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WWebRTC LUCI CQMove Destroy/Create steps for DataChannelTransport to PeerConnection.
840cf786创建于 2023年10月22日历史提交
文件最后提交记录最后更新时间
DEPS
Use WebRTC's Java VM initialization in tests. WebRTC should not depend on chromium's //base. Bug: webrtc:13662 Change-Id: Ie660aa0f2477cc747830bba611aa23ed5e732385 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256364 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36581} 4 年前
android_test_initializer.cc
Use WebRTC's Java VM initialization in tests. WebRTC should not depend on chromium's //base. Bug: webrtc:13662 Change-Id: Ie660aa0f2477cc747830bba611aa23ed5e732385 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256364 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36581} 4 年前
android_test_initializer.h
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}7 年前
fake_audio_capture_module.cc
Use SequenceChecker(SequenceChecker::kDetached) in a few places. This CL is partly a test to see if there's an impact on binary size: - Not a big difference for binaries (decrease): -776b to -4Kb - For libraries (libwebrtc.a) it actually increases the size: +40Kb Secondarily this CL is basically to introduce this pattern to the code base. In terms of LOC, this makes things slightly more compact. From: class Foo { public: Foo() { checker_.Detach(); } private: SequenceChecker checker_; }; To: class Foo { public: Foo() = default; private: SequenceChecker checker_{SequenceChecker::kDetached}; }; Bug: none Change-Id: I59fc34ccea10847e13455a349851ce9a0af458e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299020 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39664} 3 年前
fake_audio_capture_module.h
Use SequenceChecker(SequenceChecker::kDetached) in a few places. This CL is partly a test to see if there's an impact on binary size: - Not a big difference for binaries (decrease): -776b to -4Kb - For libraries (libwebrtc.a) it actually increases the size: +40Kb Secondarily this CL is basically to introduce this pattern to the code base. In terms of LOC, this makes things slightly more compact. From: class Foo { public: Foo() { checker_.Detach(); } private: SequenceChecker checker_; }; To: class Foo { public: Foo() = default; private: SequenceChecker checker_{SequenceChecker::kDetached}; }; Bug: none Change-Id: I59fc34ccea10847e13455a349851ce9a0af458e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299020 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39664} 3 年前
fake_audio_capture_module_unittest.cc
Delete TestListener and top-level thread wrapping. Instead use rtc::AutoThread in tests that need that. Bug: webrtc:9714 Change-Id: I1f33b1b2d321770d062504dd9ef86d66a345dd42 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254681 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36950} 3 年前
fake_data_channel_controller.h
Guard FakeDataChannelController state with the network thread. Tsan bots detected races since callbacks are being made on the network thread but tests checked the state from the signaling thread. Bug: none Change-Id: If854e44159c56c0d12616e0b62ad92018291ed30 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302281 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39928} 3 年前
fake_peer_connection_base.h
Move Destroy/Create steps for DataChannelTransport to PeerConnection. This moves steps from the sdp code for pc state over to the PC class and slightly simplifies the contract between the two classes. Moving forward it's easier to consolidate those steps in the PC class with other grouped operations e.g. during teardown. Also removing GetDataMid() method in favor of the sctp_mid() property. Bug: none Change-Id: I938f953099d327377abd94e6b2c9ece803d88e40 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324300 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40981} 2 年前
fake_peer_connection_for_stats.h
Split fake media channel classes This allows to remove some calls to CreateMediaChannel in the RtpTransceiver code. This removes the fake engines owning the channels and moves the responsibility to the tests themselves as it's quite hard to both return a unique_ptr to a channel and still own it. The various channel getters from the fake engine are thus also removed and tests updated accordingly, the channel is retrieved from internal structs in the tests by going through the RtpTransceiver objects as it's not possible to safely get the channels from only a sender or receiver. As some tests are running in both PlanB and Unified Plan, getting a transceiver is not working for PlanB. As PlanB has been deprecated and will eventually be removed, the problematic tests have either been removed or updated to only run with Unified Plan. Bug: webrtc:13931 Change-Id: I0571beca8b9ef2f2089d500802b7b124268d9de3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310340 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40366} 2 年前
fake_periodic_video_source.h
Deflake simulcast flow tests: stop fake source for reals. The FakePeriodicVideoSource was not actually stopping its repeated handle, which takes a raw pointer to the task queue. There could be a race here where a repeated task was being posted at the same time as the task queue was being destroyed due to the scoped safety flag being tied to the repeated task rather than the task queue. I'm still unable to repro locally, so this is a speculative fix. # No need to wait for ios/android bots, all other bots green NOTRY=True Bug: webrtc:15018 Change-Id: Id6f9bda5f4fc641abc11068f5cf8aa0f1cf36d27 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300264 Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39758} 3 年前
fake_periodic_video_track_source.h
Deflake PeerConnectionSimulcastMediaFlowTests due to unstopped sources. Only in testing environments are the task queues shut down while sources still have media flowing. It's still not clear why heap-use-after-free happens, since it should be enough to close the PC, but it is clear that the crash is happening due to frames flowing while the test is shutting down, which is not something happening outside of testing. In an attempt to deflake, this CL makes sure to manually stop the test-only sources before closing the peer connection. Bug: webrtc:15018 Change-Id: I48ee131a8994c9c4caee1bb4875580d255b97da1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299944 Commit-Queue: Henrik Boström <hbos@webrtc.org> Auto-Submit: Henrik Boström <hbos@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#39752} 3 年前
fake_rtc_certificate_generator.h
Replace RTCCertificateGeneratorCallback interface with an AnyInvocable follow up of the https://webrtc-review.googlesource.com/c/src/+/272402 Bug: None Change-Id: Ie47aff9fccdb4037c1f560801c780dd549b373ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272553 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37870} 3 年前
fake_video_track_renderer.h
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}7 年前
fake_video_track_source.h
Replace more instances of rtc::RefCountedObject with make_ref_counted. This is essentially replacing new rtc::RefCountedObject with rtc::make_ref_counted in many files. In a couple of places I made minor tweaks to make things compile such as adding parenthesis when they were missing. Bug: webrtc:12701 Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33852} 4 年前
frame_generator_capturer_video_track_source.h
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
integration_test_helpers.cc
Generalize ssrc-group check to apply to groups other than SIM BUG=chromium:1477075 Change-Id: I20f094dee11ea26a180471ce52d78d916f922f29 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322440 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40888} 2 年前
integration_test_helpers.h
Generalize ssrc-group check to apply to groups other than SIM BUG=chromium:1477075 Change-Id: I20f094dee11ea26a180471ce52d78d916f922f29 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322440 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40888} 2 年前
mock_channel_interface.h
Remove all split channels related code Bug: webrtc:13931 Change-Id: I93b8ca0ba1ec15bf260236bbc914b41fbb30aa58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310680 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40376} 2 年前
mock_data_channel.h
stats: do not expose dataChannelIdentifier before it is set filtering out the -1 value as it is done for "legacy" stats. Also change the protocol and don't use "udp" and "tcp" which are misleading since the datachannel protocol is user-supplied. BUG=webrtc:15071 Change-Id: I45d735fcf30144969630f5b8a91b40f12585bbfd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300483 Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40333} 2 年前
mock_peer_connection_internal.h
Move Destroy/Create steps for DataChannelTransport to PeerConnection. This moves steps from the sdp code for pc state over to the PC class and slightly simplifies the contract between the two classes. Moving forward it's easier to consolidate those steps in the PC class with other grouped operations e.g. during teardown. Also removing GetDataMid() method in favor of the sctp_mid() property. Bug: none Change-Id: I938f953099d327377abd94e6b2c9ece803d88e40 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324300 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40981} 2 年前
mock_peer_connection_observers.h
Update pc/ to not use implicit conversion from scoped_refptr<T> to T*. Bug: webrtc:13464 Change-Id: I768646af8ded6338ef51486b8d69db1ad71e9a2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259500 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36588} 4 年前
mock_rtp_receiver_internal.h
Reland "Remove 'trackId' dependency in stats selector algorithm." This is a reland of commit 81aab488781c1a736c9d85ff1532631be2989523 See diff between Patch Set 1 and latest Patch Set. The original CL broke this WPT[1] because getStats() with the receiver as the selector stopped working in the event of unsignalled SSRCs due to the receiver not knowing what the SSRC was. This fix is to query media_channel_ for the unsignalled SSRC in the event that the receiver does not know the SSRC. [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/simulcast/setParameters-active.https.html Original change's description: > Remove 'trackId' dependency in stats selector algorithm. > > In preparation for the deletion of deprecated 'track' stats, the > stats selector algorithm needs to be rewritten not to use 'trackId'. > > This is achieved by finding RTP stats by their SSRC, as obtained via > getParameters(). This unfortunately adds a block-invoke (in the sender > case the block-invoke happens inside GetParametersInternal and in the > receiver case the block-invoke is explicit at the calling place), but > it can't be helped and it's just once per getStats() call and only if > the selector argument is used. > > Bug: webrtc:14175 > Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38981} Bug: webrtc:14175, webrtc:14811 Change-Id: I0d16724af4efeb93d50e36dbfcc798564daff5c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290600 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39010} 3 年前
mock_rtp_sender_internal.h
Implement codec selection api The implementation covers the latest specification, but does not support mixed-codec simulcast at the moment. Changing codec for audio and video is supported. Bug: webrtc:15064 Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40616} 2 年前
mock_voice_media_receive_channel_interface.h
Rename Set(Send|Recv)Parameters Set(Sender|Receiver)Parameters following the previous change to rename the classes derived from cricket::RtpParameters Also rename ChangedRecvParameters to ChangedReceiveParameters. BUG=webrtc:13931 Change-Id: Ia51dd39905a5cbb98162c3948930e43ccaf3786d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314500 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40677} 2 年前
peer_connection_test_wrapper.cc
Implement codec selection api The implementation covers the latest specification, but does not support mixed-codec simulcast at the moment. Changing codec for audio and video is supported. Bug: webrtc:15064 Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40616} 2 年前
peer_connection_test_wrapper.h
Implement codec selection api The implementation covers the latest specification, but does not support mixed-codec simulcast at the moment. Changing codec for audio and video is supported. Bug: webrtc:15064 Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40616} 2 年前
rtc_stats_obtainer.h
Make header files self contained. This CL adds #includes to header files in order to make them self contained after the preprocessor pass. Bug: b/251890128 Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38327} 3 年前
rtp_transport_test_util.h
Fire MaybeSignalReadyToSend in a PostTask when recursive Speculative fix. Writing the test for it is more complex. Bug: chromium:1483874 Change-Id: Icfaf1524b0499c609010753e0b6f3cadbd0e98f9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321480 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40820} 2 年前
simulcast_layer_util.cc
Rename simulcast flow tests: PeerConnectionEncodingsIntegrationTest. This is a pure rename+move of PeerConnectionSimulcastMediaFlowTests. The reason for renaming is to reflect that a) this is an integration test, not a unit test, and b) not all of the tests use simulcast (some use a single encoding, i.e. singlecast or SVC). Shared helper functions between PeerConnectionEncodingsIntegrationTest and PeerConnectionSimulcastTests are placed in a test-only util file. # Already pass, no need to wait for chromium bots for webrtc testonly CL NOTRY=True Bug: webrtc:15063 Change-Id: Iec90d1a7ab712be1395c7644723422c8c6179974 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300540 Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39799} 3 年前
simulcast_layer_util.h
Rename simulcast flow tests: PeerConnectionEncodingsIntegrationTest. This is a pure rename+move of PeerConnectionSimulcastMediaFlowTests. The reason for renaming is to reflect that a) this is an integration test, not a unit test, and b) not all of the tests use simulcast (some use a single encoding, i.e. singlecast or SVC). Shared helper functions between PeerConnectionEncodingsIntegrationTest and PeerConnectionSimulcastTests are placed in a test-only util file. # Already pass, no need to wait for chromium bots for webrtc testonly CL NOTRY=True Bug: webrtc:15063 Change-Id: Iec90d1a7ab712be1395c7644723422c8c6179974 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300540 Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39799} 3 年前
srtp_test_util.h
Rename rtc_base/ssl_stream_adapter.h constants. Uppercase constants are more likely to conflict with macros (for example rtc::SRTP_AES128_CM_SHA1_80 and OpenSSL SRTP_AES128_CM_SHA1_80). This CL renames some constants and follows the C++ style guide. Bug: webrtc:12997 Change-Id: I2398232568b352f88afed571a9b698040bb81c30 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226564 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34553} 4 年前
svc_e2e_tests.cc
Use layer/encode target resolution in DropDueToSize It used input frame resolution before this change which caused unnecessary resolution adaptations when resolution scaling is used. Found that initial frame dropping was always enabled for AV1 SVC. After fixing DropDueToSize the AV1 SVC tests [1] started to fail ("number of encoded temporal layers is less than expected") on bots. The tests encode 1850x1110 in L3T3 for 5s using the default 300kbps start bitrate. Before the fix the initial frame dropping kicked in and reduced the resolution to a level that let encoder to generate all temporal layers. After the fix the resolution stayed at 1850x1110 and encoder dropped all T1 and T2 layer frames. Mitigated this by increasing test duration from 5 to 10s. This gives enough time for BWE to ramp up and for encoder to generate (stop dropping) all temporal layers. [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/test/svc_e2e_tests.cc;l=460;bpv=1 Bug: chromium:1466809 Change-Id: I16802689e234f8fc16f891f024d5f644985de01c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315142 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40536} 2 年前
test_sdp_strings.h
Reland "sdp: reject duplicate codecs with the same id but different name or clockrate" This is a reland of commit ad6807805d12e48f11c3a68b4befaf8d7c23e8b5 Original change's description: > sdp: reject duplicate codecs with the same id but different name or clockrate > > since something like > rtpmap:96 VP8/90000 > rtpmap:96 VP9/90000 > or > rtpmap:97 ISAC/32000 > rtpmap:97 ISAC/16000 > is wrong. Note that fmtp or rtcp-fb are not taken into account. > Also note that sending invalid static payload types now throws an error. > > Drive-by: replace "RtpMap" with "Rtpmap" for consistency. > > BUG=None > > Change-Id: I2574b82a6f1a0afe3edc866e514a5dbca0798e8c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263641 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> > Cr-Commit-Position: refs/heads/main@{#37028} Bug: webrtc:14140 Change-Id: I63a37aacea6b9e0a9d7570b8422849275eb69aae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264544 Reviewed-by: Christoffer Jansson <jansson@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/main@{#37066} 3 年前