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.

Scopes

You will need authorization for the https://www.googleapis.com/auth/cloud-platform scope to make a valid call.

If unset, the scope for this method defaults to https://www.googleapis.com/auth/cloud-platform. You can set the scope for this method like this: speech1 --scope <scope> speech longrunningrecognize ...

Required Request Value

The request value is a data-structure with various fields. Each field may be a simple scalar or another data-structure. In the latter case it is advised to set the field-cursor to the data-structure's field to specify values more concisely.

For example, a structure like this:

LongRunningRecognizeRequest:
  audio:
    content: string
    uri: string
  config:
    adaptation:
      phrase-set-references: [string]
    alternative-language-codes: [string]
    audio-channel-count: integer
    diarization-config:
      enable-speaker-diarization: boolean
      max-speaker-count: integer
      min-speaker-count: integer
      speaker-tag: integer
    enable-automatic-punctuation: boolean
    enable-separate-recognition-per-channel: boolean
    enable-spoken-emojis: boolean
    enable-spoken-punctuation: boolean
    enable-word-confidence: boolean
    enable-word-time-offsets: boolean
    encoding: string
    language-code: string
    max-alternatives: integer
    metadata:
      audio-topic: string
      industry-naics-code-of-audio: integer
      interaction-type: string
      microphone-distance: string
      original-media-type: string
      original-mime-type: string
      recording-device-name: string
      recording-device-type: string
    model: string
    profanity-filter: boolean
    sample-rate-hertz: integer
    use-enhanced: boolean
  output-config:
    gcs-uri: string

can be set completely with the following arguments which are assumed to be executed in the given order. Note how the cursor position is adjusted to the respective structures, allowing simple field names to be used most of the time.

  • -r .audio content=duo
    • 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=ipsum

    • 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.
  • ..config.adaptation phrase-set-references=gubergren

    • A collection of phrase set resource names to use.
    • Each invocation of this argument appends the given value to the array.
  • .. alternative-language-codes=lorem

    • 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).
    • Each invocation of this argument appends the given value to the array.
  • audio-channel-count=89
    • The number of channels in the input audio data. ONLY set this for MULTI-CHANNEL recognition. Valid values for LINEAR16 and FLAC are 1-8. Valid values for OGG_OPUS are '1'-'254'. Valid value for MULAW, AMR, AMR_WB and SPEEX_WITH_HEADER_BYTE is only 1. If 0 or omitted, defaults to one channel (mono). Note: We only recognize the first channel by default. To perform independent recognition on each channel set enable_separate_recognition_per_channel to 'true'.
  • diarization-config enable-speaker-diarization=false
    • 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.
  • max-speaker-count=97
    • 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.
  • min-speaker-count=84
    • 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.
  • speaker-tag=46

    • Output only. Unused.
  • .. enable-automatic-punctuation=false

    • 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-separate-recognition-per-channel=true
    • 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.
  • enable-spoken-emojis=true
    • The spoken emoji behavior for the call If not set, uses default behavior based on model of choice If 'true', adds spoken emoji formatting for the request. This will replace spoken emojis with the corresponding Unicode symbols in the final transcript. If 'false', spoken emojis are not replaced.
  • enable-spoken-punctuation=false
    • 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-word-confidence=true
    • 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-word-time-offsets=true
    • 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.
  • encoding=ipsum
    • 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.
  • language-code=est
    • 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.
  • max-alternatives=39
    • 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.
  • metadata audio-topic=ea
    • Description of the content. Eg. "Recordings of federal supreme court hearings from 2012".
  • industry-naics-code-of-audio=2
    • 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/.
  • interaction-type=lorem
    • The use case most closely describing the audio content to be recognized.
  • microphone-distance=eos
    • The audio type that most closely describes the audio being recognized.
  • original-media-type=labore
    • The original media the speech was recorded on.
  • original-mime-type=sed
    • 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
  • recording-device-name=duo
    • The device used to make the recording. Examples 'Nexus 5X' or 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or 'Cardioid Microphone'.
  • recording-device-type=sed

    • The type of device the speech was recorded with.
  • .. model=no

    • 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 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.
  • profanity-filter=true
    • 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.
  • sample-rate-hertz=77
    • 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.
  • use-enhanced=true

    • 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.
  • ..output-config gcs-uri=vero

    • 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.

About Cursors

The cursor position is key to comfortably set complex nested structures. The following rules apply:

  • The cursor position is always set relative to the current one, unless the field name starts with the . character. Fields can be nested such as in -r f.s.o .
  • The cursor position is set relative to the top-level structure if it starts with ., e.g. -r .s.s
  • You can also set nested fields without setting the cursor explicitly. For example, to set a value relative to the current cursor position, you would specify -r struct.sub_struct=bar.
  • You can move the cursor one level up by using ... Each additional . moves it up one additional level. E.g. ... would go three levels up.

Optional Output Flags

The method's return value a JSON encoded structure, which will be written to standard output by default.

  • -o out
    • out specifies the destination to which to write the server's result to. It will be a JSON-encoded structure. The destination may be - to indicate standard output, or a filepath that is to contain the received bytes. If unset, it defaults to standard output.

Optional General Properties

The following properties can configure any call, and are not specific to this method.

  • -p $-xgafv=string

    • V1 error format.
  • -p access-token=string

    • OAuth access token.
  • -p alt=string

    • Data format for response.
  • -p callback=string

    • JSONP
  • -p fields=string

    • Selector specifying which fields to include in a partial response.
  • -p key=string

    • API key. Your API key identifies your project and provides you with API access, quota, and reports. Required unless you provide an OAuth 2.0 token.
  • -p oauth-token=string

    • OAuth 2.0 token for the current user.
  • -p pretty-print=boolean

    • Returns response with indentations and line breaks.
  • -p quota-user=string

    • Available to use for quota purposes for server-side applications. Can be any arbitrary string assigned to a user, but should not exceed 40 characters.
  • -p upload-type=string

    • Legacy upload protocol for media (e.g. "media", "multipart").
  • -p upload-protocol=string

    • Upload protocol for media (e.g. "raw", "multipart").