Commit Graph

32830 Commits

Author SHA1 Message Date
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
64f7da0b16 Use input_state to get pixels for single active stream.
Bug: none
Change-Id: I103d2cc111ca08d1e5acde1f25c125c075502eca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208226
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33358}
2021-03-01 15:58:03 +00:00
31e06cb63d addIceCandidate: prefer ice candidate sdpMid over sdpMLineIndex
as described in JSEP
  https://tools.ietf.org/html/rfc8829#section-3.5.2.1
 If the MID field is present in a received IceCandidate, it
 MUST be used for identification; otherwise, the "m=" section
 index is used instead.

BUG=webrtc:12479

Change-Id: I688a5e59024fe8cc6a170c216c6f14536084cfb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33357}
2021-03-01 14:01:39 +00:00
df6d4ca3f5 Prevent TURN from connecting to ports < 1024 (except 443)
This should prevent TURN from being used in NAT slipstream attacks
with native clients.

Bug: webrtc:12497
Change-Id: I3c33df92e97b8e6430b72a05790c137d89661093
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208582
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33356}
2021-03-01 13:43:39 +00:00
bfc3ef0fec Mention ninja all in webrtc development documantation.
Bug: b/177971218
Change-Id: Ia49ca147ad286ef1ad8ac8ddd6b7fd8b05e384ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208942
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33355}
2021-03-01 12:14:13 +00:00
4785402475 Replace RecursiveCriticalSection with Mutex in ProcessThreadImpl
Bug: webrtc:11567
Change-Id: I03961ddc55f29a01c3e466217222fd56ba51d895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208764
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33354}
2021-03-01 12:13:09 +00:00
1e2da374b3 Update WebRTC code version (2021-02-28T04:02:30).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: If3304144dbf58dec0babaf873a5858f40539b908
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209020
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@{#33353}
2021-02-28 04:56:14 +00:00
ac615b6766 Update WebRTC code version (2021-02-27T04:03:14).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: If05da5731bedf3310c0748de02ade5c8597e94d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208920
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@{#33352}
2021-02-27 06:02:56 +00:00
c96aa30c13 Minor refactoring of RtpVideoSender
Bug: None
Change-Id: I7cc16d4e435974090310bd3a7cea9eaf0586be8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208761
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33351}
2021-02-26 13:20:52 +00:00
484acf2723 Add ability to configure sampling rate for input/output video dumps in PC level framework
Bug: b/179986638
Change-Id: I9ab960840e4b8f912abe4fb79cfd9278f4d4562a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33350}
2021-02-26 11:24:52 +00:00
9d9b8defae Reland "Rename SIGNALING and WORKER to PRIMARY and SECONDARY"
This is a reland of a37f2bd9421868e222d591d3371486a6ab939fd6

Original change's description:
> Rename SIGNALING and WORKER to PRIMARY and SECONDARY
>
> This makes the proxy macros less confusing when the secondary thread
> is sometimes the worker thread, sometimes the networking thread.
>
> Bug: none
> Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33346}

Bug: none
Change-Id: If46a6679ac0fc947797dd7be87626ef7702faca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208845
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33349}
2021-02-26 10:35:22 +00:00