v0.6.0
Added
-
Added
punctuate
andendpointing
fields toTranscriptionSettings
. -
Added dialout support with
CallClient.start_dialout()
andCallClient.stop_dialout()
. -
Added completion callbacks to
VirtualMicrophone.write_frames()
andVirtualSpeaker.read_frames()
. This change makes virtual devices completely asynchronous if they are created withnon_blocking
set toTrue
.
Changed
- Renamed
session_id
field toparticipantId
inTranscriptionMessage
.
Removed
- Removed
is_final
,user_id
anduser_name
fields fromTranscriptionMessage
.
Fixed
-
Room deletion messages from the server are now properly handled.
-
CallClient.send_app_message(None)
now properly triggers aValueError
exception. -
If an invalid participant ID is passed to
CallClient.send_app_message()
it will now trigger aValueError
exception. -
Fixed an issue that would cause audio crackling and popping when using non-blocking devices.
-
Fixed support for different audio sample rates and number of channels, other than 16000 and 1 channel.
-
Don't quote the participant ID when passing the string to video/audio renderer callbacks.
-
Fixed a potential crash on shutdown when using a virtual camera device.
-
Emit
transcription-started
event if transcription is already started when joining the room.
Other
- Added GStreamer media player demo.