downsampling. downsample_pooling_params: # Parameters of the above pooling function. pool_size: 4 strides: 2 kernel_sizes: [5, 3] # List of kernel size. filters: 16 # Number of channels of the initial conv layer. max_downsample_filters: 512 # Maximum number of channels of downsampling layers. downsample_scales: [4, 4, 4] # List of downsampling scales. nonlinear_activation: "LeakyReLU" # Nonlinear activation function. nonlinear_activation_params: # Parameters of nonlinear activation function. alpha: 0.2 is_weight_norm: false # Use weight-norm or not. hifigan_discriminator_params: out_channels: 1 # Number of output channels (number of subbands). period_scales: [3, 5, 7, 11, 17, 23, 37] # List of period scales. n_layers: 5 # Number of layer of each period discriminator. kernel_size: 5 # Kernel size. strides: 3 # Strides filters: 8 # In Conv filters of each period discriminator filter_scales: 4 # Filter scales. max_filters: 512 # maximum filters of period discriminator's conv. is_weight_norm: false # Use weight-norm or not.
STFT Loss Setting stft_loss_params: fft_lengths: [1024, 2048, 512] # List of FFT size for STFT-based loss. frame_steps: [120, 240, 50] # List of hop size for STFT-based loss frame_lengths: [600, 1200, 240] # List of window length for STFT-based loss. subband_stft_loss_params: fft_lengths: [384, 683, 171] # List of FFT size for STFT-based loss. frame_steps: [30, 60, 10] # List of hop size for STFT-based loss frame_lengths: [150, 300, 60] # List of window length for STFT-based loss.
Adversarial Loss Setting lambda_feat_match: 10.0 # Loss balancing coefficient for feature matching loss lambda_adv: 2.5 # Loss balancing coefficient for adversarial loss.
Data Loader Setting batch_size: 32 # Batch size for each GPU with assuming that gradient_accumulation_steps == 1. eval_batch_size: 16 batch_max_steps: 8192 # Length of each audio in batch for training. Make sure dividable by hop_size. batch_max_steps_valid: 8192 # Length of each audio for validation. Make sure dividable by hope_size. remove_short_samples: true # Whether to remove samples the length of which are less than batch_max_steps. allow_cache: false # Whether to allow cache in dataset. If true, it requires cpu memory. is_shuffle: false # shuffle dataset after each epoch.
Optimizer & Scheduler Setting generator_optimizer_params: lr_fn: "PiecewiseConstantDecay" lr_params: boundaries: [100000, 200000, 300000, 400000, 500000, 600000, 700000] values: [ 0.0005, 0.0005, 0.00025, 0.000125, 0.0000625, 0.00003125, 0.000015625, 0.000001, ] amsgrad: false discriminator_optimizer_params: lr_fn: "PiecewiseConstantDecay" lr_params: boundaries: [100000, 200000, 300000, 400000, 500000] values: [0.00025, 0.000125, 0.0000625, 0.00003125, 0.000015625, 0.000001] amsgrad: false gradient_accumulation_steps: 1
Interval Setting discriminator_train_start_steps: 200000 # steps begin training discriminator train_max_steps: 1000000 # Number of training steps. save_interval_steps: 20000 # Interval steps to save checkpoint. eval_interval_steps: 5000 # Interval steps to evaluate the network. log_interval_steps: 200 # Interval steps to record the training log.
Other Setting num_save_intermediate_results: 1 # Number of batch to be saved as intermediate results.
## How to Use ```py import soundfile as sf import tensorflow as tf from tensorflow_tts.inference import TFAutoModel, AutoProcessor lightspeech = TFAutoModel.from_pretrained("bookbot/lightspeech-mfa-sw-v1") processor = AutoProcessor.from_pretrained("bookbot/lightspeech-mfa-sw-v1") mb_melgan = TFAutoModel.from_pretrained("bookbot/mb-melgan-hifi-postnets-sw-v1") text, speaker_name = "Hello World.", "sw-KE-OpenBible" input_ids = processor.text_to_sequence(text) mel, _, _ = lightspeech.inference( input_ids=tf.expand_dims(tf.convert_to_tensor(input_ids, dtype=tf.int32), 0), speaker_ids=tf.convert_to_tensor( [processor.speakers_map[speaker_name]], dtype=tf.int32 ), speed_ratios=tf.convert_to_tensor([1.0], dtype=tf.float32), f0_ratios=tf.convert_to_tensor([1.0], dtype=tf.float32), energy_ratios=tf.convert_to_tensor([1.0], dtype=tf.float32), ) audio = mb_melgan.inference(mel)[0, :, 0] sf.write("./audio.wav", audio, 44100, "PCM_16") ``` ## Disclaimer Do consider the biases which came from pre-training datasets that may be carried over into the results of this model. ## Authors MB-MelGAN HiFi PostNets SW v1 was trained and evaluated by [David Samuel Setiawan](https://davidsamuell.github.io/), [Wilson Wongso](https://wilsonwongso.dev/). All computation and development are done on Scaleway. ## Framework versions - TensorFlowTTS 1.8 - TensorFlow 2.7.0