Name: livekit-libwebrtc-devel	Distribution: openSUSE Tumbleweed
Version: 137.0+git20251012.0001d84	Vendor: openSUSE
Release: 1.2	Build date: Wed Sep 16 18:57:03 2026
Group: Unspecified	Build host: reproducible
Size: 16463022	Source RPM: livekit-libwebrtc-137.0+git20251012.0001d84-1.2.src.rpm
Packager: https://bugs.opensuse.org
Url: https://github.com/webrtc-sdk/webrtc
Summary: Development files for LiveKit's build of WebRTC (M137)

The header tree and link symlink consumed by the LiveKit Rust SDK's webrtc-sys
crate, which the Zed editor uses for its calls feature.

Point webrtc-sys at it with

    LK_CUSTOM_WEBRTC=/usr/lib64/livekit-webrtc

That directory holds lib/libwebrtc.so, include/, webrtc.ninja,
desktop_capture.ninja and args.gn; webrtc-sys reads the first line of the two
.ninja files to re-apply this build's preprocessor defines to its own C++ shim.
It also has to be told to link the library dynamically, which it does not do by
itself:

    cargo:rustc-link-lib=static=webrtc -> cargo:rustc-link-lib=dylib=webrtc

Because the API and ABI change with every WebRTC snapshot, a consumer has to
require this exact version at build time and be rebuilt on every bump.

Provides

Requires

libwebrtc-suse137 = 137.0+git20251012.0001d84-1.2
pkgconfig(gio-2.0)
pkgconfig(glib-2.0)
pkgconfig(gobject-2.0)
rpmlib(CompressedFileNames) <= 3.0.4-1
rpmlib(FileDigests) <= 4.6.0-1
rpmlib(PayloadFilesHavePrefix) <= 4.0-1
rpmlib(PayloadIsZstd) <= 5.4.18-1

License

Apache-2.0 AND BSD-3-Clause AND HPND AND SUSE-Public-Domain

Changelog

* Wed Sep 16 2026 Martin Pluskal <mpluskal@suse.com>
  - Initial package, version 137.0+git20251012.0001d84: LiveKit's
    WebRTC M137 built from source as a shared library for the Zed
    editor's calls feature (webrtc-sys)
  - Source0 is maintainer-generated from a gclient checkout of
    webrtc-sdk/webrtc 0001d84; make-source.sh reproduces it
  - Build against the distribution's dav1d, ffmpeg, libaom,
    libjpeg-turbo, libvpx, openh264, opus and zlib instead of the
    bundled copies; abseil-cpp and libyuv have to stay bundled
    (README.SUSE-maint says why), as do boringssl, crc32c,
    libsrtp, perfetto, pffft, rnnoise and sigslot
  - Declare what is still bundled with Provides: bundled()
  - H.264 goes through the system openh264 and ffmpeg. H.265 is
    off: the distribution's ffmpeg-8 is built without the hevc
    decoder, so the parsers, RTP packetizers and SDP name that
    rtc_use_h265 adds would have no codec behind them
  - The bundled BoringSSL is renamed with an LK_ prefix so that it
    can neither interpose nor be interposed by the system OpenSSL
    in a process that loads both
  - Upstream ships no shared library and no soname, so the soname
    is SUSE-namespaced as libwebrtc-suse.so.137; a
    LIVEKIT_WEBRTC_137_0001d84 symbol version node pins the ABI to
    the snapshot, which changes within a milestone
  - Ship only the headers the consumer compiles against, not the
    whole source tree
  - Add add_licenses.patch: licence table entries for the
    generated LICENSE.md
  - Add ssl_verify_callback_with_native_handle.patch: pass the
    native SSL handle to a custom certificate verifier
  - Add add_deps.patch: pull zlib, log sinks, the simulcast
    encoder adapter and the frame crypto transformer into :default
  - Add disable_crel.patch: drop -Wa,--crel, which produces code
    that segfaults on aarch64
  - Add david_disable_gun_source_macro.patch: dav1d defines
    _GNU_SOURCE only for gcc on x86/x64
  - Add force_gcc.patch: the flags needed to build with GCC
    instead of Chromium's clang, plus the distribution hardening
  - Add chromium-129-rust.patch: drop the clang compiler_builtins
    config so no compiler-rt is needed
  - Add pipewire-1.5-rebased.patch: include spa/pod/iter.h, needed
    with PipeWire 1.5.81 and newer
  - Add native-gcc-toolchain.patch: drop the aarch64-linux-gnu-
    tool prefix from the arm64 gcc toolchain, which a native
    aarch64 build does not have
  - Add generate-licenses-system-gn.patch: run gn directly rather
    than through depot_tools, which is not packaged
  - Add shared-library.patch: link one shared library instead of a
    static archive, name the system ffmpeg libraries, set the
    soname and hide the PipeWire trampolines
  - Add system-openh264-shim.patch: depend on the unbundle shim so
    its forwarding headers are generated
  - Add h265-optional-in-frame-cryptor.patch: livekit's frame
    cryptor calls the H.265 parsers with no build guard, so the
    library does not link with rtc_use_h265=false

Files

/usr/lib64/livekit-webrtc
/usr/lib64/livekit-webrtc/args.gn
/usr/lib64/livekit-webrtc/desktop_capture.ninja
/usr/lib64/livekit-webrtc/include
/usr/lib64/livekit-webrtc/include/api
/usr/lib64/livekit-webrtc/include/api/adaptation
/usr/lib64/livekit-webrtc/include/api/adaptation/resource.h
/usr/lib64/livekit-webrtc/include/api/array_view.h
/usr/lib64/livekit-webrtc/include/api/async_dns_resolver.h
/usr/lib64/livekit-webrtc/include/api/audio
/usr/lib64/livekit-webrtc/include/api/audio/audio_device.h
/usr/lib64/livekit-webrtc/include/api/audio/audio_device_defines.h
/usr/lib64/livekit-webrtc/include/api/audio/audio_frame.h
/usr/lib64/livekit-webrtc/include/api/audio/audio_frame_processor.h
/usr/lib64/livekit-webrtc/include/api/audio/audio_mixer.h
/usr/lib64/livekit-webrtc/include/api/audio/audio_processing.h
/usr/lib64/livekit-webrtc/include/api/audio/audio_processing_statistics.h
/usr/lib64/livekit-webrtc/include/api/audio/audio_view.h
/usr/lib64/livekit-webrtc/include/api/audio/builtin_audio_processing_builder.h
/usr/lib64/livekit-webrtc/include/api/audio/channel_layout.h
/usr/lib64/livekit-webrtc/include/api/audio/echo_canceller3_config.h
/usr/lib64/livekit-webrtc/include/api/audio/echo_canceller3_factory.h
/usr/lib64/livekit-webrtc/include/api/audio/echo_control.h
/usr/lib64/livekit-webrtc/include/api/audio/echo_detector_creator.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs
/usr/lib64/livekit-webrtc/include/api/audio_codecs/L16
/usr/lib64/livekit-webrtc/include/api/audio_codecs/L16/audio_decoder_L16.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/L16/audio_encoder_L16.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/audio_codec_pair_id.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/audio_decoder.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/audio_decoder_factory.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/audio_decoder_factory_template.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/audio_encoder.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/audio_encoder_factory.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/audio_encoder_factory_template.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/audio_format.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/builtin_audio_decoder_factory.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/builtin_audio_encoder_factory.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/g711
/usr/lib64/livekit-webrtc/include/api/audio_codecs/g711/audio_decoder_g711.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/g711/audio_encoder_g711.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/g722
/usr/lib64/livekit-webrtc/include/api/audio_codecs/g722/audio_decoder_g722.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/g722/audio_encoder_g722.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/g722/audio_encoder_g722_config.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/opus
/usr/lib64/livekit-webrtc/include/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/opus/audio_decoder_opus.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/opus/audio_encoder_opus.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/opus/audio_encoder_opus_config.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/opus_audio_decoder_factory.h
/usr/lib64/livekit-webrtc/include/api/audio_codecs/opus_audio_encoder_factory.h
/usr/lib64/livekit-webrtc/include/api/audio_options.h
/usr/lib64/livekit-webrtc/include/api/call
/usr/lib64/livekit-webrtc/include/api/call/audio_sink.h
/usr/lib64/livekit-webrtc/include/api/call/bitrate_allocation.h
/usr/lib64/livekit-webrtc/include/api/call/transport.h
/usr/lib64/livekit-webrtc/include/api/candidate.h
/usr/lib64/livekit-webrtc/include/api/create_peerconnection_factory.h
/usr/lib64/livekit-webrtc/include/api/crypto
/usr/lib64/livekit-webrtc/include/api/crypto/crypto_options.h
/usr/lib64/livekit-webrtc/include/api/crypto/frame_crypto_transformer.h
/usr/lib64/livekit-webrtc/include/api/crypto/frame_decryptor_interface.h
/usr/lib64/livekit-webrtc/include/api/crypto/frame_encryptor_interface.h
/usr/lib64/livekit-webrtc/include/api/data_channel_event_observer_interface.h
/usr/lib64/livekit-webrtc/include/api/data_channel_interface.h
/usr/lib64/livekit-webrtc/include/api/dtls_transport_interface.h
/usr/lib64/livekit-webrtc/include/api/dtmf_sender_interface.h
/usr/lib64/livekit-webrtc/include/api/enable_media.h
/usr/lib64/livekit-webrtc/include/api/enable_media_with_defaults.h
/usr/lib64/livekit-webrtc/include/api/environment
/usr/lib64/livekit-webrtc/include/api/environment/environment.h
/usr/lib64/livekit-webrtc/include/api/environment/environment_factory.h
/usr/lib64/livekit-webrtc/include/api/fec_controller.h
/usr/lib64/livekit-webrtc/include/api/fec_controller_override.h
/usr/lib64/livekit-webrtc/include/api/field_trials.h
/usr/lib64/livekit-webrtc/include/api/field_trials_registry.h
/usr/lib64/livekit-webrtc/include/api/field_trials_view.h
/usr/lib64/livekit-webrtc/include/api/frame_transformer_factory.h
/usr/lib64/livekit-webrtc/include/api/frame_transformer_interface.h
/usr/lib64/livekit-webrtc/include/api/function_view.h
/usr/lib64/livekit-webrtc/include/api/ice_transport_factory.h
/usr/lib64/livekit-webrtc/include/api/ice_transport_interface.h
/usr/lib64/livekit-webrtc/include/api/jsep.h
/usr/lib64/livekit-webrtc/include/api/jsep_ice_candidate.h
/usr/lib64/livekit-webrtc/include/api/jsep_session_description.h
/usr/lib64/livekit-webrtc/include/api/legacy_stats_types.h
/usr/lib64/livekit-webrtc/include/api/location.h
/usr/lib64/livekit-webrtc/include/api/make_ref_counted.h
/usr/lib64/livekit-webrtc/include/api/media_stream_interface.h
/usr/lib64/livekit-webrtc/include/api/media_stream_track.h
/usr/lib64/livekit-webrtc/include/api/media_types.h
/usr/lib64/livekit-webrtc/include/api/metronome
/usr/lib64/livekit-webrtc/include/api/metronome/metronome.h
/usr/lib64/livekit-webrtc/include/api/metronome/test
/usr/lib64/livekit-webrtc/include/api/metronome/test/fake_metronome.h
/usr/lib64/livekit-webrtc/include/api/neteq
/usr/lib64/livekit-webrtc/include/api/neteq/custom_neteq_factory.h
/usr/lib64/livekit-webrtc/include/api/neteq/default_neteq_controller_factory.h
/usr/lib64/livekit-webrtc/include/api/neteq/default_neteq_factory.h
/usr/lib64/livekit-webrtc/include/api/neteq/neteq.h
/usr/lib64/livekit-webrtc/include/api/neteq/neteq_controller.h
/usr/lib64/livekit-webrtc/include/api/neteq/neteq_controller_factory.h
/usr/lib64/livekit-webrtc/include/api/neteq/neteq_factory.h
/usr/lib64/livekit-webrtc/include/api/neteq/tick_timer.h
/usr/lib64/livekit-webrtc/include/api/network_state_predictor.h
/usr/lib64/livekit-webrtc/include/api/notifier.h
/usr/lib64/livekit-webrtc/include/api/numerics
/usr/lib64/livekit-webrtc/include/api/numerics/samples_stats_counter.h
/usr/lib64/livekit-webrtc/include/api/packet_socket_factory.h
/usr/lib64/livekit-webrtc/include/api/peer_connection_interface.h
/usr/lib64/livekit-webrtc/include/api/priority.h
/usr/lib64/livekit-webrtc/include/api/ref_count.h
/usr/lib64/livekit-webrtc/include/api/ref_counted_base.h
/usr/lib64/livekit-webrtc/include/api/rtc_error.h
/usr/lib64/livekit-webrtc/include/api/rtc_event_log
/usr/lib64/livekit-webrtc/include/api/rtc_event_log/rtc_event.h
/usr/lib64/livekit-webrtc/include/api/rtc_event_log/rtc_event_log.h
/usr/lib64/livekit-webrtc/include/api/rtc_event_log/rtc_event_log_factory.h
/usr/lib64/livekit-webrtc/include/api/rtc_event_log/rtc_event_log_factory_interface.h
/usr/lib64/livekit-webrtc/include/api/rtc_event_log_output.h
/usr/lib64/livekit-webrtc/include/api/rtc_event_log_output_file.h
/usr/lib64/livekit-webrtc/include/api/rtp_headers.h
/usr/lib64/livekit-webrtc/include/api/rtp_packet_info.h
/usr/lib64/livekit-webrtc/include/api/rtp_packet_infos.h
/usr/lib64/livekit-webrtc/include/api/rtp_packet_sender.h
/usr/lib64/livekit-webrtc/include/api/rtp_parameters.h
/usr/lib64/livekit-webrtc/include/api/rtp_receiver_interface.h
/usr/lib64/livekit-webrtc/include/api/rtp_sender_interface.h
/usr/lib64/livekit-webrtc/include/api/rtp_transceiver_direction.h
/usr/lib64/livekit-webrtc/include/api/rtp_transceiver_interface.h
/usr/lib64/livekit-webrtc/include/api/scoped_refptr.h
/usr/lib64/livekit-webrtc/include/api/sctp_transport_interface.h
/usr/lib64/livekit-webrtc/include/api/sequence_checker.h
/usr/lib64/livekit-webrtc/include/api/set_local_description_observer_interface.h
/usr/lib64/livekit-webrtc/include/api/set_remote_description_observer_interface.h
/usr/lib64/livekit-webrtc/include/api/stats
/usr/lib64/livekit-webrtc/include/api/stats/attribute.h
/usr/lib64/livekit-webrtc/include/api/stats/rtc_stats.h
/usr/lib64/livekit-webrtc/include/api/stats/rtc_stats_collector_callback.h
/usr/lib64/livekit-webrtc/include/api/stats/rtc_stats_report.h
/usr/lib64/livekit-webrtc/include/api/stats/rtcstats_objects.h
/usr/lib64/livekit-webrtc/include/api/task_queue
/usr/lib64/livekit-webrtc/include/api/task_queue/default_task_queue_factory.h
/usr/lib64/livekit-webrtc/include/api/task_queue/pending_task_safety_flag.h
/usr/lib64/livekit-webrtc/include/api/task_queue/task_queue_base.h
/usr/lib64/livekit-webrtc/include/api/task_queue/task_queue_factory.h
/usr/lib64/livekit-webrtc/include/api/task_queue/task_queue_test.h
/usr/lib64/livekit-webrtc/include/api/task_queue/test
/usr/lib64/livekit-webrtc/include/api/task_queue/test/mock_task_queue_base.h
/usr/lib64/livekit-webrtc/include/api/test
/usr/lib64/livekit-webrtc/include/api/test/audio_quality_analyzer_interface.h
/usr/lib64/livekit-webrtc/include/api/test/audioproc_float.h
/usr/lib64/livekit-webrtc/include/api/test/create_frame_generator.h
/usr/lib64/livekit-webrtc/include/api/test/create_network_emulation_manager.h
/usr/lib64/livekit-webrtc/include/api/test/create_peer_connection_quality_test_frame_generator.h
/usr/lib64/livekit-webrtc/include/api/test/create_peerconnection_quality_test_fixture.h
/usr/lib64/livekit-webrtc/include/api/test/create_simulcast_test_fixture.h
/usr/lib64/livekit-webrtc/include/api/test/create_time_controller.h
/usr/lib64/livekit-webrtc/include/api/test/create_video_quality_test_fixture.h
/usr/lib64/livekit-webrtc/include/api/test/create_videocodec_test_fixture.h
/usr/lib64/livekit-webrtc/include/api/test/fake_frame_decryptor.h
/usr/lib64/livekit-webrtc/include/api/test/fake_frame_encryptor.h
/usr/lib64/livekit-webrtc/include/api/test/frame_generator_interface.h
/usr/lib64/livekit-webrtc/include/api/test/metrics
/usr/lib64/livekit-webrtc/include/api/test/metrics/chrome_perf_dashboard_metrics_exporter.h
/usr/lib64/livekit-webrtc/include/api/test/metrics/global_metrics_logger_and_exporter.h
/usr/lib64/livekit-webrtc/include/api/test/metrics/metric.h
/usr/lib64/livekit-webrtc/include/api/test/metrics/metrics_accumulator.h
/usr/lib64/livekit-webrtc/include/api/test/metrics/metrics_exporter.h
/usr/lib64/livekit-webrtc/include/api/test/metrics/metrics_logger.h
/usr/lib64/livekit-webrtc/include/api/test/metrics/metrics_set_proto_file_exporter.h
/usr/lib64/livekit-webrtc/include/api/test/metrics/print_result_proxy_metrics_exporter.h
/usr/lib64/livekit-webrtc/include/api/test/metrics/stdout_metrics_exporter.h
/usr/lib64/livekit-webrtc/include/api/test/mock_async_dns_resolver.h
/usr/lib64/livekit-webrtc/include/api/test/mock_audio_mixer.h
/usr/lib64/livekit-webrtc/include/api/test/mock_audio_sink.h
/usr/lib64/livekit-webrtc/include/api/test/mock_data_channel.h
/usr/lib64/livekit-webrtc/include/api/test/mock_dtmf_sender.h
/usr/lib64/livekit-webrtc/include/api/test/mock_encoder_selector.h
/usr/lib64/livekit-webrtc/include/api/test/mock_fec_controller_override.h
/usr/lib64/livekit-webrtc/include/api/test/mock_frame_decryptor.h
/usr/lib64/livekit-webrtc/include/api/test/mock_frame_encryptor.h
/usr/lib64/livekit-webrtc/include/api/test/mock_frame_transformer.h
/usr/lib64/livekit-webrtc/include/api/test/mock_media_stream_interface.h
/usr/lib64/livekit-webrtc/include/api/test/mock_packet_socket_factory.h
/usr/lib64/livekit-webrtc/include/api/test/mock_peer_connection_factory_interface.h
/usr/lib64/livekit-webrtc/include/api/test/mock_peerconnectioninterface.h
/usr/lib64/livekit-webrtc/include/api/test/mock_rtp_transceiver.h
/usr/lib64/livekit-webrtc/include/api/test/mock_rtpreceiver.h
/usr/lib64/livekit-webrtc/include/api/test/mock_rtpsender.h
/usr/lib64/livekit-webrtc/include/api/test/mock_session_description_interface.h
/usr/lib64/livekit-webrtc/include/api/test/mock_transformable_audio_frame.h
/usr/lib64/livekit-webrtc/include/api/test/mock_transformable_frame.h
/usr/lib64/livekit-webrtc/include/api/test/mock_transformable_video_frame.h
/usr/lib64/livekit-webrtc/include/api/test/mock_video_bitrate_allocator.h
/usr/lib64/livekit-webrtc/include/api/test/mock_video_bitrate_allocator_factory.h
/usr/lib64/livekit-webrtc/include/api/test/mock_video_decoder.h
/usr/lib64/livekit-webrtc/include/api/test/mock_video_decoder_factory.h
/usr/lib64/livekit-webrtc/include/api/test/mock_video_encoder.h
/usr/lib64/livekit-webrtc/include/api/test/mock_video_encoder_factory.h
/usr/lib64/livekit-webrtc/include/api/test/mock_video_track.h
/usr/lib64/livekit-webrtc/include/api/test/neteq_simulator.h
/usr/lib64/livekit-webrtc/include/api/test/neteq_simulator_factory.h
/usr/lib64/livekit-webrtc/include/api/test/network_emulation
/usr/lib64/livekit-webrtc/include/api/test/network_emulation/create_cross_traffic.h
/usr/lib64/livekit-webrtc/include/api/test/network_emulation/cross_traffic.h
/usr/lib64/livekit-webrtc/include/api/test/network_emulation/ecn_marking_counter.h
/usr/lib64/livekit-webrtc/include/api/test/network_emulation/network_emulation_interfaces.h
/usr/lib64/livekit-webrtc/include/api/test/network_emulation/schedulable_network_node_builder.h
/usr/lib64/livekit-webrtc/include/api/test/network_emulation_manager.h
/usr/lib64/livekit-webrtc/include/api/test/pclf
/usr/lib64/livekit-webrtc/include/api/test/pclf/media_configuration.h
/usr/lib64/livekit-webrtc/include/api/test/pclf/media_quality_test_params.h
/usr/lib64/livekit-webrtc/include/api/test/pclf/peer_configurer.h
/usr/lib64/livekit-webrtc/include/api/test/peer_network_dependencies.h
/usr/lib64/livekit-webrtc/include/api/test/peerconnection_quality_test_fixture.h
/usr/lib64/livekit-webrtc/include/api/test/rtc_error_matchers.h
/usr/lib64/livekit-webrtc/include/api/test/simulated_network.h
/usr/lib64/livekit-webrtc/include/api/test/simulcast_test_fixture.h
/usr/lib64/livekit-webrtc/include/api/test/stats_observer_interface.h
/usr/lib64/livekit-webrtc/include/api/test/test_dependency_factory.h
/usr/lib64/livekit-webrtc/include/api/test/time_controller.h
/usr/lib64/livekit-webrtc/include/api/test/track_id_stream_info_map.h
/usr/lib64/livekit-webrtc/include/api/test/video
/usr/lib64/livekit-webrtc/include/api/test/video/function_video_decoder_factory.h
/usr/lib64/livekit-webrtc/include/api/test/video/function_video_encoder_factory.h
/usr/lib64/livekit-webrtc/include/api/test/video/test_video_track_source.h
/usr/lib64/livekit-webrtc/include/api/test/video/video_frame_writer.h
/usr/lib64/livekit-webrtc/include/api/test/video_quality_analyzer_interface.h
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Fabrice Bellet, Mon Oct 26 00:13:59 2026

livekit-libwebrtc-devel-137.0+git20251012.0001d84-1.2 RPM for x86_64

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