v0.6.0

Added

• Added punctuate and endpointing fields to TranscriptionSettings.

• Added dialout support with CallClient.start_dialout() and CallClient.stop_dialout().

• Added completion callbacks to VirtualMicrophone.write_frames() and VirtualSpeaker.read_frames(). This change makes virtual devices completely asynchronous if they are created with non_blocking set to True.

Changed

• Renamed session_id field to participantId in TranscriptionMessage.

Removed

• Removed is_final, user_id and user_name fields from TranscriptionMessage.

Fixed

• Room deletion messages from the server are now properly handled.

• CallClient.send_app_message(None) now properly triggers a ValueError exception.

• If an invalid participant ID is passed to CallClient.send_app_message() it will now trigger a ValueError 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.

v0.5.4

Improvements and fixes

• Fixed another issue that could cause CallClient.join() to fail if another Daily web client was also joining at the same time.

v0.5.3

Improvements and fixes

• Fixed an issue that could cause CallClient.join() to fail if another Daily web client was also joining at the same time.

v0.5.2

Improvements and fixes

• Disabled echo cancellation, noise suppression and auto gain control by default to match the previous library behavior.

v0.5.1

Improvements and fixes

• Fixed a crash when passing audio frames to VirtualMicrophone.write_frames() that require padding (i.e. non-multiple of 10ms worth of audio frames).

v0.5.0

New features

• Support for non-blocking virtual audio devices. This allows integration with hardware devices (e.g. via PyAudio).
• Echo cancellation, noise suppression and auto gain control can now be enabled for virtual microphones via custom constraints.
• It is now possible to pass additional Deepgram settings to start_transcription() using the new extra field.
• New demos to show how to integrate with PyAudio, how to send images and other improvements in existing demos.

Improvements and fixes

• Fixed an issue on user leave (manual or by the server) that would prevent the user to rejoin.

API changes

• Transcription defaults have been removed in favor of Deepgram's defaults. This allows to simply specify {"model": "nova-2"}.
• Transcription redact can now also be a list of strings as supported by Deepgram (e.g. ["pci"]).

v0.4.0

New features

• Added support for capturing individual participant audio tracks.
• Added support for ejecting participants.
• Support python >= 3.7 and, on Linux, glibc >= 2.28.

Improvements and fixes

• Improved general video renderer performance.
• Improved media subscriptions stability and performance.
• Fixed a deadlock that would not allow receiving multiple simultaneous video renderers.
• Fixed a deadlock when a screen share was stopped.
• Fixed an issue where setting the user name could not be reflected automatically when requesting participants list.
• Fixed an issue that could cause joins/reconnects to not complete successfully.
• Fixed a sporadic crash that could occur when handling media streams.
• Added Qt demo (similar to the existing Gtk demo).
• Qt and Gtk demos can now save the selected participant audio into a WAV file and can also render screen share.

v0.3.1

Improvements and fixes

• Fixed an issue that could cause daily-python clients to join a meeting session in a different region.
• Fixed a dead-lock that could occur when a CallClient is destroyed.

v0.3.0

New features

• Support for sending chat messages to Daily Prebuilt (CallClient.send_prebuilt_chat_message()).
• Added Python type hints (helpful for editor completions).
• Support for Python 3.8.

Improvements and fixes

• Improved response time of CallClient getter functions.
• Improved low-latency performace of virtual audio devices.
• Fixed potential crash after CallClient.leave().
• Improved internal safeness of participant video renderers.
• Fixed a VirtualMicrophoneDevice memory leak.
• Properly trigger a transcription error event if transcription can't start.
• Demos have been updated to show more real live code.

API changes

• Removed detect_language from TranscriptionSettings.
• EventHandler.on_transcription_stopped can now tell if transcription was stopped by a user or because of an error occurred.

v0.2.0

New features

• Support for start/stop recordings.
• Support for start/stop transcriptions and receive transcriptions messages.

Improvements and fixes

• Fixed an issue that was causing sporadic audio gaps on macOS and in certain OS task scheduling scenarios.
• Network re-connections have been improved.

API changes

• VirtualSpeakerDevice.read_frames() has been improved and doesn't require the user to add sleeps. Therefore, it is now possible to read, for example, 10 seconds of audio in a single call. Since the timings are now controlled internally, this minimizes any potential audio issues.

The following old code:

SAMPLE_RATE = 16000
READ_INTERVAL = 0.01
FRAMES_TO_READ = int(SAMPLE_RATE * READ_INTERVAL)
SECONDS_TO_READ = 10.0

for _ in range (int(SECONDS_TO_READ / READ_INTERVAL)):
  buffer = speaker.read_frames(FRAMES_TO_READ)
  time.sleep(READ_INTERVAL)

can be replaced with:

SECONDS_TO_READ = 10
FRAMES_TO_READ = SAMPLE_RATE * SECONDS_TO_READ
buffer = speaker.read_frames(FRAMES_TO_READ)