o $ip@sUddlmZddlZddlmZmZmZmZmZm Z ddl m Z m Z m Z ddlZddlmZddlmZmZmZmZmZmZmZmZmZddlmZmZmZmZm Z dd l!m"Z"dd l#m$Z$m%Z%dd l&m'Z'm(Z(dd l)m*Z*m+Z+dd l,m-Z-ddl.m/Z/ddl0m1Z1ddl2m3Z3ddl4m5Z5ddl6m7Z7ddl8m9Z9ddl:m;Z;ddlm?Z?ddgZ@eAdZBdeCd<Gddde$ZDGddde%ZEGdddZFGd d!d!ZGGd"d#d#ZHGd$d%d%ZId,d*d+ZJdS)-) annotationsN) TYPE_CHECKINGListUnionMappingOptionalcast)Literaloverload assert_never)_legacy_response) BodyOmitQueryHeadersNotGiven FileTypesSequenceNotStromit not_given) extract_files required_argsmaybe_transformdeepcopy_minimalasync_maybe_transform)cached_property)SyncAPIResourceAsyncAPIResource)to_streamed_response_wrapper"async_to_streamed_response_wrapper)Stream AsyncStream)transcription_create_params)make_request_options) AudioModel) Transcription)AudioResponseFormat)TranscriptionInclude)TranscriptionVerbose)TranscriptionDiarized)TranscriptionStreamEvent)TranscriptionCreateResponseTranscriptionsAsyncTranscriptionszopenai.audio.transcriptionszlogging.Loggerlogc@sXeZdZedCddZedDddZeeeeeeeeeddded dEd&d'Z eeeeeeeddded( dFd+d'Z eeeeeeeddded( dGd.d'Z eeeeeeeddded/ dHd5d'Z eeeeeeeeeeddded6 dId:d'Z eeeeeeeeeeddded6 dJd=d'Z e d d ggd>eeeeeeeeeeddded?dKdBd'Z dS)Lr-returnTranscriptionsWithRawResponsecCt|Sa This property can be used as a prefix for any HTTP method call to return the raw response object instead of the parsed content. For more information, see https://www.github.com/openai/openai-python#accessing-raw-response-data-eg-headers )r1selfr6e/var/www/html/flask_server/venv/lib/python3.10/site-packages/openai/resources/audio/transcriptions.pywith_raw_response-z Transcriptions.with_raw_response#TranscriptionsWithStreamingResponsecCr2z An alternative to `.with_raw_response` that doesn't eagerly read the response body. For more information, see https://www.github.com/openai/openai-python#with_streaming_response )r:r4r6r6r7with_streaming_response7z&Transcriptions.with_streaming_responseN) chunking_strategyincludelanguagepromptresponse_formatstream temperaturetimestamp_granularities extra_headers extra_query extra_bodytimeoutfilermodelUnion[str, AudioModel]r>=Optional[transcription_create_params.ChunkingStrategy] | Omitr?!List[TranscriptionInclude] | Omitr@ str | OmitrArBUnion[Literal['json'], Omit]rCOptional[Literal[False]] | OmitrD float | OmitrE'List[Literal['word', 'segment']] | OmitrFHeaders | NonerG Query | NonerH Body | NonerI'float | httpx.Timeout | None | NotGivenr&cCdS)a Transcribes audio into the input language. Args: file: The audio file object (not file name) to transcribe, in one of these formats: flac, mp3, mp4, mpeg, mpga, m4a, ogg, wav, or webm. model: ID of the model to use. The options are `gpt-4o-transcribe`, `gpt-4o-mini-transcribe`, and `whisper-1` (which is powered by our open source Whisper V2 model). chunking_strategy: Controls how the audio is cut into chunks. When set to `"auto"`, the server first normalizes loudness and then uses voice activity detection (VAD) to choose boundaries. `server_vad` object can be provided to tweak VAD detection parameters manually. If unset, the audio is transcribed as a single block. include: Additional information to include in the transcription response. `logprobs` will return the log probabilities of the tokens in the response to understand the model's confidence in the transcription. `logprobs` only works with response_format set to `json` and only with the models `gpt-4o-transcribe` and `gpt-4o-mini-transcribe`. language: 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) (e.g. `en`) format will improve accuracy and latency. prompt: 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. response_format: The format of the output, in one of these options: `json`, `text`, `srt`, `verbose_json`, or `vtt`. For `gpt-4o-transcribe` and `gpt-4o-mini-transcribe`, the only supported format is `json`. stream: If set to true, the model response data will be streamed to the client as it is generated using [server-sent events](https://developer.mozilla.org/en-US/docs/Web/API/Server-sent_events/Using_server-sent_events#Event_stream_format). See the [Streaming section of the Speech-to-Text guide](https://platform.openai.com/docs/guides/speech-to-text?lang=curl#streaming-transcriptions) for more information. Note: Streaming is not supported for the `whisper-1` model and will be ignored. temperature: 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. timestamp_granularities: 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. extra_headers: Send extra headers extra_query: Add additional query parameters to the request Nr6)r5rJrKr>r?r@rArBrCrDrErFrGrHrIr6r6r7create@szTranscriptions.create r>r?r@rArDrErFrGrHrILiteral['verbose_json']r)c CdSNr6r5rJrKr>r?rBr@rArDrErFrGrHrIr6r6r7rYLiteral['text', 'srt', 'vtt']strc Cr\r]r6)r5rJrKr>rBr?r@rArDrErFrGrHrIr6r6r7rYr_) r>known_speaker_namesknown_speaker_referencesr@rDrErFrGrHrILiteral['diarized_json']rbSequenceNotStr[str] | Omitrcr*c Cr\r]r6)r5rJrKr>rBrbrcr@rDrErFrGrHrIr6r6r7rYr_ r>r?rbrcr@rArBrDrErFrGrHrI Literal[True] Union[AudioResponseFormat, Omit] Stream[TranscriptionStreamEvent]cCrXaS Transcribes audio into the input language. Args: file: The audio file object (not file name) to transcribe, in one of these formats: flac, mp3, mp4, mpeg, mpga, m4a, ogg, wav, or webm. model: ID of the model to use. The options are `gpt-4o-transcribe`, `gpt-4o-mini-transcribe`, `whisper-1` (which is powered by our open source Whisper V2 model), and `gpt-4o-transcribe-diarize`. stream: If set to true, the model response data will be streamed to the client as it is generated using [server-sent events](https://developer.mozilla.org/en-US/docs/Web/API/Server-sent_events/Using_server-sent_events#Event_stream_format). See the [Streaming section of the Speech-to-Text guide](https://platform.openai.com/docs/guides/speech-to-text?lang=curl#streaming-transcriptions) for more information. Note: Streaming is not supported for the `whisper-1` model and will be ignored. chunking_strategy: Controls how the audio is cut into chunks. When set to `"auto"`, the server first normalizes loudness and then uses voice activity detection (VAD) to choose boundaries. `server_vad` object can be provided to tweak VAD detection parameters manually. If unset, the audio is transcribed as a single block. Required when using `gpt-4o-transcribe-diarize` for inputs longer than 30 seconds. include: Additional information to include in the transcription response. `logprobs` will return the log probabilities of the tokens in the response to understand the model's confidence in the transcription. `logprobs` only works with response_format set to `json` and only with the models `gpt-4o-transcribe` and `gpt-4o-mini-transcribe`. This field is not supported when using `gpt-4o-transcribe-diarize`. known_speaker_names: Optional list of speaker names that correspond to the audio samples provided in `known_speaker_references[]`. Each entry should be a short identifier (for example `customer` or `agent`). Up to 4 speakers are supported. known_speaker_references: Optional list of audio samples (as [data URLs](https://developer.mozilla.org/en-US/docs/Web/HTTP/Basics_of_HTTP/Data_URLs)) that contain known speaker references matching `known_speaker_names[]`. Each sample must be between 2 and 10 seconds, and can use any of the same input audio formats supported by `file`. language: 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) (e.g. `en`) format will improve accuracy and latency. prompt: 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. This field is not supported when using `gpt-4o-transcribe-diarize`. response_format: The format of the output, in one of these options: `json`, `text`, `srt`, `verbose_json`, `vtt`, or `diarized_json`. For `gpt-4o-transcribe` and `gpt-4o-mini-transcribe`, the only supported format is `json`. For `gpt-4o-transcribe-diarize`, the supported formats are `json`, `text`, and `diarized_json`, with `diarized_json` required to receive speaker annotations. temperature: 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. timestamp_granularities: 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. This option is not available for `gpt-4o-transcribe-diarize`. extra_headers: Send extra headers extra_query: Add additional query parameters to the request extra_body: Add additional JSON properties to the request timeout: Override the client-level default timeout for this request, in seconds Nr6r5rJrKrCr>r?rbrcr@rArBrDrErFrGrHrIr6r6r7rYibool>TranscriptionCreateResponse | Stream[TranscriptionStreamEvent]cCrXrjr6rkr6r6r7rY=rlrJrKrCr>r?rbrcr@rArBrCrDrErFrGrHrI/Optional[Literal[False]] | Literal[True] | Omitestr | Transcription | TranscriptionDiarized | TranscriptionVerbose | Stream[TranscriptionStreamEvent]cCst||||||||| | | | d }tttttf|dggd}ddi| p&i} |jdt|| r2tj ntj |t | |||dt | | pCdt td S N) rJrKr>r?rbrcr@rArBrCrDrErJ)pathsz Content-Typezmultipart/form-dataz/audio/transcriptions)rFrGrHrIF)bodyfilesoptionscast_torC stream_cls)rrrrraobject_postrr#"TranscriptionCreateParamsStreaming%TranscriptionCreateParamsNonStreamingr$_get_response_format_typer!r+r5rJrKr>r?rbrcr@rArBrCrDrErFrGrHrIrurvr6r6r7rYsB)r0r1)r0r:)rJrrKrLr>rMr?rNr@rOrArOrBrPrCrQrDrRrErSrFrTrGrUrHrVrIrWr0r&rJrrKrLr>rMr?rNrBr[r@rOrArOrDrRrErSrFrTrGrUrHrVrIrWr0r))rJrrKrLr>rMrBr`r?rNr@rOrArOrDrRrErSrFrTrGrUrHrVrIrWr0ra)rJrrKrLr>rMrBrdrbrercrer@rOrDrRrErSrFrTrGrUrHrVrIrWr0r*)"rJrrKrLrCrgr>rMr?rNrbrercrer@rOrArOrBrhrDrRrErSrFrTrGrUrHrVrIrWr0ri)"rJrrKrLrCrmr>rMr?rNrbrercrer@rOrArOrBrhrDrRrErSrFrTrGrUrHrVrIrWr0rn)"rJrrKrLr>rMr?rNrbrercrer@rOrArOrBrhrCrqrDrRrErSrFrTrGrUrHrVrIrWr0rr __name__ __module__ __qualname__rr8r<r rrrYrr6r6r6r7r-,s  Rjjc@s6eZdZed>ddZed?ddZeeeeeeeeeeeddded d@d)d*Z eeeeeeeddded+ dAd.d*Z eeeeeeeddded+ dBd1d*Z eeeeeeeeeeddded2 dCd6d*Z eeeeeeeeeeddded2 dDd9d*Z e d d ggd:eeeeeeeeeeddded dEd=d*Z dS)Fr.r0"AsyncTranscriptionsWithRawResponsecCr2r3)rr4r6r6r7r8r9z%AsyncTranscriptions.with_raw_response(AsyncTranscriptionsWithStreamingResponsecCr2r;)rr4r6r6r7r<r=z+AsyncTranscriptions.with_streaming_responseNrprJrrKrLr>rMr?rNrbrercr@rOrArBrPrCrQrDrRrErSrFrTrGrUrHrVrIrWr,cdS)a Transcribes audio into the input language. Args: file: The audio file object (not file name) to transcribe, in one of these formats: flac, mp3, mp4, mpeg, mpga, m4a, ogg, wav, or webm. model: ID of the model to use. The options are `gpt-4o-transcribe`, `gpt-4o-mini-transcribe`, `whisper-1` (which is powered by our open source Whisper V2 model), and `gpt-4o-transcribe-diarize`. chunking_strategy: Controls how the audio is cut into chunks. When set to `"auto"`, the server first normalizes loudness and then uses voice activity detection (VAD) to choose boundaries. `server_vad` object can be provided to tweak VAD detection parameters manually. If unset, the audio is transcribed as a single block. Required when using `gpt-4o-transcribe-diarize` for inputs longer than 30 seconds. include: Additional information to include in the transcription response. `logprobs` will return the log probabilities of the tokens in the response to understand the model's confidence in the transcription. `logprobs` only works with response_format set to `json` and only with the models `gpt-4o-transcribe` and `gpt-4o-mini-transcribe`. This field is not supported when using `gpt-4o-transcribe-diarize`. known_speaker_names: Optional list of speaker names that correspond to the audio samples provided in `known_speaker_references[]`. Each entry should be a short identifier (for example `customer` or `agent`). Up to 4 speakers are supported. known_speaker_references: Optional list of audio samples (as [data URLs](https://developer.mozilla.org/en-US/docs/Web/HTTP/Basics_of_HTTP/Data_URLs)) that contain known speaker references matching `known_speaker_names[]`. Each sample must be between 2 and 10 seconds, and can use any of the same input audio formats supported by `file`. language: 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) (e.g. `en`) format will improve accuracy and latency. prompt: 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. This field is not supported when using `gpt-4o-transcribe-diarize`. response_format: The format of the output, in one of these options: `json`, `text`, `srt`, `verbose_json`, `vtt`, or `diarized_json`. For `gpt-4o-transcribe` and `gpt-4o-mini-transcribe`, the only supported format is `json`. For `gpt-4o-transcribe-diarize`, the supported formats are `json`, `text`, and `diarized_json`, with `diarized_json` required to receive speaker annotations. stream: If set to true, the model response data will be streamed to the client as it is generated using [server-sent events](https://developer.mozilla.org/en-US/docs/Web/API/Server-sent_events/Using_server-sent_events#Event_stream_format). See the [Streaming section of the Speech-to-Text guide](https://platform.openai.com/docs/guides/speech-to-text?lang=curl#streaming-transcriptions) for more information. Note: Streaming is not supported for the `whisper-1` model and will be ignored. temperature: 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. timestamp_granularities: 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. This option is not available for `gpt-4o-transcribe-diarize`. extra_headers: Send extra headers extra_query: Add additional query parameters to the request Nr6)r5rJrKr>r?rbrcr@rArBrCrDrErFrGrHrIr6r6r7rYszAsyncTranscriptions.createrZr[r)c dSr]r6r^r6r6r7rY`r`rac rr]r6r^r6r6r7rYurrfrgrh%AsyncStream[TranscriptionStreamEvent]crrjr6rkr6r6r7rYirmCTranscriptionCreateResponse | AsyncStream[TranscriptionStreamEvent]crrjr6rkr6r6r7rYrrorqjTranscription | TranscriptionVerbose | TranscriptionDiarized | str | AsyncStream[TranscriptionStreamEvent]cst||||||||| | | | d }tttttf|dggd}ddi| p'i} |jdt|| r3tj ntj IdH|t | |||dt | | pGdt td IdHSrs)rrrrrarzr{rr#r|r}r$r~r"r+rr6r6r7rY`sD )r0r)r0r)"rJrrKrLr>rMr?rNrbrercrer@rOrArOrBrPrCrQrDrRrErSrFrTrGrUrHrVrIrWr0r,r)rJrrKrLr>rMr?rNrBr`r@rOrArOrDrRrErSrFrTrGrUrHrVrIrWr0ra)"rJrrKrLrCrgr>rMr?rNrbrercrer@rOrArOrBrhrDrRrErSrFrTrGrUrHrVrIrWr0r)"rJrrKrLrCrmr>rMr?rNrbrercrer@rOrArOrBrhrDrRrErSrFrTrGrUrHrVrIrWr0r)"rJrrKrLr>rMr?rNrbrercrer@rOrArOrBrhrCrqrDrRrErSrFrTrGrUrHrVrIrWr0rrr6r6r6r7r.s  ejjc@eZdZdddZdS) r1transcriptionsr-r0NonecC||_t|j|_dSr])_transcriptionsr to_raw_response_wrapperrYr5rr6r6r7__init__ z&TranscriptionsWithRawResponse.__init__Nrr-r0rrrrrr6r6r6r7r1r1c@r) rrr.r0rcCrr])rr async_to_raw_response_wrapperrYrr6r6r7rrz+AsyncTranscriptionsWithRawResponse.__init__Nrr.r0rrr6r6r6r7rrrc@r) r:rr-r0rcC||_t|j|_dSr])rrrYrr6r6r7r z,TranscriptionsWithStreamingResponse.__init__Nrrr6r6r6r7r:rr:c@r) rrr.r0rcCrr])rr rYrr6r6r7rrz1AsyncTranscriptionsWithStreamingResponse.__init__Nrrr6r6r6r7rrrrBAudioResponseFormat | Omitr0Htype[Transcription | TranscriptionVerbose | TranscriptionDiarized | str]cCsvt|ts |dur tS|dkrtS|dkrtS|dkrtS|dks)|dks)|dkr+tStr3t|dSt d|tS)Njson verbose_json diarized_jsonsrttextvttz$Unexpected audio response format: %s) isinstancerr&r)r*rarr r/warn)rBr6r6r7r~s  r~)rBrr0r)K __future__rloggingtypingrrrrrrtyping_extensionsr r r httpxr _typesrrrrrrrrr_utilsrrrrr_compatr _resourcerr _responserr _streamingr!r" types.audior# _base_clientr$types.audio_modelr%types.audio.transcriptionr&types.audio_response_formatr'!types.audio.transcription_includer(!types.audio.transcription_verboser)"types.audio.transcription_diarizedr*&types.audio.transcription_stream_eventr+)types.audio.transcription_create_responser,__all__ getLoggerr/__annotations__r-r.r1rr:rr~r6r6r6r7sJ  ,           =;