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    Package google.cloud.speech.v1

    Index

    Adaptation

    Service that implements Google Cloud Speech Adaptation API.

    CreateCustomClass

    rpc CreateCustomClass( CreateCustomClassRequest ) returns ( CustomClass )

    Create a custom class.

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    CreatePhraseSet

    rpc CreatePhraseSet( CreatePhraseSetRequest ) returns ( PhraseSet )

    Create a set of phrase hints. Each item in the set can be a single word or a multi-word phrase. The items in the PhraseSet are favored by the recognition model when you send a call that includes the PhraseSet.

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    DeleteCustomClass

    rpc DeleteCustomClass( DeleteCustomClassRequest ) returns ( Empty )

    Delete a custom class.

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    DeletePhraseSet

    rpc DeletePhraseSet( DeletePhraseSetRequest ) returns ( Empty )

    Delete a phrase set.

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    GetCustomClass

    rpc GetCustomClass( GetCustomClassRequest ) returns ( CustomClass )

    Get a custom class.

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    GetPhraseSet

    rpc GetPhraseSet( GetPhraseSetRequest ) returns ( PhraseSet )

    Get a phrase set.

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    ListCustomClasses

    rpc ListCustomClasses( ListCustomClassesRequest ) returns ( ListCustomClassesResponse )

    List custom classes.

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    ListPhraseSet

    rpc ListPhraseSet( ListPhraseSetRequest ) returns ( ListPhraseSetResponse )

    List phrase sets.

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    UpdateCustomClass

    rpc UpdateCustomClass( UpdateCustomClassRequest ) returns ( CustomClass )

    Update a custom class.

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    UpdatePhraseSet

    rpc UpdatePhraseSet( UpdatePhraseSetRequest ) returns ( PhraseSet )

    Update a phrase set.

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    Speech

    Service that implements Google Cloud Speech API.

    LongRunningRecognize

    rpc LongRunningRecognize( LongRunningRecognizeRequest ) returns ( Operation )

    Performs asynchronous speech recognition: receive results via the google.longrunning.Operations interface. Returns either an Operation.error or an Operation.response which contains a LongRunningRecognizeResponse message. For more information on asynchronous speech recognition, see the how-to .

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    Recognize

    rpc Recognize( RecognizeRequest ) returns ( RecognizeResponse )

    Performs synchronous speech recognition: receive results after all audio has been sent and processed.

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    StreamingRecognize

    rpc StreamingRecognize( StreamingRecognizeRequest ) returns ( StreamingRecognizeResponse )

    Performs bidirectional streaming speech recognition: receive results while sending audio. This method is only available via the gRPC API (not REST).

    Authorization Scopes

    Requires the following OAuth scope:

    • https://www.googleapis.com/auth/cloud-platform

    For more information, see the Authentication Overview .

    CreateCustomClassRequest

    Message sent by the client for the CreateCustomClass method.

    Fields
    parent

    string

    Required. The parent resource where this custom class will be created. Format:

    projects/{project}/locations/{location}/customClasses

    Speech-to-Text supports three locations: global , us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

    Authorization requires the following IAM permission on the specified resource parent :

    • speech.customClasses.create
    custom_class_id

    string

    Required. The ID to use for the custom class, which will become the final component of the custom class' resource name.

    This value should restrict to letters, numbers, and hyphens, with the first character a letter, the last a letter or a number, and be 4-63 characters.

    custom_class

    CustomClass

    Required. The custom class to create.

    CreatePhraseSetRequest

    Message sent by the client for the CreatePhraseSet method.

    Fields
    parent

    string

    Required. The parent resource where this phrase set will be created. Format:

    projects/{project}/locations/{location}/phraseSets

    Speech-to-Text supports three locations: global , us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

    Authorization requires the following IAM permission on the specified resource parent :

    • speech.phraseSets.create
    phrase_set_id

    string

    Required. The ID to use for the phrase set, which will become the final component of the phrase set's resource name.

    This value should restrict to letters, numbers, and hyphens, with the first character a letter, the last a letter or a number, and be 4-63 characters.

    phrase_set

    PhraseSet

    Required. The phrase set to create.

    CustomClass

    A set of words or phrases that represents a common concept likely to appear in your audio, for example a list of passenger ship names. CustomClass items can be substituted into placeholders that you set in PhraseSet phrases.

    Fields
    name

    string

    The resource name of the custom class.
    custom_class_id

    string

    If this custom class is a resource, the custom_class_id is the resource id of the CustomClass. Case sensitive.
    items[]

    ClassItem

    A collection of class items.

    ClassItem

    An item of the class.

    Fields
    value

    string

    The class item's value.

    DeleteCustomClassRequest

    Message sent by the client for the DeleteCustomClass method.

    Fields
    name

    string

    Required. The name of the custom class to delete. Format:

    projects/{project}/locations/{location}/customClasses/{custom_class}

    Speech-to-Text supports three locations: global , us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

    Authorization requires the following IAM permission on the specified resource name :

    • speech.customClasses.delete

    DeletePhraseSetRequest

    Message sent by the client for the DeletePhraseSet method.

    Fields
    name

    string

    Required. The name of the phrase set to delete. Format:

    projects/{project}/locations/{location}/phraseSets/{phrase_set}

    Authorization requires the following IAM permission on the specified resource name :

    • speech.phraseSets.delete

    GetCustomClassRequest

    Message sent by the client for the GetCustomClass method.

    Fields
    name

    string

    Required. The name of the custom class to retrieve. Format:

    projects/{project}/locations/{location}/customClasses/{custom_class}

    Authorization requires the following IAM permission on the specified resource name :

    • speech.customClasses.get

    GetPhraseSetRequest

    Message sent by the client for the GetPhraseSet method.

    Fields
    name

    string

    Required. The name of the phrase set to retrieve. Format:

    projects/{project}/locations/{location}/phraseSets/{phrase_set}

    Speech-to-Text supports three locations: global , us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

    Authorization requires the following IAM permission on the specified resource name :

    • speech.phraseSets.get

    ListCustomClassesRequest

    Message sent by the client for the ListCustomClasses method.

    Fields
    parent

    string

    Required. The parent, which owns this collection of custom classes. Format:

    projects/{project}/locations/{location}/customClasses

    Speech-to-Text supports three locations: global , us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

    Authorization requires the following IAM permission on the specified resource parent :

    • speech.customClasses.list
    page_size

    int32

    The maximum number of custom classes to return. The service may return fewer than this value. If unspecified, at most 50 custom classes will be returned. The maximum value is 1000; values above 1000 will be coerced to 1000.

    page_token

    string

    A page token, received from a previous ListCustomClass call. Provide this to retrieve the subsequent page.

    When paginating, all other parameters provided to ListCustomClass must match the call that provided the page token.

    ListCustomClassesResponse

    Message returned to the client by the ListCustomClasses method.

    Fields
    custom_classes[]

    CustomClass

    The custom classes.
    next_page_token

    string

    A token, which can be sent as page_token to retrieve the next page. If this field is omitted, there are no subsequent pages.

    ListPhraseSetRequest

    Message sent by the client for the ListPhraseSet method.

    Fields
    parent

    string

    Required. The parent, which owns this collection of phrase set. Format:

    projects/{project}/locations/{location}

    Speech-to-Text supports three locations: global , us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

    Authorization requires the following IAM permission on the specified resource parent :

    • speech.phraseSets.list
    page_size

    int32

    The maximum number of phrase sets to return. The service may return fewer than this value. If unspecified, at most 50 phrase sets will be returned. The maximum value is 1000; values above 1000 will be coerced to 1000.

    page_token

    string

    A page token, received from a previous ListPhraseSet call. Provide this to retrieve the subsequent page.

    When paginating, all other parameters provided to ListPhraseSet must match the call that provided the page token.

    ListPhraseSetResponse

    Message returned to the client by the ListPhraseSet method.

    Fields
    phrase_sets[]

    PhraseSet

    The phrase set.
    next_page_token

    string

    A token, which can be sent as page_token to retrieve the next page. If this field is omitted, there are no subsequent pages.

    LongRunningRecognizeMetadata

    Describes the progress of a long-running LongRunningRecognize call. It is included in the metadata field of the Operation returned by the GetOperation call of the google::longrunning::Operations service.

    Fields
    progress_percent

    int32

    Approximate percentage of audio processed thus far. Guaranteed to be 100 when the audio is fully processed and the results are available.
    start_time

    Timestamp

    Time when the request was received.
    last_update_time

    Timestamp

    Time of the most recent processing update.
    uri

    string

    Output only. The URI of the audio file being transcribed. Empty if the audio was sent as byte content.

    LongRunningRecognizeRequest

    The top-level message sent by the client for the LongRunningRecognize method.

    Fields
    config

    RecognitionConfig

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

    RecognitionAudio

    Required. The audio data to be recognized.
    output_config

    TranscriptOutputConfig

    Optional. Specifies an optional destination for the recognition results.

    LongRunningRecognizeResponse

    The only message returned to the client by the LongRunningRecognize method. It contains the result as zero or more sequential SpeechRecognitionResult messages. It is included in the result.response field of the Operation returned by the GetOperation call of the google::longrunning::Operations service.

    Fields
    results[]

    SpeechRecognitionResult

    Sequential list of transcription results corresponding to sequential portions of audio.
    total_billed_time

    Duration

    When available, billed audio seconds for the corresponding request.
    output_config

    TranscriptOutputConfig

    Original output config if present in the request.
    output_error

    Status

    If the transcript output fails this field contains the relevant error.
    speech_adaptation_info

    SpeechAdaptationInfo

    Provides information on speech adaptation behavior in response
    request_id

    int64

    The ID associated with the request. This is a unique ID specific only to the given request.

    PhraseSet

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

    Fields
    name

    string

    The resource name of the phrase set.
    phrases[]

    Phrase

    A list of word and phrases.
    boost

    float

    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 (exclusive) and 20. We recommend using a binary search approach to finding the optimal value for your use case as well as adding phrases both with and without boost to your requests.

    Phrase

    A phrases containing words and phrase "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 include pre-built or custom classes containing groups of words that represent common concepts that occur in natural language. For example, rather than providing a phrase hint for every month of the year (e.g. "i was born in january", "i was born in febuary", ...), use the pre-built $MONTH class improves the likelihood of correctly transcribing audio that includes months (e.g. "i was born in $month"). To refer to pre-built classes, use the class' symbol prepended with $ e.g. $MONTH . To refer to custom classes that were defined inline in the request, set the class's custom_class_id to a string unique to all class resources and inline classes. Then use the class' id wrapped in $ {...} e.g. "${my-months}". To refer to custom classes resources, use the class' id wrapped in ${} (e.g. ${my-months} ).

    Speech-to-Text supports three locations: global , us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

    Fields
    value

    string

    The phrase itself.
    boost

    float

    Hint Boost. Overrides the boost set at the phrase set level. 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 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 as well as adding phrases both with and without boost to your requests.

    RecognitionAudio

    Contains audio data in the encoding specified in the RecognitionConfig . Either content or uri must be supplied. Supplying both or neither returns google.rpc.Code.INVALID_ARGUMENT . See content limits .

    Fields
    Union field audio_source . The audio source, which is either inline content or a Google Cloud Storage uri. audio_source can be only one of the following:
    content

    bytes

    The audio data bytes encoded as specified in RecognitionConfig . Note: as with all bytes fields, proto buffers use a pure binary representation, whereas JSON representations use base64.

    uri

    string

    URI that points to a file that contains audio data bytes as specified in RecognitionConfig . The file must not be compressed (for example, gzip). Currently, only Google Cloud Storage URIs are supported, which must be specified in the following format: gs://bucket_name/object_name (other URI formats return google.rpc.Code.INVALID_ARGUMENT ). For more information, see Request URIs .

    RecognitionConfig

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

    Fields
    encoding

    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

    int32

    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

    int32

    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[]

    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 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

    int32

    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.

    speech_contexts[]

    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

    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

    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.

    AudioEncoding

    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 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. sample_rate_hertz must be 8000.
    AMR_WB Adaptive Multi-Rate Wideband codec. sample_rate_hertz must be 16000.
    OGG_OPUS Opus encoded audio frames in Ogg container ( OggOpus ). sample_rate_hertz 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. sample_rate_hertz must be 16000.
    WEBM_OPUS Opus encoded audio frames in WebM container ( OggOpus ). sample_rate_hertz must be one of 8000, 12000, 16000, 24000, or 48000.

    RecognitionMetadata

    Description of audio data to be recognized.

    Fields
    interaction_type

    InteractionType

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

    uint32

    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/ .
    microphone_distance

    MicrophoneDistance

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

    OriginalMediaType

    The original media the speech was recorded on.
    recording_device_type

    RecordingDeviceType

    The type of device the speech was recorded with.
    recording_device_name

    string

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

    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

    audio_topic

    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.

    RecognizeRequest

    The top-level message sent by the client for the Recognize method.

    Fields
    config

    RecognitionConfig

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

    RecognitionAudio

    Required. The audio data to be recognized.

    RecognizeResponse

    The only message returned to the client by the Recognize method. It contains the result as zero or more sequential SpeechRecognitionResult messages.

    Fields
    results[]

    SpeechRecognitionResult

    Sequential list of transcription results corresponding to sequential portions of audio.
    total_billed_time

    Duration

    When available, billed audio seconds for the corresponding request.
    speech_adaptation_info

    SpeechAdaptationInfo

    Provides information on adaptation behavior in response
    request_id

    int64

    The ID associated with the request. This is a unique ID specific only to the given request.

    SpeakerDiarizationConfig

    Config to enable speaker diarization.

    Fields
    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.
    min_speaker_count

    int32

    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.
    max_speaker_count

    int32

    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.
    speaker_tag
    (deprecated)

    int32

    Output only. Unused.

    SpeechAdaptation

    Speech adaptation configuration.

    Fields
    phrase_sets[]

    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.
    phrase_set_references[]

    string

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

    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 custom_class_id . Refer to the inline defined class in phrase hints by its custom_class_id .
    abnf_grammar

    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

    Fields
    abnf_strings[]

    string

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

    SpeechAdaptationInfo

    Information on speech adaptation use in results

    Fields
    adaptation_timeout

    bool

    Whether there was a timeout when applying speech adaptation. If true, adaptation had no effect in the response transcript.
    timeout_message

    string

    If set, returns a message specifying which part of the speech adaptation request timed out.

    SpeechContext

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

    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

    float

    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.

    SpeechRecognitionAlternative

    Alternative hypotheses (a.k.a. n-best list).

    Fields
    transcript

    string

    Transcript text representing the words that the user spoke. In languages that use spaces to separate words, the transcript might have a leading space if it isn't the first result. You can concatenate each result to obtain the full transcript without using a separator.
    confidence

    float

    The confidence estimate between 0.0 and 1.0. A higher number indicates an estimated greater likelihood that the recognized words are correct. This field is set only for the top alternative of a non-streaming result or, of a streaming result where is_final=true . This field is not guaranteed to be accurate and users should not rely on it to be always provided. The default of 0.0 is a sentinel value indicating confidence was not set.
    words[]

    WordInfo

    A list of word-specific information for each recognized word. Note: When enable_speaker_diarization is true, you will see all the words from the beginning of the audio.

    SpeechRecognitionResult

    A speech recognition result corresponding to a portion of the audio.

    Fields
    alternatives[]

    SpeechRecognitionAlternative

    May contain one or more recognition hypotheses (up to the maximum specified in max_alternatives ). These alternatives are ordered in terms of accuracy, with the top (first) alternative being the most probable, as ranked by the recognizer.
    channel_tag

    int32

    For multi-channel audio, this is the channel number corresponding to the recognized result for the audio from that channel. For audio_channel_count = N, its output values can range from '1' to 'N'.
    result_end_time

    Duration

    Time offset of the end of this result relative to the beginning of the audio.
    language_code

    string

    Output only. The BCP-47 language tag of the language in this result. This language code was detected to have the most likelihood of being spoken in the audio.

    StreamingRecognitionConfig

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

    Fields
    config

    RecognitionConfig

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

    single_utterance

    bool

    If false or omitted, the recognizer will perform continuous recognition (continuing to wait for and process audio even if the user pauses speaking) until the client closes the input stream (gRPC API) or until the maximum time limit has been reached. May return multiple StreamingRecognitionResult s with the is_final flag set to true .

    If true , the recognizer will detect a single spoken utterance. When it detects that the user has paused or stopped speaking, it will return an END_OF_SINGLE_UTTERANCE event and cease recognition. It will return no more than one StreamingRecognitionResult with the is_final flag set to true .

    The single_utterance field can only be used with specified models, otherwise an error is thrown. The model field in [ RecognitionConfig ][] must be set to:

    • command_and_search
    • phone_call AND additional field useEnhanced = true
    • The model field is left undefined. In this case the API auto-selects a model based on any other parameters that you set in RecognitionConfig .
    interim_results

    bool

    If true , interim results (tentative hypotheses) may be returned as they become available (these interim results are indicated with the is_final=false flag). If false or omitted, only is_final=true result(s) are returned.

    enable_voice_activity_events

    bool

    If true , responses with voice activity speech events will be returned as they are detected.

    voice_activity_timeout

    VoiceActivityTimeout

    If set, the server will automatically close the stream after the specified duration has elapsed after the last VOICE_ACTIVITY speech event has been sent. The field voice_activity_events must also be set to true.

    VoiceActivityTimeout

    Events that a timeout can be set on for voice activity.

    Fields
    speech_start_timeout

    Duration

    Duration to timeout the stream if no speech begins.
    speech_end_timeout

    Duration

    Duration to timeout the stream after speech ends.

    StreamingRecognitionResult

    A streaming speech recognition result corresponding to a portion of the audio that is currently being processed.

    Fields
    alternatives[]

    SpeechRecognitionAlternative

    May contain one or more recognition hypotheses (up to the maximum specified in max_alternatives ). These alternatives are ordered in terms of accuracy, with the top (first) alternative being the most probable, as ranked by the recognizer.
    is_final

    bool

    If false , this StreamingRecognitionResult represents an interim result that may change. If true , this is the final time the speech service will return this particular StreamingRecognitionResult , the recognizer will not return any further hypotheses for this portion of the transcript and corresponding audio.
    stability

    float

    An estimate of the likelihood that the recognizer will not change its guess about this interim result. Values range from 0.0 (completely unstable) to 1.0 (completely stable). This field is only provided for interim results ( is_final=false ). The default of 0.0 is a sentinel value indicating stability was not set.
    result_end_time

    Duration

    Time offset of the end of this result relative to the beginning of the audio.
    channel_tag

    int32

    For multi-channel audio, this is the channel number corresponding to the recognized result for the audio from that channel. For audio_channel_count = N, its output values can range from '1' to 'N'.
    language_code

    string

    Output only. The BCP-47 language tag of the language in this result. This language code was detected to have the most likelihood of being spoken in the audio.

    StreamingRecognizeRequest

    The top-level message sent by the client for the StreamingRecognize method. Multiple StreamingRecognizeRequest messages are sent. The first message must contain a streaming_config message and must not contain audio_content . All subsequent messages must contain audio_content and must not contain a streaming_config message.

    Fields
    Union field streaming_request . The streaming request, which is either a streaming config or audio content. streaming_request can be only one of the following:
    streaming_config

    StreamingRecognitionConfig

    Provides information to the recognizer that specifies how to process the request. The first StreamingRecognizeRequest message must contain a streaming_config message.

    audio_content

    bytes

    The audio data to be recognized. Sequential chunks of audio data are sent in sequential StreamingRecognizeRequest messages. The first StreamingRecognizeRequest message must not contain audio_content data and all subsequent StreamingRecognizeRequest messages must contain audio_content data. The audio bytes must be encoded as specified in RecognitionConfig . Note: as with all bytes fields, proto buffers use a pure binary representation (not base64). See content limits .

    StreamingRecognizeResponse

    StreamingRecognizeResponse is the only message returned to the client by StreamingRecognize . A series of zero or more StreamingRecognizeResponse messages are streamed back to the client. If there is no recognizable audio, and single_utterance is set to false, then no messages are streamed back to the client.

    Here's an example of a series of StreamingRecognizeResponse s that might be returned while processing audio:

    1. results { alternatives { transcript: "tube" } stability: 0.01 }

    2. results { alternatives { transcript: "to be a" } stability: 0.01 }

    3. results { alternatives { transcript: "to be" } stability: 0.9 } results { alternatives { transcript: " or not to be" } stability: 0.01 }

    4. results { alternatives { transcript: "to be or not to be" confidence: 0.92 } alternatives { transcript: "to bee or not to bee" } is_final: true }

    5. results { alternatives { transcript: " that's" } stability: 0.01 }

    6. results { alternatives { transcript: " that is" } stability: 0.9 } results { alternatives { transcript: " the question" } stability: 0.01 }

    7. results { alternatives { transcript: " that is the question" confidence: 0.98 } alternatives { transcript: " that was the question" } is_final: true }

    Notes:

    • Only two of the above responses #4 and #7 contain final results; they are indicated by is_final: true . Concatenating these together generates the full transcript: "to be or not to be that is the question".

    • The others contain interim results . #3 and #6 contain two interim results : the first portion has a high stability and is less likely to change; the second portion has a low stability and is very likely to change. A UI designer might choose to show only high stability results .

    • The specific stability and confidence values shown above are only for illustrative purposes. Actual values may vary.

    • In each response, only one of these fields will be set: error , speech_event_type , or one or more (repeated) results .

    Fields
    error

    Status

    If set, returns a google.rpc.Status message that specifies the error for the operation.
    results[]

    StreamingRecognitionResult

    This repeated list contains zero or more results that correspond to consecutive portions of the audio currently being processed. It contains zero or one is_final=true result (the newly settled portion), followed by zero or more is_final=false results (the interim results).
    speech_event_type

    SpeechEventType

    Indicates the type of speech event.
    total_billed_time

    Duration

    When available, billed audio seconds for the stream. Set only if this is the last response in the stream.
    speech_adaptation_info

    SpeechAdaptationInfo

    Provides information on adaptation behavior in response
    request_id

    int64

    The ID associated with the request. This is a unique ID specific only to the given request.

    SpeechEventType

    Indicates the type of speech event.

    Enums
    SPEECH_EVENT_UNSPECIFIED No speech event specified.
    END_OF_SINGLE_UTTERANCE This event indicates that the server has detected the end of the user's speech utterance and expects no additional speech. Therefore, the server will not process additional audio (although it may subsequently return additional results). The client should stop sending additional audio data, half-close the gRPC connection, and wait for any additional results until the server closes the gRPC connection. This event is only sent if single_utterance was set to true , and is not used otherwise.
    SPEECH_ACTIVITY_BEGIN This event indicates that the server has detected the beginning of human voice activity in the stream. This event can be returned multiple times if speech starts and stops repeatedly throughout the stream. This event is only sent if voice_activity_events is set to true.
    SPEECH_ACTIVITY_END This event indicates that the server has detected the end of human voice activity in the stream. This event can be returned multiple times if speech starts and stops repeatedly throughout the stream. This event is only sent if voice_activity_events is set to true.
    SPEECH_ACTIVITY_TIMEOUT This event indicates that the user-set timeout for speech activity begin or end has exceeded. Upon receiving this event, the client is expected to send a half close. Further audio will not be processed.

    TranscriptOutputConfig

    Specifies an optional destination for the recognition results.

    Fields

    Union field output_type .

    output_type can be only one of the following:

    gcs_uri

    string

    Specifies a Cloud Storage URI for the recognition results. Must be specified in the format: gs://bucket_name/object_name , and the bucket must already exist.

    UpdateCustomClassRequest

    Message sent by the client for the UpdateCustomClass method.

    Fields
    custom_class

    CustomClass

    Required. The custom class to update.

    The custom class's name field is used to identify the custom class to be updated. Format:

    projects/{project}/locations/{location}/customClasses/{custom_class}

    Speech-to-Text supports three locations: global , us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

    Authorization requires the following IAM permission on the specified resource customClass :

    • speech.customClasses.update
    update_mask

    FieldMask

    The list of fields to be updated.

    UpdatePhraseSetRequest

    Message sent by the client for the UpdatePhraseSet method.

    Fields
    phrase_set

    PhraseSet

    Required. The phrase set to update.

    The phrase set's name field is used to identify the set to be updated. Format:

    projects/{project}/locations/{location}/phraseSets/{phrase_set}

    Speech-to-Text supports three locations: global , us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

    Authorization requires the following IAM permission on the specified resource phraseSet :

    • speech.phraseSets.update
    update_mask

    FieldMask

    The list of fields to be updated.

    WordInfo

    Word-specific information for recognized words.

    Fields
    start_time

    Duration

    Time offset relative to the beginning of the audio, and corresponding to the start of the spoken word. This field is only set if enable_word_time_offsets=true and only in the top hypothesis. This is an experimental feature and the accuracy of the time offset can vary.
    end_time

    Duration

    Time offset relative to the beginning of the audio, and corresponding to the end of the spoken word. This field is only set if enable_word_time_offsets=true and only in the top hypothesis. This is an experimental feature and the accuracy of the time offset can vary.
    word

    string

    The word corresponding to this set of information.
    confidence

    float

    The confidence estimate between 0.0 and 1.0. A higher number indicates an estimated greater likelihood that the recognized words are correct. This field is set only for the top alternative of a non-streaming result or, of a streaming result where is_final=true . This field is not guaranteed to be accurate and users should not rely on it to be always provided. The default of 0.0 is a sentinel value indicating confidence was not set.
    speaker_tag

    int32

    Output only. A distinct integer value is assigned for every speaker within the audio. This field specifies which one of those speakers was detected to have spoken this word. Value ranges from '1' to diarization_speaker_count. speaker_tag is set if enable_speaker_diarization = 'true' and only in the top alternative.
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