Commit Graph

34664 Commits

Author SHA1 Message Date
abd2ed0911 Update WebRTC code version (2021-09-18T04:02:37).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I3a7e3b5137b0b29202295dba11e0d6437150ab55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232364
Reviewed-by: 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@{#35032}
2021-09-18 05:41:19 +00:00
3bb3e4ecb3 Roll chromium_revision 991c09e243..e355563bce (922552:922741)
Change log: 991c09e243..e355563bce
Full diff: 991c09e243..e355563bce

Changed dependencies
* src/base: 74c3937fa4..d632224756
* src/build: 11ee2b49e0..67d9786cde
* src/ios: 9ddd8a846f..fba9465d9b
* src/testing: 4a78839643..7c2699b123
* src/third_party: 2425c9743f..3082261eb3
* src/third_party/androidx: VfpXiVfbJgXKD03viqELXHcjvBvx22iMSUlxHnRT-vIC..HoRt7lhvtWPJ6Dq388HPGC34ymlvXafBdnrSvb_XaIgC
* src/third_party/breakpad/breakpad: 7933ec0a69..1147c2fcf0
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/57c353ca10..7d0db1cb9e
* src/third_party/depot_tools: 5a0f43aebe..4e0cca2a31
* src/third_party/perfetto: a9118d7690..62ae508a04
* src/tools: 3cbcc67ae2..7cab6725ec
DEPS diff: 991c09e243..e355563bce/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I3b6d63c1ea42f51f9dacb06d13546ab2b34b3970
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232362
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#35031}
2021-09-18 01:03:25 +00:00
edce7b3a24 Roll chromium_revision a098e1265b..991c09e243 (921682:922552)
Change log: a098e1265b..991c09e243
Full diff: a098e1265b..991c09e243

Changed dependencies
* src/base: ecf6ff89bf..74c3937fa4
* src/build: fabb3638a7..11ee2b49e0
* src/buildtools/third_party/libc++abi/trunk: a5b6419452..c883cb129d
* src/buildtools/third_party/libunwind/trunk: 44ea7aba6a..a7e4ce09ec
* src/ios: e8d5d843cf..9ddd8a846f
* src/testing: 7adcb6a6bf..4a78839643
* src/third_party: 21d7875bec..2425c9743f
* src/third_party/androidx: QRXHawrfyCC8nr_u3VCPoilYsqHO01yv1EiduoQUPcgC..VfpXiVfbJgXKD03viqELXHcjvBvx22iMSUlxHnRT-vIC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6b072fac88..57c353ca10
* src/third_party/depot_tools: 9c67b23020..5a0f43aebe
* src/third_party/perfetto: de98afcd4a..a9118d7690
* src/third_party/r8: j4P6kTwiJeMkYRbaOqRCB-ZrYyISP0NrcyNNAd8MRM0C..ntKr-aGawyw4goIV50HidiUHTHey-mwhwgLiD3Th8jUC
* src/third_party/turbine: Om6yIEXgJxuqghErK29h9RcMH6VaymMbxwScwXmcN6EC..n2quuVAecebwosy5EPRjDGSO5EQU94aFRBg_EdrYtpsC
* src/tools: a27c09f45b..3cbcc67ae2
* src/tools/luci-go: git_revision:dc21267f89df3981e809fe9566591d6916ae758b..git_revision:028cd41e0f4b2bec99d94c780caf2f978e09b182
* src/tools/luci-go: git_revision:dc21267f89df3981e809fe9566591d6916ae758b..git_revision:028cd41e0f4b2bec99d94c780caf2f978e09b182
* src/tools/luci-go: git_revision:dc21267f89df3981e809fe9566591d6916ae758b..git_revision:028cd41e0f4b2bec99d94c780caf2f978e09b182
DEPS diff: a098e1265b..991c09e243/DEPS

Clang version changed llvmorg-14-init-3710-gd11a0c5d:llvmorg-14-init-3940-gafc45ff0
Details: a098e1265b..991c09e243/tools/clang/scripts/update.py

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I365777bb670246c421a75662740808f1dee54fb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232344
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#35030}
2021-09-17 17:23:03 +00:00
4397281f38 dcsctp: implement socket handover in the DcSctpSocket class and expose the functionality in the API
Bug: webrtc:13154
Change-Id: Idf4f4028c8e65943cb6b41fab0baef1b3584205d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232126
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35029}
2021-09-17 15:19:01 +00:00
4893dbe7f1 Update link to Chromium Modern C++ style guide
Bug: chromium:1243839
Change-Id: I4a914eec78c919347a65be4d5b5fc6447408bc39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232260
Commit-Queue: Joe Mason <joenotcharles@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35028}
2021-09-17 14:06:01 +00:00
62b340545f Default the safetyped calculation of packet interarrival times.
This defaults the calculation landed in cl 196502. The less readable legacy calculation method will be deleted in a future CL.


Bug: none
Change-Id: Ida02a5208e354835b964c69355ad1e9d5bba18aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231956
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35027}
2021-09-17 13:58:55 +00:00
1b200b93d5 APM: remove webrtc::Config
Remove the deprecated way of configuring APM.

Bug: webrtc:5298
Change-Id: Idcedf1fe4a121adfcf2881003579cd58ac42a2b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232302
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35026}
2021-09-17 11:21:32 +00:00
be1b8989d1 ExperimentalNs removed + APM not depending anymore on webrtc::Config
Thanks to the elimination of `ExperimentalNs`, there is no need anymore
to pass `webrtc::Config` to build APM.
Hence, `AudioProcessingBuilder::Create(const webrtc::Config&)` is also
removed.

Bug: webrtc:5298
Change-Id: I0a3482376a7753434486fe564681f7b9f83939c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232128
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35025}
2021-09-17 10:53:43 +00:00
25b5e08094 dcsctp: implement congestion control errata
When re-reading through the errata for RFC4960 in RFC8540, it was found
that two erratas were not applied to dcSCTP:

https://datatracker.ietf.org/doc/html/rfc8540#section-3.11
https://datatracker.ietf.org/doc/html/rfc8540#section-3.12

They are now applied. Re-running throughput tests show no difference.

Bug: webrtc:12943
Change-Id: I9d73d0d257eab8442954924dc414d8efa2c41e8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232221
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35024}
2021-09-17 10:49:13 +00:00
d67e75a45d Update m1 arm64 audio bitexactness tests hashes. Followup.
Update hashes that weren't updated at
https://webrtc-review.googlesource.com/c/src/+/232220

Bug: b/199885455
Change-Id: I4f9327fa48a9d5656f57c5dcbc6438f20f919acd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232320
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35023}
2021-09-17 10:19:01 +00:00
4420380f38 Use string_view as input type for internal string utilities
Bug: None
Change-Id: I2bfdaf4e7fac109842cc9fde8dfa28ab4961c3fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232127
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35022}
2021-09-17 09:11:21 +00:00
d14f98f635 Update m1 arm64 audio bitexactness tests hashes
Follow up for https://webrtc-review.googlesource.com/c/src/+/232061/5. Updated mac M1 tests that was missed because they are not part of CQ

Bug: b/199885455
Change-Id: I77618ac2869ba601f322857f4391b63220d20252
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232220
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35021}
2021-09-17 08:59:05 +00:00
5068820dd0 Rename "TcpFairness" in LossBasedBweV2 to "InstantUpperBound"
The new name more accurately reflects the intent of the actual implementation.


Bug: none
Change-Id: I3d2aeb561104165f9f9879854a4a210730e02ff5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232130
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35020}
2021-09-17 07:28:41 +00:00
e28c38d7fb Update WebRTC code version (2021-09-17T04:04:12).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: Iadab8056f9b14796e86451b672e1753790d8f125
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232282
Reviewed-by: 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@{#35019}
2021-09-17 05:57:39 +00:00
0a54e7a101 Verify that locks are handled correctly on RTCAudioSession
This CL is for the same behavior as before [1], to emit the NSError
when an application is not using the RTCAudioSession lock correctly.

[1] https://webrtc-review.googlesource.com/c/src/+/207432

Bug: webrtc:13091
Change-Id: I031b0e963d33c92ce1af7a306edfa6be005e043d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229461
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35018}
2021-09-17 01:41:42 +00:00
702f2a44ba [ScreenCapturerX11] Fix update-region for monitors with offsets.
This CL ensures that each DesktopFrame's updated-region is expressed in
the frame's own coordinates, where the top-left is always (0, 0).
For example, DesktopFrame::GetFrameDataAtPos() and its callers use
this coordinate system.

Previously, whenever a RANDR monitor with a non-zero offset was
selected, ScreenCapturerX11 would hit some DCHECKs when trying to
copy pixels from previous frames, or when capturing new pixels into
them from XDAMAGE regions.

Bug: None
Change-Id: I7b2e8d0449359ee7b263ad60af193e2bf89aa1f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232085
Reviewed-by: Joe Downing <joedow@chromium.org>
Commit-Queue: Joe Downing <joedow@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35017}
2021-09-16 17:12:26 +00:00
80e96de5ba dcsctp: Add more consistency checks
When there is no outstanding data, then next TSN to allocate should
always be one more than what the client has last ACKed.

Bug: None
Change-Id: Ieb8b5b23912d77d96fe3749fb53fd53652d97066
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232002
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35016}
2021-09-16 13:31:51 +00:00
ff7e1bad1f APM config: remove ExperimentalAgc
Bug: webrtc:5298,webrtc:7494
Change-Id: Ic9bcb702603ec7900fbe9ae38ab49dff8fe99318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219463
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35015}
2021-09-16 13:28:51 +00:00
54e4e35c89 dcsctp: Add consistency check for assembled msgs
The buffer of reassembled messages in ReassemblyQueue is only to be
used while processing a DATA/I-DATA or FORWARD-TSN as processing these
chunks may result in assembling messages.

When the socket is idle - between API calls - it's supposed to be empty.

Instead of having it as a member in ReassemblyQueue, it could be
provided as an argument to ReassemblyQueue::Add and
ReassemblyQueue::Handle(ForwardTSN), but that would be a quite big
refactoring. That will be investigated separately.

Bug: None
Change-Id: I41238de28f32f2a622c1d045debe3ea11e7c23f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232000
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35014}
2021-09-16 13:19:31 +00:00
fed091edf4 dcsctp: Move last_assembled_tsn_watermark further
The ReassemblyQueue will need to track which messages that have already
been delivered to the client so that they are not re-delivered on e.g.
retransmissions. It does that by tracking which TSNs that those messages
were built from. It tracks that in two variables,
`last_assembled_tsn_watermark` and `delivered_tsns_`, where the first
one represent that all TSNs including and prior this one have been
delivered and `delivered_tsns` contain additional ones when there are
gaps.

When receiving a FORWARD-TSN and asked to forget about some partially
received messages, these two variables were updated correctly, but the
`delivered_tsns_` were left in a state where it could be adjacent to the
`last_assembled_tsn_watermark` - when `last_assembled_tsn_watermark`
could actually have been moved further.

Added consistency check (that would trigger in existing tests) and
fixing the issue.

This bug is quite benign, as any received chunk would've corrected the
problem, and even at this faulty state, the ReassemblyQueue would
function completely fine.

Bug: webrtc:13154
Change-Id: Iaa7c612999c9dc609fc6e2fb3be2d0bd04534c90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232124
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35013}
2021-09-16 13:14:12 +00:00
3b08fe3dcd dssctp: support socket handover in StreamResetHandler
Bug: webrtc:13154
Change-Id: Idafbed4f3c1af8d0cca833ba983c4b4b99118335
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232121
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35012}
2021-09-16 11:33:21 +00:00
17b7a68bd2 Update h264 sps parsers and sps vui rewriter to use BitstreamReader
The new version is subjectivly cleaner
and objectively generates smaller binary size

Bug: None
Change-Id: I8d845f56f13dbc7d34e4d685f735a448c5fe8f06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232001
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35011}
2021-09-16 10:48:41 +00:00
e61d4c83ef Return proxied object in OnTransceiver
This makes it possible to invoke methods on the transceiver object
from any thread.

Also makes a few of the mock observer objects thread-safe, to allow
testing when the main thread is not the signaling thread.

Bug: webrtc:13183
Change-Id: Ic97efef71a21c3075700a028103061032f8d2bcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35010}
2021-09-16 09:40:52 +00:00
8f486f94e6 dcsctp: support socket handover in RetransmissionQueue
Bug: webrtc:13154
Change-Id: I9c73b1153b65409eb026e015804c22f3e874ff82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232022
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35009}
2021-09-16 05:18:29 +00:00
e742d8c163 Updae bitexactness tests to match new canonical results
This CL has to be submitted together with chromium roll

Roll chromium_revision 48501b3f18..a098e1265b (917742:921682)

Change log: 48501b3f18..a098e1265b
Full diff: 48501b3f18..a098e1265b

Changed dependencies
* src/base: bdbd6f899c..ecf6ff89bf
* src/build: f90eed6a5e..fabb3638a7
* src/buildtools/third_party/libc++abi/trunk: 7de86cbf37..a5b6419452
* src/buildtools/third_party/libunwind/trunk: edf77b2d2d..44ea7aba6a
* src/ios: 1b17fd57e6..e8d5d843cf
* src/testing: 59835db543..7adcb6a6bf
* src/third_party: a299c990bc..21d7875bec
* src/third_party/android_deps/libs/com_google_auto_auto_common: version:2@0.10.cr0..version:2@1.1.2.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_annotation: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_annotations: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_check_api: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_core: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_type_annotations: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_guava_guava: version:2@30.1-jre.cr0..version:2@30.1.1-jre.cr0
* src/third_party/androidx: TnotTDnWGUJDh0mSOMrgnIwzbfWFiwo2NTtr2SlUJ0MC..QRXHawrfyCC8nr_u3VCPoilYsqHO01yv1EiduoQUPcgC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/dddb60eb97..27a3328a37
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d9a9ebbe89..6b072fac88
* src/third_party/depot_tools: 789dfc223b..9c67b23020
* src/third_party/freetype/src: 7482c98f15..5b626281f1
* src/third_party/googletest/src: 955c7f837e..159c9ad23e
* src/third_party/harfbuzz-ng/src: 280366ba6a..6602cbb706
* src/third_party/icu: ece15d049f..3f443830bd
* src/third_party/libvpx/source/libvpx: 15a75b4530..65a1751e5b
* src/third_party/libyuv: 49ebc996aa..0896c34873
* src/third_party/perfetto: 8420673b4c..de98afcd4a
* src/third_party/r8: dvPOJ_8iAF6OHGO79d86VbJjyKj7Xn0SFxlVVC9LHdcC..j4P6kTwiJeMkYRbaOqRCB-ZrYyISP0NrcyNNAd8MRM0C
* src/third_party/usrsctp/usrsctplib: bdf3dd3f28..62d7d0c928
* src/tools: e3721e5cf9..a27c09f45b
* src/tools/luci-go: git_revision:7f42370cb3b75398bdb9ae0aabe215a70d40cd31..git_revision:dc21267f89df3981e809fe9566591d6916ae758b
* src/tools/luci-go: git_revision:7f42370cb3b75398bdb9ae0aabe215a70d40cd31..git_revision:dc21267f89df3981e809fe9566591d6916ae758b
* src/tools/luci-go: git_revision:7f42370cb3b75398bdb9ae0aabe215a70d40cd31..git_revision:dc21267f89df3981e809fe9566591d6916ae758b
Added dependency
* src/third_party/android_deps/libs/org_checkerframework_dataflow_errorprone
Removed dependency
* src/third_party/android_deps/libs/org_checkerframework_dataflow_shaded
DEPS diff: 48501b3f18..a098e1265b/DEPS

Clang version changed llvmorg-14-init-2175-g945cde8b:llvmorg-14-init-3710-gd11a0c5d
Details: 48501b3f18..a098e1265b/tools/clang/scripts/update.py

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,marpan@webrtc.org, jianj@chromium.org,

Bug: b/199885455
Change-Id: I64329c57949e9233f5cf65167f012db0637fe4fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232061
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35008}
2021-09-15 18:38:41 +00:00
3987e61086 GCC: fix template specialization in webrtc::BitstreamReader
GCC complains that explicit specialization in non-namespace scope
is happening for webrtc::BitstreamReader::Read(). However,
specializationvfor bool isn't used because std::is_unsigned<bool>::value
returns true. Add std::is_same for bool check and enable second
specialization only for bool types.

Bug: chromium:819294
Change-Id: I1873cd59e2737516bd4012fb952da65d6bf3172b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231561
Reviewed-by: 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@{#35007}
2021-09-15 17:20:50 +00:00
b6b29030f6 Add .vpython3 file to webrtc
It is needed to fix chromium roll. Tested here: https://webrtc-review.googlesource.com/c/src/+/232040 (the few red bots are unrelated).

Bug: b/199885455
Change-Id: I726228fe44aae1009a1934c451438a2d01002ab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232023
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35006}
2021-09-15 16:56:30 +00:00
1061686107 red: handle opus dtx 400ms timestamp gap
by not encoding redundancy. The timestamp gap of 400ms means a
rtp timestamp difference of 19200 which would overflow the 14 bit
RED timestamp difference field.

To test in Chrome, go to
  https://webrtc.github.io/samples/src/content/peerconnection/audio/
set `useDtx = true` in the console and be very quiet.

BUG=webrtc:13182,webrtc:11640

Change-Id: I1cedc7d846ac7ae821bb7cb5c0f37a17511ac727
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231940
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35005}
2021-09-15 15:09:28 +00:00
b01e6457fe Reland "Reland "Enable WebRTC-Vp9DependencyDescriptor by default""
This is a reland of b062829311bf1962a7f264cecf36d17ef41951df

> Reland "Enable WebRTC-Vp9DependencyDescriptor by default"
>
> This is a reland of 472707150662bc4e174072e445938e5c405aa884
>
> Original change's description:
> > Enable WebRTC-Vp9DependencyDescriptor by default
> >
> > Bug: chromium:1178444
> > Change-Id: I420e1e9b3c557b8b186cb08c15b962a779e1ca17
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226941
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34584}
>
> Bug: chromium:1178444
> Change-Id: I874412b41e657179be6ffbe399617e18a29ec804
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230121
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34890}

Bug: chromium:1178444
Change-Id: I5bb3e3bd2da26f9a24d5e8161bd66b447543fc8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231843
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35004}
2021-09-15 13:48:58 +00:00
b26863ed0c Reland "Handle scalability mode in QueryCodecSupport"
This reverts commit 74281bed5350af9c15f83e0b1aec5c5921dbf76f.

Reason for revert: Fixed unit test by removing VP9 profile 2 from encoder factory unit test since this is platform dependent.

Original change's description:
> Revert "Handle scalability mode in QueryCodecSupport"
>
> This reverts commit 715a14811883a642e3acca21fb6017f8a128c0a5.
>
> Reason for revert: Speculative revert. Breaks upstream project http://b/200009579
>
> Original change's description:
> > Handle scalability mode in QueryCodecSupport
> >
> > All valid scalability modes should be supported by the builtin
> > software decoder/encoder.
> >
> > Bug: chromium:1187565
> > Change-Id: If66105d210d5055019f35dae2f80a18ad4a70cdd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222642
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#34998}
>
> TBR=danilchap@webrtc.org,sprang@webrtc.org,kron@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: Ibf40d523c50791d73e2afdc3917892b859d2bcb6
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1187565
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232020
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Commit-Queue: Andrey Logvin <landrey@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35001}


Bug: chromium:1187565
Change-Id: I598a2a530b8fea22997bbb5910eb3b864d1e28a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232021
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35003}
2021-09-15 13:12:58 +00:00
9c1657cba8 dcsctp: support socket handover in ReassemblyQueue
Bug: webrtc:13154
Change-Id: I816e51dcd923ba6440480de5d5df9012e4af9e5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231958
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35002}
2021-09-15 10:49:57 +00:00
74281bed53 Revert "Handle scalability mode in QueryCodecSupport"
This reverts commit 715a14811883a642e3acca21fb6017f8a128c0a5.

Reason for revert: Speculative revert. Breaks upstream project http://b/200009579

Original change's description:
> Handle scalability mode in QueryCodecSupport
>
> All valid scalability modes should be supported by the builtin
> software decoder/encoder.
>
> Bug: chromium:1187565
> Change-Id: If66105d210d5055019f35dae2f80a18ad4a70cdd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222642
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34998}

TBR=danilchap@webrtc.org,sprang@webrtc.org,kron@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Ibf40d523c50791d73e2afdc3917892b859d2bcb6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1187565
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232020
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35001}
2021-09-15 10:45:41 +00:00
9a45e897d5 Update h264_bitstream and pps parsers to use BitstreamReader
The new version is subjectivly cleaner
and objectively generates smaller binary size

Bug: None
Change-Id: I6639b4a1455e745e0047339115b3d02ef81ce731
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231238
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35000}
2021-09-15 10:26:47 +00:00
10dc1a6d8b New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class.
Bug: webrtc:12579
Change-Id: Idea35983e204e4a3f8628d5b4eb587bbdbff5877
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227286
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34999}
2021-09-15 09:57:29 +00:00
715a148118 Handle scalability mode in QueryCodecSupport
All valid scalability modes should be supported by the builtin
software decoder/encoder.

Bug: chromium:1187565
Change-Id: If66105d210d5055019f35dae2f80a18ad4a70cdd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222642
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34998}
2021-09-15 09:38:12 +00:00
ac554ebbc5 Add VPN detection by mac-address for Windows
This patch adds VPN detection for windows
based on known MAC addresses.
- Cisco AnyConnect
- GlobalProtect Virtual Ethernet

Bug: webrtc:13097
Change-Id: Ia90ee50be0dc2dcd2e6e9de1493fdd2c5e7d9d3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230245
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34997}
2021-09-15 08:57:28 +00:00
225cd47445 dcsctp: implement handover in DataTracker
Bug: webrtc:13154
Change-Id: Ia8c41dcffd95dafd904ee630f2131b575fe833dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231955
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34996}
2021-09-15 07:30:18 +00:00
e229fb83fb Update WebRTC code version (2021-09-15T04:03:10).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I7deecf188cea2fb477d6d1f65ec04fc57ca52014
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231987
Reviewed-by: 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@{#34995}
2021-09-15 05:54:38 +00:00
bd917a13fb Don't invoke custom certificate verifier unless there is an error.
The custom callback is intended to override errors, so there's no
point in calling it if the status is ok.

Calling it during an otherwise successful verification was an
unintentional change from:
https://webrtc-review.googlesource.com/c/src/+/196941

This is misleading as the return value isn't even used.

Bug: chromium:1247577
Change-Id: Id74411f7364537a3225021e7631bc9ab962889ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231881
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34994}
2021-09-15 00:04:27 +00:00
bc503c95b5 Remove support for nack threshold.
The nack threshold feature is unlikely to provide any value, since
reordered packets are rare. This CL also removes the factory method
from the NackTracker class, since it did not add much value.

Bug: webrtc:10178
Change-Id: Ib6bece4a2d9f95bd4298799aaa15627f5c014b61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231953
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34993}
2021-09-14 16:51:17 +00:00
7ca630f8dd Fix iOS sim bot to for the new Chromium roll
Bug: b/199044368, chromium:1249456
Change-Id: I0d91abca55618d87dc8366315182729e2a2b1016
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231954
Reviewed-by: Nico Weber <thakis@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34992}
2021-09-14 16:16:33 +00:00
ad6b7a733a dcsctp: introduce handover API types and implement it for streams
Bug: webrtc:13154
Change-Id: Ifa250175af79b7adc87dbc2750054adc94b90bb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231842
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34991}
2021-09-14 13:47:03 +00:00
d0321c5e5a Deduplicate set of the rtp header extension uri constants
Bug: webrtc:7472
Change-Id: Ic0b4f2cc3374ba70a043310b5046d8bf91f0acbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231949
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34990}
2021-09-14 13:38:44 +00:00
dc8fc72204 Fix potential crash during SimulcastEncoderAdapter tear down.
On the Android and iOS platforms, occasionally crash when using the SimulcastEncoderAdapter.

The Android platform reverted,
In function `SimulcastEncoderAdapter::EncoderContext::Release`,
After executing `encoder_->RegisterEncodeCompleteCallback(nullptr)`
before execute `encoder_->Release()`

If the encoder thread is executed here,
```
// out/xxx/xxx/gen/sdk/android/generated_video_jni/VideoEncoderWrapper_jni.h
JNI_GENERATOR_EXPORT void Java_org_webrtc_VideoEncoderWrapper_nativeOnEncodedFrame(
    JNIEnv* env,
    jclass jcaller,
    jlong nativeVideoEncoderWrapper,
    jobject frame) {
  VideoEncoderWrapper* native = reinterpret_cast<VideoEncoderWrapper*>(nativeVideoEncoderWrapper);
  CHECK_NATIVE_PTR(env, jcaller, native, "OnEncodedFrame");
  return native->OnEncodedFrame(env, base::android::JavaParamRef<jobject>(env, frame)); // HERE
}
```
it will cause `native` to nullptr.

iOS also.

Bug: webrtc:13156
Change-Id: Id5563b3fa2c11606ae7b35de56bbaa6adba59b14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34989}
2021-09-14 09:15:22 +00:00
99c6ca0e66 Add danilchap@webrtc.org as owner of test/fuzzers/
Bug: None
Change-Id: I11cc6c4a17c8d0c310816c87f1b67d0a338ebe24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231941
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34988}
2021-09-14 08:48:03 +00:00
6b19d8273b Replace AV1X with AV1
* Replace "AV1X" with "AV1";
* Keep mapping of "AV1X" payload name to kVideoCodecAv1 to not break
support of injectable "AV1X".

Bug: webrtc:13166
Change-Id: I9a50481209209f3857bbf28f4ed529ee6972377e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231560
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34987}
2021-09-14 08:29:02 +00:00
52b9e1ecfb Ensure RtpVideoLayersAllocationExtension::Parse validate sanity of the output
This is tested by a simple unit test and a new fuzzer that verify that all that can be parsed also can be written.

Bug: webrtc:12000
Change-Id: I461aedf97d3dec6e8916e72110fa097c3b31c27f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231642
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34986}
2021-09-14 06:43:13 +00:00
f1384afda0 Update WebRTC code version (2021-09-14T04:04:50).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I2010ee67fc200a93732ccf67cdc82238bf75f0cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231921
Reviewed-by: 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@{#34985}
2021-09-14 05:31:20 +00:00
018cd3d6fc Avoid NACKing after DTX.
This is done by not adding missing packets to the NACK list if the number of samples per packet is too large.

Bug: webrtc:10178
Change-Id: If46398d6d05ea35f30d7028040d3b808559e950b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231841
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34984}
2021-09-13 16:31:42 +00:00
593b4d550d Pipewire: Use xdg-portal provided file descriptor
The documentation for `OpenPipeWireRemote()` says:
> Open a file descriptor to the PipeWire remote where the camera nodes
> are available. The file descriptor should be used to create a
> pw_core object, by using pw_context_connect_fd.

In `InitPipeWire()` we already successfully requested the FD, but then
went on and used the unrestricted default socket.
This does not matter in non-sandboxed environments, as the stream we
want to use is available from both FDs. In flatpak sandboxes, however,
this requires to give full Pipewire access to the application.

Fix this by simply using the right, restricted FD, and while on it,
also make sure to not leak it.

This change has already landed in downstream in Firefox, see
https://phabricator.services.mozilla.com/D122904
https://phabricator.services.mozilla.com/D124508

Bug: webrtc:13152
Change-Id: I3f8995c54c797e1a90a980f231e496a13cbe65b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230803
Reviewed-by: Joe Downing <joedow@chromium.org>
Commit-Queue: Joe Downing <joedow@chromium.org>
Cr-Commit-Position: refs/heads/main@{#34983}
2021-09-13 16:29:52 +00:00