text: // Copyright 2019 Google LLC
  //
  // Licensed under the Apache License, Version 2.0 (the "License");
  // you may not use this file except in compliance with the License.
  // You may obtain a copy of the License at
  //
  //     https://www.apache.org/licenses/LICENSE-2.0
  //
  // Unless required by applicable law or agreed to in writing, software
  // distributed under the License is distributed on an "AS IS" BASIS,
  // WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
  // See the License for the specific language governing permissions and
  // limitations under the License.
  
  // Note: this file is purely for documentation. Any contents are not expected
  // to be loaded as the JS file.
  
  /**
   * The top-level message sent by the client for the `Recognize` method.
   *
   * @property {Object} config
   *   *Required* Provides information to the recognizer that specifies how to
   *   process the request.
   *
   *   This object should have the same structure as [RecognitionConfig]{@link google.cloud.speech.v1p1beta1.RecognitionConfig}
   *
   * @property {Object} audio
   *   *Required* The audio data to be recognized.
   *
   *   This object should have the same structure as [RecognitionAudio]{@link google.cloud.speech.v1p1beta1.RecognitionAudio}
   *
   * @typedef RecognizeRequest
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.RecognizeRequest definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const RecognizeRequest = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };
  
  /**
   * The top-level message sent by the client for the `LongRunningRecognize`
   * method.
   *
   * @property {Object} config
   *   *Required* Provides information to the recognizer that specifies how to
   *   process the request.
   *
   *   This object should have the same structure as [RecognitionConfig]{@link google.cloud.speech.v1p1beta1.RecognitionConfig}
   *
   * @property {Object} audio
   *   *Required* The audio data to be recognized.
   *
   *   This object should have the same structure as [RecognitionAudio]{@link google.cloud.speech.v1p1beta1.RecognitionAudio}
   *
   * @typedef LongRunningRecognizeRequest
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.LongRunningRecognizeRequest definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const LongRunningRecognizeRequest = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };
  
  /**
   * 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` data.
   * All subsequent messages must contain `audio` data and must not contain a
   * `streaming_config` message.
   *
   * @property {Object} streamingConfig
   *   Provides information to the recognizer that specifies how to process the
   *   request. The first `StreamingRecognizeRequest` message must contain a
   *   `streaming_config`  message.
   *
   *   This object should have the same structure as [StreamingRecognitionConfig]{@link google.cloud.speech.v1p1beta1.StreamingRecognitionConfig}
   *
   * @property {Buffer} audioContent
   *   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, protobuffers use a
   *   pure binary representation (not base64). See
   *   [content limits](https://cloud.google.com/speech-to-text/quotas#content).
   *
   * @typedef StreamingRecognizeRequest
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.StreamingRecognizeRequest definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const StreamingRecognizeRequest = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };
  
  /**
   * Provides information to the recognizer that specifies how to process the
   * request.
   *
   * @property {Object} config
   *   *Required* Provides information to the recognizer that specifies how to
   *   process the request.
   *
   *   This object should have the same structure as [RecognitionConfig]{@link google.cloud.speech.v1p1beta1.RecognitionConfig}
   *
   * @property {boolean} singleUtterance
   *   *Optional* 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`.
   *
   * @property {boolean} interimResults
   *   *Optional* 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.
   *
   * @typedef StreamingRecognitionConfig
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.StreamingRecognitionConfig definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const StreamingRecognitionConfig = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };
  
  /**
   * Provides information to the recognizer that specifies how to process the
   * request.
   *
   * @property {number} encoding
   *   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.
   *
   *   The number should be among the values of [AudioEncoding]{@link google.cloud.speech.v1p1beta1.AudioEncoding}
   *
   * @property {number} sampleRateHertz
   *   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 and required
   *   for all other audio formats. For details, see
   *   AudioEncoding.
   *
   * @property {number} audioChannelCount
   *   *Optional* 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'.
   *
   * @property {boolean} enableSeparateRecognitionPerChannel
   *   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.
   *
   * @property {string} languageCode
   *   *Required* 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 [Language Support](https://cloud.google.com/speech-to-text/docs/languages)
   *   for a list of the currently supported language codes.
   *
   * @property {string[]} alternativeLanguageCodes
   *   *Optional* A list of up to 3 additional
   *   [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tags,
   *   listing possible alternative languages of the supplied audio.
   *   See [Language Support](https://cloud.google.com/speech-to-text/docs/languages)
   *   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).
   *
   * @property {number} maxAlternatives
   *   *Optional* 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.
   *
   * @property {boolean} profanityFilter
   *   *Optional* 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.
   *
   * @property {Object[]} speechContexts
   *   *Optional* array of
   *   SpeechContext. A means to
   *   provide context to assist the speech recognition. For more information, see
   *   [Phrase Hints](https://cloud.google.com/speech-to-text/docs/basics#phrase-hints).
   *
   *   This object should have the same structure as [SpeechContext]{@link google.cloud.speech.v1p1beta1.SpeechContext}
   *
   * @property {boolean} enableWordTimeOffsets
   *   *Optional* 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`.
   *
   * @property {boolean} enableWordConfidence
   *   *Optional* 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`.
   *
   * @property {boolean} enableAutomaticPunctuation
   *   *Optional* 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.
   *   Note: This is currently offered as an experimental service, complimentary
   *   to all users. In the future this may be exclusively available as a
   *   premium feature.
   *
   * @property {boolean} enableSpeakerDiarization
   *   *Optional* 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.
   *   Note: When this is true, 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.
   *
   * @property {number} diarizationSpeakerCount
   *   *Optional*
   *   If set, specifies the estimated number of speakers in the conversation.
   *   If not set, defaults to '2'.
   *   Ignored unless enable_speaker_diarization is set to true."
   *
   * @property {Object} metadata
   *   *Optional* Metadata regarding this request.
   *
   *   This object should have the same structure as [RecognitionMetadata]{@link google.cloud.speech.v1p1beta1.RecognitionMetadata}
   *
   * @property {string} model
   *   *Optional* 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.
   *   <table>
   *     <tr>
   *       <td><b>Model</b></td>
   *       <td><b>Description</b></td>
   *     </tr>
   *     <tr>
   *       <td><code>command_and_search</code></td>
   *       <td>Best for short queries such as voice commands or voice search.</td>
   *     </tr>
   *     <tr>
   *       <td><code>phone_call</code></td>
   *       <td>Best for audio that originated from a phone call (typically
   *       recorded at an 8khz sampling rate).</td>
   *     </tr>
   *     <tr>
   *       <td><code>video</code></td>
   *       <td>Best for audio that originated from 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.</td>
   *     </tr>
   *     <tr>
   *       <td><code>default</code></td>
   *       <td>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.</td>
   *     </tr>
   *   </table>
   *
   * @property {boolean} useEnhanced
   *   *Optional* 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.
   *
   * @typedef RecognitionConfig
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.RecognitionConfig definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const RecognitionConfig = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  
    /**
     * The encoding of the audio data sent in the request.
     *
     * All encodings support only 1 channel (mono) audio.
     *
     * 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`, and `SPEEX_WITH_HEADER_BYTE`.
     *
     * 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.
     *
     * @enum {number}
     * @memberof google.cloud.speech.v1p1beta1
     */
    AudioEncoding: {
  
      /**
       * Not specified.
       */
      ENCODING_UNSPECIFIED: 0,
  
      /**
       * Uncompressed 16-bit signed little-endian samples (Linear PCM).
       */
      LINEAR16: 1,
  
      /**
       * `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.
       */
      FLAC: 2,
  
      /**
       * 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
       */
      MULAW: 3,
  
      /**
       * Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
       */
      AMR: 4,
  
      /**
       * Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
       */
      AMR_WB: 5,
  
      /**
       * Opus encoded audio frames in Ogg container
       * ([OggOpus](https://wiki.xiph.org/OggOpus)).
       * `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
       */
      OGG_OPUS: 6,
  
      /**
       * 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](https://speex.org/)  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](https://tools.ietf.org/html/rfc5574).
       * 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.
       */
      SPEEX_WITH_HEADER_BYTE: 7,
  
      /**
       * MP3 audio. Support all standard MP3 bitrates (which range from 32-320
       * kbps). When using this encoding, `sample_rate_hertz` can be optionally
       * unset if not known.
       */
      MP3: 8
    }
  };
  
  /**
   * Description of audio data to be recognized.
   *
   * @property {number} interactionType
   *   The use case most closely describing the audio content to be recognized.
   *
   *   The number should be among the values of [InteractionType]{@link google.cloud.speech.v1p1beta1.InteractionType}
   *
   * @property {number} industryNaicsCodeOfAudio
   *   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/.
   *
   * @property {number} microphoneDistance
   *   The audio type that most closely describes the audio being recognized.
   *
   *   The number should be among the values of [MicrophoneDistance]{@link google.cloud.speech.v1p1beta1.MicrophoneDistance}
   *
   * @property {number} originalMediaType
   *   The original media the speech was recorded on.
   *
   *   The number should be among the values of [OriginalMediaType]{@link google.cloud.speech.v1p1beta1.OriginalMediaType}
   *
   * @property {number} recordingDeviceType
   *   The type of device the speech was recorded with.
   *
   *   The number should be among the values of [RecordingDeviceType]{@link google.cloud.speech.v1p1beta1.RecordingDeviceType}
   *
   * @property {string} recordingDeviceName
   *   The device used to make the recording.  Examples 'Nexus 5X' or
   *   'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
   *   'Cardioid Microphone'.
   *
   * @property {string} originalMimeType
   *   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
   *
   * @property {number} obfuscatedId
   *   Obfuscated (privacy-protected) ID of the user, to identify number of
   *   unique users using the service.
   *
   * @property {string} audioTopic
   *   Description of the content. Eg. "Recordings of federal supreme court
   *   hearings from 2012".
   *
   * @typedef RecognitionMetadata
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.RecognitionMetadata definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const RecognitionMetadata = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  
    /**
     * Use case categories that the audio recognition request can be described
     * by.
     *
     * @enum {number}
     * @memberof google.cloud.speech.v1p1beta1
     */
    InteractionType: {
  
      /**
       * Use case is either unknown or is something other than one of the other
       * values below.
       */
      INTERACTION_TYPE_UNSPECIFIED: 0,
  
      /**
       * 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)
       */
      DISCUSSION: 1,
  
      /**
       * One or more persons lecturing or presenting to others, mostly
       * uninterrupted.
       */
      PRESENTATION: 2,
  
      /**
       * A phone-call or video-conference in which two or more people, who are
       * not in the same room, are actively participating.
       */
      PHONE_CALL: 3,
  
      /**
       * A recorded message intended for another person to listen to.
       */
      VOICEMAIL: 4,
  
      /**
       * Professionally produced audio (eg. TV Show, Podcast).
       */
      PROFESSIONALLY_PRODUCED: 5,
  
      /**
       * Transcribe spoken questions and queries into text.
       */
      VOICE_SEARCH: 6,
  
      /**
       * Transcribe voice commands, such as for controlling a device.
       */
      VOICE_COMMAND: 7,
  
      /**
       * Transcribe speech to text to create a written document, such as a
       * text-message, email or report.
       */
      DICTATION: 8
    },
  
    /**
     * Enumerates the types of capture settings describing an audio file.
     *
     * @enum {number}
     * @memberof google.cloud.speech.v1p1beta1
     */
    MicrophoneDistance: {
  
      /**
       * Audio type is not known.
       */
      MICROPHONE_DISTANCE_UNSPECIFIED: 0,
  
      /**
       * 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.
       */
      NEARFIELD: 1,
  
      /**
       * The speaker if within 3 meters of the microphone.
       */
      MIDFIELD: 2,
  
      /**
       * The speaker is more than 3 meters away from the microphone.
       */
      FARFIELD: 3
    },
  
    /**
     * The original media the speech was recorded on.
     *
     * @enum {number}
     * @memberof google.cloud.speech.v1p1beta1
     */
    OriginalMediaType: {
  
      /**
       * Unknown original media type.
       */
      ORIGINAL_MEDIA_TYPE_UNSPECIFIED: 0,
  
      /**
       * The speech data is an audio recording.
       */
      AUDIO: 1,
  
      /**
       * The speech data originally recorded on a video.
       */
      VIDEO: 2
    },
  
    /**
     * The type of device the speech was recorded with.
     *
     * @enum {number}
     * @memberof google.cloud.speech.v1p1beta1
     */
    RecordingDeviceType: {
  
      /**
       * The recording device is unknown.
       */
      RECORDING_DEVICE_TYPE_UNSPECIFIED: 0,
  
      /**
       * Speech was recorded on a smartphone.
       */
      SMARTPHONE: 1,
  
      /**
       * Speech was recorded using a personal computer or tablet.
       */
      PC: 2,
  
      /**
       * Speech was recorded over a phone line.
       */
      PHONE_LINE: 3,
  
      /**
       * Speech was recorded in a vehicle.
       */
      VEHICLE: 4,
  
      /**
       * Speech was recorded outdoors.
       */
      OTHER_OUTDOOR_DEVICE: 5,
  
      /**
       * Speech was recorded indoors.
       */
      OTHER_INDOOR_DEVICE: 6
    }
  };
  
  /**
   * Provides "hints" to the speech recognizer to favor specific words and phrases
   * in the results.
   *
   * @property {string[]} phrases
   *   *Optional* 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](https://cloud.google.com/speech-to-text/quotas#content).
   *
   *   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.
   *
   * @property {number} boost
   *   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.
   *
   * @typedef SpeechContext
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.SpeechContext definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const SpeechContext = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };
  
  /**
   * 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](https://cloud.google.com/speech-to-text/quotas#content).
   *
   * @property {Buffer} content
   *   The audio data bytes encoded as specified in
   *   `RecognitionConfig`. Note: as with all bytes fields, protobuffers use a
   *   pure binary representation, whereas JSON representations use base64.
   *
   * @property {string} uri
   *   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](https://cloud.google.com/storage/docs/reference-uris).
   *
   * @typedef RecognitionAudio
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.RecognitionAudio definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const RecognitionAudio = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };
  
  /**
   * The only message returned to the client by the `Recognize` method. It
   * contains the result as zero or more sequential `SpeechRecognitionResult`
   * messages.
   *
   * @property {Object[]} results
   *   Output only. Sequential list of transcription results corresponding to
   *   sequential portions of audio.
   *
   *   This object should have the same structure as [SpeechRecognitionResult]{@link google.cloud.speech.v1p1beta1.SpeechRecognitionResult}
   *
   * @typedef RecognizeResponse
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.RecognizeResponse definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const RecognizeResponse = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };
  
  /**
   * 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.
   *
   * @property {Object[]} results
   *   Output only. Sequential list of transcription results corresponding to
   *   sequential portions of audio.
   *
   *   This object should have the same structure as [SpeechRecognitionResult]{@link google.cloud.speech.v1p1beta1.SpeechRecognitionResult}
   *
   * @typedef LongRunningRecognizeResponse
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.LongRunningRecognizeResponse definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const LongRunningRecognizeResponse = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };
  
  /**
   * 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.
   *
   * @property {number} progressPercent
   *   Approximate percentage of audio processed thus far. Guaranteed to be 100
   *   when the audio is fully processed and the results are available.
   *
   * @property {Object} startTime
   *   Time when the request was received.
   *
   *   This object should have the same structure as [Timestamp]{@link google.protobuf.Timestamp}
   *
   * @property {Object} lastUpdateTime
   *   Time of the most recent processing update.
   *
   *   This object should have the same structure as [Timestamp]{@link google.protobuf.Timestamp}
   *
   * @typedef LongRunningRecognizeMetadata
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.LongRunningRecognizeMetadata definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const LongRunningRecognizeMetadata = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };
  
  /**
   * `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 ten `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`.
   *
   * @property {Object} error
   *   Output only. If set, returns a google.rpc.Status
   *   message that specifies the error for the operation.
   *
   *   This object should have the same structure as [Status]{@link google.rpc.Status}
   *
   * @property {Object[]} results
   *   Output only. 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).
   *
   *   This object should have the same structure as [StreamingRecognitionResult]{@link google.cloud.speech.v1p1beta1.StreamingRecognitionResult}
   *
   * @property {number} speechEventType
   *   Output only. Indicates the type of speech event.
   *
   *   The number should be among the values of [SpeechEventType]{@link google.cloud.speech.v1p1beta1.SpeechEventType}
   *
   * @typedef StreamingRecognizeResponse
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.StreamingRecognizeResponse definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const StreamingRecognizeResponse = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  
    /**
     * Indicates the type of speech event.
     *
     * @enum {number}
     * @memberof google.cloud.speech.v1p1beta1
     */
    SpeechEventType: {
  
      /**
       * No speech event specified.
       */
      SPEECH_EVENT_UNSPECIFIED: 0,
  
      /**
       * 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.
       */
      END_OF_SINGLE_UTTERANCE: 1
    }
  };
  
  /**
   * A streaming speech recognition result corresponding to a portion of the audio
   * that is currently being processed.
   *
   * @property {Object[]} alternatives
   *   Output only. 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.
   *
   *   This object should have the same structure as [SpeechRecognitionAlternative]{@link google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative}
   *
   * @property {boolean} isFinal
   *   Output only. 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.
   *
   * @property {number} stability
   *   Output only. 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.
   *
   * @property {Object} resultEndTime
   *   Output only. Time offset of the end of this result relative to the
   *   beginning of the audio.
   *
   *   This object should have the same structure as [Duration]{@link google.protobuf.Duration}
   *
   * @property {number} channelTag
   *   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'.
   *
   * @property {string} languageCode
   *   Output only. The
   *   [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag of the
   *   language in this result. This language code was detected to have the most
   *   likelihood of being spoken in the audio.
   *
   * @typedef StreamingRecognitionResult
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.StreamingRecognitionResult definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const StreamingRecognitionResult = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };
  
  /**
   * A speech recognition result corresponding to a portion of the audio.
   *
   * @property {Object[]} alternatives
   *   Output only. 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.
   *
   *   This object should have the same structure as [SpeechRecognitionAlternative]{@link google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative}
   *
   * @property {number} channelTag
   *   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'.
   *
   * @property {string} languageCode
   *   Output only. The
   *   [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag of the
   *   language in this result. This language code was detected to have the most
   *   likelihood of being spoken in the audio.
   *
   * @typedef SpeechRecognitionResult
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.SpeechRecognitionResult definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const SpeechRecognitionResult = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };
  
  /**
   * Alternative hypotheses (a.k.a. n-best list).
   *
   * @property {string} transcript
   *   Output only. Transcript text representing the words that the user spoke.
   *
   * @property {number} confidence
   *   Output only. 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.
   *
   * @property {Object[]} words
   *   Output only. 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.
   *
   *   This object should have the same structure as [WordInfo]{@link google.cloud.speech.v1p1beta1.WordInfo}
   *
   * @typedef SpeechRecognitionAlternative
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const SpeechRecognitionAlternative = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };
  
  /**
   * Word-specific information for recognized words.
   *
   * @property {Object} startTime
   *   Output only. 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.
   *
   *   This object should have the same structure as [Duration]{@link google.protobuf.Duration}
   *
   * @property {Object} endTime
   *   Output only. 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.
   *
   *   This object should have the same structure as [Duration]{@link google.protobuf.Duration}
   *
   * @property {string} word
   *   Output only. The word corresponding to this set of information.
   *
   * @property {number} confidence
   *   Output only. 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.
   *
   * @property {number} speakerTag
   *   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.
   *
   * @typedef WordInfo
   * @memberof google.cloud.speech.v1p1beta1
   * @see [google.cloud.speech.v1p1beta1.WordInfo definition in proto format]{@link https://github.com/googleapis/googleapis/blob/master/google/cloud/speech/v1p1beta1/cloud_speech.proto}
   */
  const WordInfo = {
    // This is for documentation. Actual contents will be loaded by gRPC.
  };