sparknlp.annotator.audio.whisper_for_ctc
#
Contains classes concerning WhisperForCTC.
Module Contents#
Classes#
Whisper Model with a language modeling head on top for Connectionist Temporal Classification |
- class WhisperForCTC(classname='com.johnsnowlabs.nlp.annotators.audio.WhisperForCTC', java_model=None)[source]#
Whisper Model with a language modeling head on top for Connectionist Temporal Classification (CTC).
Whisper is an automatic speech recognition (ASR) system trained on 680,000 hours of multilingual and multitask supervised data collected from the web. It transcribe in multiple languages, as well as translate from those languages into English.
The audio needs to be provided pre-processed an array of floats.
Note that at the moment, this annotator only supports greedy search and only Spark Versions 3.4 and up are supported.
For multilingual models, the language and the task (transcribe or translate) can be set with
setLanguage
andsetTask
.Pretrained models can be loaded with
pretrained
of the companion object:speechToText = WhisperForCTC.pretrained() \ .setInputCols(["audio_assembler"]) \ .setOutputCol("text")
The default model is
"asr_whisper_tiny_opt"
, if no name is provided.For available pretrained models please see the Models Hub.
To see which models are compatible and how to import them see JohnSnowLabs/spark-nlp#5669 and to see more extended examples, see WhisperForCTCTestSpec.
References:
Robust Speech Recognition via Large-Scale Weak Supervision
Paper Abstract:
We study the capabilities of speech processing systems trained simply to predict large amounts of transcripts of audio on the internet. When scaled to 680,000 hours of multilingual and multitask supervision, the resulting models generalize well to standard benchmarks and are often competitive with prior fully supervised results but in a zero- shot transfer setting without the need for any fine- tuning. When compared to humans, the models approach their accuracy and robustness. We are releasing models and inference code to serve as a foundation for further work on robust speech processing.
Input Annotation types
Output Annotation type
AUDIO
DOCUMENT
- Parameters:
- task
The formatted task for the audio. Either <|translate|> or <|transcribe|>.
- language
The language for the audio, formatted to e.g. <|en|>. Check the model description for supported languages.
- isMultilingual
Whether the model is multilingual
- minOutputLength
Minimum length of the sequence to be generated
- maxOutputLength
Maximum length of output text
- doSample
Whether or not to use sampling; use greedy decoding otherwise
- temperature
The value used to module the next token probabilities
- topK
The number of highest probability vocabulary tokens to keep for top-k-filtering
- topP
If set to float < 1, only the most probable tokens with probabilities that add up to
top_p
or higher are kept for generation- repetitionPenalty
The parameter for repetition penalty. 1.0 means no penalty. See this paper for more details
- noRepeatNgramSize
If set to int > 0, all ngrams of that size can only occur once
- beamSize
The Number of beams for beam search
Examples
>>> import sparknlp >>> from sparknlp.base import * >>> from sparknlp.annotator import * >>> from pyspark.ml import Pipeline >>> audioAssembler = AudioAssembler() \ ... .setInputCol("audio_content") \ ... .setOutputCol("audio_assembler") >>> speechToText = WhisperForCTC.pretrained() \ ... .setInputCols(["audio_assembler"]) \ ... .setOutputCol("text") >>> pipeline = Pipeline().setStages([audioAssembler, speechToText]) >>> processedAudioFloats = spark.createDataFrame([[rawFloats]]).toDF("audio_content") >>> result = pipeline.fit(processedAudioFloats).transform(processedAudioFloats) >>> result.select("text.result").show(truncate = False) +------------------------------------------------------------------------------------------+ |result | +------------------------------------------------------------------------------------------+ |[ Mr. Quilter is the apostle of the middle classes and we are glad to welcome his gospel.]| +------------------------------------------------------------------------------------------+
- setConfigProtoBytes(b)[source]#
Sets configProto from tensorflow, serialized into byte array.
- Parameters:
- bList[int]
ConfigProto from tensorflow, serialized into byte array
- setLanguage(value)[source]#
Sets the language for the audio, formatted to e.g. <|en|>. Check the model description for supported languages.
- Parameters:
- valueString
Formatted language code
- setTask(value)[source]#
Sets the formatted task for the audio. Either <|translate|> or <|transcribe|>.
Only multilingual models can do translation.
- Parameters:
- valueString
Formatted task
- static loadSavedModel(folder, spark_session)[source]#
Loads a locally saved model.
- Parameters:
- folderstr
Folder of the saved model
- spark_sessionpyspark.sql.SparkSession
The current SparkSession
- Returns:
- WhisperForCTC
The restored model
- static pretrained(name='asr_whisper_tiny_opt', lang='xx', remote_loc=None)[source]#
Downloads and loads a pretrained model.
- Parameters:
- namestr, optional
Name of the pretrained model, by default “asr_hubert_large_ls960”
- langstr, optional
Language of the pretrained model, by default “en”
- remote_locstr, optional
Optional remote address of the resource, by default None. Will use Spark NLPs repositories otherwise.
- Returns:
- WhisperForCTC
The restored model