// File generated from our OpenAPI spec by Stainless. See CONTRIBUTING.md for details.

import { APIResource } from '../../resource';
import * as Core from '../../core';
import * as AudioAPI from './audio';

export class Transcriptions extends APIResource {
  /**
   * Transcribes audio into the input language.
   */
  create(
    body: TranscriptionCreateParams<'json' | undefined>,
    options?: Core.RequestOptions,
  ): Core.APIPromise<Transcription>;
  create(
    body: TranscriptionCreateParams<'verbose_json'>,
    options?: Core.RequestOptions,
  ): Core.APIPromise<TranscriptionVerbose>;
  create(
    body: TranscriptionCreateParams<'srt' | 'vtt' | 'text'>,
    options?: Core.RequestOptions,
  ): Core.APIPromise<string>;
  create(body: TranscriptionCreateParams, options?: Core.RequestOptions): Core.APIPromise<Transcription>;
  create(
    body: TranscriptionCreateParams,
    options?: Core.RequestOptions,
  ): Core.APIPromise<TranscriptionCreateResponse | string> {
    return this._client.post('/audio/transcriptions', Core.multipartFormRequestOptions({ body, ...options }));
  }
}

/**
 * Represents a transcription response returned by model, based on the provided
 * input.
 */
export interface Transcription {
  /**
   * The transcribed text.
   */
  text: string;
}

export interface TranscriptionSegment {
  /**
   * Unique identifier of the segment.
   */
  id: number;

  /**
   * Average logprob of the segment. If the value is lower than -1, consider the
   * logprobs failed.
   */
  avg_logprob: number;

  /**
   * Compression ratio of the segment. If the value is greater than 2.4, consider the
   * compression failed.
   */
  compression_ratio: number;

  /**
   * End time of the segment in seconds.
   */
  end: number;

  /**
   * Probability of no speech in the segment. If the value is higher than 1.0 and the
   * `avg_logprob` is below -1, consider this segment silent.
   */
  no_speech_prob: number;

  /**
   * Seek offset of the segment.
   */
  seek: number;

  /**
   * Start time of the segment in seconds.
   */
  start: number;

  /**
   * Temperature parameter used for generating the segment.
   */
  temperature: number;

  /**
   * Text content of the segment.
   */
  text: string;

  /**
   * Array of token IDs for the text content.
   */
  tokens: Array<number>;
}

/**
 * Represents a verbose json transcription response returned by model, based on the
 * provided input.
 */
export interface TranscriptionVerbose {
  /**
   * The duration of the input audio.
   */
  duration: string;

  /**
   * The language of the input audio.
   */
  language: string;

  /**
   * The transcribed text.
   */
  text: string;

  /**
   * Segments of the transcribed text and their corresponding details.
   */
  segments?: Array<TranscriptionSegment>;

  /**
   * Extracted words and their corresponding timestamps.
   */
  words?: Array<TranscriptionWord>;
}

export interface TranscriptionWord {
  /**
   * End time of the word in seconds.
   */
  end: number;

  /**
   * Start time of the word in seconds.
   */
  start: number;

  /**
   * The text content of the word.
   */
  word: string;
}

/**
 * Represents a transcription response returned by model, based on the provided
 * input.
 */
export type TranscriptionCreateResponse = Transcription | TranscriptionVerbose;

export interface TranscriptionCreateParams<
  ResponseFormat extends AudioAPI.AudioResponseFormat | undefined = AudioAPI.AudioResponseFormat | undefined,
> {
  /**
   * The audio file object (not file name) to transcribe, in one of these formats:
   * flac, mp3, mp4, mpeg, mpga, m4a, ogg, wav, or webm.
   */
  file: Core.Uploadable;

  /**
   * ID of the model to use. Only `whisper-1` (which is powered by our open source
   * Whisper V2 model) is currently available.
   */
  model: (string & {}) | AudioAPI.AudioModel;

  /**
   * The language of the input audio. Supplying the input language in
   * [ISO-639-1](https://en.wikipedia.org/wiki/List_of_ISO_639-1_codes) format will
   * improve accuracy and latency.
   */
  language?: string;

  /**
   * An optional text to guide the model's style or continue a previous audio
   * segment. The
   * [prompt](https://platform.openai.com/docs/guides/speech-to-text/prompting)
   * should match the audio language.
   */
  prompt?: string;

  /**
   * The format of the output, in one of these options: `json`, `text`, `srt`,
   * `verbose_json`, or `vtt`.
   */
  response_format?: ResponseFormat;

  /**
   * The sampling temperature, between 0 and 1. Higher values like 0.8 will make the
   * output more random, while lower values like 0.2 will make it more focused and
   * deterministic. If set to 0, the model will use
   * [log probability](https://en.wikipedia.org/wiki/Log_probability) to
   * automatically increase the temperature until certain thresholds are hit.
   */
  temperature?: number;

  /**
   * The timestamp granularities to populate for this transcription.
   * `response_format` must be set `verbose_json` to use timestamp granularities.
   * Either or both of these options are supported: `word`, or `segment`. Note: There
   * is no additional latency for segment timestamps, but generating word timestamps
   * incurs additional latency.
   */
  timestamp_granularities?: Array<'word' | 'segment'>;
}

export declare namespace Transcriptions {
  export {
    type Transcription as Transcription,
    type TranscriptionSegment as TranscriptionSegment,
    type TranscriptionVerbose as TranscriptionVerbose,
    type TranscriptionWord as TranscriptionWord,
    type TranscriptionCreateResponse as TranscriptionCreateResponse,
    type TranscriptionCreateParams as TranscriptionCreateParams,
  };
}
