├── .github ├── FUNDING.yml ├── codeql │ └── codeql_config.yml ├── images │ └── logo.png └── workflows │ ├── black.yml │ ├── build.yml │ ├── build_linux_wheels.yml │ ├── build_macos_wheels.yml │ ├── build_manylinux.yml │ ├── build_windows_wheels.yml │ ├── codeql_analysis.yml │ ├── create_sdist.yml │ └── pytest.yml ├── .gitignore ├── .gitmodules ├── .readthedocs.yaml ├── CMakeLists.txt ├── LICENSE.md ├── MANIFEST.in ├── Makefile ├── README.md ├── SECURITY.md ├── THIRD_PARTY_LICENSES.md ├── black.toml ├── docs ├── Makefile ├── links │ ├── webrtc │ └── wrtc │ │ └── __init__.py ├── requirements.txt └── source │ ├── _static │ └── css │ │ └── custom.css │ ├── conf.py │ ├── images │ ├── apple-touch-icon-180x180.png │ ├── favicon-16x16.png │ ├── favicon-32x32.png │ ├── favicon.svg │ └── logo.png │ ├── index.rst │ ├── modules.rst │ ├── templates │ └── module.rst_t │ ├── webrtc.base.rst │ ├── webrtc.functions.get_user_media.rst │ ├── webrtc.functions.rst │ ├── webrtc.interfaces.media_stream.rst │ ├── webrtc.interfaces.media_stream_track.rst │ ├── webrtc.interfaces.rst │ ├── webrtc.interfaces.rtc_audio_source.rst │ ├── webrtc.interfaces.rtc_dtls_transport.rst │ ├── webrtc.interfaces.rtc_ice_transport.rst │ ├── webrtc.interfaces.rtc_peer_connection.rst │ ├── webrtc.interfaces.rtc_rtp_receiver.rst │ ├── webrtc.interfaces.rtc_rtp_sender.rst │ ├── webrtc.interfaces.rtc_rtp_transceiver.rst │ ├── webrtc.interfaces.rtc_sctp_transport.rst │ ├── webrtc.models.rst │ ├── webrtc.models.rtc_on_data_event.rst │ ├── webrtc.models.rtc_session_description.rst │ ├── webrtc.models.rtc_session_description_init.rst │ ├── webrtc.models.rtp_encoding_parameters.rst │ ├── webrtc.models.rtp_transceiver_init.rst │ └── webrtc.rst ├── examples ├── LICENSE ├── README.md └── telegram_group_calls.py ├── pytest.ini ├── python-webrtc ├── cpp │ ├── CMakeLists.txt │ └── src │ │ ├── config.h.in │ │ ├── enums │ │ ├── enums.cpp │ │ ├── enums.h │ │ ├── python_webrtc │ │ │ ├── rtc_ice_component.cpp │ │ │ └── rtc_ice_component.h │ │ └── webrtc │ │ │ ├── cricket_ice_gathering_state.cpp │ │ │ ├── cricket_ice_gathering_state.h │ │ │ ├── dtls_transport_state.cpp │ │ │ ├── dtls_transport_state.h │ │ │ ├── ice_connection_state.cpp │ │ │ ├── ice_connection_state.h │ │ │ ├── ice_gathering_state.cpp │ │ │ ├── ice_gathering_state.h │ │ │ ├── media_type.cpp │ │ │ ├── media_type.h │ │ │ ├── peer_connection_state.cpp │ │ │ ├── peer_connection_state.h │ │ │ ├── rtc_ice_role.cpp │ │ │ ├── rtc_ice_role.h │ │ │ ├── rtc_ice_transport_state.cpp │ │ │ ├── rtc_ice_transport_state.h │ │ │ ├── rtp_transceiver_direction.cpp │ │ │ ├── rtp_transceiver_direction.h │ │ │ ├── sctp_transport_state.cpp │ │ │ ├── sctp_transport_state.h │ │ │ ├── sdp_type.cpp │ │ │ ├── sdp_type.h │ │ │ ├── source_state.cpp │ │ │ ├── source_state.h │ │ │ ├── track_state.cpp │ │ │ └── track_state.h │ │ ├── exceptions.cpp │ │ ├── exceptions.h │ │ ├── functions │ │ ├── functions.cpp │ │ ├── functions.h │ │ └── get_user_media.cpp │ │ ├── interfaces │ │ ├── create_session_description_observer.cpp │ │ ├── create_session_description_observer.h │ │ ├── interfaces.cpp │ │ ├── interfaces.h │ │ ├── media_stream.cpp │ │ ├── media_stream.h │ │ ├── media_stream_track.cpp │ │ ├── media_stream_track.h │ │ ├── peer_connection_factory.cpp │ │ ├── peer_connection_factory.h │ │ ├── rtc_audio_source.cpp │ │ ├── rtc_audio_source.h │ │ ├── rtc_audio_track_source.cpp │ │ ├── rtc_audio_track_source.h │ │ ├── rtc_dtls_transport.cpp │ │ ├── rtc_dtls_transport.h │ │ ├── rtc_ice_transport.cpp │ │ ├── rtc_ice_transport.h │ │ ├── rtc_peer_connection.cpp │ │ ├── rtc_peer_connection.h │ │ ├── rtc_rtp_receiver.cpp │ │ ├── rtc_rtp_receiver.h │ │ ├── rtc_rtp_sender.cpp │ │ ├── rtc_rtp_sender.h │ │ ├── rtc_rtp_transceiver.cpp │ │ ├── rtc_rtp_transceiver.h │ │ ├── rtc_sctp_transport.cpp │ │ ├── rtc_sctp_transport.h │ │ ├── set_session_description_observer.cpp │ │ └── set_session_description_observer.h │ │ ├── models │ │ ├── models.cpp │ │ ├── models.h │ │ ├── python_webrtc │ │ │ ├── rtc_on_data_event.cpp │ │ │ ├── rtc_on_data_event.h │ │ │ ├── rtc_session_description.cpp │ │ │ ├── rtc_session_description.h │ │ │ ├── rtc_session_description_init.cpp │ │ │ └── rtc_session_description_init.h │ │ └── webrtc │ │ │ ├── rtp_encoding_parameters.h │ │ │ └── rtp_transceiver_init.h │ │ ├── module.cpp │ │ └── utils │ │ ├── absl_optional.h │ │ └── instance_holder.h └── python │ ├── .env.example │ ├── test.py │ ├── tgcalls_test.py │ ├── webrtc │ ├── __init__.py │ ├── base.py │ ├── functions │ │ ├── __init__.py │ │ └── get_user_media.py │ ├── interfaces │ │ ├── __init__.py │ │ ├── media_stream.py │ │ ├── media_stream_track.py │ │ ├── rtc_audio_source.py │ │ ├── rtc_dtls_transport.py │ │ ├── rtc_ice_transport.py │ │ ├── rtc_peer_connection.py │ │ ├── rtc_rtp_receiver.py │ │ ├── rtc_rtp_sender.py │ │ ├── rtc_rtp_transceiver.py │ │ └── rtc_sctp_transport.py │ ├── models │ │ ├── __init__.py │ │ ├── rtc_on_data_event.py │ │ ├── rtc_session_description.py │ │ ├── rtc_session_description_init.py │ │ ├── rtp_encoding_parameters.py │ │ └── rtp_transceiver_init.py │ └── utils │ │ ├── __init__.py │ │ ├── callbacks_to_async.py │ │ └── convert_exceptions.py │ ├── wrtc.py │ └── wrtc.pyi ├── requirements-dev.txt ├── scripts ├── build-webrtc.bat ├── build-webrtc.sh ├── configure-webrtc.bat ├── configure-webrtc.sh ├── download-webrtc.bat └── download-webrtc.sh ├── setup.py ├── stubs └── wrtc │ └── __init__.pyi └── tests ├── __init__.py ├── conftest.py ├── helpers.py ├── rtc_peer_connection ├── __init__.py ├── test_add_track.py └── test_add_transceiver.py └── rtc_rtp_transceiver ├── __init__.py ├── test_direction.py └── test_stop.py /.github/FUNDING.yml: -------------------------------------------------------------------------------- 1 | github: MarshalX -------------------------------------------------------------------------------- /.github/codeql/codeql_config.yml: 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