aboutsummaryrefslogtreecommitdiff
path: root/api
AgeCommit message (Collapse)Author
2022-12-27Merge commit 'upstream-main' into masterJorge E. Moreira
Bug: 261600888 Test: none, build files to be updated in follow up cl Change-Id: Ib520938290c6bbdee4a9f73b6419b6c947a96ec4
2022-12-02Use ScopedFieldTrials in FieldTrialsTestEmil Lundmark
Resetting the global state between runs was previously handled by a RAII type, but the semantics of that type changed to remove this behavior in [1]. [1] https://webrtc-review.googlesource.com/c/src/+/276269 Bug: webrtc:14731, webrtc:14705 Change-Id: I8425cb71f49ea000434d500e0b3978324e4c3195 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285782 Commit-Queue: Emil Lundmark <lndmrk@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38800}
2022-12-02Rename test helper for registering field trial keysEmil Lundmark
This new name emphasizes that the field trial keys are only allowed within the current scope. We already have test::ScopedFieldTrials that can be used to ensure that the global field trials string itself is isolated. Bug: webrtc:14705 Change-Id: I8b66bbd9c11d97985292c334d2d3496a047074a1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284862 Commit-Queue: Emil Lundmark <lndmrk@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38796}
2022-12-01Update some audio modules with new OWNERSHenrik Lundin
Bug: b/260832909 Change-Id: I3d2ebad978988eabf228475c3fc46708e12cf5d2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285780 Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Christoffer Jansson <jansson@webrtc.org> Commit-Queue: Christoffer Jansson <jansson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38788}
2022-12-01Delete deprecated disable_ipv6 flag.Henrik Boström
M108 Stable has been released, which does not contain googIPv6 anymore, and today the last downstream dependency on this flag was removed. Let's delete! Bug: webrtc:14608 Change-Id: Ia2d201f0da04b14961f891687b6135fc69b7767e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285720 Auto-Submit: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38786}
2022-11-30Extend mocks for public typesJack Smith
Extends the mocks for rtpreceiver rtpsender and videotrack. This change allows the external HangoutsKit client to remove its own mocks of rtc types. Bug: none Change-Id: I8ba1752fe7633f9e0bba264a1279f74cc1368a2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282900 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jack Smith <jackdsmith@google.com> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38782}
2022-11-30Remove xooglers from WATCHLISTS and OWNERSChristoffer Jansson
Bug: b/260832909 Change-Id: I683c714da35c21c23404d4b1c6500da28d680ed5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285470 Commit-Queue: Christoffer Jansson <jansson@google.com> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38777}
2022-11-29Limit number of TURN servers to 32Philipp Hancke
Limit the number of TURN servers to 32 in order to allow the prioritization to assume a fixed offset for (de)prioritizing candidates. See https://github.com/w3c/webrtc-pc/pull/2679 for discussion including some data on current usage. Guarded by WebRTC-LimitTurnServers which is used as a killswitch. BUG=webrtc:13195 Change-Id: Ib12726af426ae4238aa7eb6aa062c71af52d495f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285340 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38767}
2022-11-29Rename VideoFrameMetadata tests to RTPVideoHeaderTest.Henrik Boström
This is a pure move/rename. The reason for wanting the tests in RTPVideoHeader is that it is the GetAsMetadata() function that we are testing and in a future CL we'll also want to test SetFromMetadata(). // Bots green, no need to wait for the remaining ones, just a move NOTRY=True Bug: webrtc:14709 Change-Id: Iecb938e79e7e8d55e208baea190eef4c6730158e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285460 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38764}
2022-11-29Add more information to RTPVideoHeader::GetAsMetadata().Henrik Boström
Update GetAsMetadata() to include more of the RTPVideoHeader metadata. The intent is to be able to both get and set all of these from JavaScript behind a flag. Planned follow-up CLs: 1. Also get codecs-specifics, starting with VP8. 2. Test refactoring/rename: Move tests to RTPVideoHeaderTest. 3. Add RTPVideoHeader::SetFromMetadata() covering everything gettable. 4. Chrome plumbing. Bug: webrtc:14709 Change-Id: I78679b9ff4ca749d50f309a1713e71ceabb826dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285084 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38756}
2022-11-25Add RTPVideoHeader::GetAsMetadata().Henrik Boström
In preparation of adding RTPVideoHeader::SetFromMetadata() method, the VideoFrameMetadata construct-from-RTPVideoHeader is replaced by RTPVideoHeader::GetAsMetadata(). This serves two purposes: 1. Having "GetAs" and "SetFrom" in the same file reduces the risk of these two methods getting out of sync as we expand its usage. 2. This is necessary to avoid a circular dependency that would otherwise be introduced by RTPVideoHeader::SetFromMetadata(). Bug: webrtc:14709 Change-Id: I127b3d15f9a8c6af210449a5a50d414c9ba79930 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285080 Reviewed-by: Tony Herre <herre@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38735}
2022-11-25Add a clone method to the video frame transformer API.Harald Alvestrand
This will clone an encoded video frame into a sender frame, preserving metadata as much as possible. Bug: webrtc:14708 Change-Id: I6f68d2ee65ef85c32cc3c142a41346b81ba73533 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284701 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38733}
2022-11-18[DVQA] Add QP metric to the video analyzer.Artem Titov
Bug: b/240540204 Change-Id: I43fbb779bac10e27f2607ce1545476b1389d7c69 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283763 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38686}
2022-11-18Reland "[DVQA] Create separate BUILD.gn file for video analyzer"Ilya Nikolaevskiy
This reverts commit 76793c300fdd87fa8fd8be3dd2e5faf8c1916e96. Reason for revert: Can't cleanly revert the old one. A forward fix will be provided. Original change's description: > Revert "[DVQA] Create separate BUILD.gn file for video analyzer" > > This reverts commit 116c0a53d4a35c6dee857eb4cc2b6ae233a0427c. > > Reason for revert: Breaks bot: https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_compile_dbg_ng/1415352/overview > > > Original change's description: > > [DVQA] Create separate BUILD.gn file for video analyzer > > > > Bug: None > > Change-Id: I37dd2262bf3f52b2f5abe7934b9c41eaa27ffd17 > > No-try: True > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283141 > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#38662} > > Bug: None > Change-Id: Ieeb8c569560cb9d60d0c4d3c1268fa57f56b8157 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284000 > Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38672} Bug: None Change-Id: I74506eaa6a1060bf87e651881c86b4f576f447ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38676}
2022-11-18Revert "[DVQA] Create separate BUILD.gn file for video analyzer"Ilya Nikolaevskiy
This reverts commit 116c0a53d4a35c6dee857eb4cc2b6ae233a0427c. Reason for revert: Breaks bot: https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_compile_dbg_ng/1415352/overview Original change's description: > [DVQA] Create separate BUILD.gn file for video analyzer > > Bug: None > Change-Id: I37dd2262bf3f52b2f5abe7934b9c41eaa27ffd17 > No-try: True > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283141 > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38662} Bug: None Change-Id: Ieeb8c569560cb9d60d0c4d3c1268fa57f56b8157 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284000 Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38672}
2022-11-17Reland "[ACM] iSAC audio codec removed"Alessio Bazzica
This is a reland of commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1 Original change's description: > [ACM] iSAC audio codec removed > > Note: this CL has to leave behind one part of iSAC, which is its VAD > currently used by AGC1 in APM. The target visibility has been > restricted and the VAD will be removed together with AGC1 when the > time comes. > > Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319 > > Bug: webrtc:14450 > Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421 > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38652} Bug: webrtc:14450 Change-Id: Ia22c4d7724b6022238235fede93e36e570a49376 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283843 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38665}
2022-11-17[DVQA] Create separate BUILD.gn file for video analyzerArtem Titov
Bug: None Change-Id: I37dd2262bf3f52b2f5abe7934b9c41eaa27ffd17 No-try: True Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283141 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38662}
2022-11-17Remove deprecated AddPeer method.Jeremy Leconte
Change-Id: Icd15dc4d7d79276734260fb11932d9ede8dbbf23 Bug: webrtc:14627 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283661 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#38659}
2022-11-16Revert "[ACM] iSAC audio codec removed"Alessio Bazzica
This reverts commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1. Reason for revert: breaks a downstream project Original change's description: > [ACM] iSAC audio codec removed > > Note: this CL has to leave behind one part of iSAC, which is its VAD > currently used by AGC1 in APM. The target visibility has been > restricted and the VAD will be removed together with AGC1 when the > time comes. > > Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319 > > Bug: webrtc:14450 > Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421 > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38652} Bug: webrtc:14450 Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38655}
2022-11-16[ACM] iSAC audio codec removedAlessio Bazzica
Note: this CL has to leave behind one part of iSAC, which is its VAD currently used by AGC1 in APM. The target visibility has been restricted and the VAD will be removed together with AGC1 when the time comes. Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319 Bug: webrtc:14450 Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38652}
2022-11-15pc: Add asynchronous RtpSender::SetParameters() callFlorent Castelli
As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-11-15Add parameter to control the pacer's burst outside of field trials.Henrik Boström
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}
2022-11-14Remove deprecated API for emulated network statsArtem Titov
Bug: None Change-Id: Ib70a117d67002d108474214490ed1a8bb61da463 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283140 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38619}
2022-11-14Delete api/stats_types.h in favor of api/legacy_stats_types.hHenrik Boström
The file was renamed, see https://groups.google.com/u/1/g/discuss-webrtc/c/ZQiP4f_bpw4 Bug: webrtc:14180 Change-Id: Ia76c85ba7d9da6b3a93d0a67a4b6a5187e07e230 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283084 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38616}
2022-11-12Add infrastructure stats for network emulation layerArtem Titov
Bug: b/240540204 Change-Id: I66dfd25775faa9d1bc7e75a932a36e8aa97c0f57 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38613}
2022-11-11Rename api/stats_types.h to api/legacy_stats_types.h.Henrik Boström
As to not break downstream projects, the old name api/stats_types.h is kept around to help include api/legacy_stats_types.h. We can delete this in a follow-up. NOTRY=True Bug: webrtc:14180 Change-Id: I270ca5e366ae36e324cbc9f982bbb066ab92d203 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283081 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38610}
2022-11-10Add a new PeerConnectionE2EQualityTestFixture::AddPeer method.Jeremy Leconte
Change-Id: Ic5879613db51a00e3e958931f5eda19fda1ae94a Bug: webrtc:14627 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282640 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38608}
2022-11-10Metronome: complete API migration.Markus Handell
This CL finalizes the Metronome refactor undertaken in crbug.com/1381982 and enables it again in call.cc. Fixed: chromium:1381982 Change-Id: I1642103e9c8a3f2a1f12d7635a1b27310802c1c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282920 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38605}
2022-11-10Revert "Add checks for api/test mocks to make sure they're complete"Florent Castelli
This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb. Reason for revert: Breaks upstream. Original change's description: > Add checks for api/test mocks to make sure they're complete > > Also unifies the mock inheritance if they inherited from a ref counted > interface: > - it should only inherit from the interface > - it should use make_ref_counted > > Bug: webrtc:14594 > Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38602} Bug: webrtc:14594 Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38604}
2022-11-10Remove PeerConfigurer interface.Jeremy Leconte
PeerConfigurerImpl is renamed to PeerConfigurer. Change-Id: Ie52c581126c21740536d42ff4831f0c4ed445ea4 Bug: webrtc:14627 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281883 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#38603}
2022-11-10Add checks for api/test mocks to make sure they're completeFlorent Castelli
Also unifies the mock inheritance if they inherited from a ref counted interface: - it should only inherit from the interface - it should use make_ref_counted Bug: webrtc:14594 Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38602}
2022-11-09[stats] Mark codec implementation stats as exposing hardware capabilityEvan Shrubsole
This means that these stats will be filtered out by JavaScript unless the conditions for exposing hardware capabilities are met. These conditions are described in the webrtc-stats spec at https://w3c.github.io/webrtc-stats/#limiting-exposure-of-hardware-capabilities. R=hbos@webrtc.org Bug: chromium:1369050,chromium:1369049 Change-Id: I05bdb72ef6789417488c7e786e8713ce99a91f8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279960 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38594}
2022-11-09Simplify Network Emulation stats APIArtem Titov
Bug: b/240540204 Change-Id: I669b5b01d0a10ae5d8f0bafa661dbda6fc9260b7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282420 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38590}
2022-11-08Metronome: disable & refactor for single-threaded operation.Markus Handell
The Chromium implementation unfortunately has a rare deadlock. Rather than patching that up, we're changing the metronome implementation to be able to use a single-threaded environment instead. The metronome functionality is disabled in VideoReceiveStream2 construction inside call.cc. The new design does not have listener registration or deresigstration and instead accepts and invokes callbacks, on the same sequence that requested the callback. This allows the clients to use features such as WeakPtrFactories or ScopedThreadSafety for cancellation. The CL will be followed up with cleanup CLs that removes registration APIs once downstream consumers have adapted. Bug: chromium:1381982 Change-Id: I43732d1971e2276c39b431a04365cd2fc3c55c25 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282280 Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38582}
2022-11-08Add power efficient stats to RTC statsEvan Shrubsole
As the exposure of power efficient stats to JavaScript are limited as to reduce the fingerprinting surface to getStats, a new RTCStatsMember derivation, RTCLimitedStatsMember, was added in this change. This sets the exposure criteria of the stat on the type, which keeps the size of the RTCStatsMember class the same and allows for extension in the future for new types of stat restrictions. Bug: webrtc:14483 Change-Id: Ib0303050a112441ba2416fd5f004dd8be26b47ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279021 Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38576}
2022-11-07Reland "Change default NetEq sample rate to 48k."Jakob Ivarsson
This is a reland of commit 38fcd58429b29c9474f1647efed7ebeb543c0637 Original change's description: > Change default NetEq sample rate to 48k. > > This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k). > > Bug: none > Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662 > Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38536} Bug: none Change-Id: Id634799286f6d1f1eaf315ebe8e70de669d589db Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281900 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38572}
2022-11-07Expose frame_buffer GN targetEvan Shrubsole
Bug: None Change-Id: I75068b87e95575235eb937ef73279f961d0df93e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282322 Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38571}
2022-11-07Use classes from media_configuration.h instead of the ones in ↵Jeremy Leconte
PeerConnectionE2EQualityTestFixture. Classes defined inside the class PeerConnectionE2EQualityTestFixture are replaced by the ones define in media_configuration.h. Change-Id: I1c025ff10aacf8cbc3df9bfa742a40622fe0807a Bug: webrtc:14627 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281860 Commit-Queue: Jeremy Leconte <jleconte@google.com> Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38568}
2022-11-07generateKeyFrame: add rids argumentPhilipp Hancke
and do the resolution of rids to layers. This has no effect yet since the simulcast encoder adapter (SimulcastEncoderAdapter::Encode), the VP8 encoder (LibvpxVp8Encoder::Encode) and the OpenH264 encoder (H264EncoderImpl::Encode) all generate a key frame for all layers whenever a key frame is requested on one layer. BUG=chromium:1354101 Change-Id: I13f5f1bf136839a68942b0f6bf4f2d5890415250 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280945 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38565}
2022-11-07Move media configuration classes out of PeerConnectionE2EQualityTestFixture.Jeremy Leconte
The goal is to remove the dependency between PeerConfigurerImpl and PeerConnectionE2EQualityTestFixture so that PeerConfigurerImpl can be used in PeerConnectionE2EQualityTestFixture API. Change-Id: I29ae44b9d0e39075d0c395ff9d9f8d313be12176 Bug: webrtc:14627 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281740 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#38560}
2022-11-06Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."Mirko Bonadei
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}
2022-11-04Move PeerConfigurerImpl to the test public api.Jeremy Leconte
End goal is to remove PeerConnectionE2EQualityTestFixture::PeerConfigurer interface. Change-Id: I4a6aa0ab1fb5a0d6f85154159b7da16de9b53059 Bug: webrtc:14627 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281501 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#38551}
2022-11-03[PCLF] Propagate relevant metadata to all metricsArtem Titov
Bug: None Change-Id: Ifcb67a59b68cc3468dd06e932a2a3da7b40d9845 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281680 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38545}
2022-11-02Revert "Change default NetEq sample rate to 48k."Jakob Ivarsson‎
This reverts commit 38fcd58429b29c9474f1647efed7ebeb543c0637. Reason for revert: Breaks downstream test Original change's description: > Change default NetEq sample rate to 48k. > > This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k). > > Bug: none > Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662 > Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38536} Bug: none Change-Id: I03181168ab14d2d99320767c3a25ba3cfb726b2c No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281441 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Auto-Submit: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38538}
2022-11-02Change default NetEq sample rate to 48k.Jakob Ivarsson
This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k). Bug: none Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38536}
2022-11-02stats: implement candidate-pair lastPacket(Sent|Received)TimestampPhilipp Hancke
https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-lastpacketsenttimestamp https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-lastpacketreceivedtimestamp which are useful together with the ice-restart-necessary logic mentioned in https://w3c.github.io/webrtc-pc/#dictionary-rtcofferoptions-members BUG=webrtc:14619 Change-Id: I4a8ab00a37fbd4af8b948720c83787cbdfc6b9a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281281 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#38534}
2022-11-02[Stats] Delete 'track' metrics that have previously been moved.Henrik Boström
These have all been moved to "inbound-rtp" and now that upstream projects have migrated we can delete the old location. Unblocks https://crbug.com/webrtc/14175 Bug: webrtc:14521, webrtc:14524 Change-Id: Ia2bfa399d62304cc0ead0e65c340dfad20acc530 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281183 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38532}
2022-11-02Remove unused MetricsLoggerAndExporterArtem Titov
Bug: None Change-Id: I9e05e5c29cd80bf991bd50c3bd4ee4f09ddf8134 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281420 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38531}
2022-10-31Add IWYU pragmas for some api headersDanil Chapovalov
Bug: None Change-Id: I1912e05dbc31d960f36c97151dcb387446535c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280965 Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38510}
2022-10-31Unship track.totalFramesDuration/sumSquaredFrameDurations.Henrik Boström
These metrics were not only non-standard, but residing in the non-standard "track" stats object that we want to delete. As per https://github.com/w3c/webrtc-stats/issues/695#issuecomment-1259611462 these metrics are no longer needed because we already have inbound-rtp.totalInterFrameDelay/totalSquaredInterFrameDelay which is basically the same thing. // mac_rel infra failures are unrelated NOTRY=True Bug: webrtc:14522 Change-Id: I565da42514a93f15532ba8357dd006547a5296ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278100 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38509}
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-21 22:54:24 +0000