Skip to content

alphartc
Compare
Choose a tag to compare
@Pterosaur Pterosaur released this 29 Mar 15:04
20210329.6
79d72b5
This commit was created on GitHub.com and signed with GitHub’s verified signature. The key has expired.
GPG key ID: 4AEE18F83AFDEB23
Expired
Learn about vigilant mode.

Changes:

  • 79d72b5 Add github release (#70)
  • b68237f Rename peerconnection_serverless (#69)
  • 7b4e337 Fix README typo (#68)
  • 31dc105 Add python interface to bandwidth estimator (#67)
  • f6d3b91 Fix bug: store a stack pointer at stats collection (#65)
  • 7edbe87 Update the readme to include all members. (#63)
  • 0f26786 Fix compile issue if the user is root (#61)
  • 8fffae6 Fix: Missing bwe setting from alphaRTC (#57)
  • fad8291 Add LogSink: Add FileLogSink to peerconnection (#56)
  • c24ca8b Polish Makefile to support AlphaRTC as a submodule (#49)
See More
  • 9ca3356 Add gn build flag to compile release version (#37)
  • 91fda03 Merge pull request #48 from Pterosaur/sync_upstream
  • 55c58d0 Merge branch 'upstream/webrtc' into sync_upstream
  • e0f253d Pass arguments of Makefile into Docker (#43)
  • 3feb580 Generate log at startup (#34)
  • 3d4204b Add user to Dockerfile (#41)
  • 9f6b9b7 Sleep the thread for 10ms in while loop to free up CPU (#39)
  • 4ee361e Fix a bug causing release build to fail (#36)
  • fec0471 Add option to output log to files (#32)
  • 579e121 #11 Add Windows Support (#24)
  • 2558672 Fix typo and add a new badge
  • bb8509e Add Corpus to demostrate usage (#17)
  • b32e1e3 Remove Unnecessary Readme Files (#14)
  • d4b2ff2 Fix typo: miss a comma in Readme
  • c169ae2 Revise Readme (#13)
  • cfaa6fe Revise alphacc_config: Added optional fields (#10)
  • 137a3a7 Revise .gclient: Switch git url from Azure DevOps to GitHub (#6)
  • 31743fc Merged PR 10297: OnnxInfer: Revise onnxinfer so files targeting Ubuntu 18.04
  • 0eb3812 Merged PR 10296: Add standalone build script
  • 35f7dd7 Merged PR 10285: Bandwidth Estimation: backport ONNXInfer to AlphaRTC
  • 5f98a8a Merged PR 10261: Application: Port AlphaRTC to Linux
  • 1546c9c Merged PR 10237: Application: Port AlphaRTC to Linux
  • a5e3626 Merged PR 10163: Application: Add Feature - Output AV streams to filesystem
  • 886863b Merged PR 10104: AlphaRTC: Revise Readme - change serverless-peerconnection-tutorial
  • e3d11db Merged PR 10014: AlphaRTC: Add Feature - Remove Redis interface and log data results into file
  • 4564ac3 Application: Fix Bugs - fix bugs for running in WAN of Serverless peerconnection
  • 47452d6 Merged PR 9975: AlphaRTC: Add Feature - Add OutputToFile() interface in logging
  • 2b7d969 RtpVideoSender: remove lock recursions. [ #31262 ]
  • e1b5264 RtpSenderEgress: remove lock recursions. [ #31261 ]
  • 02ba1d2 AudioProcessingImpl: remove lock recursions. [ #31260 ]
  • 28685dc Revert "Make sure that "current" rtc::Thread instances are always current for TaskQueueBase." [ #31254, #31259 ]
  • af1b9ce Revert "Revert back to using the task_queue_ for guarding access." [ #31256, #31258 ]
  • 87a6e5a objc: Export RTCStatistics and RTCStatisticsReport [ #31257 ]
  • 475006d Revert back to using the task_queue_ for guarding access. [ #31256 ]
  • 9775581 Reland "Add an optional override for AudioRecord device" [ #31130, #31255 ]
  • 46b3bc6 Make sure that "current" rtc::Thread instances are always current for TaskQueueBase. [ #31254 ]
  • f8cb70a Roll chromium_revision 9330db6b76..8ffd72401d (768590:768712) [ #31253 ]
  • a0a4480 Migrate CallStats and RtpStreamsSynchronizer timers over to RepeatingTask [ #31252 ]
  • 6ee6793 Reland "Delete unused code to handle posix signals in PhysicalSocketServer" [ #31237, #31251 ]
  • 2e2f674 Add VideoStreamEncoder tests for DegredationPreference switching. [ #31250 ]
  • 2f4b010 docs: Improve Android testing docs [ #31249 ]
  • b088a95 Roll chromium_revision 1f82a0280b..9330db6b76 (768300:768590) [ #31248 ]
  • 47173c9 Hunch that something might be broken wrt Thread and TQ [ #31247 ]
  • adaec45 Removes RepairedRtpStreamId from overhead calculation. [ #31246 ]
  • da76ad3 Roll chromium_revision 7a46078a3f..1f82a0280b (768198:768300) [ #31245 ]
  • ec02eef Android: Remove min_sdk_version in GN (reland) [ #31244 ]
  • 5bf60e4 VideoStreamEncoderTest: Wait for QP usage handled callback event. [ #31243 ]
  • 611fba4 Mark construction time members of PhysicalSocketServer as const [ #31242 ]
  • a98cea8 Remove the PendingTaskSafetyFlag::Pointer type add ScopedTaskSafety. [ #31241 ]
  • ff88a64 Revert "Delete unused code to handle posix signals in PhysicalSocketServer" [ #31237, #31240 ]
  • 21433ca Add field trial to let idle connection live longer than 30s [ #31239 ]
  • 3476e12 Revert "Add an optional override for AudioRecord device" [ #31130, #31238 ]
  • d2490ae Delete unused code to handle posix signals in PhysicalSocketServer [ #31237 ]
  • 84afe46 Don't await adaptation after deg-pref change [ #31236 ]
  • 1d6e70f Roll chromium_revision a081fb662a..7a46078a3f (768098:768198) [ #31235 ]
  • 95d40ee Remove unused VideoStream::balanced_settings() [ #31234 ]
  • 2105d64 Add field trial for using different network cost cellular types [ #31233 ]
  • f331981 Separate capturing device index from VideoConfig [ #31232 ]
  • 381d109 [Adaptation] Move adaptation logic to a separate task queue. [ #31231 ]
  • 1190f49 Roll chromium_revision cb4a800cb3..a081fb662a (767931:768098) [ #31230 ]
  • 2aa9356 Reset frame queue in ScreenCapturerX11::SelectSource to fix issues with different sized monitors. [ #31229 ]
  • b856dc1 Remove VideoGeneratorType from pc framework test api. [ #31228 ]
  • 72b7499 Roll chromium_revision 35425e2e59..cb4a800cb3 (767819:767931) [ #31227 ]
  • f531c49 Roll chromium_revision ceb47b836c..35425e2e59 (767701:767819) [ #31226 ]
  • 1aec2bf reorder sdes suites to not prefer gcm [ #31225 ]
  • 8c7384c Add field trial for surfacing cellular types in AndroidNetworkMonitor [ #31224 ]
  • cd1ba94 Roll chromium_revision a775f4aeaa..ceb47b836c (759199:767701) [ #31223 ]
  • b0bd070 Surface ResidualEchoDetector creation to API [ #31222 ]
  • 5c1a565 Revert "Disable PipeWire on Chromecast builds." [ #31205, #31221 ]
  • b28bc83 [Adaptation] Make Manager's Resources not depend on encoder queue. [ #31220 ]
  • 9438ddc [Adaptation] Give QualityScaler a pending callbacks queue and unittests [ #31219 ]
  • f9ed56b Add ability to set custom RtpEncodingParameters for each simulcast stream in PC framework [ #31218 ]
  • c55516d [Adaptation] Make Resources reference counted and add more DCHECKs. [ #31217 ]
  • cc57b93 Make video quality analyzer compatible with real SFU in the network [ #31216 ]
  • baa2c83 Introduce ability to set peer name for PC level tests [ #31215 ]
  • bf46cfe Refactors send rate statistics in RtpSenderEgress [ #31214 ]
  • 8e321cd [Adaptation] Make QuailtyScalerResourse to report underuse if quality scaling is off [ #31213 ]
  • a270250 [Adaptation] Disable inital frame drop for simuclast/svc [ #31212 ]
  • 6efc14b VideoTrackSourceInterface: make some newly introduced methods pure virtual. [ #31211 ]
  • a54ba4c Make video_loopback work with av1 [ #31210 ]
  • 3a65dba Revert "Removes lock release in PacedSender callback." [ #31206, #31209 ]
  • ffd0a84 Handle OnRttUpdate in ReceiveStatisticsProxy. [ #31208 ]
  • 09e9a83 Change the way that AecDumps are created in APM [ #31207 ]
  • 6b9c60b Removes lock release in PacedSender callback. [ #31206 ]
  • cc8c078 Disable PipeWire on Chromecast builds. [ #31205 ]
  • 1c33075 Trigger bots again. [ #31204 ]
  • 822a874 Switch CallStats to TQ interface + callbacks on the worker thread. [ #31203 ]
  • 674b0c8 Move ReceiveStatisticsProxy stats variables to the worker thread. [ #31202 ]
  • d93bf12 Call OnDecodedFrame asynchronously on the worker thread. [ #31201 ]
  • ad84d02 Remove locking from RtpStreamsSynchronizer. [ #31200 ]
  • d7e08c8 Move processing of frame meta data for OnFrame/OnRenderedFrame to the worker thread [ #31199 ]
  • 67ecb68 Trigger bots [ #31198 ]
  • 6a871d3 Revert "Remove playout delay lock." [ #31193, #31197 ]
  • 3580706 Add a RunLoop to RtpReplayer to fix fuzzers [ #31196 ]
  • fc11519 Cleanup mocks in api/test [ #31195 ]
  • 74ef940 Stop pulling binutils from WebRTC DEPS. [ #31194 ]
  • c623495 Remove playout delay lock. [ #31193 ]
  • 33d81a0 Keep OpenH264 iMaxBitrate unspecified. [ #31192 ]
  • 2454d85 Cleanup rtp_rtcp mocks [ #31191 ]
  • 804393b Removing incorrect DCHECK - breaks android [ #31190 ]
  • 28da36a Add unittest for av1 wrappers to test Encode and Decode functions [ #31189 ]
  • dcde85c Pass PeerConfigurerImpl directly into CreateTestPeer [ #31188 ]
  • 435fb9a Remove screen_share_config from the VideoConfig. [ #31187 ]
  • 553c869 Start consolidating management/querying of stats on the Call thread. [ #31186 ]
  • 04e1bab Replaces OverheadObserver with simple getter. [ #31185 ]
  • 03fade5 IWYU: uint32_t is defined in cstdint [ #31184 ]
  • b8a9630 Add a Release method for file wrapper [ #31183 ]
  • 3b6afee Add comparison methods for VideoSourceRestrictions [ #31182 ]
  • 0dcb470 [Adaptation] Adapt up requires previous underuse [ #31181 ]
  • ab866a2 Remove audio playout generator from APM API [ #31180 ]
  • b63331b Cleanup mocks for Video (en|de)coder factories [ #31179 ]
  • 1e83d34 Remove pc level test framework redundant code. [ #31178 ]
  • d6b1094 Revert "Android: Remove min_sdk_version in GN" [ #31175, #31177 ]
  • 42c5952 Create default frame generator in the AddVideoConfig method. [ #31176 ]
  • f00ed5b Android: Remove min_sdk_version in GN [ #31175 ]
  • 9af7543 Add RTC_EXPORT for NullSocketServer [ #31174 ]
  • fa95e8b fix nil RTCVideoEncoderSelector case in video encoder factory. [ #31173 ]
  • cce8643 Removed spammy log message from the FrameBuffer. [ #31172 ]
  • 5ed65b2 Add 5G detection to android_network_monitor [ #31171 ]
  • 81be421 Remove FrameTransformerInterface functions using EncodedFrame. [ #31170 ]
  • a0ff50c Reland "Improve outbound-rtp statistics for simulcast" [ #2, #31097, #31165, #31169 ]
  • c0df5fc VoIP API implementation on top of AudioIngress/Egress [ #31168 ]
  • c064467 Pass frame generator to the AddVideoConfig method in the pc framework tests. [ #31167 ]
  • 4210888 Refactors FEC in preparation for deferred packet generation. [ #31166 ]
  • 9a925c9 Revert "Improve outbound-rtp statistics for simulcast" [ #31097, #31165 ]
  • 49f574b Delete EncodedImage methods buffer(), set_buffer() and mutable_data() [ #31164 ]
  • c1aaf4c Revert "disallow pairing ICE-TCP with a local ip address" [ #31163 ]
  • dad6a94 Add helper frame generator factories for the pc framework tests. [ #31162 ]
  • 3c5450e Add support for PendingTaskSafetyFlag to ToQueuedTask. [ #31161 ]
  • 9e46cf5 Introduce a RunLoop class that supports the TaskQueue interface [ #31160 ]
  • d719708 Add unit tests for audio channel send frame transformer delegate. [ #31159 ]
  • 1b900b1 Removed unused function EncodedFrame::SetEncodedSize. [ #31158 ]
  • 701ccf9 Add unit tests for audio receive channel frame transformer delegate. [ #31157 ]
  • d9255b1 [getStats] Fix DCHECK crash in MergeInfoAboutOutboundRtpSubstreams(). [ #31156 ]
  • 455e802 Define MockTransformableFrame in test/. [ #31155 ]
  • 07ed0f4 Add more unit tests for sender video with frame transformer. [ #31154 ]
  • a81e9c8 Wrap WebRTC OBJC API types with RTC_OBJC_TYPE. [ #5, #31153 ]
  • ce1320c Add WaitForPreviouslyPostedTasks to TaskQueueForTest. [ #31152 ]
  • 1148fd5 Define MockFrameTransformer in test/. [ #31151 ]
  • 8ae18ad Remove unneeded dependency on CallStats. [ #31150 ]
  • 14a23a3 Add field trial to force playout delay [ #31149 ]
  • 3745d3f [Adaptation] Use ResourceAdaptationProcessorInterface* instead of impl. [ #31148 ]
  • 722fa4d [Adaptation] Misc tests for processor, input provider and restrictions. [ #31147 ]
  • bb826c9 Make echo metrics optional [ #31146 ]
  • 6a92e0e Android: Allow for re-assigning ScopedJavaGlobalRef [ #31145 ]
  • b5a0138 Rename done() into condition(), because it is actually condition in TimeController API [ #31144 ]
  • 91aa732 [Adaptation] Add OnAdaptationApplied(), remove ResourceListenerResponse. [ adaptation/resource.h#33, #31143 ]
  • 4d54475 ONNXINFER:Revise codes - Revise onnxinfer interface add getInfo() for debug
  • 4381af4 Change connection ASSERT into metrics for the PC level framework. [ #31142 ]
  • 3b9fe99 Add cpu_usage metrics. [ #31141 ]
  • 012aa37 Asynchronous QualityScaler: Callback-based CheckQpTask. [ #31140 ]
  • 9abc6bd Reduce audiosendstream dependency on rttstats (dead code). [ #31139 ]
  • b37e59d Add unittests for APM with submodule creation disabled [ #31138 ]
  • 62a0d64 Remove deprecated constant. [ #31137 ]
  • 74fc574 Fork a few VideoReceiveStream related classes. [ #31136 ]
  • b261118 Fix a typo for decoder naming [ #31135 ]
  • f7f7cc9 Extend IceControllerFactoryArgs with field trial string [ #31134 ]
  • cc73ed3 APM: Add build flag to allow building WebRTC without APM [ #31133 ]
  • 86bd33a Fix the name of the file generated by generate_sslroots.py. [ #31132 ]
  • 95d9a1a Update set of known root certificates. [ #31131 ]
  • 1b8ef63 Add an optional override for AudioRecord device [ #31130 ]
  • c8660b1 Open visibility of some PC level framework components [ #31129 ]
  • 3e1ac54 Refactor video dumping and rendering in PC level test. [ #31128 ]
  • 90ecee1 Make AudioEncoder::GetFrameLengthRange() pure virtual. [ #31127 ]
  • cda577f Enable simulcast statistics [ #31126 ]
  • 1fb4a05 Reland "Launch external ref control for vp9 encoder" [ #2, #31122, #31123, #31125 ]
  • e110a44 Delete uri for the Generic Frame Descriptor v1 [ #31124 ]
  • 9665b7d Revert "Launch external ref control for vp9 encoder" [ #31122, #31123 ]
  • 9427b51 Launch external ref control for vp9 encoder [ #31122 ]
  • f4ee036 [InsertableStreams] Clear callback to audio receive channel in delegate. [ #31121 ]
  • 1120f7d Makes the max padding factor default 3.0x [ #31120 ]
  • 9ce77fd Remove redundant Dcheck in vp9 decoder [ #31119 ]
  • f355e1a Add glUseProgram to the list of GL functions requiring synchronization [ #31118 ]
  • 11f92bc Audio ingress implementation for voip api. [ #31117 ]
  • f2b06ce Reduces unnecessary thread wakeups in TaskQueuePacedSender. [ #31116 ]
  • 811efc0 Trigger bots. [ #31115 ]
  • 146137a Merge OOURA BUILD.gn files. [ #31114 ]
  • 1f0677d Remove some TODOs from pc/. [ #31113 ]
  • e50c979 Exclude frame_analyzer_host build on iOS. [ #31112 ]
  • f0d64a5 Reland "Unify OOURA implementations in one directory." [ #31106, #31111 ]
  • dc4f75f [Adaptation] Make ResourceUsageState nullable, remove kStable. [ #31110 ]
  • 4c3a7db Remove RtpVideoHeader::discardable flag. [ #31109 ]
  • cbc0cba Updates RtpVideoSender to populate RtpRtcp::Config.field_trials [ #31108 ]
  • ed005be Revert "Unify OOURA implementations in one directory." [ #31106, #31107 ]
  • 09b439c Unify OOURA implementations in one directory. [ #31106 ]
  • 87eece9 [Adaptation] Introducing call/adaptation/ResourceAdaptationProcessor. [ #31105 ]
  • de8d551 [Adaptation] Rename the processor "VideoStreamEncoderResourceManager". [ #31104 ]
  • d2930c6 [Adaptation] Report AdaptationCounters OnVideoSourceRestrictionsUpdated. [ #31103 ]
  • 0003269 Clean up old GoPs when the RTP sequence number jump. [ #31102 ]
  • ee08643 Remove DetermineIceRole workaround. [ #31101 ]
  • 1d76654 [Adaptation] Move VideoStreamAdapter to call/adaptation/. [ #31100 ]
  • b613e3a [Adaptation] Resource::IsAdaptationUpAllowed() for IsBitrateConstrained. [ #31099 ]
  • d516b25 [Adaptation] Introduce VideoStreamInputState and its Provider. [ #31098 ]
  • da6cda8 Improve outbound-rtp statistics for simulcast [ #31097 ]
  • 1261b5c [Adaptation] Move EffectiveDegradationPreference to RA-Processor. [ #31096 ]
  • 2c80923 Remove WebRTC-LibvpxVp{8,9}TrustedRateController. [ #31095 ]
  • 67402f8 [Adaptation] Delete Processor Proof-of-Concept. [ #31094 ]
  • 29cf2b0 Fix link to prerequisite SW for Android development. [ #31093 ]
  • 4479a82 Remove deprecated SSLIdentity methods that return raw pointers. [ #31092 ]
  • 9acc18d Makes dynamic pacer select paddig target based on rate. [ #31091 ]
  • 6415dca Remove WebRTC-ExperimentalScreenshareSettings. [ #31090 ]
  • ec9fc22 Delete generic frame descriptor v1 trait and enum value [ #31089 ]
  • efdff53 Limit BWE drops from probes to 85% of the throughput estimate. [ #31088 ]
  • e886d2e Limits size of payload padding packets to 2x target size. [ #31087 ]
  • b33a0ca Remove deprecated ssl_identity methods [ #31086 ]
  • 5c356bb Cleanup: Removes unused BBR congestion controller. [ #31085 ]
  • ce0a11d Unify AdaptationReason and AdaptReason enums. [ #31084 ]
  • dff7925 Remove VideoStreamEncoderObserver::AdaptationReason::kNone [ #31083 ]
  • d059dfb Fix link in documentation. (take 2) [ #31082 ]
  • 3c49531 Fix link in documentation. [ #31081 ]
  • ee1e6bc Remove deprecated VideoSendStream::StreamStats data members. [ #31080 ]
  • 3d687a1 Add null checks in stacktrace.cc [ #31079 ]
  • 8ec11b8 Do not register generic frame descriptor v1 in integration tests [ #31078 ]
  • 3c9bcc1 Reland of the test portion of: [ #31077 ]
  • 4a5bce9 Change to more idiomatic map erase. [ #31076 ]
  • 708962b Roll chromium_revision 72d0f390b0..a775f4aeaa (758913:759199) [ #31075 ]
  • 43245c5 Remove excessive log spam in large conferences in downstream project. [ #31074 ]
  • f7f6870 Mark static const class/struct members as constexpr. [ #31073 ]
  • 3ebb6e9 Remove WebRTC-ExcludeTransportSequenceNumberFromFec. [ #31072 ]
  • 2136b56 Roll chromium_revision 304e175c8a..72d0f390b0 (758795:758913) [ #31071 ]
  • 491ff3d Remove AudioProcessing::UpdateHistogramsOnCallEnd. [ #31070 ]
  • f0684b5 Remove NetEq::InsertPacket deprecated method. [ #31069 ]
  • c4633fd Remove the version of Port::AddrAddress without a URL parameter. [ #31068 ]
  • b9d4685 insertable streams: include rtp_timestamp offset for audio [ #31067 ]
  • fea8b94 Reland "APM: Remove the usage of AudioFrame in the AudioProcessing interface" [ #31016, #31066 ]
  • 4395546 Fix wrong unit in log in VideoStreamEncoder [ #31065 ]
  • 3e2809f Drop support for receiving generic frame descriptor v1 [ #31064 ]
  • d2dcc58 Roll chromium_revision 41d2eaa40a..304e175c8a (758687:758795) [ #31063 ]
  • f387946 Remove WebRTC-Rfc5389StunRetransmissions. [ #31062 ]
  • dce6174 Remove deprecated SSLAdapter::SetIdentity. [ #31061 ]
  • cc34441 Remove deprecated RtpPacketInfo::RtpPacketInfo. [ #31060 ]
  • 9621377 Remove WebRTC-Audio-NewOpusPacketLossRateOptimization. [ #31059 ]
  • 39fb817 [Video, Svc] Remove inactive spatial layers in codec initializer [ #31058 ]
  • 9f297b5 Remove OpenSSLIdentity::GenerateWithExpiration. [ #31057 ]
  • 2d2c294 Remove OpenSSLAdapter restartable_ data member. [ #31056 ]
  • 8a948a3 Roll chromium_revision edcfd61be3..41d2eaa40a (758565:758687) [ #31055 ]
  • 0574822 Roll chromium_revision 94450aa986..edcfd61be3 (758458:758565) [ #31054 ]
  • 2c99264 Remove std::vector template instantiation. [ #31053 ]
  • 9ef32bb Roll chromium_revision 318b9b3c86..94450aa986 (758358:758458) [ #31052 ]
  • 790dba2 Merged PR 9208: Application: Add Feature - Enable serverless peer connection

This list of changes was auto generated.

Assets 3
Loading