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    RecognitionConfig

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

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

    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 .

    sampleRateHertz

    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 .

    audioChannelCount

    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'.

    enableSeparateRecognitionPerChannel

    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.

    languageCode

    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.

    alternativeLanguageCodes[]

    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).

    maxAlternatives

    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.

    profanityFilter

    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.

    adaptation

    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.

    transcriptNormalization

    object ( 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.

    speechContexts[]

    object ( SpeechContext )

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

    enableWordTimeOffsets

    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 .

    enableWordConfidence

    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 .

    enableAutomaticPunctuation

    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.

    enableSpokenPunctuation

    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.

    enableSpokenEmojis

    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.

    enableSpeakerDiarization
    (deprecated)

    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. Note: Use diarizationConfig instead.

    diarizationSpeakerCount
    (deprecated)

    integer

    If set, specifies the estimated number of speakers in the conversation. Defaults to '2'. Ignored unless enableSpeakerDiarization is set to true. Note: Use diarizationConfig instead.

    diarizationConfig

    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.

    metadata

    object ( 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.

    useEnhanced

    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.

    AudioEncoding

    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.

    Enums
    ENCODING_UNSPECIFIED Not specified.
    LINEAR16 Uncompressed 16-bit signed little-endian samples (Linear PCM).
    FLAC FLAC (Free Lossless Audio Codec) is the recommended encoding because it is lossless--therefore recognition is not compromised--and requires only about half the bandwidth of LINEAR16 . FLAC stream encoding supports 16-bit and 24-bit samples, however, not all fields in STREAMINFO are supported.
    MULAW 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    AMR Adaptive Multi-Rate Narrowband codec. sampleRateHertz must be 8000.
    AMR_WB Adaptive Multi-Rate Wideband codec. sampleRateHertz must be 16000.
    OGG_OPUS Opus encoded audio frames in Ogg container ( OggOpus ). sampleRateHertz must be one of 8000, 12000, 16000, 24000, or 48000.
    SPEEX_WITH_HEADER_BYTE Although the use of lossy encodings is not recommended, if a very low bitrate encoding is required, OGG_OPUS is highly preferred over Speex encoding. The Speex encoding supported by Cloud Speech API has a header byte in each block, as in MIME type audio/x-speex-with-header-byte . It is a variant of the RTP Speex encoding defined in RFC 5574 . The stream is a sequence of blocks, one block per RTP packet. Each block starts with a byte containing the length of the block, in bytes, followed by one or more frames of Speex data, padded to an integral number of bytes (octets) as specified in RFC 5574. In other words, each RTP header is replaced with a single byte containing the block length. Only Speex wideband is supported. sampleRateHertz must be 16000.
    MP3 MP3 audio. MP3 encoding is a Beta feature and only available in v1p1beta1. Support all standard MP3 bitrates (which range from 32-320 kbps). When using this encoding, sampleRateHertz has to match the sample rate of the file being used.
    WEBM_OPUS Opus encoded audio frames in WebM container ( OggOpus ). sampleRateHertz must be one of 8000, 12000, 16000, 24000, or 48000.

    SpeechAdaptation

    Speech adaptation configuration.

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

    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.
    phraseSetReferences[]

    string

    A collection of phrase set resource names to use.
    customClasses[]

    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 .
    abnfGrammar

    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

    ABNFGrammar

    JSON representation 
     { 
 "abnfStrings" 
 : 
 [ 
 string 
 ] 
 }
    Fields
    abnfStrings[]

    string

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

    TranscriptNormalization

    Transcription normalization configuration. 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.

    JSON representation 
     { 
 "entries" 
 : 
 [ 
 { 
 object (  Entry 
 
) 
 } 
 ] 
 }
    Fields
    entries[]

    object ( Entry )

    A list of replacement entries. We will perform replacement with one entry at a time. For example, the second entry in ["cat" => "dog", "mountain cat" => "mountain dog"] will never be applied because we will always process the first entry before it. At most 100 entries.

    Entry

    A single replacement configuration.

    JSON representation 
     { 
 "search" 
 : 
 string 
 , 
 "replace" 
 : 
 string 
 , 
 "caseSensitive" 
 : 
 boolean 
 }
    Fields
    search

    string

    What to replace. Max length is 100 characters.
    replace

    string

    What to replace with. Max length is 100 characters.
    caseSensitive

    boolean

    Whether the search is case sensitive.

    SpeechContext

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

    JSON representation 
     { 
 "phrases" 
 : 
 [ 
 string 
 ] 
 , 
 "boost" 
 : 
 number 
 }
    Fields
    phrases[]

    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.
    boost

    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.

    SpeakerDiarizationConfig

    Config to enable speaker diarization.

    JSON representation 
     { 
 "enableSpeakerDiarization" 
 : 
 boolean 
 , 
 "minSpeakerCount" 
 : 
 integer 
 , 
 "maxSpeakerCount" 
 : 
 integer 
 , 
 "speakerTag" 
 : 
 integer 
 }
    Fields
    enableSpeakerDiarization

    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.
    minSpeakerCount

    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.
    maxSpeakerCount

    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.
    speakerTag
    (deprecated)

    integer

    Output only. Unused.

    RecognitionMetadata

    Description of audio data to be recognized.

    JSON representation 
     { 
 "interactionType" 
 : 
 enum (  InteractionType 
 
) 
 , 
 "industryNaicsCodeOfAudio" 
 : 
 integer 
 , 
 "microphoneDistance" 
 : 
 enum (  MicrophoneDistance 
 
) 
 , 
 "originalMediaType" 
 : 
 enum (  OriginalMediaType 
 
) 
 , 
 "recordingDeviceType" 
 : 
 enum (  RecordingDeviceType 
 
) 
 , 
 "recordingDeviceName" 
 : 
 string 
 , 
 "originalMimeType" 
 : 
 string 
 , 
 "obfuscatedId" 
 : 
 string 
 , 
 "audioTopic" 
 : 
 string 
 }
    Fields
    interactionType

    enum ( InteractionType )

    The use case most closely describing the audio content to be recognized.
    industryNaicsCodeOfAudio

    integer ( uint32 format)

    The industry vertical to which this speech recognition request most closely applies. This is most indicative of the topics contained in the audio. Use the 6-digit NAICS code to identify the industry vertical - see https://www.naics.com/search/ .
    microphoneDistance

    enum ( MicrophoneDistance )

    The audio type that most closely describes the audio being recognized.
    originalMediaType

    enum ( OriginalMediaType )

    The original media the speech was recorded on.
    recordingDeviceType

    enum ( RecordingDeviceType )

    The type of device the speech was recorded with.
    recordingDeviceName

    string

    The device used to make the recording. Examples 'Nexus 5X' or 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or 'Cardioid Microphone'.
    originalMimeType

    string

    Mime type of the original audio file. For example audio/m4a , audio/x-alaw-basic , audio/mp3 , audio/3gpp . A list of possible audio mime types is maintained at http://www.iana.org/assignments/media-types/media-types.xhtml#audio

    obfuscatedId
    (deprecated)

    string ( int64 format)

    Obfuscated (privacy-protected) ID of the user, to identify number of unique users using the service.
    audioTopic

    string

    Description of the content. Eg. "Recordings of federal supreme court hearings from 2012".

    InteractionType

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

    Enums
    INTERACTION_TYPE_UNSPECIFIED Use case is either unknown or is something other than one of the other values below.
    DISCUSSION Multiple people in a conversation or discussion. For example in a meeting with two or more people actively participating. Typically all the primary people speaking would be in the same room (if not, see PHONE_CALL)
    PRESENTATION One or more persons lecturing or presenting to others, mostly uninterrupted.
    PHONE_CALL A phone-call or video-conference in which two or more people, who are not in the same room, are actively participating.
    VOICEMAIL A recorded message intended for another person to listen to.
    PROFESSIONALLY_PRODUCED Professionally produced audio (eg. TV Show, Podcast).
    VOICE_COMMAND Transcribe voice commands, such as for controlling a device.
    DICTATION Transcribe speech to text to create a written document, such as a text-message, email or report.

    MicrophoneDistance

    Enumerates the types of capture settings describing an audio file.

    Enums
    MICROPHONE_DISTANCE_UNSPECIFIED Audio type is not known.
    NEARFIELD The audio was captured from a closely placed microphone. Eg. phone, dictaphone, or handheld microphone. Generally if there speaker is within 1 meter of the microphone.
    MIDFIELD The speaker if within 3 meters of the microphone.
    FARFIELD The speaker is more than 3 meters away from the microphone.

    OriginalMediaType

    The original media the speech was recorded on.

    Enums
    ORIGINAL_MEDIA_TYPE_UNSPECIFIED Unknown original media type.
    AUDIO The speech data is an audio recording.
    VIDEO The speech data originally recorded on a video.

    RecordingDeviceType

    The type of device the speech was recorded with.

    Enums
    RECORDING_DEVICE_TYPE_UNSPECIFIED The recording device is unknown.
    SMARTPHONE Speech was recorded on a smartphone.
    PC Speech was recorded using a personal computer or tablet.
    PHONE_LINE Speech was recorded over a phone line.
    VEHICLE Speech was recorded in a vehicle.
    OTHER_OUTDOOR_DEVICE Speech was recorded outdoors.
    OTHER_INDOOR_DEVICE Speech was recorded indoors.
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