Squashed 'livekit-server/' changes from 154b4d26..46c43095
46c43095 fix goreleaser workflow, version 1.13.1 (#4577) e0815be2 chore: improve docker test shutdown reliability (#4576) bfd9deff expose TCPFallbackRTTThreshold and AllowUDPUnstableFallback via config (#4556) b93c1e16 Release v1.13.0. (#4573) fd452212 Update mediatransportutil to get ICE candidate timeout config (#4572) 8be8c74a Update github workflows (#4463) c4e41872 Update go deps to v1.17.2 (#4462) dc8e0310 Update go deps to v4 (#4482) 20fd1ad2 turn: allow for providing secret via file (#4564) 6590570d Pin pion/dtls to v3.1.2 (#4570) cdbbee1f deps: bump protocol + psrpc to latest tips (#4565) d290de81 Correct config comment (#4563) 77ecf920 rtc: report participant session end time on room move (#4561) 63be96f6 Prevent panic from nil(illegal) syncState.Subscriptions message (#4560) 835ef1b3 Metrics for participant active, i. e. fully established. (#4557) 5bd42534 Document of advertise_internal_ip and external_ip_only (#4554) 356ae211 Config documentation for advertise_internal_ip and skip_external_ip_validation (#4552) 7c319a67 rtc: prevent duration reporting for inactive participants (#4550) 2dd5e632 telemetry: split webhook-processed hook out of NewTelemetryService (#4548) 222177a9 service: prevent nil deref in validate with wrapped join request (#4547) dd7580b4 Protect against nil clientInfo (#4546) 145689e6 Start tracking Twirp method request latency in prometheus too, not just in logs (#4545) cde89627 rtc: emit per-data-track bytes via BytesTrackStats (#4540) 2e22911d Remove backwards compatibility support for TURN auth. (#4539) 062d1219 Use NACKQuueInterface type. (#4538) 7f08b04c Add IsIntentionalDisconnect helper (#4537) 1ab2bf04 Clean up packet size logging (#4536) 8ab92a80 Don't require media sections when joining (#4535) 019a6640 rtc: report participant kind code and details (#4534) 77595d38 TEL-336: fix sip error categorization (#4528) f303f499 Always enable rtx codec (#4533) e4a8a55c Check Less and LessEq in version compare. (#4532) 37eb7a32 Release v1.12.0 (#4529) 4a7b1e85 Create NACK tracker only once. (#4527) 89faaeba Apply ttl check only when authenticate allocation creating (#4526) b32933b0 Log details of RTCP packets. (#4525) 8b79ec9e Support SIP auth realm for inbound. (#4522) 4b8db3cf Add integration test for TURN auth failures (#4524) ef2e5efe Log large packets receive/send. (#4521) d1236750 feat: auto create rooms for tokens with the RoomCreate grant (#4320) 7a3e595b apply room tags from JWT grant room configuration (#4518) ab7fdeab add AssignmentHook to AssignJob; propagate websocket write errors (#4516) cf20c9cd Add expiry to TURN password. (#4515) 20d4a3a1 Populate data track loggers with context (#4514) 12fff29a allow setting agent job assignment url (#4512) ba366fc7 Fix SIP media config upgrade. (#4511) 8fbc5adf update protocol for protojson (#4510) 3de6f517 Add TURN permission handler. (#4505) 8ffcef93 Update protocol to support SIP media config. (#4509) 1ab1e072 test: verify upstream and downstream connection stats end-to-end (#4508) c4fd71a5 Fix sense check in DeltaInfo gathering (#4507) 803999ef rename agent environment to deployment (#4506) bacc21e6 add helper to check for agent worker endpoint (#4503) 253f977d add duration seconds reporting (#4500) ffab3bd3 add agent environment (#4498) ccdf23c8 Use mediatransportutil/codec package, no functional change (#4497) 680703f2 Include reception reoprts in receiver report callback. (#4496) f51798bc Fix publish-only limitations being incorrectly applied to receivers (#4495) a002337d Legacy TrackInfo.Simulcast flag. (#4493) af1dcc88 Add CloseWithReason to agent SignalConn interface (#4492) d7c2daf1 report all simulcast layers (#4491) c1ad2b22 Misc optimisations. (#4490) 19b9e8c0 Additional data tracks logging (#4489) 743d9c8b add support for client capabilities (#4461) fc47e478 Close peer connection unconditionally to unblock set local/remote (#4485) 639406eb Update module github.com/pion/ice/v4 to v4.2.3 (#4481) dc6b7505 reduce some heap use in packet path (#4478) f3b80b28 fix: wrap IPv6 addresses in brackets in UDP TURN URLs (RFC 3986) (#4476) 3a7f2628 Turn off transceiver re-use on Safari. (#4474) d84f3d7a add more types to signum (#4473) 701a37c2 Convert sort.Slice -> slices.SortFunc (#4472) 85be9d70 Avoid stream allocator event data cast to interface and back. (#4471) b43685e8 Keep a shadow copy of tracks for use by different stream allocator state (#4470) 27c2b149 Consolidate RTCP packets and do RTCP callback outside lock. (#4469) 31083307 do not log data track stats if not started (#4468) 9ee06635 feat(pion/ice): replace deprecated NAT1To1 with SetAddressRewriteRules (#4466) 8ccad68d Release v1.11.0 (#4459) dbf5cf61 Store concrete ICE candidate for remote candidates. (#4458) 2a04bc3c fix publisher frame count reporting for simulcast streams (#4457) 1d804737 fix: limit join request and WHIP request body to http.DefaultMaxHeaderBytes (#4450) 3cfb71e7 Use Muted in TrackInfo to propagated published track muted. (#4453) 69aa9479 Some drive-by clean up (#4452) 6c81f678 Add subscriber stream start event notification (#4449) ce1bf47b Revert "fix: ensure num_participants is accurate in webhook events (#4265) (#…" (#4448) cdb0769c fix: ensure num_participants is accurate in webhook events (#4265) (#4422) c91e79af Switch to stdlib maps, slices (#4445) ea7b9c6f Update module github.com/livekit/protocol to v1.45.3 (#4435) 97378368 Update go deps (major) (#3179) d6aef547 Update go deps (#3862) afc9feae Update github workflows (#4331) 4b385612 chore: pin GH commits and switch to golangci-lint (#4444) 2974ba87 Unsubscribe from data track on close (#4443) 5dc4e90d Apply IPFilter when get local ip (#4440) 88c77dc6 compute agent dispatch affinity from target load (#4442) 8fe99377 Log join duration. (#4433) 0a503a57 Add `Close` method for UpDataTrackManager and call it on participant (#4432) 55912dff Add some simple data track stats (#4431) 050909e6 Enable data tracks by default. (#4429) 72c7e65c chore: log API key during worker registration (#4428) 8a67dd1b Do not close publisher peer connection to aid migration. (#4427) 91e90c10 Add some more logging around migration. (#4426) c6ddc879 isExpectedToResume is based on whether flushing or not. (#4425) 7d06cfca Keep subscription synchronous when publisher is expected to resume. (#4424) 934f8598 Clean up data track observers on unsubscribe. (#4421) 9674ac48 Cleaning up some logs and standardising log frequency. (#4420) 7b925304 Drop time inverted packets in RED -> Opus conversion. (#4418) 4d8d232a ensure participant init is correctly serialized for logging (#4417) 4fe80877 Log time inversion between incoming packets (#4415) 248d7394 Guard against timestamp inversion in RED -> Opus conversion. (#4414) 9ab8c1d5 clear track notifier observers on subscription teardown (#4413) 397cd09a Embedded turn test (#4412) 56326654 Prepare release v1.10.1 (#4408) e9b113c8 Make the TURN bind address configurable and allow for multiple addresses. (#4315) 4bc5e6bb Address malformed H264/H265 parsing issues. (#4407) 77a0a4fc AV1 parser overflow fix. (#4405) ff7fd7ed feat(agent-dispatch): add job restart policy (#4401) 34bd1e08 do not log roll over for padding only packets (#4396) 13d02ee9 add deadline to dtls connect context (#4395) 9055a349 Path check helpers (#4392) 1f1eeb68 Fallback to servicestore if rpc is unavailable (#4391) 59e9bb41 Fix TURN server URL (#4389) 9e0a7e54 Close both peer connections to aid migration. (#4382) 9474c807 route participant reads through PSRPC instead of Redis (#4387) a5333a86 add packet trailer stripping support (#4361) 0e3d765d Release v1.10.0 (#4380) bc3aeaf3 Update grpc to address CVE-2026-33186 (#4381) 8cdd6f4c Replace deprecated io/ioutil with io in whipservice (#4375) 89410df7 handle AGENT_ERROR disconnect reason (#4339) 8f984c77 Fix repair stream ID reporting for RTX pairing. (#4369) cdfaacfc Restart nacker on OOB sequence number restart. (#4368) 4a9e0045 Update protocol. (#4367) 750d5904 Add API to restart lite stats. (#4366) c8bb2578 Rename log field "pID" to "participantID" for consistency (#4365) 77fc74a7 Do not block all ext ID determination on stream allocator listener (#4364) 90a46fab Do not kick off migration of closed participant (#4363) 5dc2e7b1 Switch data track extension to 1-byte ID/length. (#4362) 7323ad02 Sample data send error logging. (#4358) 0d34e455 Add option to not re-use transceiver in e2ee. (#4356) 95225ff2 don't require media section for dual peerconnection mode (#4354) e9639538 Refine ipv6 support (#4352) b34b0472 Add StopEgress function to the EgressLauncher interface (#4353) 69235ed2 update readme (#4340) db1a8046 defensive check for peer connection instance (#4350) cb7dc2d0 TEL-405: support originating calls from custom domains (#4349) 7eaaaada Mark last run of grow bucket outside goroutine. (#4348) caa47522 Add option to require media sections when participant joining (#4347) 087050d1 Release v1.9.12 (#4346) 493e87df Fix SIP client timeout. (#4345) 52c28a93 Log a bit more details of out-of-order TWCC feedback report. (#4343) 516aeabf Use ParticipantTelemetryListener of LocalParticipant. (#4342) b3510565 Exclude ice restart case from offer answer id mismatch warning (#4341) 303657bc feat: make INSTALL_PATH overridable in install script (#3954) 9d418689 Send participant left event after track unpublished for moved (#4334) bab91868 do not discount packets lost on duplicate packets (#4333) 939794cf mark + restart (#4329) 75f9c462 Add debug for receiver restart. (#4328) 74891f30 Protect against incorrect temporal layer. (#4327) f51b2732 Update pion/webrtc and deps to update dtls (#4326) b81bac0e Key telemetry stats worker using combination of roomID, participantID (#4323) 2d40449f Update self-hosting deployment documentation link (#4312) 03e90dd7 Fix for some CodeQL reported issues (#4314) 77c858f0 Log stats worker when it not closable. (#4313) a6035212 ESP32 client info (#4267) 478e486a Copilot suggested improvement to Github Actions permissions (#4310) cbd2f82d Copilot suggested improvement to Github Actions permissions (#4311) a9b8d40d Publish is always on publisher peer connection. (#4307) 8ae56406 generate & log egressID for start egress request (#4303) 35d7ef88 Avoid alloc in RTPStatsReceiver.Update (#4302) bb744916 More optimisation in RTPStats module. (#4298) cefd5da9 Optimise some bits in rtpstats_receiver (#4297) 52a4b848 Read client protocol from query param (#4294) 195b17f6 Populate client_protocol field in ParticipantInfo (#4293) 370e0a4d Set up audio config in audio level module when config is updated. (#4290) f3e9b688 Do not increase max expected layer on track info update. (#4285) a9849340 Avoid logger data race. (#4284) 97016627 Do not hold lock when creating buffer (#4283) 6b68e3d5 Create buffer if needed when a PLI is requested. (#4282) 3cca7180 use separate allocation for signal stats telemetry guard (#4281) d2bae34d refresh telemetry guard on participant move (#4280) 333f0349 clear reference guard when resetting signal stats (#4279) d1bab17b Add session duration and participant kind to closing log. (#4277) 1e689e1a Reducing some info level logs. (#4274) 88facc02 adds a test to ensure agent worker errors cause disconnection (#4273) 76a41a7a Generate config flags (#4268) 700e1788 require participant broadcast when metadata/attributes are set in token (#4266) b61799ec Ignore parse addr error when add remote candidate (#4264) 01bd966f Add silent frame for pcmu/a (#4258) 0c33b8c6 chore: move codecs/mime stuff to protocol (#4255) 165c1735 Update livekit protocol to v1.44.0 (#4254) 343f12b8 Wrapping SIP errors for invalid argument and not found (#4253) aea044c5 Defer setting clock rate in RTPStats module till codec is bound. (#4250) d9f716c1 FIx receiver restart race (#4248) 0508ee9f remove copy/paste left over line (#4246) 40408407 Release v1.9.11 (#4245) f8675507 Do not remove participant from cache on disconnect. (#4241) a35a6ae7 Add participant option for data track auto-subscribe. (#4240) 07572511 Wrapping the invalid request errors for CreateSipParticipant (#4239) 843d8c3e Update Pion transport package. (#4237) a05690d2 Changing field naming of data track packet (#4235) c4015008 Clear participant version cache on disconnect (#4234) 18db4ec1 Log modified timeout of API context. (#4232) ac20ccda Log timeout in API. (#4231) 541a7d6c Change some logs to debugw (#4229) 52ab3374 Return on SDP fragment read error. (#4228) 7fae5ac9 Do not restart receiver on codec change mid-session. (#4225) dafdc36e Add option to force simuclast codec. (#4226) 0a7dd40b Use only layer 0 for SVC codecs. (#4224) 4ec0f8f4 Support OpenTelemetry tracing. Add Jaeger support. (#4222) 80ba93fa Do NACK updates as soon as flow state is processed. (#4221) 4405afe2 Use atomic pointer and return interface from RED transformer constructors (#4220) b649c2fe Remove method not needed from REDTransformer. (#4219) f0080f35 Remove enable arrival time forwarding method. (#4217) 335f4c33 Swap result sink atomically rather than closing and setting. (#4216) 46651c19 Release v1.9.10 (#4214) 08ac4ecd Support preserving external supplied time. (#4212) f6efccce report video size from media data for whip (#4211) d92f6a79 return iceservers for whip (#4210) 1a4758ed Skip restart callback if external. (#4208) dde4fb49 configurable dependency descriptor restart (#4207) 08793bea Use active at time to check for track not bound timeout. (#4206) 3606ce54 Do not warn about track not bound if participant is not ready. (#4205) b8ddd0f9 Taking interface{} -> any modernize bits (#4204) b91cd2e4 Rework receiver restart. (#4202) bb00c864 Restart API on receiver. (#4200) 25ece1e9 Minor refactor in buffer base and audio level (#4198) 599002f8 ignore PLI requests for non-video (#4196) 2510b946 Taking a bunch of go modernize suggestions. (#4194) ed8e6afc Handle repair SSRC of simulcast tracks during migration. (#4193) c6bf7a27 Fix logging key and other clean up around stream restart. (#4192) 3cb9abb6 Update pion/webrtc to v4.2.1 (#4191) 8b0efb8c Resolve RTX pair via OnTrack also. (#4190) 381bce03 Return extended sequence number only and not packet. (#4189) 6bcbf54e Always instantiate nacker when using out-of-band sequence numbers. (#4187) e71184de Store buffer after creating it. (#4186) 7c8ea115 Refactor receiver and buffer into Base and higher layer. (#4185) 4104b827 update protocol (#4183) cd99fec2 Make new path for signalling v1.5 support. (#4180) 32cd0370 Flush the ext packets on restart/close and release packets. (#4179) 1df1316b Move OnDataTrackMessage to participant listener to support replay. (#4178) e7601251 Make data message naming a bit more consistent. (#4177) a04e566d Use published track for model access in data down track. (#4176) 47c86be1 Add support for TURN static auth secret credentials (#3796) 24559a28 chore(deps): update github workflows to v6 (#3866) 0a824386 add explicit room exists servicestore op (#4175) 39bd077d Release v1.9.9 (#4174) cbb2c617 Publish/Unpublish counter match. (#4173) fb849edc Minor clean up (#4172) c28e5e45 fix(kindToProto): update protocol (#4171) 47324abd Drop run away receiver reports. (#4170) 462ec324 prevent uint overflow setting packet not found count (#4169) 5c841b8e Some logging changes. (#4168) 2f2d0a57 skip lost sequence number ranges in getIntervalStats (#4166) 898ebe05 clean up manual roomservice log redaction (#4165) 3e417253 move delete to oss service store (#4164) 5964efbb Ensure subscribe data track handles are unique (#4162) a26c4830 Add support for RTP stream restart. (#4161) 386f0b38 fix typo in clearing index when removing track from room track manager (#4158) 0abfb251 deregister observability function when participant is closed (#4157) 97aba5e7 Consistently undo update to sequence number and timestamp when the (#4156) 2317c295 Fix panic while removing track from room track manager. (#4153) a0a28ac5 Avoid duplicate track add to room track manager. (#4152) f01008f8 Revert telemetry stats worker wait configuration. (#4151) ca4b56d2 Handle case of sequence number jump just after start. (#4150) 97099cae Configurable telemetry stats worker clean up wait. (#4148) d7db7cb3 chore: fix a large number of spelling issues (#4147) cadf2ad9 Release v1.9.8 (#4145) 498304cd defensive nil check (#4144) 20f6a497 Store ddParser in atomic.Pointer (#4143) 037cb906 release ext packet if patching fails (#4142) dd598ef2 Release ExtPacket if dependency descriptor or other parsing fails (#4141) 35b0e2bc update webrtc to 4.1.8 to pick up DTLS fingerprint check during handshake (#4140) 1c1a836c Mark RTCP buffer Write as noinline. (#4138) ea9b2177 protocol deps to get inactive file adjusted memory usage. (#4137) 64f3d1e9 switch participant callbacks to room to listener interface (#4136) c6e6c021 add debug metric for tracking references (#4134) a30c79fa Use isEnding to indicate if down track could be resumed. (#4132) 8c241ecf Fix RTCP reader leak in DownTrack. (#4131) 3eef869a Do not pause rid in SDP (#4129) 8e01e595 Release 1.9.7 (#4128) 7c1a0fab Fix concurrent map access. (#4127) 14446b1c Let participant close remove the published tracks. (#4125) fa0633aa move utils.WrapAround to mediatransportutil (#4124) f8706cd4 Update pion/ice to stop gather first on close (#4123) 7954748d Data tracks (#4089) 7158d983 log bucket growth (#4122) 04b35eb6 Release v1.9.6 (#4121) ebdcead5 Update mediatransportutil to get bucket packet size limit. (#4120) 411b09f6 Release v1.9.5 (#4119) 8dcf235a Update pion/ice - attempt to address tcp packet conn close hang (#4116) 64c65143 Update mediatransportutil (#4115) 0a2943bb Clean up bits added to debug peer connection close hang. (#4114) 9c483a69 Use released version v1.8.41 of pion/sctp (#4113) 35c79a57 Update SCTP hash, had the wrong one in previous PR (#4111) e0fbbef1 Update pion/sctp with RFC9260 revert (#4110) f3c80917 Try SCTP with read deadline to unblock abort. (#4109) bd5382da Splitting transport close timeout logs. (#4108) 6d4154b8 Update pion/ice. (#4107) a6418ae2 Log more peer conenction state on close timeout. (#4105) 06d99974 Check for cancel on unsubscription/source track going away. (#4104) 7f10e18b Record join/publish/subscribe cancellations. (#4102) 40293632 Clear stereo=1 if stereo is not enabled. (#4101) 70f6def3 Add checks for participant and sub-components close. (#4100) ffbabcc7 Switch forwarding latency log to Debugw (#4098) 27d82a72 Fix "address" typo in transport logs (addddress → address) (#4097) 37a06821 logger proto redaction. (#4090) 54cf7d46 Control latency of lossy data channel (#4088) 5175c1af Lock x/tools at 0.37.0 (#4085) d510fff1 Downgrade x/tools to be able to make a release (#4084) c3ea5890 Prepare release v1.9.4. (#4083) 3a128e61 protocol bump for SIP error mapping and validation (#4081) c3964ba2 Use sync.Pool for objects in packet path. (#4066) f8b994d4 Forwarding latency measurement tweaks. (#4080) f4929f09 Revert "Revert pion/transpor to v3.0.8 (#4073)" (#4074) a04d9c48 Revert pion/transpor to v3.0.8 (#4073) 2d5054ad kind details for connector (#4072) a272e28a Log raeson for subscriber not being to determine codec. (#4071) b9b4eec9 Update pion/transport to v3.1.1 (#4070) b23d093c update protocol (#4069) 4ce07bed Higher resolution forwarding latency histogram. (#4067) 858db7ab fix(deps): update module github.com/livekit/protocol to v1.43.0 (#4015) 1dc9b8fc Use buffered indicator to exclude from forwarding latency. (#4062) f117ee51 Track start up delay. (#4061) 4872f205 Return write count from WriteRTP. (#4059) d0ba46b4 Log write count atomic. (#4057) ae5fb7e8 Add packet to forwarding stats only if packet is forwarded. (#4056) f6909192 Update PsRPC to get redis pipeliner implementation. (#4055) ca3c507b Prevent invalid track access while peer connection is shutting down. (#4054) 9ca6ee00 Use replace so that x/tools does not get overridden (#4048) b9323eab chore(deps): downgrade x/tools for counterfeiter (#4047) 2f1e6c36 Prep release v1.9.3 (#4046) 9d5c351d Fix prom units for forwarding latency/jitter. (#4045) e183657c Add prom histogram for forwarding latency and jitter. (#4044) 1eefeb30 Enable AbsCaptureTimeURI in RTC configuration (#4043) 075a7576 Use simulcast codec as default policy for audio track (#4040) c264b504 Don't warn 0 payload type for PCMU (#4039) 32fc3525 Broadcast cond var on RTX write. (#4038) 061eb8b4 AddDownTrack to regressed codec after restarting forwarder. (#4037) c87eb8ed fix: add missing Unlock() in AddReceiver (#4036) 70444924 if RingingTimeout is provided, deadline should be set to that timeout. (#4018) ab906d71 Prevent leakage of previous codec after codec regression. (#4035) 79b03f97 Log queueing latency when encountering high forwarding latency (#4034) 29117b14 set max layer in allocation (#4033) 15b19ccd Remove ~ from rid which indicates disabled layer to get the actual rid (#4032) 34e16a87 Check more conditions for opportunistic alloc. (#4031) 81fbd355 Use the optimal allocation function for opportunistic allocation. (#4030) a2ce73e0 Do not bind buffer if codec is invalid. (#4028) cef6fdb7 Correct direction for request/response for prom counters. (#4027) 5042c06c Use rtp converter from protocol/utils/rtputil (#4020) 5a426d15 Use rtp converter from protocol/utils (#4019) 35fb8877 feat: use env var for GOARCH (#4012) c0397696 Issue #1 only: Fix spatial layer initialization in Forwarder (#4003) 2afbf0e8 Some golang modernisation bits. (#4016) 484f784a Prepare release v1.9.2 (#4011) ad074ed2 counterfeiter needs an older version of x/tools (#4009) e63e8b6f Include mid -> trackID in both SDP offer and answer. (#4007) 781dfede Do not call receiver methods under settings lock. (#4006) 69ff25a0 Use answer with mid -> trackID mapping when in single peer connection (#4005) fe912acf Update pion/webrtc to prevent GetStats panic. (#4004) 7930dcde Do not try to read stats from peer connection after close. (#4002) ca0d5ee9 Count request/response packets on both client and server side. (#4001) dd62eb00 Resort to full search for requested quality is not available. (#4000) f6ca82d1 Revert to using silence packets for audio dummy start. (#3999) 0e2c59c8 Sort codec layers when adding track (#3998) 100bb46a Adding ProviderInfo to GetSIPTrunkAuthenticationResponse (#3993) a8d4df66 "Power of Two Random Choices" option for node selection (#3785) a20bbe34 Log RPC details. (#3991) 158496bc Increment RTP timestamp on padding when using dummy start. (#3989) 4f6ed65d Limit check to red + opus when looking for primary codec match. (#3988) a87f6c4b Allow passing inline trunk for outbound calls. (#3987) bf06596f Support Opus mixed with RED when encrypted. (#3986) ea208a1c Add encryption datapacket type (#3869) 2a6adbe8 Use padding only packets for dummy start of audio. (#3984) be018f97 Provide the InputVideo/AudioState to Ingress in WHIPRTCConnectionNotify (#3982) 146bd969 Do not panic of redis is not configured (#3981) 01337ba7 Do not start forawarding on out-of-order packet. (#3985) c7f625d6 Do not force codec regression between opus and red. (#3980) 3bd20ddb Revert unintentional change to not handle transport fallback on (#3970) 89a2f46c Update deps to fix redis issue when 1 cluster address is provided (#3969) 060719d1 add config for user data recording (#3966) b3ee219c fix stats worker closed condition (#3965) 3d737031 add idempotent reference count to telemetry stats worker (#3964) 735c663a Update protocol for EventKey helper. (#3963) 646b9de8 Add node_ip to config-sample.yaml (#3960) 0bf7b178 avoid logging on small values (#3958) 00ff2ab9 Adjust for hold time when fowarding RTCP report. (#3956) e180be06 short circuit participant broadcast filter in livestream mode (#3955) bfba6fee Adjust stream allocator ping interval based on state. (#3951) 3837006b Revert "Switch ops queue a singly linked list. (#3949)" (#3950) 990c5faf feat: server rpc apis (#3904) 80b11662 Switch ops queue a singly linked list. (#3949) 56ee2328 handle terminated job requests (#3948) 49f9b9c8 Flush stats when there are no packets. (#3947) e6a3df1e ForwarStats.GetStats needs to be public (#3946) 824d116b Tweaks tresholds for logging high forwarding latency/jitter. (#3945) 408492e0 Log some information around high forwarding latency. (#3944) 6a41fae5 Use microseconds for forwarding stats. (#3943) 856e0871 mediatransportutil to log local address when validating external IP (#3942) 40101cf7 Update protocol for SipCreateParticipant (#3939) b07e7a38 Use difference in key frame counter to stop seeder. (#3936) d7f92878 Avoid matching on empty track id. (#3937) 56fb2885 Do DD restart only if DD structure is present. (#3935) 86facce9 More debugging of DD jump (#3934) 6058a3f6 Add debugging from DD frame number wrap around. (#3933) dc6825c0 mediatransportutil crash fix for logging local address (#3930) d6f0588f Update mediatransportutil to log external IP found via STUN. (#3929) 2c30a064 Fix dynacast subscriber node clearing on move participant. (#3926) 6489237e Simulcast audio fixes (#3925) 9f0ab870 Wait for `SetRemoteDescription` before configuring senders. (#3924) df6c26db Subscrbed audio codecs - update from remote nodes. (#3921) 798fa761 Support simulcasting of audio (#3920) f4a06cf0 Clean code as there is no oss sweeper for ingress (#3918) 5f561b4f Include agent_name as a participant attribute (#3914) 782a35e8 update protocol for psrpc (#3915) eee2001a Set publisher codec preferences after setting remote description (#3913) fc995533 add incoming request id to request response message (#3912) 76645fad Rpcs for ingress proxy WHIP (#3911) 991a4a4f Refactor subscribedTrack + mediaTrackSubscriptions. (#3908) e16b3ba9 Use gzip reader pool (#3903) 17c34921 update protocol for sip api change (#3902) 2f43a575 Release candidate for v1.9.1 (#3899) 07c40cf3 Use `RequestResponse` to report protocol handling errors (#3895) 98352fd0 Prevent race in determining BWE type. (#3891) f7291fda Do not send both asb-send-time and twcc. (#3890) 21b42fa6 Do not advertise NACK for RED. (#3889) 6633bf93 Use departure timeout from room preset. (#3888) 38f7906e Handle migration better in single peer connection case. (#3886) 5026de2b handle frame number wrap back in svc (#3885) 091e3c13 Revert to using answer for migration case. (#3884) 2aeadf14 init ua parser once (#3883) 998a9f94 Switch known rids from 012 -> 210, used by OBS. (#3882) 890fd942 Single peer connection mode (#3873) bfe98eaa fix: ensure the participant kind is set on refresh tokens (#3881) 8d270e2a chunk room updates (#3880) b4e146c5 update mediatransport util for ice port 3478 (#3877) dc3a7753 Fix timeout handing in StopEgress (#3876) d62336e1 Remove unnecessary check (#3870) c58e5d23 Update golang Docker tag to v1.25 (#3864) 98d577ee Update module github.com/livekit/protocol to v1.40.0 (#3865) afbf541e Update pion deps (#3863) b660c3b5 Extract video size from media stream (#3856) 456b8709 Fix missed unlock (#3861) d500806e Handle no codecs in track info. (#3859) 11b240d6 Log track settings more. (#3853) 1aa0f963 Log signal messages on media node. (#3852) b182d07b Log signal messages as debug. (#3851) a370bb20 Support G.711 A-law and U-law (#3849) fa5f4ef3 Populate SDP cid in track info when available. (#3845) eed27885 Send `participant_connection_aborted` when participant session is closed (#3848) 61e59346 Update go deps (#3439) 1b228913 Support video layer mode from client and make most of the code mime aware (#3843) f2da4444 Support per simulcast codec layers. (#3840) f275f592 handle SyncState in join request (#3839) 5d44cf6d Use wrapped join request to be able to support compressed and uncompressed. (#3838) 5ca16264 Support join request as proto + base64 encoded query param (#3836) 7dea1012 Clean up missed v2 pieces (#3837) 34a49130 Delete v2 signalling (#3835) 1fe33716 Fix: RingingTimeout was being skipped for transferParticipant (#3831) 5751692a deps (#3829) db4bc127 Get to the point of connecting publisher PC and using it for async signalling (#3822) 5e483e75 update readme (#3809) e3155b14 Get to the point of establishing subscriber peer connection. (#3821) a7ce1382 HTTP DELETE of participant session (#3819) 01de0e36 Do not send leave if nil (to older clients) (#3817) 10103449 Add country label to edge prom stats. (#3816) 68387b41 Minor tweak to keep RPC type at service level. (#3815) a75295fc More v2 signalling changes (#3814) b20db94d Validation end point for v2 signalling. (#3811) f2f595f4 update readme (#3808) fffc2ac0 Use signalling utils from protocol (#3807) f5fc82d3 Filling out messages unlikely to change in v2. (#3806) 1c99b9ad Split signal segmenter and reassembler. (#3805) 0a1bfd30 Signal handling interfaces and participant specific HTTP PATCH. (#3804) 7837c8e5 starting signaller interface (#3802) 18ce5244 Handle Metadata field from RoomConfig (#3798) 2a6a9b8a Grouping all signal messages into participant_signal. (#3801) 078c01fa Signal v2: envelope and fragments as wire message format. (#3800) b9a44c3f Signalling V2 protocol implementation start (#3794) ba702a53 forward agent id to job state (#3786) 1f31d430 Map ErrNoResponse to ErrRequestTimedOut in StopEgress to avoid returning 503 (#3788) 51bbe8c5 Set participant active when peerconnection connected (#3790) 40028dc3 Normalize known rids. (#3779) ddd92329 Return default layer for invalid rid + track info combination. (#3778) 8c033ce9 Enable H265 by default (#3773) 7678e087 Set rids for all codecs. (#3772) 5d636acf Limit taking rids from SDP only in WHIP path. (#3771) 4d09e5b5 Log SDP rids to understand the mapping better. (#3770) c69f1aae Revert "Temporary change: use pre-defined rids" (#3769) 8197438e bounds check layer index (#3768) d11da5f5 Temporary change: use pre-defined rids (#3767) cb4da533 fix(deps): update module github.com/livekit/protocol to v1.39.3 (#3733) d6d2b6d8 feat(cli-flags): add option for cpu profiling (#3765) 9fc4ddbe ClearAllReceivers interface is used to pause relay tracks. (#3761) 1216113b Do not need to just clean up receivers. Remove that interface. (#3760) ef6c38ce Log previous allocation to see changes. (#3759) 01bf9685 SVC with RID -> spatial layer mapping (#3754) c481396f offer could be nil when migrating. (#3752) 8c2fc0bc Fix svc encoding for chrome mobile on iOS (#3751) e467daa0 move egress roomID load to launcher (#3748) 3783ebb3 feat(cli): update to urfave/cli/v3 (#3745) 03d3fcab Fix data packet ParticipantIdentity override logic in participant.go (#3735) 068b4366 reuse compiled client config scripts (#3743) e754a860 return error when moving egree/agent participant (#3741) 7542cf07 remove unused code (#3740) 9d569e2f Take ClientInfo from request. (#3738) 80774263 chore: set workerid on job creation (#3737) 5549ab55 Revert clearing RIDs. (#3732) ae967313 Clear rids if not present in SDP. (#3731) 0e033907 Return highest available layer if requested quality is higher than (#3729) 9ce737db Add log for dropping out of order reliable message (#3728) 1b95e818 Don't check bindState on downtrack.Bind (#3726) 670f927f Set and use rid/spatial layer in TrackInfo. (#3724) a9e29116 Add Id to SDP signalling messages. (#3722) 4ec828ce Fix bug with SDP rid, clear only overflow. (#3723) 8f6c3a9b Clear rids from default for layers not published. (#3721) ce07740e Add simulcast support for WHIP. (#3719) e98fb94f Create client config manager in room manager constructor. (#3718) fdf9b852 e2e reliability for data channel (#3716) 35dda8ea swap pub/sub track metrics (#3717) 1d9a4366 Do not require create permission for WHIP participant. (#3715) e0aea17a Flush stats on close (#3713) 630aa7d9 implement observability for room metrics (#3712) e7f0294e remove unused ws signal read loop (#3709) 77f70b18 for real, pick up protocol change for webhooks queue length bucnkets (#3707) 7b180646 protocol dep for webhook stats buckets (#3706) b0ab95ba warn about credentials when used in tokens (#3705) a72ce30f Small changes to add/use helper functions for length checks. (#3704) 425f6bb3 Allow passing extra attributes to RTC endpoint. (#3693) 758e1762 Add a trend check before declaring joint queuing region. (#3701) fe81e411 Adds Devin to readme so it auto updates DeepWiki weekly (#3699) 09dede35 version bump v1.9.0 (#3698) fc867c5b Webhook prom stats (#3697) 0e17916f Do not use Redis pipeline for SIP delete. Fixes Redis clustering support. (#3694) 1b760393 WHIP support. (#3692) e4f7d81b add client ip to agent worker registration (#3675) 6b849a87 update mediatransportutil for sctp congestion control (#3673) 83b189b0 Add ServerInfo to ReconnectResponse (#3671) 5f87a35b Prevent operating on swapped out map. (#3670) 13b55a80 move agent token (#3669) c9385edd handle agent worker jwt (#3668) 3b359d8b Use logger resolver reset to reset contexts. (#3665) dbb70e0f Fix dynacast quality for moving out tracks (#3664) 0a5f3c2a resolve new room name logger earlier when moving participant (#3662) 2df05517 Revert unbound transceiver stop. (#3661) 5172af15 ~Send initial participant update only after a participant becomes active.~ - General clean up (#3655) 5d4d86f8 protocol to pick resolver values replace (#3659) 7f8e6323 Send self participant update immediately. (#3656) 11630878 Use unordered for lossy data channel. (#3653) aee34ffe log request for agent dispatch api (#3650) 793b383a Add Moving participant to another room (#3648) 2fff36cb Stub MoveParticipant so that cloud can include the latest protocol. (#3646) d4ab1142 Redact address (#3643) b83190a3 protocol update to fix memory stats path (#3642) a1f4e88e Update protocol to latest, got bit by tag (#3641) 8d3902af Protocol to pick up cgroups v2 memory path fix (#3640) 2f002388 Use participant close reason in remove. (#3639) 58822c26 Include clientInfo in connectivity logs. (#3638) 6d6393a6 Take AudioFeatures from AddTrack. (#3635) 08670412 Limit buffer queue before Bind. (#3634) 9f5bc9b9 Avoid synthesising duplicate feature. (#3632) 847239c3 Disable vp9 for safari 18.4 (#3631) f69ab680 Populate the sender identity when translating to user packet. (#3628) e1490558 Forward data between WHIP client and non-WHIP client (#3627) 6739e7bc Broadcast inside lock (#3626) f24152b4 Call Broadcast in lock scope. (#3625) b760918a Use logger from request context. (#3623) 34a2e2c1 Check for multiple layers for managed track. (#3622) 4955ebe4 Forward transfer headers to internal request (#3615) d9ee9214 Set up RTX for WHIP publish (#3619) d8cf5439 Determine TURN connection type and no fallback for TURN/TLS. (#3612) d0d212fd Fix WHIP ICE restart. (#3616) 5e7f8a12 Update mediatransportutil for max sctp message (65K) (#3611) 28dfac14 Use exported GetEgressNotifyOptions (#3604) 75236bef protocol update to fix IPv6 SDP fragment parsing (#3603) 2130980d Add basic video support to WHIP. (#3602) e5cbb227 Allow specifying extra webhooks with egress requests (#3597) 7e16106a Add OnSubscirberReady callback on LocalParticipant. (#3600) 5c2d96b9 Check DestinationRoom of VideoGrant for participant forwarding (#3599) 2e236a19 Revert participant state ACTIVE change. (#3598) 35ac5f56 Add support for WHIP ICE Trickle/Restart. (#3596) ec2dff96 Fix SIP updates when replacing slices. (#3592) e24fe77b map PEER_CONNECTION_DISCONNECTED -> CONNECTION_TIMEOUT (#3591) 6ee6eb43 Do not drop audio codecs (#3590) 68357ba6 List audio codecs after video codecs. (#3589) 05a891ff Fix rule (had an extra bracket) (#3588) d7c41091 Exclude RED from enabled codecs for Flutter + 2.4.2 + Android. (#3587) ee08aede skip out of order participant state updates (#3583) 15a8d9a2 Break track published fuse when there are no tracks (#3581) 35f83c51 Replace Promise with Fuse. (#3580) 07fe9b72 Prevent migration race. (#3579) ac8082ff Use older SDP module to accommodate bad SDP. (#3578) 1c8307c7 Use cgroup for memstats. (#3573) e9be0fca log SDP offer on error (#3577) 3238ab8d Calculate rates for memory used and total. (#3570) d08487bf Unlabeled (pass through) data channels. (#3567) 52ce18d5 fix: revert recent changes to determine simulcast from sdp (#3565) cdfbb106 Audio uses signal SignalCid and SdpCid. (#3564) ed5e2f16 Keep simulcast information tied to receiver. (#3563) ad010cfc chore(logs): log VLS type for VP9/AV1 (#3561) 8cc17f8f Rework node stats a bit. (#3555) 15f56551 fix(video): determine svc/simulcast from SDP for advanced codecs (#3549) 2b6a46f4 Handle `prefer_regression` for backup codec (#3554) b0abb0ae Add option to use different pacer with send side bwe. (#3552) 26822b6b ParseUsername utility for TURN user name. (#3547) 55909ed7 log the initial join response (#3546) 97fcb82a Fix: Return NotFoundErr instead of Unavailable when the participant does not exist in UpdateParticipant. (#3543) 75d0e18e Implement SIP update API. (#3141) e118aff1 Fire track subscribed when the subscriber connected (#3540) 13417c01 Send mute event only on change (#3537) 7f4c4597 Stubs for SIP update API. (#3533) fe673bb2 Send regressed codec upstream stats to analytics. (#3532) 8eb81388 Use a generation to counter to stop key frame seeder on codec change (#3531) 188470a2 Do not accept unsupported track type in AddTrack (#3530) 507fc9cf Do not instantiate 0 sized sequencer. (#3529) 20bddfea Clean up published track on participant removal. (#3527) 65d8aa28 Handle subscribe race with track close better. (#3526) a6cb00b3 Reduce seeder duration to 30s and also do not force send PLI. (#3525) c8233205 Add a key frame seeder in up track. (#3524) 0f61ff3a Remove redundant log (#3523) 7685cd25 Log ParticipantInit on signal start to get a picture of join params (#3522) ac9e62ef add server agent load threshold config (#3520) cd5d32f0 Add pID and connID to log context to make it easier to search using pID. (#3518) 2d9aa6dd Update api call info method (#3515) b3779a90 WebHookConfig (#3517) 6121b9af Check ForwardParticipant room name (#3514) 9a7c9442 mediatransportutil update (#3511) 50ab47c1 Log packet drops/forward. (#3510) 139d1b13 Add ForwardParticipant method to room service (#3507) 6c04909f Use atomic to store codec. (#3505) 7f6afe05 Prevent bind lock deadlock on muted. (#3504) 48063df5 load mime type before calling writeBlankFrameRTP (#3502) d2e6cd15 Do not bind lock across flush which could take time (#3501) 47896f50 Update protocol and IO service. (#3499) 1dc42eef Bump github.com/go-jose/go-jose/v3 from 3.0.3 to 3.0.4 (#3497) 3a35cbc4 Log migration complete only when coming from sync (#3496) c2f17a10 refactor: using slices.Contains to simplify the code (#3495) 01e51dbd fix: fix the wrong error return value (#3493) ff9115b2 Disable dd parser for vp8 if extension is not found (#3492) c3e06f05 Do not attempt to create objects for URL ingresses as the ingress service will do so (#3491) f0edfbba Fix receiver rtt/jitter. (#3487) 05dfd30d Take RTT and jitter from receiver view while reporting track stats for (#3483) 04ed5683 Don't issue TrackPublished/Unpublished event on migrated track (#3482) 1cffe30c Use a RED transformer to consolidate both RED -> Opus OR Opus -> RED (#3481) 591888f7 Fix missing RTCP sender report when forwarding RED as Opus. (#3480) git-subtree-dir: livekit-server git-subtree-split: 46c4309554d37d23ee8da88a8a7e02a68fba09c1
This commit is contained in:
+91
-678
@@ -15,181 +15,38 @@
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package sfu
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import (
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"errors"
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"io"
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"strings"
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"sync"
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"time"
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"github.com/pion/rtcp"
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"github.com/pion/webrtc/v4"
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"go.uber.org/atomic"
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"github.com/livekit/mediatransportutil/pkg/bucket"
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"github.com/livekit/protocol/codecs/mime"
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"github.com/livekit/protocol/livekit"
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"github.com/livekit/protocol/logger"
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"github.com/livekit/protocol/utils"
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"github.com/livekit/livekit-server/pkg/sfu/audio"
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"github.com/livekit/livekit-server/pkg/sfu/buffer"
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"github.com/livekit/livekit-server/pkg/sfu/connectionquality"
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"github.com/livekit/livekit-server/pkg/sfu/mime"
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dd "github.com/livekit/livekit-server/pkg/sfu/rtpextension/dependencydescriptor"
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"github.com/livekit/livekit-server/pkg/sfu/rtpstats"
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"github.com/livekit/livekit-server/pkg/sfu/streamtracker"
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)
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var (
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ErrReceiverClosed = errors.New("receiver closed")
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ErrDownTrackAlreadyExist = errors.New("DownTrack already exist")
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ErrBufferNotFound = errors.New("buffer not found")
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ErrDuplicateLayer = errors.New("duplicate layer")
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)
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// --------------------------------------
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type PLIThrottleConfig struct {
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LowQuality time.Duration `yaml:"low_quality,omitempty"`
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MidQuality time.Duration `yaml:"mid_quality,omitempty"`
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HighQuality time.Duration `yaml:"high_quality,omitempty"`
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}
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var (
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DefaultPLIThrottleConfig = PLIThrottleConfig{
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LowQuality: 500 * time.Millisecond,
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MidQuality: time.Second,
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HighQuality: time.Second,
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}
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)
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// --------------------------------------
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type AudioConfig struct {
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audio.AudioLevelConfig `yaml:",inline"`
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// enable red encoding downtrack for opus only audio up track
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ActiveREDEncoding bool `yaml:"active_red_encoding,omitempty"`
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// enable proxying weakest subscriber loss to publisher in RTCP Receiver Report
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EnableLossProxying bool `yaml:"enable_loss_proxying,omitempty"`
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}
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var (
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DefaultAudioConfig = AudioConfig{
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AudioLevelConfig: audio.DefaultAudioLevelConfig,
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}
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)
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// --------------------------------------
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type AudioLevelHandle func(level uint8, duration uint32)
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type Bitrates [buffer.DefaultMaxLayerSpatial + 1][buffer.DefaultMaxLayerTemporal + 1]int64
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type ReceiverCodecState int
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const (
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ReceiverCodecStateNormal ReceiverCodecState = iota
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ReceiverCodecStateSuspended
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ReceiverCodecStateInvalid
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)
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// TrackReceiver defines an interface receive media from remote peer
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type TrackReceiver interface {
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TrackID() livekit.TrackID
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StreamID() string
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// returns the initial codec of the receiver, it is determined by the track's codec
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// and will not change if the codec changes during the session (publisher changes codec)
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Codec() webrtc.RTPCodecParameters
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Mime() mime.MimeType
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HeaderExtensions() []webrtc.RTPHeaderExtensionParameter
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IsClosed() bool
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ReadRTP(buf []byte, layer uint8, esn uint64) (int, error)
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GetLayeredBitrate() ([]int32, Bitrates)
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GetAudioLevel() (float64, bool)
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SendPLI(layer int32, force bool)
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SetUpTrackPaused(paused bool)
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SetMaxExpectedSpatialLayer(layer int32)
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AddDownTrack(track TrackSender) error
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DeleteDownTrack(participantID livekit.ParticipantID)
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GetDownTracks() []TrackSender
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DebugInfo() map[string]interface{}
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TrackInfo() *livekit.TrackInfo
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UpdateTrackInfo(ti *livekit.TrackInfo)
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// Get primary receiver if this receiver represents a RED codec; otherwise it will return itself
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GetPrimaryReceiverForRed() TrackReceiver
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// Get red receiver for primary codec, used by forward red encodings for opus only codec
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GetRedReceiver() TrackReceiver
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GetTemporalLayerFpsForSpatial(layer int32) []float32
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GetTrackStats() *livekit.RTPStats
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// AddOnReady adds a function to be called when the receiver is ready, the callback
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// could be called immediately if the receiver is ready when the callback is added
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AddOnReady(func())
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AddOnCodecStateChange(func(webrtc.RTPCodecParameters, ReceiverCodecState))
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CodecState() ReceiverCodecState
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}
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type redPktWriteFunc func(pkt *buffer.ExtPacket, spatialLayer int32) int
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var _ TrackReceiver = (*WebRTCReceiver)(nil)
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// WebRTCReceiver receives a media track
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type WebRTCReceiver struct {
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logger logger.Logger
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*ReceiverBase
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pliThrottleConfig PLIThrottleConfig
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audioConfig AudioConfig
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trackID livekit.TrackID
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streamID string
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kind webrtc.RTPCodecType
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receiver *webrtc.RTPReceiver
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codec webrtc.RTPCodecParameters
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codecState ReceiverCodecState
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codecStateLock sync.Mutex
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onCodecStateChange []func(webrtc.RTPCodecParameters, ReceiverCodecState)
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isSVC bool
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isRED bool
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onCloseHandler func()
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closeOnce sync.Once
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closed atomic.Bool
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useTrackers bool
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trackInfo atomic.Pointer[livekit.TrackInfo]
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receiver *webrtc.RTPReceiver
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onCloseHandler func()
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onRTCP func([]rtcp.Packet)
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bufferMu sync.RWMutex
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buffers [buffer.DefaultMaxLayerSpatial + 1]*buffer.Buffer
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upTracks [buffer.DefaultMaxLayerSpatial + 1]TrackRemote
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rtt uint32
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lbThreshold int
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streamTrackerManager *StreamTrackerManager
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downTrackSpreader *DownTrackSpreader
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upTracksMu sync.Mutex
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upTracks [buffer.DefaultMaxLayerSpatial + 1]TrackRemote
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connectionStats *connectionquality.ConnectionStats
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onStatsUpdate func(w *WebRTCReceiver, stat *livekit.AnalyticsStat)
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onMaxLayerChange func(maxLayer int32)
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primaryReceiver atomic.Pointer[RedPrimaryReceiver]
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redReceiver atomic.Pointer[RedReceiver]
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redPktWriter atomic.Value // redPktWriteFunc
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forwardStats *ForwardStats
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onStatsUpdate func(w *WebRTCReceiver, stat *livekit.AnalyticsStat)
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}
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type ReceiverOpts func(w *WebRTCReceiver) *WebRTCReceiver
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@@ -197,7 +54,7 @@ type ReceiverOpts func(w *WebRTCReceiver) *WebRTCReceiver
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// WithPliThrottleConfig indicates minimum time(ms) between sending PLIs
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func WithPliThrottleConfig(pliThrottleConfig PLIThrottleConfig) ReceiverOpts {
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return func(w *WebRTCReceiver) *WebRTCReceiver {
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w.pliThrottleConfig = pliThrottleConfig
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w.ReceiverBase.SetPLIThrottleConfig(pliThrottleConfig)
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return w
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}
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}
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@@ -205,15 +62,14 @@ func WithPliThrottleConfig(pliThrottleConfig PLIThrottleConfig) ReceiverOpts {
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// WithAudioConfig sets up parameters for active speaker detection
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func WithAudioConfig(audioConfig AudioConfig) ReceiverOpts {
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return func(w *WebRTCReceiver) *WebRTCReceiver {
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w.audioConfig = audioConfig
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w.ReceiverBase.SetAudioConfig(audioConfig)
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return w
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}
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}
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// WithStreamTrackers enables StreamTracker use for simulcast
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func WithStreamTrackers() ReceiverOpts {
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func WithEnableRTPStreamRestartDetection(enable bool) ReceiverOpts {
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return func(w *WebRTCReceiver) *WebRTCReceiver {
|
||||
w.useTrackers = true
|
||||
w.ReceiverBase.SetEnableRTPStreamRestartDetection(enable)
|
||||
return w
|
||||
}
|
||||
}
|
||||
@@ -225,14 +81,14 @@ func WithStreamTrackers() ReceiverOpts {
|
||||
// Set to 0 (disabled) by default.
|
||||
func WithLoadBalanceThreshold(downTracks int) ReceiverOpts {
|
||||
return func(w *WebRTCReceiver) *WebRTCReceiver {
|
||||
w.lbThreshold = downTracks
|
||||
w.ReceiverBase.SetLBThreshold(downTracks)
|
||||
return w
|
||||
}
|
||||
}
|
||||
|
||||
func WithForwardStats(forwardStats *ForwardStats) ReceiverOpts {
|
||||
return func(w *WebRTCReceiver) *WebRTCReceiver {
|
||||
w.forwardStats = forwardStats
|
||||
w.ReceiverBase.SetForwardStats(forwardStats)
|
||||
return w
|
||||
}
|
||||
}
|
||||
@@ -248,243 +104,119 @@ func NewWebRTCReceiver(
|
||||
opts ...ReceiverOpts,
|
||||
) *WebRTCReceiver {
|
||||
w := &WebRTCReceiver{
|
||||
logger: logger,
|
||||
receiver: receiver,
|
||||
trackID: livekit.TrackID(track.ID()),
|
||||
streamID: track.StreamID(),
|
||||
codec: track.Codec(),
|
||||
codecState: ReceiverCodecStateNormal,
|
||||
kind: track.Kind(),
|
||||
onRTCP: onRTCP,
|
||||
isSVC: mime.IsMimeTypeStringSVC(track.Codec().MimeType),
|
||||
isRED: mime.IsMimeTypeStringRED(track.Codec().MimeType),
|
||||
receiver: receiver,
|
||||
onRTCP: onRTCP,
|
||||
}
|
||||
|
||||
w.ReceiverBase = NewReceiverBase(
|
||||
ReceiverBaseParams{
|
||||
TrackID: livekit.TrackID(track.ID()),
|
||||
StreamID: track.StreamID(),
|
||||
Kind: track.Kind(),
|
||||
Codec: track.Codec(),
|
||||
HeaderExtensions: receiver.GetParameters().HeaderExtensions,
|
||||
Logger: logger,
|
||||
StreamTrackerManagerConfig: streamTrackerManagerConfig,
|
||||
StreamTrackerManagerListener: w,
|
||||
IsSelfClosing: true,
|
||||
OnClosed: w.onClosed,
|
||||
},
|
||||
trackInfo,
|
||||
ReceiverCodecStateNormal,
|
||||
)
|
||||
|
||||
for _, opt := range opts {
|
||||
w = opt(w)
|
||||
}
|
||||
w.trackInfo.Store(utils.CloneProto(trackInfo))
|
||||
|
||||
w.downTrackSpreader = NewDownTrackSpreader(DownTrackSpreaderParams{
|
||||
Threshold: w.lbThreshold,
|
||||
Logger: logger,
|
||||
})
|
||||
|
||||
w.connectionStats = connectionquality.NewConnectionStats(connectionquality.ConnectionStatsParams{
|
||||
ReceiverProvider: w,
|
||||
Logger: w.logger.WithValues("direction", "up"),
|
||||
Logger: logger.WithValues("direction", "up"),
|
||||
})
|
||||
w.connectionStats.OnStatsUpdate(func(_cs *connectionquality.ConnectionStats, stat *livekit.AnalyticsStat) {
|
||||
if w.onStatsUpdate != nil {
|
||||
w.onStatsUpdate(w, stat)
|
||||
}
|
||||
})
|
||||
codec := track.Codec()
|
||||
w.connectionStats.Start(
|
||||
mime.NormalizeMimeType(w.codec.MimeType),
|
||||
mime.NormalizeMimeType(codec.MimeType),
|
||||
// TODO: technically not correct to declare FEC on when RED. Need the primary codec's fmtp line to check.
|
||||
mime.IsMimeTypeStringRED(w.codec.MimeType) || strings.Contains(strings.ToLower(w.codec.SDPFmtpLine), "useinbandfec=1"),
|
||||
mime.IsMimeTypeStringRED(codec.MimeType) || strings.Contains(strings.ToLower(codec.SDPFmtpLine), "useinbandfec=1"),
|
||||
)
|
||||
|
||||
w.streamTrackerManager = NewStreamTrackerManager(logger, trackInfo, w.isSVC, w.codec.ClockRate, streamTrackerManagerConfig)
|
||||
w.streamTrackerManager.SetListener(w)
|
||||
// SVC-TODO: Handle DD for non-SVC cases???
|
||||
if w.isSVC {
|
||||
for _, ext := range receiver.GetParameters().HeaderExtensions {
|
||||
if ext.URI == dd.ExtensionURI {
|
||||
w.streamTrackerManager.AddDependencyDescriptorTrackers()
|
||||
break
|
||||
}
|
||||
}
|
||||
}
|
||||
w.UpdateTrackInfo(trackInfo)
|
||||
|
||||
return w
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) TrackInfo() *livekit.TrackInfo {
|
||||
return w.trackInfo.Load()
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) UpdateTrackInfo(ti *livekit.TrackInfo) {
|
||||
w.trackInfo.Store(utils.CloneProto(ti))
|
||||
w.streamTrackerManager.UpdateTrackInfo(ti)
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) OnStatsUpdate(fn func(w *WebRTCReceiver, stat *livekit.AnalyticsStat)) {
|
||||
w.onStatsUpdate = fn
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) OnMaxLayerChange(fn func(maxLayer int32)) {
|
||||
w.bufferMu.Lock()
|
||||
w.onMaxLayerChange = fn
|
||||
w.bufferMu.Unlock()
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) getOnMaxLayerChange() func(maxLayer int32) {
|
||||
w.bufferMu.RLock()
|
||||
defer w.bufferMu.RUnlock()
|
||||
|
||||
return w.onMaxLayerChange
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) GetConnectionScoreAndQuality() (float32, livekit.ConnectionQuality) {
|
||||
return w.connectionStats.GetScoreAndQuality()
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) IsClosed() bool {
|
||||
return w.closed.Load()
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) SetRTT(rtt uint32) {
|
||||
w.bufferMu.Lock()
|
||||
if w.rtt == rtt {
|
||||
w.bufferMu.Unlock()
|
||||
return
|
||||
}
|
||||
|
||||
w.rtt = rtt
|
||||
buffers := w.buffers
|
||||
w.bufferMu.Unlock()
|
||||
|
||||
for _, buff := range buffers {
|
||||
if buff == nil {
|
||||
continue
|
||||
}
|
||||
|
||||
buff.SetRTT(rtt)
|
||||
}
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) StreamID() string {
|
||||
return w.streamID
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) TrackID() livekit.TrackID {
|
||||
return w.trackID
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) ssrc(layer int) uint32 {
|
||||
w.upTracksMu.Lock()
|
||||
defer w.upTracksMu.Unlock()
|
||||
|
||||
if track := w.upTracks[layer]; track != nil {
|
||||
return uint32(track.SSRC())
|
||||
}
|
||||
return 0
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) Codec() webrtc.RTPCodecParameters {
|
||||
return w.codec
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) Mime() mime.MimeType {
|
||||
return mime.NormalizeMimeType(w.codec.MimeType)
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) HeaderExtensions() []webrtc.RTPHeaderExtensionParameter {
|
||||
return w.receiver.GetParameters().HeaderExtensions
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) Kind() webrtc.RTPCodecType {
|
||||
return w.kind
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) AddUpTrack(track TrackRemote, buff *buffer.Buffer) error {
|
||||
if w.closed.Load() {
|
||||
if w.isClosed.Load() {
|
||||
return ErrReceiverClosed
|
||||
}
|
||||
|
||||
layer := int32(0)
|
||||
if w.Kind() == webrtc.RTPCodecTypeVideo && !w.isSVC {
|
||||
layer = buffer.RidToSpatialLayer(track.RID(), w.trackInfo.Load())
|
||||
if w.Kind() == webrtc.RTPCodecTypeVideo && w.videoLayerMode != livekit.VideoLayer_MULTIPLE_SPATIAL_LAYERS_PER_STREAM {
|
||||
layer = buffer.GetSpatialLayerForRid(w.Mime(), track.RID(), w.ReceiverBase.TrackInfo())
|
||||
}
|
||||
buff.SetLogger(w.logger.WithValues("layer", layer))
|
||||
buff.SetAudioLevelParams(audio.AudioLevelParams{
|
||||
Config: w.audioConfig.AudioLevelConfig,
|
||||
})
|
||||
buff.SetAudioLossProxying(w.audioConfig.EnableLossProxying)
|
||||
buff.OnRtcpFeedback(w.sendRTCP)
|
||||
buff.OnRtcpSenderReport(func() {
|
||||
srData := buff.GetSenderReportData()
|
||||
w.downTrackSpreader.Broadcast(func(dt TrackSender) {
|
||||
_ = dt.HandleRTCPSenderReportData(w.codec.PayloadType, w.isSVC, layer, srData)
|
||||
})
|
||||
})
|
||||
|
||||
if w.Kind() == webrtc.RTPCodecTypeVideo && layer == 0 {
|
||||
buff.OnCodecChange(w.handleCodecChange)
|
||||
if layer < 0 {
|
||||
w.ReceiverBase.Logger().Warnw(
|
||||
"invalid layer", nil,
|
||||
"rid", track.RID(),
|
||||
"trackInfo", logger.Proto(w.ReceiverBase.TrackInfo()),
|
||||
)
|
||||
return ErrInvalidLayer
|
||||
}
|
||||
|
||||
var duration time.Duration
|
||||
switch layer {
|
||||
case 2:
|
||||
duration = w.pliThrottleConfig.HighQuality
|
||||
case 1:
|
||||
duration = w.pliThrottleConfig.MidQuality
|
||||
case 0:
|
||||
duration = w.pliThrottleConfig.LowQuality
|
||||
default:
|
||||
duration = w.pliThrottleConfig.MidQuality
|
||||
}
|
||||
if duration != 0 {
|
||||
buff.SetPLIThrottle(duration.Nanoseconds())
|
||||
}
|
||||
|
||||
w.bufferMu.Lock()
|
||||
w.upTracksMu.Lock()
|
||||
if w.upTracks[layer] != nil {
|
||||
w.bufferMu.Unlock()
|
||||
w.upTracksMu.Unlock()
|
||||
return ErrDuplicateLayer
|
||||
}
|
||||
w.upTracks[layer] = track
|
||||
w.buffers[layer] = buff
|
||||
rtt := w.rtt
|
||||
w.bufferMu.Unlock()
|
||||
w.upTracksMu.Unlock()
|
||||
|
||||
buff.SetRTT(rtt)
|
||||
buff.SetPaused(w.streamTrackerManager.IsPaused())
|
||||
|
||||
if w.Kind() == webrtc.RTPCodecTypeVideo && w.useTrackers {
|
||||
w.streamTrackerManager.AddTracker(layer)
|
||||
}
|
||||
|
||||
go w.forwardRTP(layer, buff)
|
||||
w.ReceiverBase.AddBuffer(buff, layer)
|
||||
buff.OnRtcpFeedback(w.sendRTCP)
|
||||
w.ReceiverBase.StartBuffer(buff, layer)
|
||||
return nil
|
||||
}
|
||||
|
||||
// SetUpTrackPaused indicates upstream will not be sending any data.
|
||||
// this will reflect the "muted" status and will pause streamtracker to ensure we don't turn off
|
||||
// the layer
|
||||
func (w *WebRTCReceiver) SetUpTrackPaused(paused bool) {
|
||||
w.streamTrackerManager.SetPaused(paused)
|
||||
func (w *WebRTCReceiver) NumUpTracks() int {
|
||||
numUpTracks := 0
|
||||
|
||||
w.bufferMu.RLock()
|
||||
for _, buff := range w.buffers {
|
||||
if buff == nil {
|
||||
continue
|
||||
w.upTracksMu.Lock()
|
||||
for _, track := range w.upTracks {
|
||||
if track != nil {
|
||||
numUpTracks++
|
||||
}
|
||||
|
||||
buff.SetPaused(paused)
|
||||
}
|
||||
w.bufferMu.RUnlock()
|
||||
w.upTracksMu.Unlock()
|
||||
|
||||
w.connectionStats.UpdateMute(paused)
|
||||
return numUpTracks
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) AddDownTrack(track TrackSender) error {
|
||||
if w.closed.Load() {
|
||||
return ErrReceiverClosed
|
||||
}
|
||||
|
||||
if w.downTrackSpreader.HasDownTrack(track.SubscriberID()) {
|
||||
w.logger.Infow("subscriberID already exists, replacing downtrack", "subscriberID", track.SubscriberID())
|
||||
}
|
||||
|
||||
track.UpTrackMaxPublishedLayerChange(w.streamTrackerManager.GetMaxPublishedLayer())
|
||||
track.UpTrackMaxTemporalLayerSeenChange(w.streamTrackerManager.GetMaxTemporalLayerSeen())
|
||||
|
||||
w.downTrackSpreader.Store(track)
|
||||
w.logger.Debugw("downtrack added", "subscriberID", track.SubscriberID())
|
||||
return nil
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) GetDownTracks() []TrackSender {
|
||||
return w.downTrackSpreader.GetDownTracks()
|
||||
func (w *WebRTCReceiver) UpdateTrackInfo(ti *livekit.TrackInfo) {
|
||||
w.ReceiverBase.UpdateTrackInfo(ti)
|
||||
w.connectionStats.UpdateMute(ti.GetMuted())
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) notifyMaxExpectedLayer(layer int32) {
|
||||
@@ -499,9 +231,8 @@ func (w *WebRTCReceiver) notifyMaxExpectedLayer(layer int32) {
|
||||
}
|
||||
|
||||
expectedBitrate := int64(0)
|
||||
for _, vl := range ti.Layers {
|
||||
l := buffer.VideoQualityToSpatialLayer(vl.Quality, ti)
|
||||
if l <= layer {
|
||||
for _, vl := range buffer.GetVideoLayersForMimeType(w.Mime(), ti) {
|
||||
if vl.SpatialLayer <= layer {
|
||||
expectedBitrate += int64(vl.Bitrate)
|
||||
}
|
||||
}
|
||||
@@ -510,89 +241,54 @@ func (w *WebRTCReceiver) notifyMaxExpectedLayer(layer int32) {
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) SetMaxExpectedSpatialLayer(layer int32) {
|
||||
w.streamTrackerManager.SetMaxExpectedSpatialLayer(layer)
|
||||
w.ReceiverBase.SetMaxExpectedSpatialLayer(layer)
|
||||
|
||||
w.notifyMaxExpectedLayer(layer)
|
||||
|
||||
if layer == buffer.InvalidLayerSpatial {
|
||||
w.connectionStats.UpdateLayerMute(true)
|
||||
} else {
|
||||
w.connectionStats.UpdateLayerMute(false)
|
||||
w.connectionStats.AddLayerTransition(w.streamTrackerManager.DistanceToDesired())
|
||||
w.connectionStats.AddLayerTransition(w.ReceiverBase.StreamTrackerManager().DistanceToDesired())
|
||||
}
|
||||
}
|
||||
|
||||
// StreamTrackerManagerListener.OnAvailableLayersChanged
|
||||
func (w *WebRTCReceiver) OnAvailableLayersChanged() {
|
||||
w.downTrackSpreader.Broadcast(func(dt TrackSender) {
|
||||
dt.UpTrackLayersChange()
|
||||
})
|
||||
|
||||
w.connectionStats.AddLayerTransition(w.streamTrackerManager.DistanceToDesired())
|
||||
w.connectionStats.AddLayerTransition(w.ReceiverBase.StreamTrackerManager().DistanceToDesired())
|
||||
}
|
||||
|
||||
// StreamTrackerManagerListener.OnBitrateAvailabilityChanged
|
||||
func (w *WebRTCReceiver) OnBitrateAvailabilityChanged() {
|
||||
w.downTrackSpreader.Broadcast(func(dt TrackSender) {
|
||||
dt.UpTrackBitrateAvailabilityChange()
|
||||
})
|
||||
}
|
||||
|
||||
// StreamTrackerManagerListener.OnMaxPublishedLayerChanged
|
||||
func (w *WebRTCReceiver) OnMaxPublishedLayerChanged(maxPublishedLayer int32) {
|
||||
w.downTrackSpreader.Broadcast(func(dt TrackSender) {
|
||||
dt.UpTrackMaxPublishedLayerChange(maxPublishedLayer)
|
||||
})
|
||||
|
||||
w.notifyMaxExpectedLayer(maxPublishedLayer)
|
||||
w.connectionStats.AddLayerTransition(w.streamTrackerManager.DistanceToDesired())
|
||||
w.connectionStats.AddLayerTransition(w.ReceiverBase.StreamTrackerManager().DistanceToDesired())
|
||||
}
|
||||
|
||||
// StreamTrackerManagerListener.OnMaxTemporalLayerSeenChanged
|
||||
func (w *WebRTCReceiver) OnMaxTemporalLayerSeenChanged(maxTemporalLayerSeen int32) {
|
||||
w.downTrackSpreader.Broadcast(func(dt TrackSender) {
|
||||
dt.UpTrackMaxTemporalLayerSeenChange(maxTemporalLayerSeen)
|
||||
})
|
||||
|
||||
w.connectionStats.AddLayerTransition(w.streamTrackerManager.DistanceToDesired())
|
||||
w.connectionStats.AddLayerTransition(w.ReceiverBase.StreamTrackerManager().DistanceToDesired())
|
||||
}
|
||||
|
||||
// StreamTrackerManagerListener.OnMaxAvailableLayerChanged
|
||||
func (w *WebRTCReceiver) OnMaxAvailableLayerChanged(maxAvailableLayer int32) {
|
||||
if onMaxLayerChange := w.getOnMaxLayerChange(); onMaxLayerChange != nil {
|
||||
onMaxLayerChange(maxAvailableLayer)
|
||||
}
|
||||
}
|
||||
|
||||
// StreamTrackerManagerListener.OnBitrateReport
|
||||
func (w *WebRTCReceiver) OnBitrateReport(availableLayers []int32, bitrates Bitrates) {
|
||||
w.downTrackSpreader.Broadcast(func(dt TrackSender) {
|
||||
dt.UpTrackBitrateReport(availableLayers, bitrates)
|
||||
})
|
||||
|
||||
w.connectionStats.AddLayerTransition(w.streamTrackerManager.DistanceToDesired())
|
||||
w.connectionStats.AddLayerTransition(w.ReceiverBase.StreamTrackerManager().DistanceToDesired())
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) GetLayeredBitrate() ([]int32, Bitrates) {
|
||||
return w.streamTrackerManager.GetLayeredBitrate()
|
||||
}
|
||||
|
||||
// OnCloseHandler method to be called on remote tracked removed
|
||||
// OnCloseHandler method to be called on remote track removed
|
||||
func (w *WebRTCReceiver) OnCloseHandler(fn func()) {
|
||||
w.onCloseHandler = fn
|
||||
}
|
||||
|
||||
// DeleteDownTrack removes a DownTrack from a Receiver
|
||||
func (w *WebRTCReceiver) DeleteDownTrack(subscriberID livekit.ParticipantID) {
|
||||
if w.closed.Load() {
|
||||
return
|
||||
}
|
||||
|
||||
w.downTrackSpreader.Free(subscriberID)
|
||||
w.logger.Debugw("downtrack deleted", "subscriberID", subscriberID)
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) sendRTCP(packets []rtcp.Packet) {
|
||||
if packets == nil || w.closed.Load() {
|
||||
if packets == nil || w.isClosed.Load() {
|
||||
return
|
||||
}
|
||||
|
||||
@@ -601,93 +297,10 @@ func (w *WebRTCReceiver) sendRTCP(packets []rtcp.Packet) {
|
||||
}
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) SendPLI(layer int32, force bool) {
|
||||
// SVC-TODO : should send LRR (Layer Refresh Request) instead of PLI
|
||||
buff := w.getBuffer(layer)
|
||||
if buff == nil {
|
||||
return
|
||||
}
|
||||
|
||||
buff.SendPLI(force)
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) getBuffer(layer int32) *buffer.Buffer {
|
||||
w.bufferMu.RLock()
|
||||
defer w.bufferMu.RUnlock()
|
||||
|
||||
return w.getBufferLocked(layer)
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) getBufferLocked(layer int32) *buffer.Buffer {
|
||||
// for svc codecs, use layer = 0 always.
|
||||
// spatial layers are in-built and handled by single buffer
|
||||
if w.isSVC {
|
||||
layer = 0
|
||||
}
|
||||
|
||||
if layer < 0 || int(layer) >= len(w.buffers) {
|
||||
return nil
|
||||
}
|
||||
|
||||
return w.buffers[layer]
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) ReadRTP(buf []byte, layer uint8, esn uint64) (int, error) {
|
||||
b := w.getBuffer(int32(layer))
|
||||
if b == nil {
|
||||
return 0, ErrBufferNotFound
|
||||
}
|
||||
|
||||
return b.GetPacket(buf, esn)
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) GetTrackStats() *livekit.RTPStats {
|
||||
w.bufferMu.RLock()
|
||||
defer w.bufferMu.RUnlock()
|
||||
|
||||
stats := make([]*livekit.RTPStats, 0, len(w.buffers))
|
||||
for _, buff := range w.buffers {
|
||||
if buff == nil {
|
||||
continue
|
||||
}
|
||||
|
||||
sswl := buff.GetStats()
|
||||
if sswl == nil {
|
||||
continue
|
||||
}
|
||||
|
||||
stats = append(stats, sswl)
|
||||
}
|
||||
|
||||
return rtpstats.AggregateRTPStats(stats)
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) GetAudioLevel() (float64, bool) {
|
||||
if w.Kind() == webrtc.RTPCodecTypeVideo {
|
||||
return 0, false
|
||||
}
|
||||
|
||||
w.bufferMu.RLock()
|
||||
defer w.bufferMu.RUnlock()
|
||||
|
||||
for _, buff := range w.buffers {
|
||||
if buff == nil {
|
||||
continue
|
||||
}
|
||||
|
||||
return buff.GetAudioLevel()
|
||||
}
|
||||
|
||||
return 0, false
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) GetDeltaStats() map[uint32]*buffer.StreamStatsWithLayers {
|
||||
w.bufferMu.RLock()
|
||||
defer w.bufferMu.RUnlock()
|
||||
|
||||
deltaStats := make(map[uint32]*buffer.StreamStatsWithLayers, len(w.buffers))
|
||||
|
||||
for layer, buff := range w.buffers {
|
||||
buffers := w.ReceiverBase.GetAllBuffers()
|
||||
deltaStats := make(map[uint32]*buffer.StreamStatsWithLayers, len(buffers))
|
||||
for layer, buff := range buffers {
|
||||
if buff == nil {
|
||||
continue
|
||||
}
|
||||
@@ -709,11 +322,9 @@ func (w *WebRTCReceiver) GetDeltaStats() map[uint32]*buffer.StreamStatsWithLayer
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) GetLastSenderReportTime() time.Time {
|
||||
w.bufferMu.RLock()
|
||||
defer w.bufferMu.RUnlock()
|
||||
|
||||
buffers := w.ReceiverBase.GetAllBuffers()
|
||||
latestSRTime := time.Time{}
|
||||
for _, buff := range w.buffers {
|
||||
for _, buff := range buffers {
|
||||
if buff == nil {
|
||||
continue
|
||||
}
|
||||
@@ -727,120 +338,22 @@ func (w *WebRTCReceiver) GetLastSenderReportTime() time.Time {
|
||||
return latestSRTime
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) forwardRTP(layer int32, buff *buffer.Buffer) {
|
||||
defer func() {
|
||||
w.closeOnce.Do(func() {
|
||||
w.closed.Store(true)
|
||||
w.closeTracks()
|
||||
if pr := w.primaryReceiver.Load(); pr != nil {
|
||||
pr.Close()
|
||||
}
|
||||
if pr := w.redReceiver.Load(); pr != nil {
|
||||
pr.Close()
|
||||
}
|
||||
})
|
||||
|
||||
w.streamTrackerManager.RemoveTracker(layer)
|
||||
if w.isSVC {
|
||||
w.streamTrackerManager.RemoveAllTrackers()
|
||||
}
|
||||
}()
|
||||
|
||||
var spatialTrackers [buffer.DefaultMaxLayerSpatial + 1]streamtracker.StreamTrackerWorker
|
||||
if layer < 0 || int(layer) >= len(spatialTrackers) {
|
||||
w.logger.Errorw("invalid layer", nil, "layer", layer)
|
||||
return
|
||||
}
|
||||
spatialTrackers[layer] = w.streamTrackerManager.GetTracker(layer)
|
||||
|
||||
pktBuf := make([]byte, bucket.MaxPktSize)
|
||||
for {
|
||||
pkt, err := buff.ReadExtended(pktBuf)
|
||||
if err == io.EOF {
|
||||
return
|
||||
}
|
||||
|
||||
if pkt.Packet.PayloadType != uint8(w.codec.PayloadType) {
|
||||
// drop packets as we don't support codec fallback directly
|
||||
continue
|
||||
}
|
||||
|
||||
spatialLayer := layer
|
||||
if pkt.Spatial >= 0 {
|
||||
// svc packet, take spatial layer info from packet
|
||||
spatialLayer = pkt.Spatial
|
||||
}
|
||||
if int(spatialLayer) >= len(spatialTrackers) {
|
||||
w.logger.Errorw(
|
||||
"unexpected spatial layer", nil,
|
||||
"spatialLayer", spatialLayer,
|
||||
"pktSpatialLayer", pkt.Spatial,
|
||||
)
|
||||
continue
|
||||
}
|
||||
|
||||
writeCount := w.downTrackSpreader.Broadcast(func(dt TrackSender) {
|
||||
_ = dt.WriteRTP(pkt, spatialLayer)
|
||||
})
|
||||
|
||||
if f := w.redPktWriter.Load(); f != nil {
|
||||
writeCount += f.(redPktWriteFunc)(pkt, spatialLayer)
|
||||
}
|
||||
|
||||
// track delay/jitter
|
||||
if writeCount > 0 && w.forwardStats != nil {
|
||||
w.forwardStats.Update(pkt.Arrival, time.Now().UnixNano())
|
||||
}
|
||||
|
||||
// track video layers
|
||||
if w.Kind() == webrtc.RTPCodecTypeVideo {
|
||||
if spatialTrackers[spatialLayer] == nil {
|
||||
spatialTrackers[spatialLayer] = w.streamTrackerManager.GetTracker(spatialLayer)
|
||||
if spatialTrackers[spatialLayer] == nil {
|
||||
spatialTrackers[spatialLayer] = w.streamTrackerManager.AddTracker(spatialLayer)
|
||||
}
|
||||
}
|
||||
if spatialTrackers[spatialLayer] != nil {
|
||||
spatialTrackers[spatialLayer].Observe(
|
||||
pkt.Temporal,
|
||||
len(pkt.RawPacket),
|
||||
len(pkt.Packet.Payload),
|
||||
pkt.Packet.Marker,
|
||||
pkt.Packet.Timestamp,
|
||||
pkt.DependencyDescriptor,
|
||||
)
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// closeTracks close all tracks from Receiver
|
||||
func (w *WebRTCReceiver) closeTracks() {
|
||||
func (w *WebRTCReceiver) onClosed() {
|
||||
w.connectionStats.Close()
|
||||
w.streamTrackerManager.Close()
|
||||
|
||||
closeTrackSenders(w.downTrackSpreader.ResetAndGetDownTracks())
|
||||
|
||||
if w.onCloseHandler != nil {
|
||||
w.onCloseHandler()
|
||||
}
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) DebugInfo() map[string]interface{} {
|
||||
isSimulcast := !w.isSVC
|
||||
if ti := w.trackInfo.Load(); ti != nil {
|
||||
isSimulcast = isSimulcast && len(ti.Layers) > 1
|
||||
}
|
||||
info := map[string]interface{}{
|
||||
"SVC": w.isSVC,
|
||||
"Simulcast": isSimulcast,
|
||||
}
|
||||
func (w *WebRTCReceiver) DebugInfo() map[string]any {
|
||||
info := w.ReceiverBase.DebugInfo()
|
||||
|
||||
w.bufferMu.RLock()
|
||||
upTrackInfo := make([]map[string]interface{}, 0, len(w.upTracks))
|
||||
w.upTracksMu.Lock()
|
||||
upTrackInfo := make([]map[string]any, 0, len(w.upTracks))
|
||||
for layer, ut := range w.upTracks {
|
||||
if ut != nil {
|
||||
upTrackInfo = append(upTrackInfo, map[string]interface{}{
|
||||
upTrackInfo = append(upTrackInfo, map[string]any{
|
||||
"Layer": layer,
|
||||
"SSRC": ut.SSRC(),
|
||||
"Msid": ut.Msid(),
|
||||
@@ -848,110 +361,10 @@ func (w *WebRTCReceiver) DebugInfo() map[string]interface{} {
|
||||
})
|
||||
}
|
||||
}
|
||||
w.bufferMu.RUnlock()
|
||||
w.upTracksMu.Unlock()
|
||||
info["UpTracks"] = upTrackInfo
|
||||
|
||||
return info
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) GetPrimaryReceiverForRed() TrackReceiver {
|
||||
if !w.isRED || w.closed.Load() {
|
||||
return w
|
||||
}
|
||||
|
||||
if w.primaryReceiver.Load() == nil {
|
||||
pr := NewRedPrimaryReceiver(w, DownTrackSpreaderParams{
|
||||
Threshold: w.lbThreshold,
|
||||
Logger: w.logger,
|
||||
})
|
||||
if w.primaryReceiver.CompareAndSwap(nil, pr) {
|
||||
w.redPktWriter.Store(redPktWriteFunc(pr.ForwardRTP))
|
||||
}
|
||||
}
|
||||
return w.primaryReceiver.Load()
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) GetRedReceiver() TrackReceiver {
|
||||
if w.isRED || w.closed.Load() {
|
||||
return w
|
||||
}
|
||||
|
||||
if w.redReceiver.Load() == nil {
|
||||
pr := NewRedReceiver(w, DownTrackSpreaderParams{
|
||||
Threshold: w.lbThreshold,
|
||||
Logger: w.logger,
|
||||
})
|
||||
if w.redReceiver.CompareAndSwap(nil, pr) {
|
||||
w.redPktWriter.Store(redPktWriteFunc(pr.ForwardRTP))
|
||||
}
|
||||
}
|
||||
return w.redReceiver.Load()
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) GetTemporalLayerFpsForSpatial(layer int32) []float32 {
|
||||
b := w.getBuffer(layer)
|
||||
if b == nil {
|
||||
return nil
|
||||
}
|
||||
|
||||
if !w.isSVC {
|
||||
return b.GetTemporalLayerFpsForSpatial(0)
|
||||
}
|
||||
|
||||
return b.GetTemporalLayerFpsForSpatial(layer)
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) AddOnReady(fn func()) {
|
||||
// webRTCReceiver is always ready after created
|
||||
fn()
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) handleCodecChange(newCodec webrtc.RTPCodecParameters) {
|
||||
// we don't support the codec fallback directly, set the codec state to invalid once it happens
|
||||
w.SetCodecState(ReceiverCodecStateInvalid)
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) AddOnCodecStateChange(f func(webrtc.RTPCodecParameters, ReceiverCodecState)) {
|
||||
w.codecStateLock.Lock()
|
||||
w.onCodecStateChange = append(w.onCodecStateChange, f)
|
||||
w.codecStateLock.Unlock()
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) CodecState() ReceiverCodecState {
|
||||
w.codecStateLock.Lock()
|
||||
defer w.codecStateLock.Unlock()
|
||||
|
||||
return w.codecState
|
||||
}
|
||||
|
||||
func (w *WebRTCReceiver) SetCodecState(state ReceiverCodecState) {
|
||||
w.codecStateLock.Lock()
|
||||
if w.codecState == state || w.codecState == ReceiverCodecStateInvalid {
|
||||
w.codecStateLock.Unlock()
|
||||
return
|
||||
}
|
||||
|
||||
w.codecState = state
|
||||
fns := w.onCodecStateChange
|
||||
w.codecStateLock.Unlock()
|
||||
|
||||
for _, f := range fns {
|
||||
f(w.codec, state)
|
||||
}
|
||||
}
|
||||
|
||||
// -----------------------------------------------------------
|
||||
|
||||
// closes all track senders in parallel, returns when all are closed
|
||||
func closeTrackSenders(senders []TrackSender) {
|
||||
wg := sync.WaitGroup{}
|
||||
for _, dt := range senders {
|
||||
dt := dt
|
||||
wg.Add(1)
|
||||
go func() {
|
||||
defer wg.Done()
|
||||
dt.Close()
|
||||
}()
|
||||
}
|
||||
wg.Wait()
|
||||
}
|
||||
|
||||
Reference in New Issue
Block a user