5265b9367a
Add build-id to libjingle_peerconnection_so.so
...
This is required by Firebase Crashlytics:
https://firebase.google.com/docs/crashlytics/ndk-reports#enable-symbol-uploading
Bug: None
Change-Id: Ie0d2c2e92477df78b26b7e1fc2273589b71efa81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210965
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org >
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33408}
2021-03-09 20:21:23 +00:00
cb7ff13432
Roll chromium_revision e1b9354ff4..74090df66c (854007:861144)
...
Change log: e1b9354ff4..74090df66ce1b9354ff4..74090df66c242517a010..b64741b2d4251bfc5101..6e85f7f8aa228b1c1883..368c7dd2c9196ba1aaa8..c93be42a40d999d54f4b..cc80b4ac98d365fb8897..171bf5f5ff7dee089ce5..f8f63f141ad15b5c89b3..68e9de796chttps://boringssl.googlesource.com/boringssl.git/+log/ce9b002ebd..dfe0b01b3e
* src/third_party/breakpad/breakpad: e3d485f73f..d6a6f52606https://chromium.googlesource.com/catapult.git/+log/35976bfcac..4f82770e78
* src/third_party/depot_tools: 7386a1e04f..d071f2a8a97849316ccf..fedd945c3e1b0cdaae57..07f486922153806e5b83..7686ff854b12825ed8aa..e6379c8802https://aomedia.googlesource.com/aom.git/+log/e72d81ffbc..fb9f22ce8c
* src/third_party/libyuv: 93b1b332cd..a8c181050c7945040041..37518c39e3a6647318b5..6e00fe20dfbd77bcca0d..6cf14f8442e1b9354ff4..74090df66ce1b9354ff4..74090df66cTBR=titovartem@webrtc.org ,
BUG=None
Change-Id: Id2b5745ddc858a9cc4aa0ee7c4009fff62fc7fca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210962
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33407}
2021-03-09 17:09:11 +00:00
dc08aea610
Fix chromium roll into WebRTC
...
Add missing org_robolectric_shadows_multidex dependency as it is done
in chromium DEPS file.
Tried on: https://webrtc-review.googlesource.com/c/src/+/211004
Bug: chromium:1184780
Change-Id: Ib8f9e794b36c8da008672d989271f368df9203df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211040
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33406}
2021-03-09 14:35:39 +00:00
c84a887b8a
Allow port 80 for TURN servers
...
Bug: webrtc:12537
Change-Id: Id6a419b4b435284f743314956945783d4db94a7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210960
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Jonas Oreland <jonaso@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33405}
2021-03-09 09:58:58 +00:00
668dbf66ce
[Stats] Populate "frames" stats for video source.
...
Spec: https://www.w3.org/TR/webrtc-stats/#dom-rtcvideosourcestats-frames
Wiring up the "frames" stats with the cumulative fps counter on the video source.
Tests:
./out/Default/peerconnection_unittests
./out/Default/video_engine_tests
Bug: webrtc:12499
Change-Id: I4103f56ed04cb464f5f7e70fbf2d77c25a867a68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208782
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33404}
2021-03-09 08:54:38 +00:00
5ab9b32ee6
Update WebRTC code version (2021-03-09T04:03:43).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: I8915d8f307b09348e8879027d47a2a546103db54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210920
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/master@{#33403}
2021-03-09 06:13:31 +00:00
92d12707e0
Expose PeerConnection.restartIce in iOS SDK.
...
Bug: None
Change-Id: I76b95b3182e6b384fd68aecf4210c23459f76d2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209709
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com >
Cr-Commit-Position: refs/heads/master@{#33402}
2021-03-08 17:38:54 +00:00
d67253532f
Expose parameterless setLocalDescription() in iOS SDK.
...
Parameterless setLocalDescription is used to implement perfect
negotiation algorithm.
Bug: None
Change-Id: I89c39ee175fec5b09d9ca1700ef682e3cf20fe94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209700
Commit-Queue: Kári Helgason <kthelgason@webrtc.org >
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33401}
2021-03-08 16:39:54 +00:00
f00cd533ae
Do more actions on SDP fuzzing.
...
This will wait for completion of the setRemoteDescription() call,
and if the setRemoteDescription() is successful, it will do a
setLocalDescription() (default description).
Bug: none
Change-Id: Id1cb6b1ecbdc90d4f2c5b46a7f8e92b7491ff401
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210682
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33400}
2021-03-08 12:18:10 +00:00
a113d231a6
srtp: use srtp_crypto_policy_set_from_profile_for_* from libsrtp
...
use the helper functions
srtp_crypto_policy_set_from_profile_for_rtp
and
srtp_crypto_policy_set_from_profile_for_rtcp
provided by libsrtp to initialize the rtp and rtcp policies.
BUG=None
Change-Id: Ib1560c0fc1c06d9e79c1f871b028555b3b4d66d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208480
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Cr-Commit-Position: refs/heads/master@{#33399}
2021-03-08 10:41:29 +00:00
c81665cf9d
Change AndroidNetworkMonitor::NotifyConnectionTypeChanged to use Invoke
...
This is consistent with other Notify methods in this class, which
handle callbacks from java using blocking invokes to the network
thread.
This eliminates the use of the deprecated AsyncInvoker class.
Bug: webrtc:12339
Change-Id: Ib2d19b37b8f669df5b97e89d720f6eb6fc9e5517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209181
Reviewed-by: Taylor <deadbeef@webrtc.org >
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33398}
2021-03-08 09:04:29 +00:00
82a94125a4
Reland "Add a fuzzer test that tries to connect a PeerConnection."
...
This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38.
Reason for revert: Added Chromium compile guards
Original change's description:
> Revert "Add a fuzzer test that tries to connect a PeerConnection."
>
> This reverts commit c67b77eee4b08c05638a219723a9141a65015da4.
>
> Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll.
>
> Original change's description:
> > Add a fuzzer test that tries to connect a PeerConnection.
> >
> > Bug: none
> > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org >
> > Reviewed-by: Henrik Boström <hbos@webrtc.org >
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#33369}
>
> NOPRESUBMIT=true
>
> Bug: none
> Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640
> Commit-Queue: Evan Shrubsole <eshr@google.com >
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#33380}
Bug: none
Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 08:58:09 +00:00
be66d95ab7
srtp: document rationale for srtp overhead calculation
...
documents why it is safe to not follow libsrtp's advice
to ensure additional SRTP_MAX_TRAILER_LEN bytes are available
when calling srtp_protect (and similar srtcp functions).
BUG=None
Change-Id: I504645d21553160f06133fd8bb3ee79e178247da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209064
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Cr-Commit-Position: refs/heads/master@{#33396}
2021-03-08 08:50:09 +00:00
456a2642d3
Update WebRTC code version (2021-03-08T04:01:44).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: Ia9c79e506989ca773c47399b3f1eca56875b096c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210640
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/master@{#33395}
2021-03-08 05:59:19 +00:00
aedb0f0c66
Update WebRTC code version (2021-03-07T04:02:41).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: I1f4e7859d1fcca9154840ad72bac68e152456ef9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210380
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/master@{#33394}
2021-03-07 05:50:46 +00:00
eb449a979b
Revert "Reland "Enable quality scaling when allowed""
...
This reverts commit 83be84bb74133343358bba22e4e5106ecc385721.
Reason for revert: Suspect of crbug.com/1185276
Original change's description:
> Reland "Enable quality scaling when allowed"
>
> This reverts commit 609b524dd3ff36719b5c4470b85d37dcdadfb1f8.
>
> Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.
>
> Original change's description:
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020
> Bug: webrtc:12511
> Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
> Reviewed-by: Åsa Persson <asapersson@webrtc.org >
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#33385}
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I7004014c5936176f8c125aeb55da91ce095b266e
TBR: ssilkin@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209708
Reviewed-by: Guido Urdaneta <guidou@webrtc.org >
Commit-Queue: Guido Urdaneta <guidou@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33393}
2021-03-06 09:40:50 +00:00
77f97ec066
Update WebRTC code version (2021-03-06T04:02:56).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: Ied0c453f05c0d2d172661b712d63660ce1227988
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210120
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/master@{#33392}
2021-03-06 05:40:47 +00:00
964a88fce1
Prevent possible out-of-bounds access
...
It read from simulcast_layers[0] even if vector is empty.
Bug: none
Change-Id: I293890feda70022beae4247ab10cf8182b4cf4a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209706
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33391}
2021-03-05 17:48:48 +00:00
4a4273bf05
VP9 ResolutionBitrateLimits: If bitrates are configured, use intersection.
...
Bug: none
Change-Id: I58ada41d7a196837b35df4cf61f7e7561998cf13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209704
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
Commit-Queue: Åsa Persson <asapersson@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33390}
2021-03-05 17:39:04 +00:00
a86b29be01
Add VP9-specific default resolution bitrate limits
...
Bug: none
Change-Id: Ifb6f962f04b1f05d20f80a721b1f41904e0a7e99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209702
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33389}
2021-03-05 14:28:14 +00:00
1a89bc8b62
build: improve rtc_include_tests documentation
...
"including" them in the standalone checkout is ambigious,
this might mean not checking them out from git.
BUG=None
NOTRY=true
Change-Id: I53334a718c5535f63faa87d9e0fe6575012e0d2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209223
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Cr-Commit-Position: refs/heads/master@{#33388}
2021-03-05 13:54:20 +00:00
1413e2db4c
Update WebRTC code version (2021-03-05T04:03:29).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: I8fbcbf192da90593ad03fc8da0392957d33753ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209861
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/master@{#33387}
2021-03-05 05:57:29 +00:00
8a38b1cf68
Revert "Fix race between destroying SctpTransport and receiving notification on timer thread."
...
This reverts commit a88fe7be146b9b85575504d4d5193c007f2e3de4.
Reason for revert: Breaks downstream test, still investigating.
Original change's description:
> Fix race between destroying SctpTransport and receiving notification on timer thread.
>
> This gets rid of the SctpTransportMap::Retrieve method and forces
> everything to go through PostToTransportThread, which behaves safely
> with relation to the transport's destruction.
>
> Bug: webrtc:12467
> Change-Id: Id4a723c2c985be2a368d2cc5c5e62deb04c509ab
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208800
> Reviewed-by: Niels Moller <nisse@webrtc.org >
> Commit-Queue: Taylor <deadbeef@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#33364}
TBR=nisse@webrtc.org
Bug: webrtc:12467
Change-Id: Ib5d815a2cbca4feb25f360bff7ed62c02d1910a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209820
Reviewed-by: Taylor <deadbeef@webrtc.org >
Commit-Queue: Taylor <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33386}
2021-03-05 03:06:38 +00:00
83be84bb74
Reland "Enable quality scaling when allowed"
...
This reverts commit 609b524dd3ff36719b5c4470b85d37dcdadfb1f8.
Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.
Original change's description:
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).
This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33385}
2021-03-04 16:01:23 +00:00
d140c8f43b
Added missing nullable annotations to iOS SDK.
...
Some of the PCF and PC methods are actually return nil, but was by
default annotated as nonnull via NS_ASSUME_NONNULL_BEGIN.
Bug: None
No-Presubmit: True
Change-Id: Ib8b9263452a61241c9e7a16c1807d87bd597c093
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209180
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com >
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33384}
2021-03-04 13:22:26 +00:00
854d59f750
Temporarily disable remaining Opus bit exactness tests.
...
Bug: webrtc:12518
Change-Id: Ia006c4258404a6e124101cd4ebfd399008f82227
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209645
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33383}
2021-03-04 13:02:17 +00:00
3208bf102b
Allow uninstantiated parametrized tests in data_channel_integration_tests.cc
...
Bug: webrtc:12525
Change-Id: I79244a3a6de29bcf77677dbc19368618116f8f9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209642
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33382}
2021-03-04 10:20:34 +00:00
609b524dd3
Revert "Enable quality scaling when allowed"
...
This reverts commit 752cbaba907de077e5f1b24a232e71feb479dccb.
Reason for revert: The test VideoStreamEncoderTest.QualityScalingAllowed_QualityScalingEnabled seems to fail on iOS.
Original change's description:
> Enable quality scaling when allowed
>
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020, webrtc:12511
> Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
> Reviewed-by: Åsa Persson <asapersson@webrtc.org >
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#33373}
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: Icabf2d9a034d359f79491f2c37f1044f17a7445d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209641
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33381}
2021-03-04 10:11:36 +00:00
ae44fde188
Revert "Add a fuzzer test that tries to connect a PeerConnection."
...
This reverts commit c67b77eee4b08c05638a219723a9141a65015da4.
Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll.
Original change's description:
> Add a fuzzer test that tries to connect a PeerConnection.
>
> Bug: none
> Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182
> Commit-Queue: Harald Alvestrand <hta@webrtc.org >
> Reviewed-by: Henrik Boström <hbos@webrtc.org >
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#33369}
NOPRESUBMIT=true
Bug: none
Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640
Commit-Queue: Evan Shrubsole <eshr@google.com >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33380}
2021-03-04 09:42:34 +00:00
d744c5a9e4
Update WebRTC code version (2021-03-04T04:02:19).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: Ie58acd9eee4556c17fd62cb65f146030e2724ab9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209580
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/master@{#33379}
2021-03-04 05:55:43 +00:00
8cfb287735
Add AV1 encoder&decoder wrappers for iOS SDK.
...
It is now possible to use AV1 encoder and decoder on iOS and test
them in apps like AppRTCMobile.
Bug: None
Change-Id: Ifae221020e5abf3809010676862eecd9ffeec5e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208400
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33378}
2021-03-03 20:08:17 +00:00
662d4c328f
AV1 test: change ssim threshold
...
Bug: webrtc:12519
Change-Id: Ibdcaa08800d03d289f86e14cc7d94b5f2d3b7117
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209480
Reviewed-by: Marco Paniconi <marpan@webrtc.org >
Commit-Queue: Jerome Jiang <jianj@google.com >
Cr-Commit-Position: refs/heads/master@{#33377}
2021-03-03 18:22:45 +00:00
83e6eceac7
Comment out uninstantiated parametrized PC full stack test
...
Bug: None
Change-Id: If4756fd30df5788fdbe8bfcb36f5333167c50669
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209460
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33376}
2021-03-03 17:31:46 +00:00
e7a55813f9
Temporarily disable some Opus bit exactness tests.
...
This is required to be able to update the Opus version and will be
rolled back after.
Bug: webrtc:12518
Change-Id: Icc649039787db44bd55a0dc8e5ba4089df3a9566
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209363
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Reviewed-by: Minyue Li <minyue@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33375}
2021-03-03 15:30:46 +00:00
b6b782da68
Fix potential unsafe access to VCMTimestampMap::data
...
The access to |_timestampMap| was guarded by a lock but
not the access to the data pointer stored in |_timestampMap|.
There was a potential race condition if new data was added
in VCMGenericDecoder::Decode() while the data pointer
retrieved from _timestampMap.Pop() was being used in
VCMDecodedFrameCallback::Decoded().
This CL moves the storage of data to within |_timestampMap|,
instead of being a pointer so that it's guarded by the same
lock.
Bug: webrtc:11229
Change-Id: I3f2afb568ed724db5719d508a73de402c4531dec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209361
Commit-Queue: Johannes Kron <kron@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33374}
2021-03-03 14:21:17 +00:00
752cbaba90
Enable quality scaling when allowed
...
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).
This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.
Bug: chromium:1179020, webrtc:12511
Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33373}
2021-03-03 13:57:22 +00:00
db0b4a8935
Do not crash if codec is not available
...
Check if codec was successfully created and exit from RunTest if not
before creating VideoProcessor.
Bug: none
Change-Id: Ia6d7171650dbc9824fb78f4a8e2851f755cfd63b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209362
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33372}
2021-03-03 13:56:17 +00:00
652ada5029
Enabling a safe fall-back functionality for overruns in the runtime settings
...
Bug: b/177830919
Change-Id: I9369f6fc004ceb2b626d33b36262bc8aeabdb1a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206988
Commit-Queue: Per Åhgren <peah@webrtc.org >
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33371}
2021-03-03 12:06:54 +00:00
99bcf60a41
Check MID for illegal token characters.
...
Bug: webrtc:12516
Change-Id: I311dc984aa1dc8784d3ba3394676337b35cc92d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209360
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33370}
2021-03-03 08:55:39 +00:00
c67b77eee4
Add a fuzzer test that tries to connect a PeerConnection.
...
Bug: none
Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33369}
2021-03-02 19:28:23 +00:00
94f77ebbc5
Allow using test code in fuzzers
...
This makes certain fuzzers easier to write.
Bug: none
Change-Id: If1dad462ff71075c9cc81d35f19556b4c1b7108f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209221
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33368}
2021-03-02 15:30:21 +00:00
09932cded8
Manually update jdk DEPS.
...
This should hopefully unblock the WebRTC DEPS autoroller
https://ci.chromium.org/p/webrtc/builders/cron/Auto-roll%20-%20WebRTC%20DEPS
Bug: None
Change-Id: I0b6df2e4ca7df5a3ca1f58e86b3f95a493eddef1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209183
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33367}
2021-03-02 12:56:27 +00:00
07d83c8a9a
Modified STUN verification functions
...
The new verification makes verification a function on a message.
It also stores the password used in the request message, so that
it is easily accessible when verifying the response.
Bug: chromium:1177125
Change-Id: I505df4b54214643a28a6b292c4e2262b9d97b097
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209060
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33366}
2021-03-02 10:34:17 +00:00
d99da804d7
Parent CL.
...
Change-Id: I0221159dc334f8c358fcc73c8e2cbe41da89799c
No-Try: True
TBR: mbonadei@webrtc.org
Bug: webrtc:12512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209080
Reviewed-by: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org >
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33365}
2021-03-02 09:43:11 +00:00
a88fe7be14
Fix race between destroying SctpTransport and receiving notification on timer thread.
...
This gets rid of the SctpTransportMap::Retrieve method and forces
everything to go through PostToTransportThread, which behaves safely
with relation to the transport's destruction.
Bug: webrtc:12467
Change-Id: Id4a723c2c985be2a368d2cc5c5e62deb04c509ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208800
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Taylor <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33364}
2021-03-01 22:53:13 +00:00
88a51b2902
Populate "total_round_trip_time" and "round_trip_time_measurements" for remote inbound RTP streams
...
Spec: https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict *
Adding them into the stats definition as well.
Bug: webrtc:12507
Change-Id: Id467a33fe7bb256655b68819e3ce87ca9af5b25f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209000
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33363}
2021-03-01 20:49:22 +00:00
625e1d9b16
VP8 ref finder unittest cleanup
...
Change-Id: I627dda1229ceb4b2da3f37f7418da7b7653e4d04
Bug: webrtc:12221
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208482
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33362}
2021-03-01 17:23:38 +00:00
86f04ad135
Populate “fractionLost” stats for remote inbound rtp streams
...
Tests:
./out/Default/peerconnection_unittests
Manually tested with Chromium to see the data populated
Spec: https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict *
Bug: webrtc:12506
Change-Id: I60ef8061fb31deab06ca5f115246ceb5a8cdc5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208960
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33361}
2021-03-01 16:48:37 +00:00
6512af04ba
Add root doc site definition for WebRTC documentation
...
Bug: None
Change-Id: I64f5a356e0358360bd8326c39c8b0a898b879ebf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208641
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33360}
2021-03-01 16:40:58 +00:00
f57f2cd3d2
doc: add M88/M89 release notes
...
BUG=None
Change-Id: Ia5bb63a0825a4dd5aaeca305fb7796f36dd06e92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Cr-Commit-Position: refs/heads/master@{#33359}
2021-03-01 16:16:03 +00:00
