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use unbounded buffer for audio_stream
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livekit-rtc/livekit/rtc/audio_stream.py

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@@ -239,6 +239,7 @@ def _create_owned_stream(self) -> Any:
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new_audio_stream.track_handle = self._track._ffi_handle.handle
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new_audio_stream.sample_rate = self._sample_rate
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new_audio_stream.num_channels = self._num_channels
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new_audio_stream.queue_size_frames = 0
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if self._frame_size_ms:
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new_audio_stream.frame_size_ms = self._frame_size_ms
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new_audio_stream.type = proto_audio_frame.AudioStreamType.AUDIO_STREAM_NATIVE
@@ -257,6 +258,7 @@ def _create_owned_stream_from_participant(
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audio_stream_from_participant.participant_handle = participant._ffi_handle.handle
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audio_stream_from_participant.sample_rate = self._sample_rate
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audio_stream_from_participant.num_channels = self._num_channels
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audio_stream_from_participant.queue_size_frames = 0
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audio_stream_from_participant.type = proto_audio_frame.AudioStreamType.AUDIO_STREAM_NATIVE
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audio_stream_from_participant.track_source = track_source
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if self._frame_size_ms:

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