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path: root/call/BUILD.gn
AgeCommit message (Expand)Author
2020-07-17Delete callbacks from RtpDemuxer on ssrc bindingDanil Chapovalov
2020-07-08Delete RtcpDemuxer as unusedDanil Chapovalov
2020-07-06Migrate call/ to webrtc::Mutex.Markus Handell
2020-07-02[Adaptation] Multi-processor support for injected Resources.Henrik Boström
2020-06-23Fix missing dependencies.Mirko Bonadei
2020-06-11[Adaptation] Adding adaptation resources from Call.Henrik Boström
2020-06-08Use absl_deps in order to preapre to the Abseil component build release.Mirko Bonadei
2020-05-26Remove send_crit_, bitrate_crit_ and last_bandwidth_bps_crit_ locks.Tommi
2020-05-20Calculate chain_diff for DependencyDescriptor from GenericFrameInfoDanil Chapovalov
2020-05-04Define MockFrameTransformer in test/.Marina Ciocea
2020-04-20Updates RtpVideoSender to populate RtpRtcp::Config.field_trialsErik Språng
2020-03-31Insert audio frame transformer between encoder and packetizer.Marina Ciocea
2020-03-26Allow setting a bandwidth cap for relayed connections.Christoffer Rodbro
2020-03-24[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.Henrik Boström
2020-02-28Insert frame transformer between Encoded and Packetizer.Marina Ciocea
2020-02-21Refactoring mock_transport to be used separatelyTim Na
2020-02-12Populate generic descriptor based on GenericFrameInfo when available.Danil Chapovalov
2020-01-21Reformat GN files.Mirko Bonadei
2020-01-14Adds scenario test for transport wide feedback based retransmission.Sebastian Jansson
2020-01-13Refactoring AudioSender api out of AudioSendStream for more abstraction to re...Tim Na
2020-01-07Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing.Danil Chapovalov
2019-12-07Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator classArtem Titov
2019-12-03VideoReceiveStream: Enable encoded frame sink.Markus Handell
2019-12-03Trials should always be populated in call config.Erik Språng
2019-11-26Delete media transport integration.Bjorn A Mellem
2019-11-25Makes all of RtpVideoSenderTest use simulated timeErik Språng
2019-11-21Adds injectable trials from peerconnection down to transport controller.Erik Språng
2019-11-01Enable injection of a custom NetEqFactory into PeerConnectionFactory.Ivo Creusen
2019-10-21Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueFor...Danil Chapovalov
2019-10-21Replace SingleThreadedTaskQueueForTesting::SendTask usage with ::webrtc::Send...Danil Chapovalov
2019-10-17Use source_sets in component builds and static_library in release builds.Mirko Bonadei
2019-10-17Delete unused method PacedSender::QueueSizePacketsNiels Möller
2019-10-15Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSenderSebastian Jansson
2019-10-03Reduces locking in RtpSenderVideo.Erik Språng
2019-09-19Propagating TargetRate struct to BitrateAllocator.Sebastian Jansson
2019-09-18Using struct for bitrate allocation limits.Sebastian Jansson
2019-09-18Remove api/bitrate_constraints.h.Mirko Bonadei
2019-09-17Reusing MediaStreamAllocationConfig struct in ObserverConfig.Sebastian Jansson
2019-09-17Use std::make_unique instead of absl::make_unique.Mirko Bonadei
2019-09-13Introduce api/crypto/BUILD.gn.Mirko Bonadei
2019-09-13Move MediaTransportInterface out of the libjingle_peerconnection_api targetNiels Möller
2019-09-11Move rtc_error.{h,cc} to its own build target.Mirko Bonadei
2019-09-02Split out RtpSource from libjingle_peerconnection_apiNiels Möller
2019-08-29Make the RtpHeaderParserImpl available to tests and tools only.Tommi
2019-08-29Delete unneeded dependencies on libjingle_peerconnection_apiNiels Möller
2019-08-29New target for api/rtp_parameters.h and api/media_types.h.Niels Möller
2019-08-07Delete deprecated rtc_event_log headerDanil Chapovalov
2019-07-29Add RtpPacketPacer interface for pacer controlErik Språng
2019-07-19Migrate WebRTC test infra to ABSL_FLAG.Mirko Bonadei
2019-07-03Make TaskQueueFactory required construction parameter for CallDanil Chapovalov