WebRTC library built from beta release M95 https://chromium.googlesource.com/external/webrtc/+log/branch-heads/4638 (cbad18b147e06f27082e0ff9312aeed86e6632b6)
Manually added:
- modules/audio_device/audio_device_impl.h
- modules/audio_device/audio_device_buffer.h
- modules/video_coding/encoded_frame.h
- modules/video_coding/codecs/interface/common_constants.h
- common_video/generic_frame_descriptor/generic_frame_info.h
- call/rtp_transport_controller_send_factory_interface.h
- call/rtp_transport_config.h
- call/rtp_transport_controller_send_interface.h
- call/rtp_config.h
- common_video/frame_counts.h
- call/audio_state.h
- modules/async_audio_processing/async_audio_processing.h
- call/video_receive_stream.h
- call/receive_stream.h
- p2p/base/port.h
- logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h
- logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h
- logging/rtc_event_log/ice_logger.h
- p2p/base/candidate_pair_interface.h
- p2p/base/connection.h
- p2p/base/connection_info.h
- p2p/base/p2p_transport_channel_ice_field_trials.h
- p2p/base/stun_request.h
- p2p/base/transport_description.h
- p2p/base/p2p_constants.h
- p2p/base/port_interface.h
- p2p/base/port_allocator.h