b6i#ddlZddlZddlZddlZddlmZej rddlmZGdde eZ GddZ y)N)Enum)ClientConnectionc\eZdZdZdZdZdZdZdZdZ dZ d Z d Z d Z d Zd ZdZdZdZdZdZdZdZy)RealtimeEventsz(Events emitted by the RealtimeConnectionopenclosesession_startedpartial_transcriptcommitted_transcript$committed_transcript_with_timestampserror auth_errorquota_exceededcommit_throttledtranscriber_errorunaccepted_terms_error rate_limited input_errorqueue_overflowresource_exhaustedsession_time_limit_exceededchunk_size_exceededinsufficient_audio_activityN)__name__ __module__ __qualname____doc__OPENCLOSESESSION_STARTEDPARTIAL_TRANSCRIPTCOMMITTED_TRANSCRIPT$COMMITTED_TRANSCRIPT_WITH_TIMESTAMPSERROR AUTH_ERRORQUOTA_EXCEEDEDCOMMIT_THROTTLEDTRANSCRIBER_ERRORUNACCEPTED_TERMS_ERROR RATE_LIMITED INPUT_ERRORQUEUE_OVERFLOWRESOURCE_EXHAUSTEDSESSION_TIME_LIMIT_EXCEEDEDCHUNK_SIZE_EXCEEDEDINSUFFICIENT_AUDIO_ACTIVITYi/root/.openclaw/workspace/visionaryfx/venv/lib/python3.12/site-packages/elevenlabs/realtime/connection.pyrr sn2 D E'O-1+Q( EJ%N)+5!LK%N-"?/"?r2rceZdZdZddddedej ejfdZ de d ejd dfd Z de d dfd Z dd Zdeje ej fd dfdZddZddZddZy)RealtimeConnectiona  A WebSocket connection for real-time speech-to-text transcription. This class handles bidirectional WebSocket communication with the ElevenLabs speech-to-text API, managing audio streaming and receiving transcription results. Example: ```python connection = await client.speech_to_text.realtime.connect({ "audio_format": AudioFormat.PCM_16000, "sample_rate": 16000 }) connection.on(RealtimeEvents.PARTIAL_TRANSCRIPT, lambda data: print(data)) connection.on(RealtimeEvents.COMMITTED_TRANSCRIPT, lambda data: print(data)) # Send audio connection.send({"audioBase64": audio_chunk}) # When done connection.commit() await connection.close() ``` N websocketrcurrent_sample_rateffmpeg_processcJ||_||_||_i|_d|_yN)r6r7r8_event_handlers _message_task)selfr6r7r8s r3__init__zRealtimeConnection.__init__=s)"#6 ,OQ<@r2eventcallbackreturncz||jvrg|j|<|j|j|y)a Register an event handler for a specific event type. Args: event: The event type to listen for (from RealtimeEvents enum) callback: The function to call when the event occurs Example: ```python def handle_transcript(data): print(f"Transcript: {data['transcript']}") connection.on(RealtimeEvents.PARTIAL_TRANSCRIPT, handle_transcript) ``` N)r;append)r=r?r@s r3onzRealtimeConnection.onDs; ,, ,*,D  ' U#**84r2c||jvr|j|D]} || yy#t$r}td|d|Yd}~,d}~wwxYw)z(Emit an event to all registered handlerszError in event handler for z: N)r; Exceptionprint)r=r?argshandleres r3_emitzRealtimeConnection._emitXsk D(( (//6 FFTN F )!F7wbDEEFs+ AA  Ac 6K |j23d{} tj|}|jd} t |}|j ||tj tjtjtjtjtjtjtjtjtjtj tj"h }||vr |j tj$|*7&#t&$rYwxYw#tj($r0}|j tj$dd|iYd}~zd}~wwxYw6nA#t*$r5}|j tj$dt-|iYd}~nd}~wwxYw|j tj.y#|j tj.wxYww)z*Start handling incoming WebSocket messagesN message_typer zFailed to parse message: )r6jsonloadsgetrrKr%r&r'r(r)r*r+r,r-r.r/r0r$ ValueErrorJSONDecodeErrorrFstrr)r=messagedatarMr? error_eventsrJs r3_start_message_handlerz)RealtimeConnection._start_message_handleras$ -!% a aga::g.D#'88N#;L .| < 5$/+55*99*;;*<<*AA*77*66*99*==*FF*>>*FF ( !L0 JJ~';';TB5 a6&++aJJ~33gAZ[\Z]?^5_``a="0@ @ JJ~++gs1v-> ? ? @ JJ~++ ,DJJ~++ ,sH FFD:FF&E C7D=8F:F= E E E  E  F%F F FFG5 G+G G5 GG5 H5!HHrUc K|js tdd|jddd|j|jdd}|jj t j |d{y7w) a Send an audio chunk to the server for transcription. Args: data: Dictionary containing the following keys: - audio_base_64 (str): Base64-encoded audio data to transcribe - previous_text (str, optional): Previous transcript text to provide context for more accurate transcription Raises: RuntimeError: If the WebSocket connection is not open Example: ```python # Send audio chunk connection.send({ "audio_base_64": base64_encoded_audio }) # Send audio chunk with context - can only be sent with the first chunk of audio connection.send({ "audio_base_64": base64_encoded_audio, "previous_text": "Previously transcribed text for context" }) ``` WebSocket is not connectedinput_audio_chunk audio_base_64F previous_text)rMr[commit sample_rater]N)r6 RuntimeErrorrPr7sendrNdumps)r=rUrTs r3razRealtimeConnection.sendsp6~~;< <0!XXor:33!XXo6  nn!!$**W"5666sA:B<B=BcK|js tdddd|jd}|jjt j |d{y7w)a Commits the segment, triggering a COMMITTED_TRANSCRIPT event and clearing the buffer. It's recommend to commit often when using CommitStrategy.MANUAL to keep latency low. Raises: RuntimeError: If the WebSocket connection is not open Remarks: Only needed when using CommitStrategy.MANUAL. When using CommitStrategy.VAD, commits are handled automatically by the server. Example: ```python # Send all audio chunks for chunk in audio_chunks: connection.send({"audioBase64": chunk}) # Commit the audio segment await connection.commit() ``` rYrZr\T)rMr[r^r_N)r6r`r7rarNrb)r=rTs r3r^zRealtimeConnection.commitsY,~~;< <033  nn!!$**W"5666sAA$A"A$cK|jd{|jr$|jjddd{|jrK|jj s0|jj  |jd{yyy77^7 #t j$rYywxYww)aG Closes the WebSocket connection and cleans up resources. This will terminate any ongoing transcription and stop ffmpeg processes if running. Remarks: After calling close(), this connection cannot be reused. Create a new connection if you need to start transcribing again. Example: ```python connection.on(RealtimeEvents.COMMITTED_TRANSCRIPT, async lambda data: ( print("Committed:", data["transcript"]), await connection.close() )) ``` NizUser ended conversation)_cleanupr6rr<donecancelasyncioCancelledErrorr=s r3rzRealtimeConnection.closes"mmo >>..&&t-FG G G   d&8&8&=&=&?    % % ' (((('@   G)))  sXCB#/CB%AC B)B'B) C%C'B))B?<C>B??CcK|jr?|jj |jjdd|_yy#tj$r|jjY8wxYww)z(Clean up resources like ffmpeg processes)timeoutN)r8killwait subprocessTimeoutExpiredrjs r3rezRealtimeConnection._cleanupst       $ $ & +##(((3#'D  ,, +##((* +s('BA B-A?<B>A??Br:)rAN)rrrrinttypingOptionalrpPopenr>rSCallablerDrKrWDictAnyrar^rrer1r2r3r5r5#s2A"4A3A`f`o`opzqAqAaBA55v545(F3F$F&-P&7v{{3 ?;&7&7P 7D6'r2r5) rhrNrprsenumr TYPE_CHECKINGwebsockets.asyncio.clientrrSrr5r1r2r3r|s<   :@S$@0S'S'r2