gcloud-help/gcloud/alpha/ml/speech/recognize

NAME
    gcloud alpha ml speech recognize - get transcripts of short
        (less than 60 seconds) audio from an audio file

SYNOPSIS
    gcloud alpha ml speech recognize AUDIO --language-code=LANGUAGE_CODE
        [--additional-language-codes=[LANGUAGE_CODE,...]]
        [--enable-automatic-punctuation]
        [--encoding=ENCODING; default="encoding-unspecified"]
        [--filter-profanity] [--hints=[HINT,...]] [--include-word-confidence]
        [--include-word-time-offsets]
        [--max-alternatives=MAX_ALTERNATIVES; default=1] [--model=MODEL]
        [--sample-rate=SAMPLE_RATE]
        [--audio-channel-count=AUDIO_CHANNEL_COUNT
          --separate-channel-recognition]
        [--audio-topic=AUDIO_TOPIC --interaction-type=INTERACTION_TYPE
          --microphone-distance=MICROPHONE_DISTANCE --naics-code=NAICS_CODE
          --original-media-type=ORIGINAL_MEDIA_TYPE
          --original-mime-type=ORIGINAL_MIME_TYPE
          --recording-device-name=RECORDING_DEVICE_NAME
          --recording-device-type=RECORDING_DEVICE_TYPE]
        [--enable-speaker-diarization
          : --max-diarization-speaker-count=MAX_DIARIZATION_SPEAKER_COUNT
          --min-diarization-speaker-count=MIN_DIARIZATION_SPEAKER_COUNT]
        [GCLOUD_WIDE_FLAG ...]

DESCRIPTION
    (ALPHA) Get a transcript of an audio file that is less than 60 seconds. You
    can use an audio file that is on your local drive or a Google Cloud Storage
    URL.

    If the audio is longer than 60 seconds, you will get an error. Please use
    gcloud alpha ml speech recognize-long-running instead.

EXAMPLES
    To get a transcript of an audio file 'my-recording.wav':

        $ gcloud alpha ml speech recognize 'my-recording.wav' \
          --language-code=en-US

    To get a transcript of an audio file in bucket 'gs://bucket/myaudio' with a
    custom sampling rate and encoding that uses hints and filters profanity:

        $ gcloud alpha ml speech recognize 'gs://bucket/myaudio' \
          --language-code=es-ES --sample-rate=2200 --hints=Bueno \
          --encoding=OGG_OPUS --filter-profanity

POSITIONAL ARGUMENTS
     AUDIO
        The location of the audio file to transcribe. Must be a local path or a
        Google Cloud Storage URL (in the format gs://bucket/object).

REQUIRED FLAGS
     --language-code=LANGUAGE_CODE
        The language of the supplied audio as a BCP-47
        (https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag. Example:
        "en-US". See https://cloud.google.com/speech/docs/languages for a list
        of the currently supported language codes.

OPTIONAL FLAGS
     --additional-language-codes=[LANGUAGE_CODE,...]
        The BCP-47 language tags of other languages that the speech may be in.
        Up to 3 can be provided.

        If alternative languages are listed, recognition result will contain
        recognition in the most likely language detected including the main
        language-code.

     --enable-automatic-punctuation
        Adds punctuation to recognition result hypotheses.

     --encoding=ENCODING; default="encoding-unspecified"
        The type of encoding of the file. Required if the file format is not
        WAV or FLAC. ENCODING must be one of: alaw, amr, amr-wb,
        encoding-unspecified, flac, linear16, mp3, mulaw, ogg-opus,
        speex-with-header-byte, webm-opus.

     --filter-profanity
        If True, the server will attempt to filter out profanities, replacing
        all but the initial character in each filtered word with asterisks,
        e.g. f***.

     --hints=[HINT,...]
        A list of strings containing word 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
        https://cloud.google.com/speech/limits#content.

     --include-word-confidence
        Include a list of words and the confidence for those words in the top
        result.

     --include-word-time-offsets
        If True, the top result includes a list of words with the start and end
        time offsets (timestamps) for those words. If False, no word-level time
        offset information is returned.

     --max-alternatives=MAX_ALTERNATIVES; default=1
        Maximum number of recognition hypotheses to be returned. 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.

     --model=MODEL
        Select the model best suited to your domain to get best results. If you
        do not explicitly specify a model, Speech-to-Text will auto-select a
        model based on your other specified parameters. Some models are premium
        and cost more than standard models (although you can reduce the price
        by opting into
        https://cloud.google.com/speech-to-text/docs/data-logging). MODEL must
        be one of:

         command_and_search
            short queries such as voice commands or voice search.
         default
            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.
         latest_long
            Use this model for any kind of long form content such as media or
            spontaneous speech and conversations. Consider using this model in
            place of the video model, especially if the video model is not
            available in your target language. You can also use this in place
            of the default model.
         latest_short
            Use this model for short utterances that are a few seconds in
            length. It is useful for trying to capture commands or other single
            shot directed speech use cases. Consider using this model instead
            of the command and search model.
         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.
         phone_call
            audio that originated from a phone call (typically recorded at an
            8khz sampling rate).
         phone_call_enhanced
            audio that originated from a phone call (typically recorded at an
            8khz sampling rate). This is a premium model and can produce better
            results but costs more than the standard rate.
         telephony
            Improved version of the "phone_call" model, best for audio that
            originated from a phone call, typically recorded at an 8kHz
            sampling rate.
         telephony_short
            Dedicated version of the modern "telephony" model for short or even
            single-word utterances for audio that originated from a phone call,
            typically recorded at an 8kHz sampling rate.
         video_enhanced
            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.

     --sample-rate=SAMPLE_RATE
        The sample rate in Hertz. 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).

     Audio channel settings.

       --audio-channel-count=AUDIO_CHANNEL_COUNT
          The number of channels in the input audio data. Set this for
          separate-channel-recognition. Valid values are: 1)LINEAR16 and FLAC
          are 1-8 2)OGG_OPUS are 1-254 3) MULAW, AMR, AMR_WB and
          SPEEX_WITH_HEADER_BYTE is only 1.

          This flag argument must be specified if any of the other arguments in
          this group are specified.

       --separate-channel-recognition
          Recognition result will contain a channel_tag field to state which
          channel that result belongs to. If this is not true, only the first
          channel will be recognized.

          This flag argument must be specified if any of the other arguments in
          this group are specified.

     Description of audio data to be recognized. Note that the Google Cloud
     Speech-to-text-api does not use this information, and only passes it
     through back into response.

       --audio-topic=AUDIO_TOPIC
          (DEPRECATED) Description of the content, e.g. "Recordings of federal
          supreme court hearings from 2012".

          The audio-topic flag is deprecated and will be removed. The Google
          Cloud Speech-to-text api does not use it, and only passes it through
          back into response.

       --interaction-type=INTERACTION_TYPE
          (DEPRECATED) Determining the interaction type in the conversation.

          The interaction-type flag is deprecated and will be removed. The
          Google Cloud Speech-to-text api does not use it, and only passes it
          through back into response. INTERACTION_TYPE must be one of:

           dictation
              Transcribe speech to text to create a written document, such as a
              text-message, email or report.
           discussion
              Multiple people in a conversation or discussion.
           phone-call
              A phone-call or video-conference in which two or more people, who
              are not in the same room, are actively participating.
           presentation
              One or more persons lecturing or presenting to others, mostly
              uninterrupted.
           professionally-produced
              Professionally produced audio (eg. TV Show, Podcast).
           voicemail
              A recorded message intended for another person to listen to.
           voice-command
              Transcribe voice commands, such as for controlling a device.
           voice-search
              Transcribe spoken questions and queries into text.

       --microphone-distance=MICROPHONE_DISTANCE
          (DEPRECATED) The distance at which the audio device is placed to
          record the conversation.

          The microphone-distance flag is deprecated and will be removed. The
          Google Cloud Speech-to-text api does not use it, and only passes it
          through back into response. MICROPHONE_DISTANCE must be one of:

           farfield
              The speaker is more than 3 meters away from the microphone.
           midfield
              The speaker is within 3 meters of the microphone.
           nearfield
              The audio was captured from a microphone close to the speaker,
              generally within 1 meter. Examples include a phone, dictaphone,
              or handheld microphone.

       --naics-code=NAICS_CODE
          (DEPRECATED) The industry vertical to which this speech recognition
          request most closely applies.

          The naics-code flag is deprecated and will be removed. The Google
          Cloud Speech-to-text api does not use it, and only passes it through
          back into response.

       --original-media-type=ORIGINAL_MEDIA_TYPE
          (DEPRECATED) The media type of the original audio conversation.

          The original-media-type flag is deprecated and will be removed. The
          Google Cloud Speech-to-text api does not use it, and only passes it
          through back into response. ORIGINAL_MEDIA_TYPE must be one of:

           audio
              The speech data is an audio recording.
           video
              The speech data originally recorded on a video.

       --original-mime-type=ORIGINAL_MIME_TYPE
          (DEPRECATED) Mime type of the original audio file. Examples:
          audio/m4a, audio/mp3.

          The original-mime-type flag is deprecated and will be removed. The
          Google Cloud Speech-to-text api does not use it, and only passes it
          through back into response.

       --recording-device-name=RECORDING_DEVICE_NAME
          (DEPRECATED) The device used to make the recording. Examples: Nexus
          5X, Polycom SoundStation IP 6000

          The recording-device-name flag is deprecated and will be removed. The
          Google Cloud Speech-to-text api does not use it, and only passes it
          through back into response.

       --recording-device-type=RECORDING_DEVICE_TYPE
          (DEPRECATED) The device type through which the original audio was
          recorded on.

          The recording-device-type flag is deprecated and will be removed. The
          Google Cloud Speech-to-text api does not use it, and only passes it
          through back into response. RECORDING_DEVICE_TYPE must be one of:

           indoor
              Speech was recorded indoors.
           outdoor
              Speech was recorded outdoors.
           pc
              Speech was recorded using a personal computer or tablet.
           phone-line
              Speech was recorded over a phone line.
           smartphone
              Speech was recorded on a smartphone.
           vehicle
              Speech was recorded in a vehicle.

     --enable-speaker-diarization
        Enable speaker detection for each recognized word in the top
        alternative of the recognition result using an integer speaker_tag
        provided in the WordInfo.

     --max-diarization-speaker-count=MAX_DIARIZATION_SPEAKER_COUNT
        Maximum estimated number of speakers in the conversation being
        recognized.

     --min-diarization-speaker-count=MIN_DIARIZATION_SPEAKER_COUNT
        Minimum estimated number of speakers in the conversation being
        recognized.

GCLOUD WIDE FLAGS
    These flags are available to all commands: --access-token-file, --account,
    --billing-project, --configuration, --flags-file, --flatten, --format,
    --help, --impersonate-service-account, --log-http, --project, --quiet,
    --trace-token, --user-output-enabled, --verbosity.

    Run $ gcloud help for details.

API REFERENCE
    This command uses the speech/v1p1beta1 API. The full documentation for this
    API can be found at:
    https://cloud.google.com/speech-to-text/docs/quickstart-protocol

NOTES
    This command is currently in alpha and might change without notice. If this
    command fails with API permission errors despite specifying the correct
    project, you might be trying to access an API with an invitation-only early
    access allowlist. These variants are also available:

        $ gcloud ml speech recognize

        $ gcloud beta ml speech recognize