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Follow the following steps to update this project:
1- Git merge aosp/upstream-main, resolving any conflicts, to obtain a recent enough version of the code.
2- Use an upstream checkout to generate the json files.
2.1 - Follow instructions to download and build upstream webrtc from https://webrtc.googlesource.com/src/+/main/docs/native-code/development/index.md.
2.2 - Generate the json files for each architecture.
2.2.1 - Disable libaom (search for enable_libaom in all .gni/.gn files and set it to false).
2.2.2 - Disable x11 (set rtc_use_x11 to false in webrtc.gni).
2.2.3 - Disable pipewire (rtc_use_pipewire to false in webrtc.gni).
2.2.4 - Enable dummy audio file (rtc_use_dummy_audio_file_devices to true in webrtc.gni).
2.2.5 - Disable dav1d decoder (rtc_include_dav1d_in_internal_decoder_factory to false in webrtc.gni).
2.2.6 - Disable protobuf (rtc_enable_protobuf to false in webrtc.gni).
2.2.7 - Run this command:
for arch in x64 x86 arm64 arm; do
gn gen out/Debug --args="target_os=\"linux\" target_cpu=\"${arch}\"" --json-file-name=project_${arch}.json --ide=json
done
2.3 Copy out/Debug/project_*.json into android_tools.
3- Run android_tools/generate_android_bp.sh.
4- Build cuttlefish and run it in all architectures to ensure webrtc is working correctly (connect to it, press a few buttons and make sure audio works).
5- The build in step number 4 likely failed, fix any errors. Make sure to not edit external/webrtc/Android.bp directly, but modify the generating script instead.
6- Update these instructions with any new steps needed.
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