ibm-watson/speech-to-text/v1-generated.d.ts

Client library to use the IBM Watson Services

/**
 * (C) Copyright IBM Corp. 2024.
 *
 * 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
 *
 *      http://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.
 */
/// <reference types="node" />
/// <reference types="node" />
import { IncomingHttpHeaders, OutgoingHttpHeaders } from 'http';
import { BaseService, UserOptions } from 'ibm-cloud-sdk-core';
/**
 * The IBM Watson&trade; Speech to Text service provides APIs that use IBM's speech-recognition capabilities to produce
 * transcripts of spoken audio. The service can transcribe speech from various languages and audio formats. In addition
 * to basic transcription, the service can produce detailed information about many different aspects of the audio. It
 * returns all JSON response content in the UTF-8 character set.
 *
 * The service supports two types of models: previous-generation models that include the terms `Broadband` and
 * `Narrowband` in their names, and next-generation models that include the terms `Multimedia` and `Telephony` in their
 * names. Broadband and multimedia models have minimum sampling rates of 16 kHz. Narrowband and telephony models have
 * minimum sampling rates of 8 kHz. The next-generation models offer high throughput and greater transcription accuracy.
 *
 *
 * Effective **31 July 2023**, all previous-generation models will be removed from the service and the documentation.
 * Most previous-generation models were deprecated on 15 March 2022. You must migrate to the equivalent large speech
 * model or next-generation model by 31 July 2023. For more information, see [Migrating to large speech
 * models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-migrate).{: deprecated}
 *
 * For speech recognition, the service supports synchronous and asynchronous HTTP Representational State Transfer (REST)
 * interfaces. It also supports a WebSocket interface that provides a full-duplex, low-latency communication channel:
 * Clients send requests and audio to the service and receive results over a single connection asynchronously.
 *
 * The service also offers two customization interfaces. Use language model customization to expand the vocabulary of a
 * base model with domain-specific terminology. Use acoustic model customization to adapt a base model for the acoustic
 * characteristics of your audio. For language model customization, the service also supports grammars. A grammar is a
 * formal language specification that lets you restrict the phrases that the service can recognize.
 *
 * Language model customization and grammars are available for most previous- and next-generation models. Acoustic model
 * customization is available for all previous-generation models.
 *
 * API Version: 1.0.0
 * See: https://cloud.ibm.com/docs/speech-to-text
 */
declare class SpeechToTextV1 extends BaseService {
    static DEFAULT_SERVICE_URL: string;
    static DEFAULT_SERVICE_NAME: string;
    /**
     * Construct a SpeechToTextV1 object.
     *
     * @param {Object} options - Options for the service.
     * @param {string} [options.serviceUrl] - The base URL for the service
     * @param {OutgoingHttpHeaders} [options.headers] - Default headers that shall be included with every request to the service.
     * @param {string} [options.serviceName] - The name of the service to configure
     * @param {Authenticator} [options.authenticator] - The Authenticator object used to authenticate requests to the service. Defaults to environment if not set
     * @constructor
     * @returns {SpeechToTextV1}
     */
    constructor(options: UserOptions);
    /*************************
     * models
     ************************/
    /**
     * List models.
     *
     * Lists all language models that are available for use with the service. The information includes the name of the
     * model and its minimum sampling rate in Hertz, among other things. The ordering of the list of models can change
     * from call to call; do not rely on an alphabetized or static list of models.
     *
     * **See also:** [Listing all
     * models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-list#models-list-all).
     *
     * @param {Object} [params] - The parameters to send to the service.
     * @param {OutgoingHttpHeaders} [params.headers] - Custom request headers
     * @returns {Promise<SpeechToTextV1.Response<SpeechToTextV1.SpeechModels>>}
     */
    listModels(params?: SpeechToTextV1.ListModelsParams): Promise<SpeechToTextV1.Response<SpeechToTextV1.SpeechModels>>;
    /**
     * Get a model.
     *
     * Gets information for a single specified language model that is available for use with the service. The information
     * includes the name of the model and its minimum sampling rate in Hertz, among other things.
     *
     * **See also:** [Listing a specific
     * model](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-list#models-list-specific).
     *
     * @param {Object} params - The parameters to send to the service.
     * @param {string} params.modelId - The identifier of the model in the form of its name from the output of the [List
     * models](#listmodels) method.
     * @param {OutgoingHttpHeaders} [params.headers] - Custom request headers
     * @returns {Promise<SpeechToTextV1.Response<SpeechToTextV1.SpeechModel>>}
     */
    getModel(params: SpeechToTextV1.GetModelParams): Promise<SpeechToTextV1.Response<SpeechToTextV1.SpeechModel>>;
    /*************************
     * synchronous
     ************************/
    /**
     * Recognize audio.
     *
     * Sends audio and returns transcription results for a recognition request. You can pass a maximum of 100 MB and a
     * minimum of 100 bytes of audio with a request. The service automatically detects the endianness of the incoming
     * audio and, for audio that includes multiple channels, downmixes the audio to one-channel mono during transcoding.
     * The method returns only final results; to enable interim results, use the WebSocket API. (With the `curl` command,
     * use the `--data-binary` option to upload the file for the request.)
     *
     * **See also:** [Making a basic HTTP
     * request](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-http#HTTP-basic).
     *
     * ### Streaming mode
     *
     *  For requests to transcribe live audio as it becomes available, you must set the `Transfer-Encoding` header to
     * `chunked` to use streaming mode. In streaming mode, the service closes the connection (status code 408) if it does
     * not receive at least 15 seconds of audio (including silence) in any 30-second period. The service also closes the
     * connection (status code 400) if it detects no speech for `inactivity_timeout` seconds of streaming audio; use the
     * `inactivity_timeout` parameter to change the default of 30 seconds.
     *
     * **See also:**
     * * [Audio transmission](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-input#transmission)
     * * [Timeouts](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-input#timeouts)
     *
     * ### Audio formats (content types)
     *
     *  The service accepts audio in the following formats (MIME types).
     * * For formats that are labeled **Required**, you must use the `Content-Type` header with the request to specify the
     * format of the audio.
     * * For all other formats, you can omit the `Content-Type` header or specify `application/octet-stream` with the
     * header to have the service automatically detect the format of the audio. (With the `curl` command, you can specify
     * either `"Content-Type:"` or `"Content-Type: application/octet-stream"`.)
     *
     * Where indicated, the format that you specify must include the sampling rate and can optionally include the number
     * of channels and the endianness of the audio.
     * * `audio/alaw` (**Required.** Specify the sampling rate (`rate`) of the audio.)
     * * `audio/basic` (**Required.** Use only with narrowband models.)
     * * `audio/flac`
     * * `audio/g729` (Use only with narrowband models.)
     * * `audio/l16` (**Required.** Specify the sampling rate (`rate`) and optionally the number of channels (`channels`)
     * and endianness (`endianness`) of the audio.)
     * * `audio/mp3`
     * * `audio/mpeg`
     * * `audio/mulaw` (**Required.** Specify the sampling rate (`rate`) of the audio.)
     * * `audio/ogg` (The service automatically detects the codec of the input audio.)
     * * `audio/ogg;codecs=opus`
     * * `audio/ogg;codecs=vorbis`
     * * `audio/wav` (Provide audio with a maximum of nine channels.)
     * * `audio/webm` (The service automatically detects the codec of the input audio.)
     * * `audio/webm;codecs=opus`
     * * `audio/webm;codecs=vorbis`
     *
     * The sampling rate of the audio must match the sampling rate of the model for the recognition request: for broadband
     * models, at least 16 kHz; for narrowband models, at least 8 kHz. If the sampling rate of the audio is higher than
     * the minimum required rate, the service down-samples the audio to the appropriate rate. If the sampling rate of the
     * audio is lower than the minimum required rate, the request fails.
     *
     *  **See also:** [Supported audio
     * formats](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-audio-formats).
     *
     * ### Large speech models and Next-generation models
     *
     *  The service supports large speech models and next-generation `Multimedia` (16 kHz) and `Telephony` (8 kHz) models
     * for many languages. Large speech models and next-generation models have higher throughput than the service's
     * previous generation of `Broadband` and `Narrowband` models. When you use large speech models and next-generation
     * models, the service can return transcriptions more quickly and also provide noticeably better transcription
     * accuracy.
     *
     * You specify a large speech model or next-generation model by using the `model` query parameter, as you do a
     * previous-generation model. Only the next-generation models support the `low_latency` parameter, and all large
     * speech models and next-generation models support the `character_insertion_bias` parameter. These parameters are not
     * available with previous-generation models.
     *
     * Large speech models and next-generation models do not support all of the speech recognition parameters that are
     * available for use with previous-generation models. Next-generation models do not support the following parameters:
     * * `acoustic_customization_id`
     * * `keywords` and `keywords_threshold`
     * * `processing_metrics` and `processing_metrics_interval`
     * * `word_alternatives_threshold`
     *
     * **Important:** Effective **31 July 2023**, all previous-generation models will be removed from the service and the
     * documentation. Most previous-generation models were deprecated on 15 March 2022. You must migrate to the equivalent
     * large speech model or next-generation model by 31 July 2023. For more information, see [Migrating to large speech
     * models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-migrate).
     *
     * **See also:**
     * * [Large speech languages and
     * models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-large-speech-languages)
     * * [Supported features for large speech
     * models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-large-speech-languages#models-lsm-supported-features)
     * * [Next-generation languages and models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-ng)
     * * [Supported features for next-generation
     * models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-ng#models-ng-features)
     *
     * ### Multipart speech recognition
     *
     *  **Note:** The asynchronous HTTP interface, WebSocket interface, and Watson SDKs do not support multipart speech
     * recognition.
     *
     * The HTTP `POST` method of the service also supports multipart speech recognition. With multipart requests, you pass
     * all audio data as multipart form data. You specify some parameters as request headers and query parameters, but you
     * pass JSON metadata as form data to control most aspects of the transcription. You can use multipart recognition to
     * pass multiple audio files with a single request.
     *
     * Use the multipart approach with browsers for which JavaScript is disabled or when the parameters used with the
     * request are greater than the 8 KB limit imposed by most HTTP servers and proxies. You can encounter this limit, for
     * example, if you want to spot a very large number of keywords.
     *
     * **See also:** [Making a multipart HTTP
     * request](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-http#HTTP-multi).
     *
     * @param {Object} params - The parameters to send to the service.
     * @param {NodeJS.ReadableStream | Buffer} params.audio - The audio to transcribe.
     * @param {string} [params.contentType] - The format (MIME type) of the audio. For more information about specifying
     * an audio format, see **Audio formats (content types)** in the method description.
     * @param {string} [params.model] - The model to use for speech recognition. If you omit the `model` parameter, the
     * service uses the US English `en-US_BroadbandModel` by default.
     *
     * _For IBM Cloud Pak for Data,_ if you do not install the `en-US_BroadbandModel`, you must either specify a model
     * with the request or specify a new default model for your installation of the service.
     *
     * **See also:**
     * * [Using a model for speech recognition](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-use)
     * * [Using the default
     * model](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-use#models-use-default).
     * @param {boolean} [params.speechBeginEvent] - If `true`, the service returns a response object `SpeechActivity`
     * which contains the time when a speech activity is detected in the stream. This can be used both in standard and low
     * latency mode. This feature enables client applications to know that some words/speech has been detected and the
     * service is in the process of decoding. This can be used in lieu of interim results in standard mode. See [Using
     * speech recognition
     * parameters](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-service-features#features-parameters).
     * @param {string} [params.languageCustomizationId] - The customization ID (GUID) of a custom language model that is
     * to be used with the recognition request. The base model of the specified custom language model must match the model
     * specified with the `model` parameter. You must make the request with credentials for the instance of the service
     * that owns the custom model. By default, no custom language model is used. See [Using a custom language model for
     * speech recognition](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-languageUse).
     *
     * **Note:** Use this parameter instead of the deprecated `customization_id` parameter.
     * @param {string} [params.acousticCustomizationId] - The customization ID (GUID) of a custom acoustic model that is
     * to be used with the recognition request. The base model of the specified custom acoustic model must match the model
     * specified with the `model` parameter. You must make the request with credentials for the instance of the service
     * that owns the custom model. By default, no custom acoustic model is used. See [Using a custom acoustic model for
     * speech recognition](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-acousticUse).
     * @param {string} [params.baseModelVersion] - The version of the specified base model that is to be used with the
     * recognition request. Multiple versions of a base model can exist when a model is updated for internal improvements.
     * The parameter is intended primarily for use with custom models that have been upgraded for a new base model. The
     * default value depends on whether the parameter is used with or without a custom model. See [Making speech
     * recognition requests with upgraded custom
     * models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-custom-upgrade-use#custom-upgrade-use-recognition).
     * @param {number} [params.customizationWeight] - If you specify the customization ID (GUID) of a custom language
     * model with the recognition request, the customization weight tells the service how much weight to give to words
     * from the custom language model compared to those from the base model for the current request.
     *
     * Specify a value between 0.0 and 1.0. Unless a different customization weight was specified for the custom model
     * when the model was trained, the default value is:
     * * 0.5 for large speech models
     * * 0.3 for previous-generation models
     * * 0.2 for most next-generation models
     * * 0.1 for next-generation English and Japanese models
     *
     * A customization weight that you specify overrides a weight that was specified when the custom model was trained.
     * The default value yields the best performance in general. Assign a higher value if your audio makes frequent use of
     * OOV words from the custom model. Use caution when setting the weight: a higher value can improve the accuracy of
     * phrases from the custom model's domain, but it can negatively affect performance on non-domain phrases.
     *
     * See [Using customization
     * weight](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-languageUse#weight).
     * @param {number} [params.inactivityTimeout] - The time in seconds after which, if only silence (no speech) is
     * detected in streaming audio, the connection is closed with a 400 error. The parameter is useful for stopping audio
     * submission from a live microphone when a user simply walks away. Use `-1` for infinity. See [Inactivity
     * timeout](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-input#timeouts-inactivity).
     * @param {string[]} [params.keywords] - An array of keyword strings to spot in the audio. Each keyword string can
     * include one or more string tokens. Keywords are spotted only in the final results, not in interim hypotheses. If
     * you specify any keywords, you must also specify a keywords threshold. Omit the parameter or specify an empty array
     * if you do not need to spot keywords.
     *
     * You can spot a maximum of 1000 keywords with a single request. A single keyword can have a maximum length of 1024
     * characters, though the maximum effective length for double-byte languages might be shorter. Keywords are
     * case-insensitive.
     *
     * See [Keyword spotting](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-spotting#keyword-spotting).
     * @param {number} [params.keywordsThreshold] - A confidence value that is the lower bound for spotting a keyword. A
     * word is considered to match a keyword if its confidence is greater than or equal to the threshold. Specify a
     * probability between 0.0 and 1.0. If you specify a threshold, you must also specify one or more keywords. The
     * service performs no keyword spotting if you omit either parameter. See [Keyword
     * spotting](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-spotting#keyword-spotting).
     * @param {number} [params.maxAlternatives] - The maximum number of alternative transcripts that the service is to
     * return. By default, the service returns a single transcript. If you specify a value of `0`, the service uses the
     * default value, `1`. See [Maximum
     * alternatives](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-metadata#max-alternatives).
     * @param {number} [params.wordAlternativesThreshold] - A confidence value that is the lower bound for identifying a
     * hypothesis as a possible word alternative (also known as "Confusion Networks"). An alternative word is considered
     * if its confidence is greater than or equal to the threshold. Specify a probability between 0.0 and 1.0. By default,
     * the service computes no alternative words. See [Word
     * alternatives](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-spotting#word-alternatives).
     * @param {boolean} [params.wordConfidence] - If `true`, the service returns a confidence measure in the range of 0.0
     * to 1.0 for each word. By default, the service returns no word confidence scores. See [Word
     * confidence](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-metadata#word-confidence).
     * @param {boolean} [params.timestamps] - If `true`, the service returns time alignment for each word. By default, no
     * timestamps are returned. See [Word
     * timestamps](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-metadata#word-timestamps).
     * @param {boolean} [params.profanityFilter] - If `true`, the service filters profanity from all output except for
     * keyword results by replacing inappropriate words with a series of asterisks. Set the parameter to `false` to return
     * results with no censoring.
     *
     * **Note:** The parameter can be used with US English and Japanese transcription only. See [Profanity
     * filtering](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-formatting#profanity-filtering).
     * @param {boolean} [params.smartFormatting] - If `true`, the service converts dates, times, series of digits and
     * numbers, phone numbers, currency values, and internet addresses into more readable, conventional representations in
     * the final transcript of a recognition request. For US English, the service also converts certain keyword strings to
     * punctuation symbols. By default, the service performs no smart formatting.
     *
     * **Note:** The parameter can be used with US English, Japanese, and Spanish (all dialects) transcription only.
     *
     * See [Smart formatting](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-formatting#smart-formatting).
     * @param {number} [params.smartFormattingVersion] - Smart formatting version for large speech models and
     * next-generation models is supported in US English, Brazilian Portuguese, French, German, Spanish and French
     * Canadian languages.
     * @param {boolean} [params.speakerLabels] - If `true`, the response includes labels that identify which words were
     * spoken by which participants in a multi-person exchange. By default, the service returns no speaker labels. Setting
     * `speaker_labels` to `true` forces the `timestamps` parameter to be `true`, regardless of whether you specify
     * `false` for the parameter.
     * * _For previous-generation models,_ the parameter can be used with Australian English, US English, German,
     * Japanese, Korean, and Spanish (both broadband and narrowband models) and UK English (narrowband model)
     * transcription only.
     * * _For large speech models and next-generation models,_ the parameter can be used with all available languages.
     *
     * See [Speaker labels](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-speaker-labels).
     * @param {string} [params.grammarName] - The name of a grammar that is to be used with the recognition request. If
     * you specify a grammar, you must also use the `language_customization_id` parameter to specify the name of the
     * custom language model for which the grammar is defined. The service recognizes only strings that are recognized by
     * the specified grammar; it does not recognize other custom words from the model's words resource.
     *
     * See [Using a grammar for speech
     * recognition](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-grammarUse).
     * @param {boolean} [params.redaction] - If `true`, the service redacts, or masks, numeric data from final
     * transcripts. The feature redacts any number that has three or more consecutive digits by replacing each digit with
     * an `X` character. It is intended to redact sensitive numeric data, such as credit card numbers. By default, the
     * service performs no redaction.
     *
     * When you enable redaction, the service automatically enables smart formatting, regardless of whether you explicitly
     * disable that feature. To ensure maximum security, the service also disables keyword spotting (ignores the
     * `keywords` and `keywords_threshold` parameters) and returns only a single final transcript (forces the
     * `max_alternatives` parameter to be `1`).
     *
     * **Note:** The parameter can be used with US English, Japanese, and Korean transcription only.
     *
     * See [Numeric
     * redaction](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-formatting#numeric-redaction).
     * @param {boolean} [params.audioMetrics] - If `true`, requests detailed information about the signal characteristics
     * of the input audio. The service returns audio metrics with the final transcription results. By default, the service
     * returns no audio metrics.
     *
     * See [Audio metrics](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-metrics#audio-metrics).
     * @param {number} [params.endOfPhraseSilenceTime] - Specifies the duration of the pause interval at which the service
     * splits a transcript into multiple final results. If the service detects pauses or extended silence before it
     * reaches the end of the audio stream, its response can include multiple final results. Silence indicates a point at
     * which the speaker pauses between spoken words or phrases.
     *
     * Specify a value for the pause interval in the range of 0.0 to 120.0.
     * * A value greater than 0 specifies the interval that the service is to use for speech recognition.
     * * A value of 0 indicates that the service is to use the default interval. It is equivalent to omitting the
     * parameter.
     *
     * The default pause interval for most languages is 0.8 seconds; the default for Chinese is 0.6 seconds.
     *
     * See [End of phrase silence
     * time](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-parsing#silence-time).
     * @param {boolean} [params.splitTranscriptAtPhraseEnd] - If `true`, directs the service to split the transcript into
     * multiple final results based on semantic features of the input, for example, at the conclusion of meaningful
     * phrases such as sentences. The service bases its understanding of semantic features on the base language model that
     * you use with a request. Custom language models and grammars can also influence how and where the service splits a
     * transcript.
     *
     * By default, the service splits transcripts based solely on the pause interval. If the parameters are used together
     * on the same request, `end_of_phrase_silence_time` has precedence over `split_transcript_at_phrase_end`.
     *
     * See [Split transcript at phrase
     * end](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-parsing#split-transcript).
     * @param {number} [params.speechDetectorSensitivity] - The sensitivity of speech activity detection that the service
     * is to perform. Use the parameter to suppress word insertions from music, coughing, and other non-speech events. The
     * service biases the audio it passes for speech recognition by evaluating the input audio against prior models of
     * speech and non-speech activity.
     *
     * Specify a value between 0.0 and 1.0:
     * * 0.0 suppresses all audio (no speech is transcribed).
     * * 0.5 (the default) provides a reasonable compromise for the level of sensitivity.
     * * 1.0 suppresses no audio (speech detection sensitivity is disabled).
     *
     * The values increase on a monotonic curve. Specifying one or two decimal places of precision (for example, `0.55`)
     * is typically more than sufficient.
     *
     * The parameter is supported with all large speech models, next-generation models and with most previous-generation
     * models. See [Speech detector
     * sensitivity](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-detection#detection-parameters-sensitivity)
     * and [Language model
     * support](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-detection#detection-support).
     * @param {number} [params.backgroundAudioSuppression] - The level to which the service is to suppress background
     * audio based on its volume to prevent it from being transcribed as speech. Use the parameter to suppress side
     * conversations or background noise.
     *
     * Specify a value in the range of 0.0 to 1.0:
     * * 0.0 (the default) provides no suppression (background audio suppression is disabled).
     * * 0.5 provides a reasonable level of audio suppression for general usage.
     * * 1.0 suppresses all audio (no audio is transcribed).
     *
     * The values increase on a monotonic curve. Specifying one or two decimal places of precision (for example, `0.55`)
     * is typically more than sufficient.
     *
     * The parameter is supported with all large speech models, next-generation models and with most previous-generation
     * models. See [Background audio
     * suppression](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-detection#detection-parameters-suppression)
     * and [Language model
     * support](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-detection#detection-support).
     * @param {boolean} [params.lowLatency] - If `true` for next-generation `Multimedia` and `Telephony` models that
     * support low latency, directs the service to produce results even more quickly than it usually does. Next-generation
     * models produce transcription results faster than previous-generation models. The `low_latency` parameter causes the
     * models to produce results even more quickly, though the results might be less accurate when the parameter is used.
     *
     * The parameter is not available for large speech models and previous-generation `Broadband` and `Narrowband` models.
     * It is available for most next-generation models.
     * * For a list of next-generation models that support low latency, see [Supported next-generation language
     * models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-ng#models-ng-supported).
     * * For more information about the `low_latency` parameter, see [Low
     * latency](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-interim#low-latency).
     * @param {number} [params.characterInsertionBias] - For large speech models and next-generation models, an indication
     * of whether the service is biased to recognize shorter or longer strings of characters when developing transcription
     * hypotheses. By default, the service is optimized to produce the best balance of strings of different lengths.
     *
     * The default bias is 0.0. The allowable range of values is -1.0 to 1.0.
     * * Negative values bias the service to favor hypotheses with shorter strings of characters.
     * * Positive values bias the service to favor hypotheses with longer strings of characters.
     *
     * As the value approaches -1.0 or 1.0, the impact of the parameter becomes more pronounced. To determine the most
     * effective value for your scenario, start by setting the value of the parameter to a small increment, such as -0.1,
     * -0.05, 0.05, or 0.1, and assess how the value impacts the transcription results. Then experiment with different
     * values as necessary, adjusting the value by small increments.
     *
     * The parameter is not available for previous-generation models.
     *
     * See [Character insertion
     * bias](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-parsing#insertion-bias).
     * @param {OutgoingHttpHeaders} [params.headers] - Custom request headers
     * @returns {Promise<SpeechToTextV1.Response<SpeechToTextV1.SpeechRecognitionResults>>}
     */
    recognize(params: SpeechToTextV1.RecognizeParams): Promise<SpeechToTextV1.Response<SpeechToTextV1.SpeechRecognitionResults>>;
    /*************************
     * asynchronous
     ************************/
    /**
     * Register a callback.
     *
     * Registers a callback URL with the service for use with subsequent asynchronous recognition requests. The service
     * attempts to register, or allowlist, the callback URL if it is not already registered by sending a `GET` request to
     * the callback URL. The service passes a random alphanumeric challenge string via the `challenge_string` parameter of
     * the request. The request includes an `Accept` header that specifies `text/plain` as the required response type.
     *
     * To be registered successfully, the callback URL must respond to the `GET` request from the service. The response
     * must send status code 200 and must include the challenge string in its body. Set the `Content-Type` response header
     * to `text/plain`. Upon receiving this response, the service responds to the original registration request with
     * response code 201.
     *
     * The service sends only a single `GET` request to the callback URL. If the service does not receive a reply with a
     * response code of 200 and a body that echoes the challenge string sent by the service within five seconds, it does
     * not allowlist the URL; it instead sends status code 400 in response to the request to register a callback. If the
     * requested callback URL is already allowlisted, the service responds to the initial registration request with
     * response code 200.
     *
     * If you specify a user secret with the request, the service uses it as a key to calculate an HMAC-SHA1 signature of
     * the challenge string in its response to the `POST` request. It sends this signature in the `X-Callback-Signature`
     * header of its `GET` request to the URL during registration. It also uses the secret to calculate a signature over
     * the payload of every callback notification that uses the URL. The signature provides authentication and data
     * integrity for HTTP communications.
     *
     * After you successfully register a callback URL, you can use it with an indefinite number of recognition requests.
     * You can register a maximum of 20 callback URLS in a one-hour span of time.
     *
     * **See also:** [Registering a callback
     * URL](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-async#register).
     *
     * @param {Object} params - The parameters to send to the service.
     * @param {string} params.callbackUrl - An HTTP or HTTPS URL to which callback notifications are to be sent. To be
     * allowlisted, the URL must successfully echo the challenge string during URL verification. During verification, the
     * client can also check the signature that the service sends in the `X-Callback-Signature` header to verify the
     * origin of the request.
     * @param {string} [params.userSecret] - A user-specified string that the service uses to generate the HMAC-SHA1
     * signature that it sends via the `X-Callback-Signature` header. The service includes the header during URL
     * verification and with every notification sent to the callback URL. It calculates the signature over the payload of
     * the notification. If you omit the parameter, the service does not send the header.
     * @param {OutgoingHttpHeaders} [params.headers] - Custom request headers
     * @returns {Promise<SpeechToTextV1.Response<SpeechToTextV1.RegisterStatus>>}
     */
    registerCallback(params: SpeechToTextV1.RegisterCallbackParams): Promise<SpeechToTextV1.Response<SpeechToTextV1.RegisterStatus>>;
    /**
     * Unregister a callback.
     *
     * Unregisters a callback URL that was previously allowlisted with a [Register a callback](#registercallback) request
     * for use with the asynchronous interface. Once unregistered, the URL can no longer be used with asynchronous
     * recognition requests.
     *
     * **See also:** [Unregistering a callback
     * URL](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-async#unregister).
     *
     * @param {Object} params - The parameters to send to the service.
     * @param {string} params.callbackUrl - The callback URL that is to be unregistered.
     * @param {OutgoingHttpHeaders} [params.headers] - Custom request headers
     * @returns {Promise<SpeechToTextV1.Response<SpeechToTextV1.EmptyObject>>}
     */
    unregisterCallback(params: SpeechToTextV1.UnregisterCallbackParams): Promise<SpeechToTextV1.Response<SpeechToTextV1.EmptyObject>>;
    /**
     * Create a job.
     *
     * Creates a job for a new asynchronous recognition request. The job is owned by the instance of the service whose
     * credentials are used to create it. How you learn the status and results of a job depends on the parameters you
     * include with the job creation request:
     * * By callback notification: Include the `callback_url` parameter to specify a URL to which the service is to send
     * callback notifications when the status of the job changes. Optionally, you can also include the `events` and
     * `user_token` parameters to subscribe to specific events and to specify a string that is to be included with each
     * notification for the job.
     * * By polling the service: Omit the `callback_url`, `events`, and `user_token` parameters. You must then use the
     * [Check jobs](#checkjobs) or [Check a job](#checkjob) methods to check the status of the job, using the latter to
     * retrieve the results when the job is complete.
     *
     * The two approaches are not mutually exclusive. You can poll the service for job status or obtain results from the
     * service manually even if you include a callback URL. In both cases, you can include the `results_ttl` parameter to
     * specify how long the results are to remain available after the job is complete. Using the HTTPS [Check a
     * job](#checkjob) method to retrieve results is more secure than receiving them via callback notification over HTTP
     * because it provides confidentiality in addition to authentication and data integrity.
     *
     * The method supports the same basic parameters as other HTTP and WebSocket recognition requests. It also supports
     * the following parameters specific to the asynchronous interface:
     * * `callback_url`
     * * `events`
     * * `user_token`
     * * `results_ttl`
     *
     * You can pass a maximum of 1 GB and a minimum of 100 bytes of audio with a request. The service automatically
     * detects the endianness of the incoming audio and, for audio that includes multiple channels, downmixes the audio to
     * one-channel mono during transcoding. The method returns only final results; to enable interim results, use the
     * WebSocket API. (With the `curl` command, use the `--data-binary` option to upload the file for the request.)
     *
     * **See also:** [Creating a job](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-async#create).
     *
     * ### Streaming mode
     *
     *  For requests to transcribe live audio as it becomes available, you must set the `Transfer-Encoding` header to
     * `chunked` to use streaming mode. In streaming mode, the service closes the connection (status code 408) if it does
     * not receive at least 15 seconds of audio (including silence) in any 30-second period. The service also closes the
     * connection (status code 400) if it detects no speech for `inactivity_timeout` seconds of streaming audio; use the
     * `inactivity_timeout` parameter to change the default of 30 seconds.
     *
     * **See also:**
     * * [Audio transmission](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-input#transmission)
     * * [Timeouts](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-input#timeouts)
     *
     * ### Audio formats (content types)
     *
     *  The service accepts audio in the following formats (MIME types).
     * * For formats that are labeled **Required**, you must use the `Content-Type` header with the request to specify the
     * format of the audio.
     * * For all other formats, you can omit the `Content-Type` header or specify `application/octet-stream` with the
     * header to have the service automatically detect the format of the audio. (With the `curl` command, you can specify
     * either `"Content-Type:"` or `"Content-Type: application/octet-stream"`.)
     *
     * Where indicated, the format that you specify must include the sampling rate and can optionally include the number
     * of channels and the endianness of the audio.
     * * `audio/alaw` (**Required.** Specify the sampling rate (`rate`) of the audio.)
     * * `audio/basic` (**Required.** Use only with narrowband models.)
     * * `audio/flac`
     * * `audio/g729` (Use only with narrowband models.)
     * * `audio/l16` (**Required.** Specify the sampling rate (`rate`) and optionally the number of channels (`channels`)
     * and endianness (`endianness`) of the audio.)
     * * `audio/mp3`
     * * `audio/mpeg`
     * * `audio/mulaw` (**Required.** Specify the sampling rate (`rate`) of the audio.)
     * * `audio/ogg` (The service automatically detects the codec of the input audio.)
     * * `audio/ogg;codecs=opus`
     * * `audio/ogg;codecs=vorbis`
     * * `audio/wav` (Provide audio with a maximum of nine channels.)
     * * `audio/webm` (The service automatically detects the codec of the input audio.)
     * * `audio/webm;codecs=opus`
     * * `audio/webm;codecs=vorbis`
     *
     * The sampling rate of the audio must match the sampling rate of the model for the recognition request: for broadband
     * models, at least 16 kHz; for narrowband models, at least 8 kHz. If the sampling rate of the audio is higher than
     * the minimum required rate, the service down-samples the audio to the appropriate rate. If the sampling rate of the
     * audio is lower than the minimum required rate, the request fails.
     *
     *  **See also:** [Supported audio
     * formats](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-audio-formats).
     *
     * ### Large speech models and Next-generation models
     *
     *  The service supports large speech models and next-generation `Multimedia` (16 kHz) and `Telephony` (8 kHz) models
     * for many languages. Large speech models and next-generation models have higher throughput than the service's
     * previous generation of `Broadband` and `Narrowband` models. When you use large speech models and next-generation
     * models, the service can return transcriptions more quickly and also provide noticeably better transcription
     * accuracy.
     *
     * You specify a large speech model or next-generation model by using the `model` query parameter, as you do a
     * previous-generation model. Only the next-generation models support the `low_latency` parameter, and all large
     * speech models and next-generation models support the `character_insertion_bias` parameter. These parameters are not
     * available with previous-generation models.
     *
     * Large speech models and next-generation models do not support all of the speech recognition parameters that are
     * available for use with previous-generation models. Next-generation models do not support the following parameters:
     * * `acoustic_customization_id`
     * * `keywords` and `keywords_threshold`
     * * `processing_metrics` and `processing_metrics_interval`
     * * `word_alternatives_threshold`
     *
     * **Important:** Effective **31 July 2023**, all previous-generation models will be removed from the service and the
     * documentation. Most previous-generation models were deprecated on 15 March 2022. You must migrate to the equivalent
     * large speech model or next-generation model by 31 July 2023. For more information, see [Migrating to large speech
     * models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-migrate).
     *
     * **See also:**
     * * [Large speech languages and
     * models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-large-speech-languages)
     * * [Supported features for large speech
     * models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-large-speech-languages#models-lsm-supported-features)
     * * [Next-generation languages and models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-ng)
     * * [Supported features for next-generation
     * models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-ng#models-ng-features).
     *
     * @param {Object} params - The parameters to send to the service.
     * @param {NodeJS.ReadableStream | Buffer} params.audio - The audio to transcribe.
     * @param {string} [params.contentType] - The format (MIME type) of the audio. For more information about specifying
     * an audio format, see **Audio formats (content types)** in the method description.
     * @param {string} [params.model] - The model to use for speech recognition. If you omit the `model` parameter, the
     * service uses the US English `en-US_BroadbandModel` by default.
     *
     * _For IBM Cloud Pak for Data,_ if you do not install the `en-US_BroadbandModel`, you must either specify a model
     * with the request or specify a new default model for your installation of the service.
     *
     * **See also:**
     * * [Using a model for speech recognition](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-use)
     * * [Using the default
     * model](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-use#models-use-default).
     * @param {string} [params.callbackUrl] - A URL to which callback notifications are to be sent. The URL must already
     * be successfully allowlisted by using the [Register a callback](#registercallback) method. You can include the same
     * callback URL with any number of job creation requests. Omit the parameter to poll the service for job completion
     * and results.
     *
     * Use the `user_token` parameter to specify a unique user-specified string with each job to differentiate the
     * callback notifications for the jobs.
     * @param {string} [params.events] - If the job includes a callback URL, a comma-separated list of notification events
     * to which to subscribe. Valid events are
     * * `recognitions.started` generates a callback notification when the service begins to process the job.
     * * `recognitions.completed` generates a callback notification when the job is complete. You must use the [Check a
     * job](#checkjob) method to retrieve the results before they time out or are deleted.
     * * `recognitions.completed_with_results` generates a callback notification when the job is complete. The
     * notification includes the results of the request.
     * * `recognitions.failed` generates a callback notification if the service experiences an error while processing the
     * job.
     *
     * The `recognitions.completed` and `recognitions.completed_with_results` events are incompatible. You can specify
     * only of the two events.
     *
     * If the job includes a callback URL, omit the parameter to subscribe to the default events: `recognitions.started`,
     * `recognitions.completed`, and `recognitions.failed`. If the job does not include a callback URL, omit the
     * parameter.
     * @param {string} [params.userToken] - If the job includes a callback URL, a user-specified string that the service
     * is to include with each callback notification for the job; the token allows the user to maintain an internal
     * mapping between jobs and notification events. If the job does not include a callback URL, omit the parameter.
     * @param {number} [params.resultsTtl] - The number of minutes for which the results are to be available after the job
     * has finished. If not delivered via a callback, the results must be retrieved within this time. Omit the parameter
     * to use a time to live of one week. The parameter is valid with or without a callback URL.
     * @param {string} [params.languageCustomizationId] - The customization ID (GUID) of a custom language model that is
     * to be used with the recognition request. The base model of the specified custom language model must match the model
     * specified with the `model` parameter. You must make the request with credentials for the instance of the service
     * that owns the custom model. By default, no custom language model is used. See [Using a custom language model for
     * speech recognition](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-languageUse).
     *
     * **Note:** Use this parameter instead of the deprecated `customization_id` parameter.
     * @param {