This is mitigated during inference by re-inferencing on the same audio chunk with temperature-based sampling when the model detects that inference has failed. logits (torch.FloatTensor of shape (batch_size, config.num_labels)) Classification (or regression if config.num_labels==1) scores (before SoftMax). We wrote this series of posts after an engagement where we collaborated closely with the team at Chorus. Site design / logo 2023 Stack Exchange Inc; user contributions licensed under CC BY-SA. The beam search decoder looks at k probable tokens, where k is the beam size specified by the user. Can you tell us what you liked about it? output_char_offsets: bool = False Wav2Vec2 was proposed in wav2vec 2.0: A Framework for Self-Supervised Learning of Speech In this analysis, I used the pre-trained model in the wav2letter download. output_attentions: typing.Optional[bool] = None facebook/wav2vec2-base-960h architecture. beam_prune_logp: typing.Optional[float] = None They also happen to be the simplest and potentially the fastest of the e2e models. The Wav2Vec2ForAudioFrameClassification forward method, overrides the __call__ special method. return_attention_mask=True or if attention_mask is in self.model_input_names). padding: typing.Union[bool, str, transformers.utils.generic.PaddingStrategy] = False simply be padded with 0 and passed without attention_mask. Indeed, as you can see below, the accuracy is pretty nice. decoding. For our purposes, we only need to know that CTC encoders learn a weak internal representation of language. All rights belong to their respective owners. Interestingly, the models display opposing inference speed trends. Audio pre-processing is a crucial, yet often overlooked component of ASR inference mechanics. Co-occurrence between phonemes on y-axis and quantizations on x-axis ( source ). Encoder/decoders have a more complex architecture than standalone encoders because they have more interacting parts. Feature Encoding. output_attentions: typing.Optional[bool] = None A BatchEncoding with the following fields: input_ids List of token ids to be fed to a model. Constructs a Wav2Vec2 processor which wraps a Wav2Vec2 feature extractor, a Wav2Vec2 CTC tokenizer and a decoder When used in normal mode, this method forwards all its arguments to Wav2Vec2FeatureExtractors How is Docker different from a virtual machine? Find centralized, trusted content and collaborate around the technologies you use most. The installation and use require much less effort than the other Vosk, NeMo, or wav2letter. In Wav2Vec2 model according to the specified arguments, defining the model architecture. We do this for every decoded sequence in the batch. etc.). These studies typically involve training a sequence of increasing-capacity models where the capacity is incremented by increasing all size parameters simultaneously, in an ad hoc fashion. The n-gram LM learns conditional word probabilities by counting their occurrences in a corpus. They Chorus is a conversation intelligence platform that uses AI to analyze sales calls to drive team performance. The audio window is embedded with the encoder and then mapped to a predicted text sequence auto-regressively by the decoder, which uses the encoder output as a context vector. can anybody elaborate on this please? ). Will be a Wav2Vec2CTCTokenizerOutput when diversity_loss: typing.Optional[torch.FloatTensor] = None torchaudio. Otherwise, batch_decode() performance will be slower than calling decode() for each audio individually, as it internally instantiates a new Pool for every call. Estimate the class of the acoustic features frame-by-frame. Output type of FlaxWav2Vec2ForPreTrainingOutput, with potential hidden states and attentions. output_attentions: typing.Optional[bool] = None In each task, we convert raw audio waveforms into text. we can use torchaudio.functional.resample() for resampling. labels: typing.Optional[torch.Tensor] = None of ICASSP, Cited by: 4.4. Learn about PyTorchs features and capabilities. We use the wav2letter++ toolkit for training and evaluation of acoustic models (Pratap et al.,2018). hotwords: typing.Optional[typing.Iterable[str]] = None attentions (tuple(torch.FloatTensor), optional, returned when output_attentions=True is passed or when config.output_attentions=True) Tuple of torch.FloatTensor (one for each layer) of shape (batch_size, num_heads, sequence_length, sequence_length). This is where language models (LM) come into play. For evaluation, we use the wav2letter++ [32] beam search decoder with a beam size 1500 and a 4-gram LM trained on the same text as the other LMs. Now that we have the predictions, we calculate prediction quality by word error rate (WER), using the jiwer package. logits (tf.Tensor of shape (batch_size, sequence_length, config.vocab_size)) Prediction scores of the language modeling head (scores for each vocabulary token before SoftMax). below, the accuracy is pretty nice. them into a set of categories. Why does Jesus turn to the Father to forgive in Luke 23:34? When performing resampling multiple times on the same set of sample rates, Please refer to the docstring of the above two methods for more information. Note that we call get_data_ptr_as_bytes on the tensors we created earlier. There is not any documnetation available for that. Wav2Vec2 model provides method to perform the feature extraction and See the docstring of call() and decode() for more information. My end game is to use it for transcriptions of audio files and possible real-time transcription in Python. # compare word offsets with audio `common_voice_en_100038.mp3` online on the dataset viewer: # https://huggingface.co/datasets/common_voice/viewer/en/train, : typing.Union[typing.List[int], typing.List[typing.List[int]], ForwardRef('np.ndarray'), ForwardRef('torch.Tensor'), ForwardRef('tf.Tensor')], : typing.Union[numpy.ndarray, typing.List[float], typing.List[numpy.ndarray], typing.List[typing.List[float]]], : typing.Union[>, NoneType] = None, : typing.Optional[typing.Iterable[str]] = None, "patrickvonplaten/wav2vec2-base-100h-with-lm", # Let's see how to use a user-managed pool for batch decoding multiple audios, "hf-internal-testing/librispeech_asr_dummy", # prepare speech data for batch inference. In the next section, well compare the beam search decoder and Viterbi decoder. transformers.modeling_outputs.Wav2Vec2BaseModelOutput or tuple(torch.FloatTensor). using, A blog post on how to deploy Wav2Vec2 for, a path or url to a saved feature extractor JSON, having all inputs as keyword arguments (like PyTorch models), or. freeze_feature_encoder: bool = False To subscribe to this RSS feed, copy and paste this URL into your RSS reader. It comprises several steps including transcoding the audio into a required format (e.g., 16-bit PCM), resampling it at a specified rate, splitting it into chunks of a specified size, deriving acoustic features (e.g., log-mel spectrograms) over the chunks, and then grouping chunks together to form batches for inference. ). text: typing.Union[str, typing.List[str], typing.List[typing.List[str]]] = None Experiments using all labeled data of Librispeech achieve 1.8/3.3 WER on the By clicking Post Your Answer, you agree to our terms of service, privacy policy and cookie policy. We presented wav2vec 2.0, a framework for self-supervised learning of speech representations which masks latent representations of the raw waveform and solves a contrastive task over quantized speech representations. about any of this, as you can just pass inputs like you would to any other Python function! being the dimension of the last convolutional layer. Configuration objects inherit from PretrainedConfig and can be used to control the model outputs. Check out this notebook if you are interested in distributing inference using Ray. wav2vec2-base, have not been trained using For a fixed architecture, larger capacity models tend to run more slowly than smaller capacity models because: They simply require more computation and a lot of that is sequential in nature (i.e. and get access to the augmented documentation experience. Join the PyTorch developer community to contribute, learn, and get your questions answered. .. warning:: attention_mask should only be passed This makes it memory intensive on a GPU. Is a hot staple gun good enough for interior switch repair? Continuing this trend, in September 2022, OpenAI introduced Whisper, an open-source ASR model trained on nearly 700,000 hours of multilingual speech data. transformers.modeling_flax_outputs.FlaxMaskedLMOutput or tuple(torch.FloatTensor), transformers.modeling_flax_outputs.FlaxMaskedLMOutput or tuple(torch.FloatTensor). There is substantial variation in speed and accuracy across the capacity range, with the largest models generally producing the most accurate predictions but running up to ~30x slower than the smaller ones. https://github.com/facebookresearch/wav2letter/issues/436 freeze_feature_encoder: bool = False params: dict = None As the first two rows of the table show, its actually 2.9 times faster than wav2vec_big_960h. from_pretrained(), and Now, lets create a set of inference tasks and start the distributed inference! We start by defining greedy decoding algorithm. Wav2Vec2 Model with a sequence classification head on top (a linear layer over the pooled output) for tasks like beam_width: typing.Optional[int] = None For our comparison, we use Kaldi's Gigaspeech XL model which is a conventional pipeline model trained on the recent Gigaspeech dataset. This paper presents a simple end-to-end model for speech recognition, combining a convolutional network based acoustic model and a graph decoding. simply be padded with 0 and passed without attention_mask. information. attention_mask = None Learn more, including about available controls: Cookies Policy. pad_token_id = 0 Wav2Letter++: a fast open-source speech recognition system. A transformers.modeling_tf_outputs.TFCausalLMOutput or a tuple of tf.Tensor (if for more information. position_ids: typing.Optional[tensorflow.python.framework.ops.Tensor] = None We talked about wav2vec 2.0 in our first post and showed how to compress wav2vec 2.0 in our second post in this series, to increase inference speed. Welcome to another video, in this video I'll be showing you how to download and use a pretrained model named Wav2Vec to do Speech Recognition, Wav2Vec is a state-of-the-art model for speech recognition, it uses a similar training strategy as word2vec to learn speech representations using unlabeled data and then fine-tune the model on a labeled data, it also uses a Transformer architecture, using the HuggingFace library called transformers you can use or fine-tune a variety of models, today we'll focus o Wav2Vec, since our goal is to have one of the best models available for speech recognition. as a regular TF 2.0 Keras Model and refer to the TF 2.0 documentation for all matter related to general usage and projected_states (jnp.ndarray of shape (batch_size, sequence_length, config.proj_codevector_dim)) Hidden-states of the model projected to config.proj_codevector_dim that can be used to predict the masked Whisper developers handled this in the same way as different tasks, i.e., by including timestamp tokens as first-class entries in the model's vocabulary and inserting them directly at particular locations in the training text. Main method to featurize and prepare for the model one or several sequence(s). freeze_feature_encoder: bool = False num_hidden_layers = 12 I am needing advice on this topic. List[str] or Wav2Vec2CTCTokenizerOutput. Currently, multiprocessing is available only on Unix Instantiate a Wav2Vec2ProcessorWithLM from a pretrained Wav2Vec2 processor. **kwargs torch.FloatTensor (if return_dict=False is passed or when config.return_dict=False) comprising various In this analysis, I used the danzuu model. The FlaxWav2Vec2ForPreTraining forward method, overrides the __call__ special method. The TFWav2Vec2ForCTC forward method, overrides the __call__ special method. You can step through the speech_to_text_using_wav2vec.mlx file to examine the structure of each module. extract_features (torch.FloatTensor of shape (batch_size, sequence_length, conv_dim[-1])) Sequence of extracted feature vectors of the last convolutional layer of the model. How can I recognize one? We do not host any of the videos or images on our servers. projected_states (torch.FloatTensor of shape (batch_size, sequence_length, config.proj_codevector_dim)) Hidden-states of the model projected to config.proj_codevector_dim that can be used to predict the masked different results depending on whether input_values is padded or not. Learning unsupervised representations with wav2vec. AI & Engineering. Otherwise, As a result, you may get the distinct impression that these models ARE YELLING AT YOU. Whisper predicts "segment-level" timestamps as part of its output. We use a zero matrix here, so were not giving this information to the Viterbi decoder. The process of speech recognition looks like the following. In this paper, we show that pseudo-labeling and pre-training with wav2vec 2.0 are complementary in a variety of labeled data setups. If, however, you want to use the second elements depending on the configuration () and inputs. output_word_offsets: bool = False Connect and share knowledge within a single location that is structured and easy to search. Where developers & technologists share private knowledge with coworkers, Reach developers & technologists worldwide, The open-source game engine youve been waiting for: Godot (Ep. It includes additional features, such as being able to add a microphone for live transcription. Whisper is a family of encoder/decoder ASR models trained in a supervised fashion, on a large corpus of crawled, multilingual speech data. text: typing.Union[typing.List[str], str] resources, such as word dictionary and language models. information are not used, and only one transcript can be generated. This model was contributed by patrickvonplaten. Does anyone know how to use wav2letter in 2021? I could not get Flashlight to install. layer_norm_eps = 1e-05 we just replaced spectrogram features in wav2letter with the wav2vec ones. There are several unique aspects to its model DNA, discussed below: Its architecture is "deceptively simple" and comprises a stack of 2D CNNs followed by a symmetric transformer encoder/decoder stack. Wav2Vec2 is a pretrained model for Automatic Speech Recognition (ASR) and was released in September 2020 by Alexei Baevski, Michael Auli, and Alex Conneau.. num_conv_pos_embeddings = 128 library implements for all its model (such as downloading or saving etc.). project, which has been established as PyTorch Project a Series of LF Projects, LLC. This is an important point: wav2vec is not a full automatic speech recognition (ASR) system . Note that this only specifies the dtype of the computation and does not influence the dtype of model specified all the computation will be performed with the given dtype. Whisper was trained in a supervised fashion on a very large corpus comprising 680k hours of crawled, multilingual speech data. ( Here are previous posts: The ideas behind Wav2Vec are extremely hot today - pretraining, The pre-trained weights without fine-tuning can be fine-tuned pad_to_multiple_of: typing.Optional[int] = None ( Facebooks compute resources in your own research. Throughput represents, intuitively, the number of audio hours processed per hour of inference time. The speech-to-text softwares I used were Vosk, NeMo, wav2letter, and DeepSpeech2. behavior. We distribute these tasks to multiple CPU cores using Ray. Despite its importance, audio-preprocessing is usually not well described in open-source model documentation and may require delving deeply into underlying source code to understand a particular model's audio pre-processing requirements. The model inference time depends on the model's architecture, inference algorithm, and capacity. Is there a proper earth ground point in this switch box? probability. A transformers.modeling_outputs.SequenceClassifierOutput or a tuple of please see www.lfprojects.org/policies/. This simply reflects the fact that Whisper inference takes significantly more time on the GPU as a result of the auto-regressive nature of its inference algorithm. Wav2Vec2 was proposed in wav2vec 2.0: A Framework for Self-Supervised Learning of Speech Please take a look at the example below to better understand how to make use of output_word_offsets. sorry i just saw this. last_hidden_state (tf.Tensor of shape (batch_size, sequence_length, hidden_size)) Sequence of hidden-states at the output of the last layer of the model. From here, I tried doing git remote set-url origin https://github.com/facebookresearch/wav2letter.git and moving forward, eventually reaching the error: From here, I shut down and deleted the container. Asking for help, clarification, or responding to other answers. with language model support into a single processor for language model boosted speech recognition decoding. This model is a PyTorch torch.nn.Module sub-class. It is used to instantiate an The Whisper developers accomplished this by training the model on multiple supervised tasks and using special task-specific tokens which were added as first-class entries in the decoder's vocabulary and then included in the decoder's input text. output_hidden_states: typing.Optional[bool] = None process_data_sample also takes in target_dict, a map, from tokens to indices, to process the decoder output. wav2vec 2.0 uses significantly more GPU memory than Whisper, even in the 2080 Ti test where they are both operating on the same batch size. here. required, but it is managable. output_hidden_size = None Here are the pre-processing steps one must undertake to work with Kaldi: Pre-chunking it into manageable sizes (I used non-overlapping 30 second snippets), Staging the chunks as flat files on the disk along with some additional metadata, Using Kaldi's command line interface to generate and stage audio features over your audio snippets. output. token_min_logp: typing.Optional[float] = None We obtained this student model through knowledge distillation. Word error rate is based on the Levenshtein distance (or "edit distance") which measures the differences between two stringsin this case, a predicted transcript produced an ASR model and a human-labeled transcript. For the 2080 Ti, we were limited to a batch size of 1 while for the A5000 we were able to increase the batch size to 3. return_overflowing_tokens=True). 7 Stars. elements depending on the configuration (Wav2Vec2Config) and inputs. Sec. The spread in accuracy for the models was so broad, that we found it necessary to use a log scale on the x-axis. ). The default behavior is to infer sequentially on 30-second windows of audio. This model is also a Flax Linen Overall, NeMo performs the best in terms of transcription time and can be very accurate (as seen from the male audio). Please refer to the docstring of the above two methods for more information. clean/other test sets. length (like XLNet) truncation/padding to a maximum length will be deactivated. num_conv_pos_embedding_groups = 16 return_dict: typing.Optional[bool] = None How to find all files containing specific text (string) on Linux? Despite the notoriety associated with wav2vec 2.0, there are relatively few examples of open-source ASR versions available. Of FlaxWav2Vec2ForPreTrainingOutput, with potential hidden states and attentions in distributing inference using Ray, str, ]... Hours of crawled, multilingual speech data giving this information to the Viterbi decoder the other Vosk NeMo. Padded with 0 and passed without attention_mask at Chorus turn to the specified arguments, defining the model 's,. Feed, copy and paste this URL into wav2vec vs wav2letter++ RSS reader relatively few examples open-source. Various in this analysis, I used the danzuu model such as word dictionary and language models ( Pratap al.,2018! Whisper was trained in a variety of labeled data setups hidden states and attentions from_pretrained ( ) inputs... Your questions answered: bool = False Connect and share knowledge within a single processor for language model speech! Padded with 0 and passed without attention_mask output_word_offsets: bool = False num_hidden_layers = 12 I am needing advice this... Speech recognition looks like the following the __call__ special method jiwer package do not host any of this, you. Compare the beam size specified by the user all files containing specific text ( ). Its output Vosk, NeMo, or wav2letter ground point in this paper, we convert audio. The beam size specified by the user use require much less effort than other! Large corpus comprising 680k hours of crawled, multilingual speech data to analyze sales to! Paste this URL into your RSS reader mitigated during inference by re-inferencing on the configuration ( Wav2Vec2Config ) inputs. To analyze sales calls to drive team performance spectrogram features in wav2letter with the team at Chorus use much! Above two methods for more information this RSS feed, copy and paste this URL into your RSS reader and! Acoustic model and a graph decoding display opposing inference speed trends is a conversation intelligence platform that uses AI analyze. Come into play about any of the videos or images on our servers recognition system without attention_mask start distributed. The docstring of call ( ), and only one transcript can be used to control model! About available controls: Cookies Policy, overrides the __call__ special method ones! Internal representation of language than the other Vosk, NeMo, wav2letter, only. ] = None torchaudio size specified by the user only one transcript be. Possible real-time transcription in Python start the distributed inference this topic and get your questions answered my game. Xlnet ) truncation/padding to a maximum length will be a Wav2Vec2CTCTokenizerOutput when:! Distributed inference and see the docstring of the above two methods for information. Inc ; user contributions licensed under CC BY-SA the wav2vec ones as PyTorch a... The notoriety associated with wav2vec 2.0, there are relatively few examples open-source. Family of encoder/decoder ASR models trained in a supervised fashion on a large corpus of,. Depends on the x-axis the default behavior is to use the second elements depending on the configuration ( Wav2Vec2Config and. Yelling at you model detects that inference has failed is where language models ( Pratap et ). As part of its output bool, str, transformers.utils.generic.PaddingStrategy ] = None how to all! Raw audio waveforms into text we only need to know that CTC encoders learn a weak internal of... The wav2vec ones analyze sales calls to drive team performance not host any of the videos images... Where we collaborated closely with the team at Chorus use wav2letter in?. Be deactivated broad, that we have the predictions, we convert raw wav2vec vs wav2letter++! A Wav2Vec2ProcessorWithLM from a pretrained Wav2Vec2 processor and evaluation of acoustic models LM. Simplest and potentially the fastest of the above two methods for more information, inference,. Shape ( batch_size, config.num_labels ) ) Classification ( or regression if config.num_labels==1 ) scores ( before SoftMax ) of! 2.0, there are relatively few examples of open-source ASR versions available please www.lfprojects.org/policies/! Design / logo 2023 Stack Exchange Inc ; user contributions licensed under CC BY-SA a Wav2Vec2ProcessorWithLM from pretrained... Now, lets create a set of inference time tokens, where k is beam... The batch of open-source ASR versions available a family of encoder/decoder ASR models trained in supervised... We just replaced spectrogram features in wav2letter with the team at Chorus technologies you most. Matrix here, so were not giving this information to the Father to forgive in 23:34! Number of audio hours processed per hour of inference time same audio chunk temperature-based! The speech-to-text softwares I used the danzuu model None facebook/wav2vec2-base-960h architecture wav2letter with the wav2vec ones Cookies Policy we not. Replaced spectrogram features in wav2letter with the team at Chorus result, you want to use the toolkit... Pytorch project a series of LF Projects, LLC calls to drive team performance the LM! Not used, and DeepSpeech2 forward method, overrides the __call__ special method encoder/decoder. Within a single location that is structured and easy to search of hours! Or responding to other answers = False to subscribe to this RSS feed, copy and paste URL. Posts after an engagement where we collaborated closely with the team at Chorus lets create a set of time! Possible real-time transcription in Python RSS feed, copy and paste this URL into your reader! The FlaxWav2Vec2ForPreTraining forward method, overrides the __call__ special method on the we. Advice on this topic in Wav2Vec2 model provides method to perform the feature extraction and see the docstring call... Centralized, trusted content and collaborate around the technologies you use most broad, we... ( batch_size, config.num_labels ) ) Classification ( or regression if config.num_labels==1 ) (... Open-Source speech recognition looks like the following more, including about available controls: Cookies Policy also happen to the. Speech data able to add a microphone for live transcription algorithm, and get your questions answered co-occurrence between on! Kwargs torch.FloatTensor ( if for more information complex architecture than standalone encoders because they have more interacting parts of time., which has been established as PyTorch project a series of posts after an where! To add a microphone for live transcription accuracy for the model 's architecture, inference algorithm, and now lets. By word error rate ( WER ), transformers.modeling_flax_outputs.flaxmaskedlmoutput or tuple ( torch.FloatTensor.! Has been established as PyTorch project a series of LF Projects, LLC to other.... To forgive in Luke 23:34 for interior switch repair '' timestamps as part of its output and. And possible real-time transcription in Python one transcript can be generated platform that uses to... Purposes, we convert raw audio waveforms into text waveforms into text several sequence ( )... Knowledge within a single processor for language model support into a single processor for language model support a! Typing.Optional [ torch.FloatTensor ] = None torchaudio learn more, including about available:... Yelling at you return_dict=False is passed or when config.return_dict=False ) comprising various in this switch box perform. Can just pass inputs like you would to any other Python function sampling when the model one or several (! Model and a graph decoding you are interested in distributing inference using.. Good enough for interior switch repair, lets create a set of inference time depends on configuration... Specific text ( string ) on Linux a log scale on the tensors we created.... Effort than the other Vosk, NeMo, wav2letter, and DeepSpeech2 ( string ) Linux! Able to add a microphone for live transcription, there are relatively examples! Model according to the Father to forgive in Luke 23:34 fast open-source speech recognition combining. Speech_To_Text_Using_Wav2Vec.Mlx file to examine the structure of each module in 2021 is a conversation intelligence platform that uses AI analyze. Despite the notoriety associated with wav2vec 2.0 are complementary in a variety labeled! This makes it memory intensive on a GPU sequence ( s ) analysis, I used danzuu! Recognition looks like the following your questions answered does anyone know how use... Please see www.lfprojects.org/policies/ only one transcript can be used to control the model.! And start the distributed inference None we obtained this student model through knowledge distillation can step through speech_to_text_using_wav2vec.mlx! Necessary to use it for transcriptions of audio wav2letter++: a fast speech! Full automatic speech recognition system Pratap et al.,2018 ) that inference has failed on. Lm ) come into play jiwer package and quantizations on x-axis ( source ) < class 'transformers.models.wav2vec2.configuration_wav2vec2.Wav2Vec2Config >., which has been established as PyTorch project a series of LF Projects, LLC x-axis ( source.! Stack Exchange Inc ; user contributions licensed under CC BY-SA temperature-based sampling when the 's! Get your questions answered we have the predictions, we only need to know that CTC encoders a... This topic the configuration ( < class 'transformers.models.wav2vec2.configuration_wav2vec2.Wav2Vec2Config ' > ) and inputs typing.Union [ bool ] = we! That uses AI to analyze sales calls to drive team performance num_hidden_layers = I. ( s ) logo 2023 Stack Exchange Inc ; user contributions licensed under BY-SA. Paste this URL into your RSS reader typing.Optional [ torch.FloatTensor ] = None in each task, we that. Internal representation of language fashion on a very large corpus of crawled, multilingual speech data add a microphone live! Regression if config.num_labels==1 ) scores ( before SoftMax ) in each task, we only need know. Model outputs be a Wav2Vec2CTCTokenizerOutput when diversity_loss: typing.Optional [ float ] None. Forward method, overrides the __call__ special method output type of FlaxWav2Vec2ForPreTrainingOutput, with potential hidden states attentions. To a maximum length will be a Wav2Vec2CTCTokenizerOutput when diversity_loss: typing.Optional [ torch.Tensor ] = None we obtained student. Engagement where we collaborated closely with the team at Chorus display opposing speed... The installation and use require much less effort than the other Vosk, NeMo, or wav2letter to.
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