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Implementation of AudioLM, a SOTA Language Modeling Approach to Audio Generation out of Google Research, in Pytorch

AudioLM - Pytorch

Implementation of AudioLM, a Language Modeling Approach to Audio Generation out of Google Research, in Pytorch

It also extends the work for conditioning with classifier free guidance with T5. This allows for one to do text-to-audio or TTS, not offered in the paper. Yes, this means VALL-E can be trained from this repository. It is essentially the same.

Please join Join us on Discord if you are interested in replicating this work in the open

This repository now also contains a MIT licensed version of SoundStream. It is also compatible with EnCodec, which is also MIT-licensed at the time of writing.

Update: AudioLM was essentially used to 'solve' music generation in the new MusicLM

In the future, this movie clip would no longer make any sense. You would just prompt an AI instead.

Appreciation

  • Stability.ai for the generous sponsorship to work and open source cutting edge artificial intelligence research

  • šŸ¤— Huggingface for their amazing accelerate and transformers libraries

  • MetaAI for Fairseq and the liberal license

  • @eonglints and Joseph for offering their professional advice and expertise as well as pull requests!

  • @djqualia, @yigityu, @inspirit, and @BlackFox1197 for helping with the debugging of soundstream

  • Allen and LWprogramming for reviewing the code and submitting bug fixes!

  • Ilya for finding an issue with multi-scale discriminator downsampling and for soundstream trainer improvements

  • Andrey for identifying a missing loss in soundstream and guiding me through the proper mel spectrogram hyperparameters

  • Alejandro and Ilya for sharing their results with training soundstream, and for working through a few issues with the local attention positional embeddings

  • LWprogramming for adding Encodec compatibility!

  • LWprogramming for finding an issue with handling of the EOS token when sampling from the FineTransformer!

  • @YoungloLee for identifying a big bug in the 1d causal convolution for soundstream related to padding not accounting for strides!

  • Hayden for pointing out some discrepancies in the multi-scale discriminator for Soundstream

Install

$ pip install audiolm-pytorch

Usage

SoundStream & Encodec

There are two options for the neural codec. If you want to use the pretrained 24kHz Encodec, just create an Encodec object as follows:

from audiolm_pytorch import EncodecWrapper
encodec = EncodecWrapper()
# Now you can use the encodec variable in the same way you'd use the soundstream variables below.

Otherwise, to stay more true to the original paper, you can use SoundStream. First, SoundStream needs to be trained on a large corpus of audio data

from audiolm_pytorch import SoundStream, SoundStreamTrainer

soundstream = SoundStream(
    codebook_size = 1024,
    rq_num_quantizers = 8,
    rq_groups = 2,                # this paper proposes using multi-headed residual vector quantization - https://arxiv.org/abs/2305.02765
    attn_window_size = 128,       # local attention receptive field at bottleneck
    attn_depth = 2                # 2 local attention transformer blocks - the soundstream folks were not experts with attention, so i took the liberty to add some. encodec went with lstms, but attention should be better
)

trainer = SoundStreamTrainer(
    soundstream,
    folder = '/path/to/audio/files',
    batch_size = 4,
    grad_accum_every = 8,         # effective batch size of 32
    data_max_length_seconds = 2,  # train on 2 second audio
    num_train_steps = 1_000_000
).cuda()

trainer.train()

# after a lot of training, you can test the autoencoding as so

audio = torch.randn(10080).cuda()
recons = soundstream(audio, return_recons_only = True) # (1, 10080) - 1 channel

You can also use soundstreams that are specific to AudioLM and MusicLM by importing AudioLMSoundStream and MusicLMSoundStream respectively

from audiolm_pytorch import AudioLMSoundStream, MusicLMSoundStream

soundstream = AudioLMSoundStream(...) # say you want the hyperparameters as in Audio LM paper

# rest is the same as above

As of version 0.17.0, you can now invoke the class method on SoundStream to load from checkpoint files, without having to remember your configurations.

from audiolm_pytorch import SoundStream

soundstream = SoundStream.init_and_load_from('./path/to/checkpoint.pt')

Hierarchical Transformers

Then three separate transformers (SemanticTransformer, CoarseTransformer, FineTransformer) need to be trained

ex. SemanticTransformer

import torch
from audiolm_pytorch import HubertWithKmeans, SemanticTransformer, SemanticTransformerTrainer

# hubert checkpoints can be downloaded at
# https://github.com/facebookresearch/fairseq/tree/main/examples/hubert

wav2vec = HubertWithKmeans(
    checkpoint_path = './hubert/hubert_base_ls960.pt',
    kmeans_path = './hubert/hubert_base_ls960_L9_km500.bin'
)

semantic_transformer = SemanticTransformer(
    num_semantic_tokens = wav2vec.codebook_size,
    dim = 1024,
    depth = 6,
    flash_attn = True
).cuda()


trainer = SemanticTransformerTrainer(
    transformer = semantic_transformer,
    wav2vec = wav2vec,
    folder ='/path/to/audio/files',
    batch_size = 1,
    data_max_length = 320 * 32,
    num_train_steps = 1
)

trainer.train()

ex. CoarseTransformer

import torch
from audiolm_pytorch import HubertWithKmeans, SoundStream, CoarseTransformer, CoarseTransformerTrainer

wav2vec = HubertWithKmeans(
    checkpoint_path = './hubert/hubert_base_ls960.pt',
    kmeans_path = './hubert/hubert_base_ls960_L9_km500.bin'
)

soundstream = SoundStream.init_and_load_from('/path/to/trained/soundstream.pt')

coarse_transformer = CoarseTransformer(
    num_semantic_tokens = wav2vec.codebook_size,
    codebook_size = 1024,
    num_coarse_quantizers = 3,
    dim = 512,
    depth = 6,
    flash_attn = True
)

trainer = CoarseTransformerTrainer(
    transformer = coarse_transformer,
    codec = soundstream,
    wav2vec = wav2vec,
    folder = '/path/to/audio/files',
    batch_size = 1,
    data_max_length = 320 * 32,
    num_train_steps = 1_000_000
)

trainer.train()

ex. FineTransformer

import torch
from audiolm_pytorch import SoundStream, FineTransformer, FineTransformerTrainer

soundstream = SoundStream.init_and_load_from('/path/to/trained/soundstream.pt')

fine_transformer = FineTransformer(
    num_coarse_quantizers = 3,
    num_fine_quantizers = 5,
    codebook_size = 1024,
    dim = 512,
    depth = 6,
    flash_attn = True
)

trainer = FineTransformerTrainer(
    transformer = fine_transformer,
    codec = soundstream,
    folder = '/path/to/audio/files',
    batch_size = 1,
    data_max_length = 320 * 32,
    num_train_steps = 1_000_000
)

trainer.train()

All together now

from audiolm_pytorch import AudioLM

audiolm = AudioLM(
    wav2vec = wav2vec,
    codec = soundstream,
    semantic_transformer = semantic_transformer,
    coarse_transformer = coarse_transformer,
    fine_transformer = fine_transformer
)

generated_wav = audiolm(batch_size = 1)

# or with priming

generated_wav_with_prime = audiolm(prime_wave = torch.randn(1, 320 * 8))

# or with text condition, if given

generated_wav_with_text_condition = audiolm(text = ['chirping of birds and the distant echos of bells'])

Text Conditioned Audio Synthesis

Update: Looks like this will work, given 'VALL-E'

ex. Semantic Transformer

import torch
from audiolm_pytorch import HubertWithKmeans, SemanticTransformer, SemanticTransformerTrainer

wav2vec = HubertWithKmeans(
    checkpoint_path = './hubert/hubert_base_ls960.pt',
    kmeans_path = './hubert/hubert_base_ls960_L9_km500.bin'
)

semantic_transformer = SemanticTransformer(
    num_semantic_tokens = 500,
    dim = 1024,
    depth = 6,
    has_condition = True,               # this will have to be set to True
    cond_as_self_attn_prefix = True     # whether to condition as prefix to self attention, instead of cross attention, as was done in 'VALL-E' paper
).cuda()

# mock text video dataset (as an example)

# you will have to extend your own from `Dataset`, and return an audio tensor as well as a string (the audio description) in any order (the framework will autodetect and route it into the transformer)

from torch.utils.data import Dataset

class MockTextAudioDataset(Dataset):
    def __init__(self, length = 100, audio_length = 320 * 32):
        super().__init__()
        self.audio_length = audio_length
        self.len = length

    def __len__(self):
        return self.len

    def __getitem__(self, idx):
        mock_audio = torch.randn(self.audio_length)
        mock_caption = 'audio caption'
        return mock_caption, mock_audio

dataset = MockTextAudioDataset()

# instantiate semantic transformer trainer and train

trainer = SemanticTransformerTrainer(
    transformer = semantic_transformer,
    wav2vec = wav2vec,
    dataset = dataset,
    batch_size = 4,
    grad_accum_every = 8,
    data_max_length = 320 * 32,
    num_train_steps = 1_000_000
)

trainer.train()

# after much training above

sample = trainer.generate(text = ['sound of rain drops on the rooftops'], batch_size = 1, max_length = 2) # (1, < 128) - may terminate early if it detects [eos]

Multi-GPU

Because all the trainer classes uses šŸ¤— Accelerator, you can easily do multi gpu training by using the accelerate command as so

At the project root

$ accelerate config

Then, in the same directory

$ accelerate launch train.py

Todo

  • complete CoarseTransformer

  • use fairseq vq-wav2vec for embeddings

  • add conditioning

  • add classifier free guidance

  • add unique consecutive for

  • incorporate ability to use hubert intermediate features as semantic tokens, recommended by eonglints

  • accommodate variable lengthed audio, bring in eos token

  • make sure unique consecutive works with coarse transformer

  • pretty printing all discriminator losses to log

  • handle when generating semantic tokens, that last logits may not be necessarily the last in the sequence given unique consecutive processing

  • complete sampling code for both Coarse and Fine Transformers, which will be tricky

  • make sure full inference with or without prompting works on the AudioLM class

  • complete full training code for soundstream, taking care of discriminator training

  • add efficient gradient penalty for discriminators for soundstream

  • wire up sample hz from sound dataset -> transformers, and have proper resampling within during training - think about whether to allow for dataset to have sound files of varying or enforce same sample hz

  • full transformer training code for all three transformers

  • refactor so semantic transformer has a wrapper to that handles unique consecutives as well as wav to hubert or vq-wav2vec

  • simply not self attend to eos token on the prompting side (semantic for coarse transformer, coarse for fine transformer)

  • add structured dropout from forgetful causal masking, far better than traditional dropouts

  • figure out how to suppress logging in fairseq

  • assert that all three transformers passed into audiolm is compatible

  • allow for specialized relative positional embeddings in fine transformer based on absolute matching positions of quantizers between coarse and fine

  • allow for grouped residual vq in soundstream (use GroupedResidualVQ from vector-quantize-pytorch lib), from hifi-codec

  • add flash attention with NoPE

  • accept prime wave in AudioLM as a path to an audio file, and auto resample for semantic vs acoustic

  • design a hierarchical coarse and fine transformer

  • investigate spec decoding, first test in x-transformers, then port over if applicable

  • redo the positional embeddings in the presence of groups in residual vq

  • test with speech synthesis for starters

  • cli tool, something like audiolm generate <wav.file | text> and save generated wav file to local directory

  • return a list of waves in the case of variable lengthed audio

  • just take care of the edge case in coarse transformer text conditioned training, where the raw wave is resampled at different frequencies. autodetermine how to route based on length

Citations

@inproceedings{Borsos2022AudioLMAL,
  title  = {AudioLM: a Language Modeling Approach to Audio Generation},
  author = {Zal{\'a}n Borsos and Rapha{\"e}l Marinier and Damien Vincent and Eugene Kharitonov and Olivier Pietquin and Matthew Sharifi and Olivier Teboul and David Grangier and Marco Tagliasacchi and Neil Zeghidour},
  year   = {2022}
}
@misc{https://doi.org/10.48550/arxiv.2107.03312,
  title  = {SoundStream: An End-to-End Neural Audio Codec},
  author = {Zeghidour, Neil and Luebs, Alejandro and Omran, Ahmed and Skoglund, Jan and Tagliasacchi, Marco},
  publisher = {arXiv},
  url    = {https://arxiv.org/abs/2107.03312},
  year   = {2021}
}
@misc{shazeer2020glu,
    title   = {GLU Variants Improve Transformer},
    author  = {Noam Shazeer},
    year    = {2020},
    url     = {https://arxiv.org/abs/2002.05202}
}
@article{Shazeer2019FastTD,
    title   = {Fast Transformer Decoding: One Write-Head is All You Need},
    author  = {Noam M. Shazeer},
    journal = {ArXiv},
    year    = {2019},
    volume  = {abs/1911.02150}
}
@article{Ho2022ClassifierFreeDG,
    title   = {Classifier-Free Diffusion Guidance},
    author  = {Jonathan Ho},
    journal = {ArXiv},
    year    = {2022},
    volume  = {abs/2207.12598}
}
@misc{crowson2022,
    author  = {Katherine Crowson},
    url     = {https://twitter.com/rivershavewings}
}
@misc{ding2021cogview,
    title   = {CogView: Mastering Text-to-Image Generation via Transformers},
    author  = {Ming Ding and Zhuoyi Yang and Wenyi Hong and Wendi Zheng and Chang Zhou and Da Yin and Junyang Lin and Xu Zou and Zhou Shao and Hongxia Yang and Jie Tang},
    year    = {2021},
    eprint  = {2105.13290},
    archivePrefix = {arXiv},
    primaryClass = {cs.CV}
}
@article{Liu2022FCMFC,
    title   = {FCM: Forgetful Causal Masking Makes Causal Language Models Better Zero-Shot Learners},
    author  = {Hao Liu and Xinyang Geng and Lisa Lee and Igor Mordatch and Sergey Levine and Sharan Narang and P. Abbeel},
    journal = {ArXiv},
    year    = {2022},
    volume  = {abs/2210.13432}
}
@inproceedings{anonymous2022normformer,
    title   = {NormFormer: Improved Transformer Pretraining with Extra Normalization},
    author  = {Anonymous},
    booktitle = {Submitted to The Tenth International Conference on Learning Representations },
    year    = {2022},
    url     = {https://openreview.net/forum?id=GMYWzWztDx5},
    note    = {under review}
}
@misc{liu2021swin,
    title   = {Swin Transformer V2: Scaling Up Capacity and Resolution},
    author  = {Ze Liu and Han Hu and Yutong Lin and Zhuliang Yao and Zhenda Xie and Yixuan Wei and Jia Ning and Yue Cao and Zheng Zhang and Li Dong and Furu Wei and Baining Guo},
    year    = {2021},
    eprint  = {2111.09883},
    archivePrefix = {arXiv},
    primaryClass = {cs.CV}
}
@article{Li2021LocalViTBL,
    title   = {LocalViT: Bringing Locality to Vision Transformers},
    author  = {Yawei Li and K. Zhang and Jie Cao and Radu Timofte and Luc Van Gool},
    journal = {ArXiv},
    year    = {2021},
    volume  = {abs/2104.05707}
}
@article{Defossez2022HighFN,
    title   = {High Fidelity Neural Audio Compression},
    author  = {Alexandre D'efossez and Jade Copet and Gabriel Synnaeve and Yossi Adi},
    journal = {ArXiv},
    year    = {2022},
    volume  = {abs/2210.13438}
}
@article{Hu2017SqueezeandExcitationN,
    title   = {Squeeze-and-Excitation Networks},
    author  = {Jie Hu and Li Shen and Gang Sun},
    journal = {2018 IEEE/CVF Conference on Computer Vision and Pattern Recognition},
    year    = {2017},
    pages   = {7132-7141}
}
@inproceedings{Yang2023HiFiCodecGV,
    title   = {HiFi-Codec: Group-residual Vector quantization for High Fidelity Audio Codec},
    author  = {Dongchao Yang and Songxiang Liu and Rongjie Huang and Jinchuan Tian and Chao Weng and Yuexian Zou},
    year    = {2023}
}
@article{Kazemnejad2023TheIO,
    title   = {The Impact of Positional Encoding on Length Generalization in Transformers},
    author  = {Amirhossein Kazemnejad and Inkit Padhi and Karthikeyan Natesan Ramamurthy and Payel Das and Siva Reddy},
    journal = {ArXiv},
    year    = {2023},
    volume  = {abs/2305.19466}
}
@inproceedings{dao2022flashattention,
    title   = {Flash{A}ttention: Fast and Memory-Efficient Exact Attention with {IO}-Awareness},
    author  = {Dao, Tri and Fu, Daniel Y. and Ermon, Stefano and Rudra, Atri and R{\'e}, Christopher},
    booktitle = {Advances in Neural Information Processing Systems},
    year    = {2022}
}

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Python
311
star
62

slot-attention

Implementation of Slot Attention from GoogleAI
Python
303
star
63

q-transformer

Implementation of Q-Transformer, Scalable Offline Reinforcement Learning via Autoregressive Q-Functions, out of Google Deepmind
Python
293
star
64

BS-RoFormer

Implementation of Band Split Roformer, SOTA Attention network for music source separation out of ByteDance AI Labs
Python
289
star
65

classifier-free-guidance-pytorch

Implementation of Classifier Free Guidance in Pytorch, with emphasis on text conditioning, and flexibility to include multiple text embedding models
Python
282
star
66

transformer-in-transformer

Implementation of Transformer in Transformer, pixel level attention paired with patch level attention for image classification, in Pytorch
Python
277
star
67

axial-attention

Implementation of Axial attention - attending to multi-dimensional data efficiently
Python
273
star
68

conformer

Implementation of the convolutional module from the Conformer paper, for use in Transformers
Python
272
star
69

mixture-of-experts

A Pytorch implementation of Sparsely-Gated Mixture of Experts, for massively increasing the parameter count of language models
Python
264
star
70

deformable-attention

Implementation of Deformable Attention in Pytorch from the paper "Vision Transformer with Deformable Attention"
Python
258
star
71

magic3d-pytorch

Implementation of Magic3D, Text to 3D content synthesis, in Pytorch
Python
258
star
72

x-unet

Implementation of a U-net complete with efficient attention as well as the latest research findings
Python
252
star
73

routing-transformer

Fully featured implementation of Routing Transformer
Python
251
star
74

Adan-pytorch

Implementation of the Adan (ADAptive Nesterov momentum algorithm) Optimizer in Pytorch
Python
245
star
75

spear-tts-pytorch

Implementation of Spear-TTS - multi-speaker text-to-speech attention network, in Pytorch
Python
241
star
76

st-moe-pytorch

Implementation of ST-Moe, the latest incarnation of MoE after years of research at Brain, in Pytorch
Python
237
star
77

perfusion-pytorch

Implementation of Key-Locked Rank One Editing, from Nvidia AI
Python
229
star
78

equiformer-pytorch

Implementation of the Equiformer, SE3/E3 equivariant attention network that reaches new SOTA, and adopted for use by EquiFold for protein folding
Python
227
star
79

segformer-pytorch

Implementation of Segformer, Attention + MLP neural network for segmentation, in Pytorch
Python
227
star
80

sinkhorn-transformer

Sinkhorn Transformer - Practical implementation of Sparse Sinkhorn Attention
Python
222
star
81

pixel-level-contrastive-learning

Implementation of Pixel-level Contrastive Learning, proposed in the paper "Propagate Yourself", in Pytorch
Python
220
star
82

lumiere-pytorch

Implementation of Lumiere, SOTA text-to-video generation from Google Deepmind, in Pytorch
Python
216
star
83

local-attention

An implementation of local windowed attention for language modeling
Python
216
star
84

CoLT5-attention

Implementation of the conditionally routed attention in the CoLT5 architecture, in Pytorch
Python
216
star
85

natural-speech-pytorch

Implementation of the neural network proposed in Natural Speech, a text-to-speech generator that is indistinguishable from human recordings for the first time, from Microsoft Research
Python
215
star
86

soft-moe-pytorch

Implementation of Soft MoE, proposed by Brain's Vision team, in Pytorch
Python
211
star
87

se3-transformer-pytorch

Implementation of SE3-Transformers for Equivariant Self-Attention, in Pytorch. This specific repository is geared towards integration with eventual Alphafold2 replication.
Python
211
star
88

block-recurrent-transformer-pytorch

Implementation of Block Recurrent Transformer - Pytorch
Python
205
star
89

Mega-pytorch

Implementation of Mega, the Single-head Attention with Multi-headed EMA architecture that currently holds SOTA on Long Range Arena
Python
201
star
90

simple-hierarchical-transformer

Experiments around a simple idea for inducing multiple hierarchical predictive model within a GPT
Python
198
star
91

med-seg-diff-pytorch

Implementation of MedSegDiff in Pytorch - SOTA medical segmentation using DDPM and filtering of features in fourier space
Python
195
star
92

triton-transformer

Implementation of a Transformer, but completely in Triton
Python
195
star
93

jax2torch

Use Jax functions in Pytorch
Python
194
star
94

flash-cosine-sim-attention

Implementation of fused cosine similarity attention in the same style as Flash Attention
Cuda
194
star
95

halonet-pytorch

Implementation of the šŸ˜‡ Attention layer from the paper, Scaling Local Self-Attention For Parameter Efficient Visual Backbones
Python
193
star
96

attention

This repository will house a visualization that will attempt to convey instant enlightenment of how Attention works to someone not working in artificial intelligence, with 3Blue1Brown as inspiration
HTML
189
star
97

recurrent-interface-network-pytorch

Implementation of Recurrent Interface Network (RIN), for highly efficient generation of images and video without cascading networks, in Pytorch
Python
188
star
98

electra-pytorch

A simple and working implementation of Electra, the fastest way to pretrain language models from scratch, in Pytorch
Python
186
star
99

PaLM-jax

Implementation of the specific Transformer architecture from PaLM - Scaling Language Modeling with Pathways - in Jax (Equinox framework)
Python
184
star
100

unet-stylegan2

A Pytorch implementation of Stylegan2 with UNet Discriminator
Python
182
star