+ ~j-*^RIHt^RIt^RIt^RIHt^RIt^RIt^RI H t ^RI H t ^RI Ht^RIHt]] P$,] P$,t] P(t] P(tRR lt!R R 4tRR R llt!RR4tR#)) annotationsN)AsyncGenerator) DTypeLike)rtc)logger)cancel_and_waitc V^8dQhRRRR/#)frames AudioBufferreturnfloat)formats"o/Users/mitch_tango/dev/rabbit-r1-livekit/agent/.venv/lib/python3.14/site-packages/livekit/agents/utils/audio.py __annotate__rs[Ucj\V\4'd\RV44#VP#)a Calculate the total duration of audio frames. This function computes the total duration of audio frames in seconds. It accepts either a list of `rtc.AudioFrame` objects or a single `rtc.AudioFrame` object. Parameters: - frames (AudioBuffer): A list of `rtc.AudioFrame` instances or a single `rtc.AudioFrame` instance. Returns: - float: The total duration in seconds of all frames provided. c38"TFqPxK R#5iN)duration).0frames& r +calculate_audio_duration..%s6ve>>vs) isinstancelistsumr)r s&rcalculate_audio_durationrs,&$6v666rcV]tRt^*tRt^tR RRlltRRlt]tRRlt R R lt R t R#) AudioByteStreamuBuffer and chunk audio byte data into fixed-size frames. Accepts variable-sized byte chunks (e.g. from a network stream or file) and emits consistently-sized ``rtc.AudioFrame`` objects. Two modes of operation: * **Fixed** (``progressive=False``, the default): every emitted frame is exactly ``samples_per_channel`` samples long. * **Progressive** (``progressive=True``): the *first* emitted frame is only 20 ms of audio. Each subsequent frame doubles in size until ``samples_per_channel`` is reached. This minimises time-to-first-audio while giving the pipeline a brief warm-up before reaching full frame sizes. Example with ``sample_rate=16000, samples_per_channel=3200`` (200 ms) and ``progressive=True``:: Frame 1: 320 samples ( 20 ms) Frame 2: 640 samples ( 40 ms) Frame 3: 1280 samples ( 80 ms) Frame 4: 2560 samples (160 ms) Frame 5: 3200 samples (200 ms) ← target reached Frame 6: 3200 samples (200 ms) ... Nc ,V^8dQhRRRRRRRRRR /#) r sample_rateint num_channelssamples_per_channelz int | None progressiveboolr Noner)rs"rrAudioByteStream.__annotate__HsC"F"F"F"F( "F  "F  "Frc WnW nVf V^ ,pV\P!\P4,VnW0P ,Vn\4VnV'dFWP,R,p\WPP ,VP 4Vn MVP Vn VPVn R#)a Args: sample_rate: Audio sample rate in Hz. num_channels: Number of audio channels. samples_per_channel: Target samples per channel in each emitted frame. Defaults to ``sample_rate // 10`` (100 ms). progressive: When *True*, start with a small 20 ms frame and double the frame size on each subsequent emission until ``samples_per_channel`` is reached. Ni) _sample_rate _num_channelsctypessizeofc_int16_bytes_per_sample_target_bytes_per_frame bytearray_buf_MIN_PROGRESSIVE_MSmin_initial_bytes_per_frame_current_bytes_per_frame)selfr#r%r&r' min_sampless&&&&& r__init__AudioByteStream.__init__Hs"()  &"-"3 !- fnn0M!M':=S=S'S$K %(@(@@DHK,/444d6R6R-D )-1,H,HD )(,(E(E%rc V^8dQhRRRR/#)r datazbytes | memoryviewr list[rtc.AudioFrame]r)rs"rrr*ls))+)0D)rc  0VPPV4.p\VP4VP8dVPRVPpVPRVP1VP \ P !VVPVP\V4VP,R74VPVP8gK\VP^,VP4VnKV#)a{ Add audio data to the buffer and retrieve fixed-size frames. Parameters: data (bytes): The incoming audio data to buffer. Returns: list[rtc.AudioFrame]: A list of `AudioFrame` objects of fixed size. The method appends the incoming data to the internal buffer. While the buffer contains enough data to form complete frames, it extracts the data for each frame, creates an `AudioFrame` object, and appends it to the list of frames to return. This allows you to feed in variable-sized chunks of audio data (e.g., from a stream or file) and receive back a list of fixed-size audio frames ready for processing or transmission. Nr>r#r%r&) r4extendlenr8appendr AudioFramer,r-r1r2r6)r9r>r frame_datas&& rpushAudioByteStream.pushls& $))n = ==#BT%B%BCJ 9D999: MM# $ 1 1!%!3!3(+J4;Q;Q(Q  ,,t/K/KK0311A5t7S7S1- rcV^8dQhRR/#)r r r?r)rs"rrr*s  + rc \VP4^8Xd.#\VP4VP,^8wd\P!R4.#\ P !VPP4VPVP\VP4VP,R7.pVPP4V#)a^ Flush the buffer and retrieve any remaining audio data as a frame. Returns: list[rtc.AudioFrame]: A list containing any remaining `AudioFrame` objects. This method processes any remaining data in the buffer that does not fill a complete frame. If the remaining data forms a partial frame (i.e., its size is not a multiple of the expected sample size), a warning is logged and an empty list is returned. Otherwise, it returns the final `AudioFrame` containing the remaining data. Use this method when you have no more data to push and want to ensure that all buffered audio data has been processed. z8AudioByteStream: incomplete frame during flush, droppingrA) rCr4r1rwarningrrEcopyr,r-clear)r9r s& rflushAudioByteStream.flushs tyy>Q I tyy>D22 2a 7 NNU VI NNYY^^% --!//$' Nd6L6L$L     rcV^8dQhRR/#r r r)r)rs"rrr*sFFtFrc \VPP4VPVnR#)zDiscard all buffered data and reset progressive frame sizing. After clearing, the next :meth:`push` will start from the initial (small) frame size again, ensuring low latency on the first frame after an interruption. N)r4rMr7r8r9s&rrMAudioByteStream.clears  (,(E(E%r)r4r1r8r7r-r,r2)NF) __name__ __module__ __qualname____firstlineno____doc__r5r;rGwriterNrM__static_attributes__rrrr!r!*s56"FH)V E DFFrr!c(V^8dQhRRRRRRRR/#)r file_pathstrr#r$r%r z$AsyncGenerator[rtc.AudioFrame, None]r)rs"rrrs,(((!$(.__annotate__s   t  rc~<"\P!SRR7;_uu_4GRjxL pVPR4GRjxL pV'gMSPV4K6RRR4GRjxL SP 4R#L\LDL +GRjxL 'giL1;i SP 4i;i5i)rb)modeNi)aiofilesopenreadrG end_input)fchunkdecoderr]s r file_reader+audio_frames_from_file..file_readers }}YT:::a"#&&,.E LL' ;:    ;.;:::    sB=$B(BB(B B B B $ B(/B 0B(4B=B(B  B( B% B B% B% B((B::B=N) codecsr`asyncio create_taskraclosedone cancelled exception) r]r#r%r`rm reader_taskrexcrls f&& @raudio_frames_from_filerxs+ [TG   %%km4K"  %K 7k***nnk***nn+"7"7"9"9'') )3 )I *#:s5C=A?AA A A? AA?C=A" C=8A;9C=?B.BB.'B*(B..C=C=C=5C=c~]tRt^tR]P RR/RRlltRRltRR ltR R lt R R lt RRlt Rt R#)AudioArrayBufferdtyper#i>c$V^8dQhRRRRRR/#)r buffer_sizer$r{rr#r)rs"rrAudioArrayBuffer.__annotate__s" ( (s (9 (VY (rc ~WnW n\P!WR7Vn^VnRVnW0nR#)zCreate a fixed-size buffer for audio array data. Args: buffer_size: The size of the buffer in samples. dtype: The dtype of the buffer. sample_rate: The sample rate of the buffer. )r{N) _buffer_size_dtypenpzeros_buffer _start_idx _resamplerr,)r9r}r{r#s&$$$rr;AudioArrayBuffer.__init__s3( xx 9 59'rc V^8dQhRRRR/#)r rzrtc.AudioFramer r$r)rs"rrr~s.).).)3.)rc VPVP8d \R4h.pVPfqVPVP 8wdV\ P!VPVP VP\ PPR7VnVP'd\VPVPP8wd \R4hVPVPPV44MVPV4\V4pVP VP,VP, ;p^8dVP#V4VP$VP VP VP,pVP^8d\&P(!VP*\&P,VPVP,R7P/RVP4p\&P0!V^\&P2R7VP,P5\&P,4VR&M>\&P(!VP*\&P,VPR7VR&V;P VP, unVP#)zPush an audio frame to the buffer. Args: frame: The audio frame to push. Returns: The number of samples written to the buffer. Raises: ValueError: If the frame samples are greater than the buffer size. z.frame samples are greater than the buffer size) input_rate output_rater%qualityz#frame sample rates are inconsistent)r{count)axisr{:NNN)r&r ValueErrorrr#r,rAudioResamplerr%AudioResamplerQualityQUICK _input_raterBrGrD merge_framesrshiftrr frombufferr>int16reshaperint32astype)r9rr shift_sizeptrarr_i16s&& r push_frameAudioArrayBuffer.push_frames   $ $t'8'8 8MN N') ?? "u'8'8Dstrc J^VnVPP^4R#)rN)rrfillrSs&rresetAudioArrayBuffer.reset>s !rcV^8dQhRR/#)r r r$r)rs"rrr~Bsrc VP#r)rrSs&r__len__AudioArrayBuffer.__len__Bs r)rrrrr,rN) rUrVrWrXrrr;rrrhrrr[rrrrzrzs9 (rxx (\a (.)` 6rrz)i) __future__rrpr.collections.abcrrfnumpyr numpy.typingrlivekitrlogr aio.utilsrrrEr combine_audio_framescombine_framesrrr!rxrzrrrrsx" *"&3>>"S^^3 ))'' &YFYFx(VRRr