Class RecognitionConfig (2.16.2)

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RecognitionConfig(mapping=None, *, ignore_unknown_fields=False, **kwargs)

Provides information to the recognizer that specifies how to process the request.

Attributes

NameDescription
encoding google.cloud.speech_v1p1beta1.types.RecognitionConfig.AudioEncoding
Encoding of audio data sent in all RecognitionAudio messages. This field is optional for FLAC and WAV audio files and required for all other audio formats. For details, see AudioEncoding.
sample_rate_hertz int
Sample rate in Hertz of the audio data sent in all RecognitionAudio messages. Valid values are: 8000-48000. 16000 is optimal. For best results, set the sampling rate of the audio source to 16000 Hz. If that's not possible, use the native sample rate of the audio source (instead of re-sampling). This field is optional for FLAC and WAV audio files, but is required for all other audio formats. For details, see AudioEncoding.
audio_channel_count int
The number of channels in the input audio data. ONLY set this for MULTI-CHANNEL recognition. Valid values for LINEAR16 and FLAC are 1-8. Valid values for OGG_OPUS are '1'-'254'. Valid value for MULAW, AMR, AMR_WB and SPEEX_WITH_HEADER_BYTE is only 1. If 0 or omitted, defaults to one channel (mono). Note: We only recognize the first channel by default. To perform independent recognition on each channel set enable_separate_recognition_per_channel to 'true'.
enable_separate_recognition_per_channel bool
This needs to be set to true explicitly and audio_channel_count > 1 to get each channel recognized separately. The recognition result will contain a channel_tag field to state which channel that result belongs to. If this is not true, we will only recognize the first channel. The request is billed cumulatively for all channels recognized: audio_channel_count multiplied by the length of the audio.
language_code str
Required. The language of the supplied audio as a BCP-47 __ language tag. Example: "en-US". See `Language Support
alternative_language_codes Sequence[str]
A list of up to 3 additional BCP-47 __ language tags, listing possible alternative languages of the supplied audio. See `Language Support
max_alternatives int
Maximum number of recognition hypotheses to be returned. Specifically, the maximum number of SpeechRecognitionAlternative messages within each SpeechRecognitionResult. The server may return fewer than max_alternatives. Valid values are 0-30. A value of 0 or 1 will return a maximum of one. If omitted, will return a maximum of one.
profanity_filter bool
If set to true, the server will attempt to filter out profanities, replacing all but the initial character in each filtered word with asterisks, e.g. "f***". If set to false or omitted, profanities won't be filtered out.
adaptation google.cloud.speech_v1p1beta1.types.SpeechAdaptation
Speech adaptation configuration improves the accuracy of speech recognition. For more information, see the `speech adaptation
transcript_normalization google.cloud.speech_v1p1beta1.types.TranscriptNormalization
Use transcription normalization to automatically replace parts of the transcript with phrases of your choosing. For StreamingRecognize, this normalization only applies to stable partial transcripts (stability > 0.8) and final transcripts.
speech_contexts Sequence[google.cloud.speech_v1p1beta1.types.SpeechContext]
Array of SpeechContext. A means to provide context to assist the speech recognition. For more information, see `speech adaptation
enable_word_time_offsets bool
If true, the top result includes a list of words and the start and end time offsets (timestamps) for those words. If false, no word-level time offset information is returned. The default is false.
enable_word_confidence bool
If true, the top result includes a list of words and the confidence for those words. If false, no word-level confidence information is returned. The default is false.
enable_automatic_punctuation bool
If 'true', adds punctuation to recognition result hypotheses. This feature is only available in select languages. Setting this for requests in other languages has no effect at all. The default 'false' value does not add punctuation to result hypotheses.
enable_spoken_punctuation google.protobuf.wrappers_pb2.BoolValue
The spoken punctuation behavior for the call If not set, uses default behavior based on model of choice e.g. command_and_search will enable spoken punctuation by default If 'true', replaces spoken punctuation with the corresponding symbols in the request. For example, "how are you question mark" becomes "how are you?". See https://cloud.google.com/speech-to-text/docs/spoken-punctuation for support. If 'false', spoken punctuation is not replaced.
enable_spoken_emojis google.protobuf.wrappers_pb2.BoolValue
The spoken emoji behavior for the call If not set, uses default behavior based on model of choice If 'true', adds spoken emoji formatting for the request. This will replace spoken emojis with the corresponding Unicode symbols in the final transcript. If 'false', spoken emojis are not replaced.
enable_speaker_diarization bool
If 'true', enables speaker detection for each recognized word in the top alternative of the recognition result using a speaker_tag provided in the WordInfo. Note: Use diarization_config instead.
diarization_speaker_count int
If set, specifies the estimated number of speakers in the conversation. Defaults to '2'. Ignored unless enable_speaker_diarization is set to true. Note: Use diarization_config instead.
diarization_config google.cloud.speech_v1p1beta1.types.SpeakerDiarizationConfig
Config to enable speaker diarization and set additional parameters to make diarization better suited for your application. Note: When this is enabled, we send all the words from the beginning of the audio for the top alternative in every consecutive STREAMING responses. This is done in order to improve our speaker tags as our models learn to identify the speakers in the conversation over time. For non-streaming requests, the diarization results will be provided only in the top alternative of the FINAL SpeechRecognitionResult.
metadata google.cloud.speech_v1p1beta1.types.RecognitionMetadata
Metadata regarding this request.
model str
Which model to select for the given request. Select the model best suited to your domain to get best results. If a model is not explicitly specified, then we auto-select a model based on the parameters in the RecognitionConfig. .. raw:: html
ModelDescription
latest_longBest for long form content like media or conversation.
latest_shortBest for short form content like commands or single shot directed speech.
command_and_searchBest for short queries such as voice commands or voice search.
phone_callBest for audio that originated from a phone call (typically recorded at an 8khz sampling rate).
videoBest for audio that originated from video or includes multiple speakers. Ideally the audio is recorded at a 16khz or greater sampling rate. This is a premium model that costs more than the standard rate.
defaultBest for audio that is not one of the specific audio models. For example, long-form audio. Ideally the audio is high-fidelity, recorded at a 16khz or greater sampling rate.
medical_conversationBest for audio that originated from a conversation between a medical provider and patient.
medical_dictationBest for audio that originated from dictation notes by a medical provider.
use_enhanced bool
Set to true to use an enhanced model for speech recognition. If use_enhanced is set to true and the model field is not set, then an appropriate enhanced model is chosen if an enhanced model exists for the audio. If use_enhanced is true and an enhanced version of the specified model does not exist, then the speech is recognized using the standard version of the specified model.

Inheritance

builtins.object > proto.message.Message > RecognitionConfig

Classes

AudioEncoding

AudioEncoding(value)

The encoding of the audio data sent in the request.

All encodings support only 1 channel (mono) audio, unless the audio_channel_count and enable_separate_recognition_per_channel fields are set.

For best results, the audio source should be captured and transmitted using a lossless encoding (FLAC or LINEAR16). The accuracy of the speech recognition can be reduced if lossy codecs are used to capture or transmit audio, particularly if background noise is present. Lossy codecs include MULAW, AMR, AMR_WB, OGG_OPUS, SPEEX_WITH_HEADER_BYTE, MP3, and WEBM_OPUS.

The FLAC and WAV audio file formats include a header that describes the included audio content. You can request recognition for WAV files that contain either LINEAR16 or MULAW encoded audio. If you send FLAC or WAV audio file format in your request, you do not need to specify an AudioEncoding; the audio encoding format is determined from the file header. If you specify an AudioEncoding when you send send FLAC or WAV audio, the encoding configuration must match the encoding described in the audio header; otherwise the request returns an google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error code.