v0.19.9

Fixed

• Fixed an issue where custom audio tracks won't send silence from the very beginning. This is necessary to avoid issues with browsers that expect RTP packets from the start.

v0.19.8

Fixed

• Fixed an issue where the application could keep attempting to send messages after the signaling channel was closed preventing the application to finish.

v0.19.7

Added

• Added callerId field to CallClient.sip_call_transfer() properties.

v0.19.6

Fixed

• Fixed a race condition that would cause audio to never be processed when received.

v0.19.5

Added

• It is now possible to set a logging interval when creating an audio or video renderer. This is useful to check if the renderers are being called correctly (specially on startup). By default, the interval is set to 10 seconds but can be changed using the logging_interval_ms argument. Note that these logs are only visible when setting the environment variable RUST_LOG=debug.

Fixed

• Fixed an issue where the newly introduce Daily.set_log_level() was only setting the log level in debug builds.

v0.19.4

Added

• Added Daily.set_log_level() as well as a new log_level argument to Daily.init(). The available log levels are: LogLevel.Off, LogLevel.Error, LogLevel.Warn, LogLevel.Info, LogLevel.Debug, LogLevel.Trace.

v0.19.3

Added

• Added option to ignore audio level when creating a custom audio track.

v0.19.2

Fixed

• Fixed an issue where an audio source could cause a segmentation fault on application exit if audio was still being written.

v0.19.1

Fixed

• Fixed an audio renderer issue that would cause garbled audio if remote audio was stereo.

• Handle meeting session ID changes which occur once the session has started.

v0.19.0

Added

• Added a new CustomAudioTrack. This new track can be used as an additional custom audio track (i.e. with custom names) or as the main microphone track.

audio_source = CustomAudioSource(...)

audio_track = CustomAudioTrack(audio_source)

client.join("YOUR_DAILY_ROOM", client_settings={
    "inputs": {
        "microphone": {
            "isEnabled": True,
            "settings": {
               "customTrack": {
                   "id": audio_track.id
               }
            }
        }
    }
})

• Added support for CallClient.start_dialout() new fields: displayName, userId, video, codecs, and permissions.

• CallClient.set_audio_renderer() can receive two additional arguments: sample_rate and callback_interval_ms. With sample_rate you can now specify the desired sample rate of the incoming audio data. With callback_interval_ms you can control how often the provided callback is called (with 10ms intervals).

Changed

• CallClient.add_custom_audio_track() and CallClient.update_custom_audio_track() now receive a CustomAudioTrack instead of a CustomAudioSource.

• System certificates are now loaded on macOS, Linux and Windows platforms instead of the embedded Mozilla's root certificates.

Fixed

• Fixed an issue that would cause a blocking virtual microphone to not send any audio in some cases.