Cloud Speech V1 Client - Class RecognitionConfig (1.19.2)

Reference documentation and code samples for the Cloud Speech V1 Client class RecognitionConfig.

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

Generated from protobuf message google.cloud.speech.v1.RecognitionConfig

Namespace

Google \ Cloud \ Speech \ V1

Methods

__construct

Constructor.

Parameters

Name Description

data

array

Optional. Data for populating the Message object.

↳ encoding

int

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, OGG_OPUS and FLAC are 1-8. 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

string

Required. The language of the supplied audio as a BCP-47 language tag. Example: "en-US". See Language Support for a list of the currently supported language codes.

↳ alternative_language_codes

array

A list of up to 3 additional BCP-47 language tags, listing possible alternative languages of the supplied audio. See Language Support for a list of the currently supported language codes. If alternative languages are listed, recognition result will contain recognition in the most likely language detected including the main language_code. The recognition result will include the language tag of the language detected in the audio. Note: This feature is only supported for Voice Command and Voice Search use cases and performance may vary for other use cases (e.g., phone call transcription).

↳ 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

SpeechAdaptation

Speech adaptation configuration improves the accuracy of speech recognition. For more information, see the speech adaptation documentation. When speech adaptation is set it supersedes the speech_contexts field.

↳ transcript_normalization

TranscriptNormalization

Optional. 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

array<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\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\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.

↳ diarization_config

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

RecognitionMetadata

Metadata regarding this request.

↳ model

string

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.

Model	Description
`latest_long`	Best for long form content like media or conversation.
`latest_short`	Best for short form content like commands or single shot directed speech.
`command_and_search`	Best for short queries such as voice commands or voice search.
`phone_call`	Best for audio that originated from a phone call (typically recorded at an 8khz sampling rate).
`video`	Best 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.
`default`	Best 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_conversation`	Best for audio that originated from a conversation between a medical provider and patient.
`medical_dictation`	Best 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.

getEncoding

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.

Returns
Type	Description
`int`

setEncoding

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.

Parameter
Name	Description
`var`	`int`

Returns
Type	Description
`$this`

getSampleRateHertz

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.

Returns
Type	Description
`int`

setSampleRateHertz

Sample rate in Hertz of the audio data sent in all RecognitionAudio messages. Valid values are: 8000-48000.

Parameter
Name	Description
`var`	`int`

Returns
Type	Description
`$this`

getAudioChannelCount

The number of channels in the input audio data.

ONLY set this for MULTI-CHANNEL recognition. Valid values for LINEAR16, OGG_OPUS and FLAC are 1-8. 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'.

Returns
Type	Description
`int`

setAudioChannelCount

The number of channels in the input audio data.

Parameter
Name	Description
`var`	`int`

Returns
Type	Description
`$this`

getEnableSeparateRecognitionPerChannel

Returns
Type	Description
`bool`

setEnableSeparateRecognitionPerChannel

Parameter
Name	Description
`var`	`bool`

Returns
Type	Description
`$this`

getLanguageCode

Required. The language of the supplied audio as a BCP-47 language tag.

Example: "en-US". See Language Support for a list of the currently supported language codes.

Returns
Type	Description
`string`

setLanguageCode

Required. The language of the supplied audio as a BCP-47 language tag.

Example: "en-US". See Language Support for a list of the currently supported language codes.

Parameter
Name	Description
`var`	`string`

Returns
Type	Description
`$this`

getAlternativeLanguageCodes

A list of up to 3 additional BCP-47 language tags, listing possible alternative languages of the supplied audio.

See Language Support for a list of the currently supported language codes. If alternative languages are listed, recognition result will contain recognition in the most likely language detected including the main language_code. The recognition result will include the language tag of the language detected in the audio. Note: This feature is only supported for Voice Command and Voice Search use cases and performance may vary for other use cases (e.g., phone call transcription).

Returns
Type	Description
`Google\Protobuf\Internal\RepeatedField`

setAlternativeLanguageCodes

A list of up to 3 additional BCP-47 language tags, listing possible alternative languages of the supplied audio.

Parameter
Name	Description
`var`	`string[]`

Returns
Type	Description
`$this`

getMaxAlternatives

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.

Returns
Type	Description
`int`

setMaxAlternatives

Maximum number of recognition hypotheses to be returned.

Parameter
Name	Description
`var`	`int`

Returns
Type	Description
`$this`

getProfanityFilter

Returns
Type	Description
`bool`

setProfanityFilter

Parameter
Name	Description
`var`	`bool`

Returns
Type	Description
`$this`

getAdaptation

Speech adaptation configuration improves the accuracy of speech recognition. For more information, see the speech adaptation documentation.

When speech adaptation is set it supersedes the speech_contexts field.

Returns
Type	Description
`SpeechAdaptation\|null`

hasAdaptation

clearAdaptation

setAdaptation

Speech adaptation configuration improves the accuracy of speech recognition. For more information, see the speech adaptation documentation.

When speech adaptation is set it supersedes the speech_contexts field.

Parameter
Name	Description
`var`	`SpeechAdaptation`

Returns
Type	Description
`$this`

getTranscriptNormalization

Returns
Type	Description
`TranscriptNormalization\|null`

hasTranscriptNormalization

clearTranscriptNormalization

setTranscriptNormalization

Parameter
Name	Description
`var`	`TranscriptNormalization`

Returns
Type	Description
`$this`

getSpeechContexts

Array of SpeechContext.

A means to provide context to assist the speech recognition. For more information, see speech adaptation.

Returns
Type	Description
`Google\Protobuf\Internal\RepeatedField`

setSpeechContexts

Array of SpeechContext.

A means to provide context to assist the speech recognition. For more information, see speech adaptation.

Parameter
Name	Description
`var`	`array<SpeechContext>`

Returns
Type	Description
`$this`

getEnableWordTimeOffsets

Returns
Type	Description
`bool`

setEnableWordTimeOffsets

Parameter
Name	Description
`var`	`bool`

Returns
Type	Description
`$this`

getEnableWordConfidence

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.

Returns
Type	Description
`bool`

setEnableWordConfidence

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.

Parameter
Name	Description
`var`	`bool`

Returns
Type	Description
`$this`

getEnableAutomaticPunctuation

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.

Returns
Type	Description
`bool`

setEnableAutomaticPunctuation

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.

Parameter
Name	Description
`var`	`bool`

Returns
Type	Description
`$this`

getEnableSpokenPunctuation

Returns
Type	Description
`Google\Protobuf\BoolValue\|null`

hasEnableSpokenPunctuation

clearEnableSpokenPunctuation

getEnableSpokenPunctuationValue

Returns the unboxed value from getEnableSpokenPunctuation()

Returns
Type	Description
`bool\|null`

setEnableSpokenPunctuation

Parameter
Name	Description
`var`	`Google\Protobuf\BoolValue`

Returns
Type	Description
`$this`

setEnableSpokenPunctuationValue

Sets the field by wrapping a primitive type in a Google\Protobuf\BoolValue object.

Parameter
Name	Description
`var`	`bool\|null`

Returns
Type	Description
`$this`

getEnableSpokenEmojis

Returns
Type	Description
`Google\Protobuf\BoolValue\|null`

hasEnableSpokenEmojis

clearEnableSpokenEmojis

getEnableSpokenEmojisValue

Returns the unboxed value from getEnableSpokenEmojis()

Returns
Type	Description
`bool\|null`

setEnableSpokenEmojis

Parameter
Name	Description
`var`	`Google\Protobuf\BoolValue`

Returns
Type	Description
`$this`

setEnableSpokenEmojisValue

Sets the field by wrapping a primitive type in a Google\Protobuf\BoolValue object.

Parameter
Name	Description
`var`	`bool\|null`

Returns
Type	Description
`$this`

getDiarizationConfig

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.

Returns
Type	Description
`SpeakerDiarizationConfig\|null`

hasDiarizationConfig

clearDiarizationConfig

setDiarizationConfig

Config to enable speaker diarization and set additional parameters to make diarization better suited for your application.

Parameter
Name	Description
`var`	`SpeakerDiarizationConfig`

Returns
Type	Description
`$this`

getMetadata

Metadata regarding this request.

Returns
Type	Description
`RecognitionMetadata\|null`

hasMetadata

clearMetadata

setMetadata

Metadata regarding this request.

Parameter
Name	Description
`var`	`RecognitionMetadata`

Returns
Type	Description
`$this`

getModel

Model	Description
`latest_long`	Best for long form content like media or conversation.
`latest_short`	Best for short form content like commands or single shot directed speech.
`command_and_search`	Best for short queries such as voice commands or voice search.
`phone_call`	Best for audio that originated from a phone call (typically recorded at an 8khz sampling rate).
`video`	Best 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.
`default`	Best 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_conversation`	Best for audio that originated from a conversation between a medical provider and patient.
`medical_dictation`	Best for audio that originated from dictation notes by a medical provider.

Returns
Type	Description
`string`

setModel

Model	Description
`latest_long`	Best for long form content like media or conversation.
`latest_short`	Best for short form content like commands or single shot directed speech.
`command_and_search`	Best for short queries such as voice commands or voice search.
`phone_call`	Best for audio that originated from a phone call (typically recorded at an 8khz sampling rate).
`video`	Best 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.
`default`	Best 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_conversation`	Best for audio that originated from a conversation between a medical provider and patient.
`medical_dictation`	Best for audio that originated from dictation notes by a medical provider.

Parameter
Name	Description
`var`	`string`

Returns
Type	Description
`$this`

getUseEnhanced

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.

Returns
Type	Description
`bool`

setUseEnhanced

Set to true to use an enhanced model for speech recognition.

Parameter
Name	Description
`var`	`bool`

Returns
Type	Description
`$this`