v0.33.0

Added

• Added support to automatically start transcription when joining a room if the auto_start_transcription meeting token property is set, using the room's auto_transcription_settings.

• Added candidate-pair round-trip time to call metrics, so latency is measured consistently, including for receive-only participants.

Changed

• Upgraded to pyo3 0.29.2.

• Upgraded libwebrtc to branch-heads/8010 (M153). Its vendored media and crypto dependencies (ffmpeg, libvpx and BoringSSL among them) are substantially newer.

• ⚠️ Breaking change ⚠️: macOS wheels now require macOS 13.0 or later (previously 10.15), following the deployment target libwebrtc 8010 is built against. Linux wheels are unaffected: they continue to require glibc 2.28.

Fixed

• Fixed an issue where videoBitrate and audioBitrate in a meeting token's start_cloud_recording_opts were ignored, causing recordings started from the token to use the default bitrates.

• Fixed a potential crash in Daily.deinit() when a joined CallClient had been released without leave() being called first: the call's peer connection and microphone stayed alive and were used after deinit() had destroyed the WebRTC threads.

• Fixed a rare deadlock that could freeze the process while a CallClient was being released shortly after another one.

• Fixed a native memory leak that grew with the length of a call, about 0.7 MB per minute with one published and one subscribed track. Processes that run many calls, such as bots, no longer grow by several MB per call.