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Showing content from https://cloud.google.com/speech-to-text/docs/reference/rest/v1/RecognitionConfig below:

RecognitionConfig | Cloud Speech-to-Text Documentation

• JSON representation
• AudioEncoding
• SpeechAdaptation
  • JSON representation
• ABNFGrammar
  • JSON representation
• SpeechContext
  • JSON representation
• SpeakerDiarizationConfig
  • JSON representation
• RecognitionMetadata
  • JSON representation
• InteractionType
• MicrophoneDistance
• OriginalMediaType
• RecordingDeviceType

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

{
  "encoding": enum (AudioEncoding),
  "sampleRateHertz": integer,
  "audioChannelCount": integer,
  "enableSeparateRecognitionPerChannel": boolean,
  "languageCode": string,
  "alternativeLanguageCodes": [
    string
  ],
  "maxAlternatives": integer,
  "profanityFilter": boolean,
  "adaptation": {
    object (SpeechAdaptation)
  },
  "speechContexts": [
    {
      object (SpeechContext)
    }
  ],
  "enableWordTimeOffsets": boolean,
  "enableWordConfidence": boolean,
  "enableAutomaticPunctuation": boolean,
  "enableSpokenPunctuation": boolean,
  "enableSpokenEmojis": boolean,
  "diarizationConfig": {
    object (SpeakerDiarizationConfig)
  },
  "metadata": {
    object (RecognitionMetadata)
  },
  "model": string,
  "useEnhanced": boolean
}

enum (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.

integer

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.

integer

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 enableSeparateRecognitionPerChannel to 'true'.

boolean

This needs to be set to true explicitly and audioChannelCount > 1 to get each channel recognized separately. The recognition result will contain a channelTag 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: audioChannelCount multiplied by the length of the audio.

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.

string

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 languageCode. 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).

integer

Maximum number of recognition hypotheses to be returned. Specifically, the maximum number of SpeechRecognitionAlternative messages within each SpeechRecognitionResult. The server may return fewer than maxAlternatives. 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.

boolean

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.

object (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 speechContexts field.

object (SpeechContext)

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

boolean

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.

boolean

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.

boolean

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.

boolean

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.

boolean

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.

object (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.

object (RecognitionMetadata)

Metadata regarding this request.

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.

latest_long

latest_short

command_and_search

phone_call

video

default

medical_conversation

medical_dictation

boolean

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

The encoding of the audio data sent in the request.

All encodings support only 1 channel (mono) audio, unless the audioChannelCount and enableSeparateRecognitionPerChannel 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 error code.

Speech adaptation configuration.

{
  "phraseSets": [
    {
      object (PhraseSet)
    }
  ],
  "phraseSetReferences": [
    string
  ],
  "customClasses": [
    {
      object (CustomClass)
    }
  ],
  "abnfGrammar": {
    object (ABNFGrammar)
  }
}

object (PhraseSet)

A collection of phrase sets. To specify the hints inline, leave the phrase set's name blank and fill in the rest of its fields. Any phrase set can use any custom class.

string

A collection of phrase set resource names to use.

object (CustomClass)

A collection of custom classes. To specify the classes inline, leave the class' name blank and fill in the rest of its fields, giving it a unique customClassId. Refer to the inline defined class in phrase hints by its customClassId.

object (ABNFGrammar)

Augmented Backus-Naur form (ABNF) is a standardized grammar notation comprised by a set of derivation rules. See specifications: https://www.w3.org/TR/speech-grammar

{
  "abnfStrings": [
    string
  ]
}

string

All declarations and rules of an ABNF grammar broken up into multiple strings that will end up concatenated.

Provides "hints" to the speech recognizer to favor specific words and phrases in the results.

{
  "phrases": [
    string
  ],
  "boost": number
}

string

A list of strings containing words and phrases "hints" so that the speech recognition is more likely to recognize them. This can be used to improve the accuracy for specific words and phrases, for example, if specific commands are typically spoken by the user. This can also be used to add additional words to the vocabulary of the recognizer. See usage limits.

List items can also be set to classes for groups of words that represent common concepts that occur in natural language. For example, rather than providing phrase hints for every month of the year, using the $MONTH class improves the likelihood of correctly transcribing audio that includes months.

number

Hint Boost. Positive value will increase the probability that a specific phrase will be recognized over other similar sounding phrases. The higher the boost, the higher the chance of false positive recognition as well. Negative boost values would correspond to anti-biasing. Anti-biasing is not enabled, so negative boost will simply be ignored. Though boost can accept a wide range of positive values, most use cases are best served with values between 0 and 20. We recommend using a binary search approach to finding the optimal value for your use case.

Config to enable speaker diarization.

{
  "enableSpeakerDiarization": boolean,
  "minSpeakerCount": integer,
  "maxSpeakerCount": integer,
  "speakerTag": integer
}

boolean

If 'true', enables speaker detection for each recognized word in the top alternative of the recognition result using a speakerTag provided in the WordInfo.

integer

Minimum number of speakers in the conversation. This range gives you more flexibility by allowing the system to automatically determine the correct number of speakers. If not set, the default value is 2.

integer

Maximum number of speakers in the conversation. This range gives you more flexibility by allowing the system to automatically determine the correct number of speakers. If not set, the default value is 6.

integer

This item is deprecated!

Output only. Unused.

This item is deprecated!

Description of audio data to be recognized.

Use case categories that the audio recognition request can be described by.

Enumerates the types of capture settings describing an audio file.

The original media the speech was recorded on.

The type of device the speech was recorded with.

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