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WWebRTC LUCI CQUpdate WebRTC code version (2023-10-30T04:03:42).
43074257创建于 2023年10月30日历史提交
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adaptation
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ half of the remaining folders git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -E "^(api|call|common_audio|examples|media|net|p2p|pc)/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I8b2cac13f4587d3ce9b2fccc7362967283f57ea2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302062 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39977} 3 年前
test
Revert "Clean up last_packet_received_time_ as it's no longer used." This reverts commit 2f4bc6416651be40ef8f95a4695e6b7c41f18666. Reason for revert: Breaks downstream test Original change's description: > Clean up last_packet_received_time_ as it's no longer used. > > Bug: webrtc:15377 > Change-Id: I5453b9fd572a04dbea3241a2eb1c8ad8bb8b1186 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320560 > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Ying Wang <yinwa@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#40792} Bug: webrtc:15377 Change-Id: Ifa57671cc479cdd86f543c4edc236221beb76f90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321340 Auto-Submit: Björn Terelius <terelius@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Owners-Override: Björn Terelius <terelius@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40797} 2 年前
BUILD.gn
Cleanup Call construction Return unique_ptr to clearly communicate ownership is transfered. Remove Call::Config alias Bug: None Change-Id: Ie3aa1da383ad65fae490d218fced443d44961eab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323160 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40934} 2 年前
DEPS
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated. Results from test (CallPerfTest.TestEncodeFramerateVp8Simulcast): Simulcast streams: 0: max_fps:20 -> StreamStats.encode_frame_rate:15 (before), 20 (after) 1: max_fps:30 Bug: webrtc:13031 Change-Id: I30e6b2dcb2746859bd3e21b098bfa7b0fb3b2dda Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230120 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34867} 4 年前
OWNERS
Update OWNERS for call/ No-try: True Bug: None Change-Id: I53103a75c3c5e1d9c798c68bcca7c2041129d32a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265001 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37112} 3 年前
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface Bug: webrtc:7484 Change-Id: I22eaa7a9e082fc575cf7471d7a2f4f706564d54f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262805 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36965} 3 年前
audio_receive_stream.h
Propagate time of the last received packet with Timestamp type Bug: webrtc:13757 Change-Id: I446fc10ad6a90ab9ecaac337b9f2ad4ccad37cbd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307020 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40211} 2 年前
audio_send_stream.cc
Reland "Wire up non-sender RTT for audio, and implement related standardized stats." This reverts commit 2c41cbae37cac548a1133589b9d2c2e8614fa6cb. Reason for revert: The breaking test in Chromium has been temporarily disabled in https://chromium-review.googlesource.com/c/chromium/src/+/3139794/2. Original change's description: > Revert "Wire up non-sender RTT for audio, and implement related standardized stats." > > This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e. > > Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium. > > Original change's description: > > Wire up non-sender RTT for audio, and implement related standardized stats. > > > > The implemented stats are: > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements > > > > Bug: webrtc:12951, webrtc:12714 > > Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956 > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#34861} > > # Not skipping CQ checks because original CL landed > 1 day ago. > > TBR=hta,hbos,minyue > > Bug: webrtc:12951, webrtc:12714 > Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Olga Sharonova <olka@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#34897} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12951, webrtc:12714 Change-Id: I786b06933d85bdffc5e879bf52436bb3469b7f3a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231181 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34930} 4 年前
audio_send_stream.h
Replace "rcvd" with "received" for readability following guidance in https://google.github.io/styleguide/cppguide.html#General_Naming_Rules BUG=None Change-Id: I105591a7f709d65a3d3320f7f44137d432cf7ce0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302282 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#39937} 3 年前
audio_sender.h
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api Bug: webrtc:11251 Change-Id: Id3b6ff1814931d8250c4aaac59e494521fbe93ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164560 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tim Na <natim@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30238} 6 年前
audio_state.cc
Remove chromium clang style errors affecting sdk/android/media_jni Bug: webrtc:163 Change-Id: I1e98174817ca032ee13f9a6a386803382843389d Reviewed-on: https://webrtc-review.googlesource.com/67360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Paulina Hensman <phensman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22796}7 年前
audio_state.h
Async audio processing API API to injecting a heavy audio processing operation into WebRTC audio capture pipeline Bug: webrtc:12003 Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32291} 5 年前
bitrate_allocator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1 This cl/ 1) move WebRtcKeyValueConfig from api/transport to api/ directory. 2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials). 3) removes usage of webrtc::field_trial:: from the pc/ directory. 4) removes a few unused includes of system_wrappers/field_trial.h. Bug: webrtc:10335 Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36160} 4 年前
bitrate_allocator.h
Use backticks not vertical bars to denote variables in comments for /call Bug: webrtc:12338 Change-Id: I8f92127b61352bd4b98a0690e9a0435bb6c6f870 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226943 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34569} 4 年前
bitrate_allocator_unittest.cc
Adopt absl::string_view in call/ Bug: webrtc:13579 Change-Id: Ib616eb3372da341fafb55c23038182751b9da5a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262780 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36910} 3 年前
bitrate_estimator_tests.cc
Reland: FrameGeneratorCapturer: don't generate video before Start is called It is partial reland, which adds call to Start() to all relevant places, but doesn't actually switches frame generator to not produce frames from the moment it was created. Bug: b/272350185 Change-Id: I6e3bd7af6f5cd8d9baff79c2aada7b2ddfae1c8d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310782 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40379} 2 年前
call.cc
Delete unnesseccary Call::RegisterReceiveStream and Call::DeregisterReceiveStream methods. Bug: webrtc:7135 Change-Id: I12e417b9bc5ed8bfae64e4591c37f882ead04092 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291481 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40959} 2 年前
call.h
Cleanup Call construction Return unique_ptr to clearly communicate ownership is transfered. Remove Call::Config alias Bug: None Change-Id: Ie3aa1da383ad65fae490d218fced443d44961eab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323160 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40934} 2 年前
call_config.cc
Add parameter to control the pacer's burst outside of field trials. BurstyPacer is currently controlled via field trials. In order for Chrome to be able to have burst without relying on a field trial, this parameter is added. When all burst experiments have concluded we may be able to have a hardcoded constant instead, but for now the parameter is added to RTCConfiguration. NOTRY=True Bug: chromium:1354491 Change-Id: I386c1651dbbcbf309c15ea3d3380cf8f632b5429 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283420 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38621} 3 年前
call_config.h
Introduce support for video packet batching. This CL introduces a new feature enabling video packet send batches. The feature is enabled via PeerConnectionInterface ::RTCConfiguration ::MediaConfig ::enable_send_packet_batching. PacketOptions have been augmented with attribute "batchable" (set for all video packets) and attribute "last_packet_in_batch" which gives injected AsyncPacketSockets a chance to understand when a batch begins and ends. When the feature is on, packets are collected in RtpSenderEgress. On reception of OnBatchComplete from PacingController, RtpSenderEgress sends the collected batch, setting "last_packet_in_batch" to true in the last packet. Bug: chromium:1439830 Change-Id: I1846b9d4a8a0efd227d617691213a2e048bdc8a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303720 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40012} 3 年前
call_factory.cc
Cleanup Call construction Return unique_ptr to clearly communicate ownership is transfered. Remove Call::Config alias Bug: None Change-Id: Ie3aa1da383ad65fae490d218fced443d44961eab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323160 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40934} 2 年前
call_factory.h
Cleanup Call construction Return unique_ptr to clearly communicate ownership is transfered. Remove Call::Config alias Bug: None Change-Id: Ie3aa1da383ad65fae490d218fced443d44961eab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323160 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40934} 2 年前
call_perf_tests.cc
Cleanup Call construction Return unique_ptr to clearly communicate ownership is transfered. Remove Call::Config alias Bug: None Change-Id: Ie3aa1da383ad65fae490d218fced443d44961eab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323160 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40934} 2 年前
call_unittest.cc
Cleanup Call construction Return unique_ptr to clearly communicate ownership is transfered. Remove Call::Config alias Bug: None Change-Id: Ie3aa1da383ad65fae490d218fced443d44961eab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323160 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40934} 2 年前
degraded_call.cc
Remove internal overrides using old SendRtp and SendRtcp interfaces. This CL takes away all usages except for Android code. Low-Coverage-Reason: Refactoring old code Bug: webrtc:15410 Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40554} 2 年前
degraded_call.h
Remove internal overrides using old SendRtp and SendRtcp interfaces. This CL takes away all usages except for Android code. Low-Coverage-Reason: Refactoring old code Bug: webrtc:15410 Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40554} 2 年前
fake_network_pipe.cc
Delete unused constructor of FakeNetworkPipe Bug: None Change-Id: I960f9d3988e10fa22f3379d071818ad44e36d456 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316861 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40569} 2 年前
fake_network_pipe.h
Delete unused constructor of FakeNetworkPipe Bug: None Change-Id: I960f9d3988e10fa22f3379d071818ad44e36d456 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316861 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40569} 2 年前
fake_network_pipe_unittest.cc
Reland "Delete PacketReceiver::DeliverPacket from all implementations" This reverts commit f2a083f262d86737893e774c696716742fcab3e3. Reason for revert: Test problem fixed in https://webrtc-review.googlesource.com/c/src/+/291333. Original change's description: > Revert "Delete PacketReceiver::DeliverPacket from all implementations" > > This reverts commit 897ea04db5db2e591e28bd884191be58d9bcdc63. > > Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200 > > Original change's description: > > Delete PacketReceiver::DeliverPacket from all implementations > > > > And fix tests that still depend on extensions to be known by the receiver. > > > > Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3 > > > > Bug: webrtc:7135,webrtc:14795 > > Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996 > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39184} > > Bug: webrtc:7135,webrtc:14795,b/266658815 > Change-Id: I9d03f4952938d176ffee110a707acadc1846457c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400 > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39189} Bug: webrtc:7135,webrtc:14795,b/266658815 Change-Id: Ia640f4342a1f42012ba5295003e17aef7613ad80 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291440 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39199} 3 年前
flexfec_receive_stream.cc
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}7 年前
flexfec_receive_stream.h
stats: implement flexfec fecBytesReceived stats for FlexFEC specified in https://github.com/w3c/webrtc-stats/pull/762 and take FlexFEC into account for receive statistics. BUG=webrtc:15250 Change-Id: Id85775ab1f29487d5b8bf478da6e22071005901a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294881 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40325} 2 年前
flexfec_receive_stream_impl.cc
stats: implement flexfec fecBytesReceived stats for FlexFEC specified in https://github.com/w3c/webrtc-stats/pull/762 and take FlexFEC into account for receive statistics. BUG=webrtc:15250 Change-Id: Id85775ab1f29487d5b8bf478da6e22071005901a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294881 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40325} 2 年前
flexfec_receive_stream_impl.h
stats: implement flexfec fecBytesReceived stats for FlexFEC specified in https://github.com/w3c/webrtc-stats/pull/762 and take FlexFEC into account for receive statistics. BUG=webrtc:15250 Change-Id: Id85775ab1f29487d5b8bf478da6e22071005901a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294881 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40325} 2 年前
flexfec_receive_stream_unittest.cc
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ half of the remaining folders git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -E "^(api|call|common_audio|examples|media|net|p2p|pc)/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I8b2cac13f4587d3ce9b2fccc7362967283f57ea2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302062 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39977} 3 年前
packet_receiver.h
Allow injecting packets of type Any to Call::DeliverRtpPacket MediaType::Any will be used by packets that can not be demuxed by RtpTransport. Bug: webrtc:14928 Change-Id: Ib759e65c7eede29defdad8073fd1ed6be814ab81 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299280 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39710} 3 年前
rampup_tests.cc
Replace WebRTC-QuickPerfTest field trial with a flag This field trial is configured via command line flag, so may use flag system directly, reducing dependency on global field trial string. Bug: webrtc:7101, webrtc:10335 Change-Id: I1e48e0e3fdc251b73a375c6d7f1a46fa4f8a179b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322624 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40897} 2 年前
rampup_tests.h
Stop overriding extensions in rampup tests Instead, ensure extensions are registered so that both transport and send streams are aware. Bug: webrtc:7135,webrtc:14795,b/266658815 Change-Id: I7710113893e2c5e23c1365de6aa3b761e3408308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291333 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39193} 3 年前
receive_stream.h
Remove rtp header extension from config of Call audio and video receivers These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived. Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d Bug: webrtc:7135 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480 Owners-Override: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39236} 3 年前
receive_time_calculator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365} 4 年前
receive_time_calculator.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365} 4 年前
receive_time_calculator_unittest.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2 convert call/ (and the collaterals) Bug: webrtc:10335 Change-Id: I8f6bc13c032713aa2a947724b464f6f35454d39a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36165} 4 年前
rtp_bitrate_configurator.cc
Allow setting a bandwidth cap for relayed connections. For now the capping is experimental and applied via a field trial. Bug: webrtc:11434 Change-Id: Id8e6e9b948f099a0940974a9a431b5b0a43c32f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171226 Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30909} 6 年前
rtp_bitrate_configurator.h
Remove RTC_DISALLOW_COPY_AND_ASSIGN more. Bug: webrtc:13555, webrtc:13082 Change-Id: I9c07708108da0a26f5e228384fd56cef4d1540b3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247300 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#35749} 4 年前
rtp_bitrate_configurator_unittest.cc
Revert "In RtpBitrateConfigurator ignore new parameters when set to default values." This reverts commit bcbdeedd432198c3d48effb2162af6344d885b14. Reason for revert: Speculative revert after a perf regression. Original change's description: > In RtpBitrateConfigurator ignore new parameters when set to default values. > > Bug: webrtc:11263 > Change-Id: Ia7539c7c142b059d0295849b916439bb647f112d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162207 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30191} TBR=danilchap@webrtc.org,srte@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:11263 Change-Id: I17804655465b27523c462d2aba44519c820b8e04 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165687 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30213} 6 年前
rtp_config.cc
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED. The new macros will replace the old one when old one's usage will be removed. The idea of the renaming to provide a clear signal that this is debug build only macros and will be stripped in the production build. Bug: webrtc:9065 Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35348} 4 年前
rtp_config.h
Introduce support for video packet batching. This CL introduces a new feature enabling video packet send batches. The feature is enabled via PeerConnectionInterface ::RTCConfiguration ::MediaConfig ::enable_send_packet_batching. PacketOptions have been augmented with attribute "batchable" (set for all video packets) and attribute "last_packet_in_batch" which gives injected AsyncPacketSockets a chance to understand when a batch begins and ends. When the feature is on, packets are collected in RtpSenderEgress. On reception of OnBatchComplete from PacingController, RtpSenderEgress sends the collected batch, setting "last_packet_in_batch" to true in the last packet. Bug: chromium:1439830 Change-Id: I1846b9d4a8a0efd227d617691213a2e048bdc8a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303720 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40012} 3 年前
rtp_demuxer.cc
Adopt absl::string_view in call/ Bug: webrtc:13579 Change-Id: Ib616eb3372da341fafb55c23038182751b9da5a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262780 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36910} 3 年前
rtp_demuxer.h
Adopt absl::string_view in call/ Bug: webrtc:13579 Change-Id: Ib616eb3372da341fafb55c23038182751b9da5a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262780 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36910} 3 年前
rtp_demuxer_unittest.cc
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ half of the remaining folders git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -E "^(api|call|common_audio|examples|media|net|p2p|pc)/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I8b2cac13f4587d3ce9b2fccc7362967283f57ea2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302062 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39977} 3 年前
rtp_packet_sink_interface.h
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}8 年前
rtp_payload_params.cc
Add codec name H265 to support H265 in WebRTC Bug: webrtc:13485 Change-Id: I352b15a65867f0d56fc8e9a9e03081bd3258108e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316283 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40773} 2 年前
rtp_payload_params.h
For VP9 assume max number of spatial layers to simulate generic descriptor VP9 allows to increase number of spatial layers on delta frame, which is not supported by dependency descriptor. Thus to generate DD compatible generic header, simulator would set max number of spatial layers, while number of active spatial layers would be communicated with active_decode_target bitmask Bug: webrtc:14042 Change-Id: I4da63fa7c38b0f17758a7a6243640f444470b40c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265164 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37151} 3 年前
rtp_payload_params_unittest.cc
When simulating chains from VP9 codec specific info support first_active_layer > 0 Bug: webrtc:11999 Change-Id: Ie2bae8113968fdab330f2c89e5f5416a79f14dc7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314900 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40507} 2 年前
rtp_stream_receiver_controller.cc
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived Instead of getting header extension mapping from a receiver object, get the mapping from the received packet. The purpose is to be able to remove extension information from webrtc/call/receive_stream.h. Header extensions are negotiated per mid, not per receive stream. The goal is to reduce the number of places where packets are parsed and demuxed. Bug: webrtc:7135, webrtc:14795 Change-Id: I8944bc06a11dc572d9e14e7d7ee446a841096295 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288968 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38944} 3 年前
rtp_stream_receiver_controller.h
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived Instead of getting header extension mapping from a receiver object, get the mapping from the received packet. The purpose is to be able to remove extension information from webrtc/call/receive_stream.h. Header extensions are negotiated per mid, not per receive stream. The goal is to reduce the number of places where packets are parsed and demuxed. Bug: webrtc:7135, webrtc:14795 Change-Id: I8944bc06a11dc572d9e14e7d7ee446a841096295 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288968 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38944} 3 年前
rtp_stream_receiver_controller_interface.h
Demote RtpStreamReceiverController AddSink/RemoveSink to private Bug: webrtc:7135, webrtc:10198, webrtc:14256 Change-Id: I47dc9034170b1868ad442d36c74c5380964b476b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267827 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37457} 3 年前
rtp_transport_config.h
Add parameter to control the pacer's burst outside of field trials. BurstyPacer is currently controlled via field trials. In order for Chrome to be able to have burst without relying on a field trial, this parameter is added. When all burst experiments have concluded we may be able to have a hardcoded constant instead, but for now the parameter is added to RTCConfiguration. NOTRY=True Bug: chromium:1354491 Change-Id: I386c1651dbbcbf309c15ea3d3380cf8f632b5429 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283420 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38621} 3 年前
rtp_transport_controller_send.cc
Change expectation of GoogCCNetworkController::OnNetworkAvailability Expect OnNetworkAvailabability to be invoked when the transport becomes writable. Before this change, ProbeController in GoogCC was expected to be created when the transport is writable or explicitly notifed after creation that network is not writable. Bug: None Change-Id: I623b1c34e40a82e912f85b92fea49629e7e72d4e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323463 Reviewed-by: Diep Bui <diepbp@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40975} 2 年前
rtp_transport_controller_send.h
Revert "Clean up last_packet_received_time_ as it's no longer used." This reverts commit 2f4bc6416651be40ef8f95a4695e6b7c41f18666. Reason for revert: Breaks downstream test Original change's description: > Clean up last_packet_received_time_ as it's no longer used. > > Bug: webrtc:15377 > Change-Id: I5453b9fd572a04dbea3241a2eb1c8ad8bb8b1186 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320560 > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Ying Wang <yinwa@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#40792} Bug: webrtc:15377 Change-Id: Ifa57671cc479cdd86f543c4edc236221beb76f90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321340 Auto-Submit: Björn Terelius <terelius@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Owners-Override: Björn Terelius <terelius@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40797} 2 年前
rtp_transport_controller_send_factory.h
Refactor some config plumbing in call/. Address perkj's comments left in https://webrtc-review.googlesource.com/c/src/+/283420. I was a bit trigger-happy with the submit button. Bug: chromium:1354491 Change-Id: Ifd052f75af3763b0b52807c31ea790e3efee921d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283521 Reviewed-by: Erik Språng <sprang@webrtc.org> Auto-Submit: Henrik Boström <hbos@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38638} 3 年前
rtp_transport_controller_send_factory_interface.h
Delete Call dependency on ProcessThread as unused Last usage or ProcessThread was removed in https://webrtc-review.googlesource.com/c/src/+/265921 Bug: webrtc:7219 Change-Id: Ia46d9e2530cd0dbf56a5c0ca6e1bf0936fd62672 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266363 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37287} 3 年前
rtp_transport_controller_send_interface.h
Revert "Clean up last_packet_received_time_ as it's no longer used." This reverts commit 2f4bc6416651be40ef8f95a4695e6b7c41f18666. Reason for revert: Breaks downstream test Original change's description: > Clean up last_packet_received_time_ as it's no longer used. > > Bug: webrtc:15377 > Change-Id: I5453b9fd572a04dbea3241a2eb1c8ad8bb8b1186 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320560 > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Ying Wang <yinwa@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#40792} Bug: webrtc:15377 Change-Id: Ifa57671cc479cdd86f543c4edc236221beb76f90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321340 Auto-Submit: Björn Terelius <terelius@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Owners-Override: Björn Terelius <terelius@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40797} 2 年前
rtp_video_sender.cc
Make field trial string DisableRtxRateLimiter enabled by default. Bug: webrtc:15184 Change-Id: Ie2a20892b71defe2a3b744ae5b631a76f9a8712c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325120 Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41016} 2 年前
rtp_video_sender.h
Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController. Currently FecController knows about network conditions, these information can be used to control RTX settings in-call. Change-Id: I8f84164aeac48ea13b7f1cf82fd7424431f98ada Bug: webrtc:15167 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304800 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40192} 2 年前
rtp_video_sender_interface.h
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream" This reverts commit 75170be4acc90fece7c65f1a5b9bef03a5cc3880. Reason for revert: Perf regression not affecting open source. Original change's description: > Revert "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream" > > This reverts commit d8c4de71722c9de38f942932be21d4015f32a3bc. > > Reason for revert: Tentative revert due to possible perf regression. b/260123362 > > Original change's description: > > Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream > > > > VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream. > > Therefore this cl: > > - Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience. > > - Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing. > > - RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used. > > > > Bug: none > > Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#38698} > > Bug: none > Change-Id: I4f0d27679e51361b9ec54d2ae8e4d972527875d1 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284940 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38725} Bug: b/260400659 Change-Id: Ie8e545edcad85284a7d612183a8e4201672d0b5e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285900 Auto-Submit: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38794} 3 年前
rtp_video_sender_unittest.cc
Rename EncodedImage property Timetamp to RtpTimestamp To avoid name collision with Timestamp type, To avoid confusion with capture time represented as Timestamp Bug: webrtc:9378 Change-Id: I8438a9cf4316e5f81d98c2af9dc9454c21c78e70 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320601 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40796} 2 年前
rtx_receive_stream.cc
Updated associated payload types without recreating receive streams. Bug: webrtc:11993 Change-Id: I49c61653b296b1b3ca6a12fa75ac699ee58f096c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271543 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37799} 3 年前
rtx_receive_stream.h
Updated associated payload types without recreating receive streams. Bug: webrtc:11993 Change-Id: I49c61653b296b1b3ca6a12fa75ac699ee58f096c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271543 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37799} 3 年前
rtx_receive_stream_unittest.cc
Store RtpPacketReceived::arrival_time as Timestamp. Previously this value was rounded up to a millisecond value. This change is complementary to another change: https://webrtc-review.googlesource.com/c/src/+/216398 Bug: webrtc:12722 Change-Id: I0fd2baceb4608132615fb6ad241ec863e343edb1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217521 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33928} 4 年前
simulated_network.cc
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork." This is a reland of commit c1d5fda22c8ae456950c5549d22d099b478c67e2 Original change's description: > Add documentation, tests and simplify webrtc::SimulatedNetwork. > > This CL increases the test coverage for webrtc::SimualtedNetwork, adds > some more comments to the class and the interface it implements and > simplify the logic around capacity and delay management in the > simulated network. > > More CLs will follow to continue the refactoring but this is the > ground work to make this more modular in the future. > > Bug: webrtc:14525, b/243202138 > Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042 > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38388} Bug: webrtc:14525, b/243202138, b/256595485 Change-Id: Iaf8160eb8f8e29034b8f98e81ce07eb608663d30 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280963 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38557} 3 年前
simulated_network.h
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork." This is a reland of commit c1d5fda22c8ae456950c5549d22d099b478c67e2 Original change's description: > Add documentation, tests and simplify webrtc::SimulatedNetwork. > > This CL increases the test coverage for webrtc::SimualtedNetwork, adds > some more comments to the class and the interface it implements and > simplify the logic around capacity and delay management in the > simulated network. > > More CLs will follow to continue the refactoring but this is the > ground work to make this more modular in the future. > > Bug: webrtc:14525, b/243202138 > Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042 > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38388} Bug: webrtc:14525, b/243202138, b/256595485 Change-Id: Iaf8160eb8f8e29034b8f98e81ce07eb608663d30 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280963 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38557} 3 年前
simulated_network_unittest.cc
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork." This is a reland of commit c1d5fda22c8ae456950c5549d22d099b478c67e2 Original change's description: > Add documentation, tests and simplify webrtc::SimulatedNetwork. > > This CL increases the test coverage for webrtc::SimualtedNetwork, adds > some more comments to the class and the interface it implements and > simplify the logic around capacity and delay management in the > simulated network. > > More CLs will follow to continue the refactoring but this is the > ground work to make this more modular in the future. > > Bug: webrtc:14525, b/243202138 > Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042 > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38388} Bug: webrtc:14525, b/243202138, b/256595485 Change-Id: Iaf8160eb8f8e29034b8f98e81ce07eb608663d30 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280963 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38557} 3 年前
simulated_packet_receiver.h
Calculate next process time in simulated network. Currently there's an implicit requirement that users of SimulatedNetwork should call it repeatedly, even if the return value of NextDeliveryTimeUs is unset. With this change, it will indicate that there might be a delivery in 5 ms at any time there are packets in queue. Which results in unchanged behavior compared to current usage but allows new users to expect robust behavior. Bug: webrtc:9510 Change-Id: I45b8b5f1f0d3d13a8ec9b163d4011c5f01a53069 Reviewed-on: https://webrtc-review.googlesource.com/c/120402 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26617}7 年前
syncable.cc
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}8 年前
syncable.h
Rename AudioReceiveStream to AudioReceiveStreamInterface Bug: webrtc:7484 Change-Id: I22eaa7a9e082fc575cf7471d7a2f4f706564d54f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262805 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36965} 3 年前
version.cc
Update WebRTC code version (2023-10-30T04:03:42). Bug: None Change-Id: I1b1218b506fb691aad569af1c7b1aa185d33e2ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325202 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#41033} 2 年前
version.h
Add WebRTC code freshness version string. This CL adds a string to the resulting WebRTC library (trying to make sure the version string will be there no matter how WebRTC is packaged). This CL should be followed by some process to regularly and automatically update the version string. No-Try: True No-Presubmit: True Bug: webrtc:12159 Change-Id: I9143aeae2cd54d0d4048c138772888100d7873cb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191223 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32825} 5 年前
video_receive_stream.cc
Add missing comma in VideoReceiveStreamInterface::Stats::ToString Bug: None Change-Id: I665fd120bdfe3e93e51f11f9035e30d09381db75 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323800 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40947} 2 年前
video_receive_stream.h
Remove default "unknown" encoderImplementation/decoderImplementation which means this will not show up in getStats inbound-rtp/outbound-rtp until the encoder/decoder is known. This has implications in particular for inbound-rtp where the value is currently "unknown" until video frames have been received. This is safe to change as the previous change to gate decoderImplementation behind getUserMedia access already broke the assumption that the field is always string. BUG=webrtc:14906 Change-Id: Ie6040ada3656e80f792c0c32c1b86ad1d6609d3c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293600 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40334} 2 年前
video_send_stream.cc
Cleanup usasge of ReportBlockData::report_block accessor This reduces dependency on the struct RTCPReportBlock and would allow to delete it in favor of class ReportBlockData Bug: None Change-Id: Ia46a2516e26453724eed2e499f475f65df6cd3fa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304163 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39990} 3 年前
video_send_stream.h
Remove default "unknown" encoderImplementation/decoderImplementation which means this will not show up in getStats inbound-rtp/outbound-rtp until the encoder/decoder is known. This has implications in particular for inbound-rtp where the value is currently "unknown" until video frames have been received. This is safe to change as the previous change to gate decoderImplementation behind getUserMedia access already broke the assumption that the field is always string. BUG=webrtc:14906 Change-Id: Ie6040ada3656e80f792c0c32c1b86ad1d6609d3c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293600 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40334} 2 年前